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Development => Amsn development related issues => Topic started by: kakaroto on February 10, 2008, 07:00:45 am



Title: Audio/Video conversation
Post by: kakaroto on February 10, 2008, 07:00:45 am
Hi,
For the interested, you can now do an audio call in aMSN... The audio+video call is still being worked on but it's not complete though...
Here's a small summary of what you need for aMSN to have audio calls working :
1 - get all the dependencies installed.. here's how : http://amsn-project.net/wiki/Farsight
2 - take the latest SVN version of amsn, then ./configure && make (make sure it detected farsight installed during the configure)
3 - run amsn and from a chat window : actions->start audio call


For those interested in the audio+video feature, here's the original post I had in here :

Hi,
here's a copy of a mail I sent to the amsn-devel mailing list :
Quote
Hi all,
I don't know if many of you are following the amsn-commits mailing list, but for those who don't, this email is for you!
I want to let you guys know that I'm working on the audio/video conferencing and it's going pretty good so far!
Basically what this is for those who aren't aware, is the 'video call' feature of WLM, which is audio AND video AND bidirectional. In other
words, one click and you start sending and receiving both audio and video.. so no more 2 invites to send/receive webcam + use voice clips...

Here's the current status :
first, it's far from being done, but I can get audio + video decoded. The problem is... to play them...
So here's how it works,
once you do the a/v conf you see a dump.av file in your amsn dir.. then you can do ...
Code:
./demux.tcl dump.av dump
which will create 3 files :
dump.siren, dump.raw and dump.wmv3.
then do
Code:
./wmv3_dec dump.wmv3 out.rgb
which will decode it in rgb and open display to show you each image.
and for the sound.. simply do
Code:
cat dump.raw | ./stream_audio.tcl


Problems are :
1 - i don't send yet
2 - no tk extension to decode the wmv3 yet (it's a little C program)
3 - can't record/play audio unless using snack and snack doesn't like real time streaming (it opens/closes the audio device on every chunk
(every 50ms) so the sound gets chopped so it looks like garbage, not real sound...
4 - i can send sound (not written yet) but there's no wmv3 encoder in open source anywhere..
So that's the current status..

About sending video, implementing a WMV3 encoder is too complicated and can't/won't be done. So our choices are :
1 - don't send video...
2 - encode/send the video with libmimic..
(libmimic is the codec currently used for the 'webcam' feature)

there's an FCC (Four Character Code) being sent with every frame, so I can send ML20 in the FCC instead of WMV3 and hopefully, WLM reads the
FCC and chooses the decoder accordingly. if it does, then we're good, if the wmv3 encoder is hardcoded, then we won't send video


That's about it.. you can test it if you want, it's in the 'video_conf' branch of SVN... you can use the status log to see if you are
receiving video/audio frames... and that's it!
SVN revision 9556 is necessary...

I'd like to know if it works for anyone besides me...

KaKaRoTo


So this is to announce that it's being worked on... and it requires MSNP12, so you don't need to go into MSNP15 to make it work (but it works just as good on MSNP15 so if you already use that, no need to go back to MSNP12 either).
You would need to ask someone to send you an invite, or if you want to 'send' the invite yourself, just do this in the console (ctrl-shift-C) :
Code:
::MSNCAM::StartVideoConferenceQueue $email


You are free to test, no bugs to report please, but just have fun with it.. hopefully, it will evolve into a much better/fuller implementation in the next few days...


Title: Audio/Video conversation
Post by: kakaroto on February 10, 2008, 08:49:18 am
by the way, the window that appears showing you the webcam is imagemagick's 'display' command, it will show you one frame at a time, you must press the spacebar to see the next frame, etc... In the latest SVN, I just gave it an additional parameter (-delay 1) so that it will show you the next frame after 1sec delay, but it's very slow...
also, you might notice that you can hear the first few seconds of audio in aMSN when you start the A/V conversation, then it stops, it's normal.. to hear it fully, just use the command I gave you above in your xterm.


Title: Audio/Video conversation
Post by: iron_maiden89b on February 10, 2008, 08:06:18 pm
Good!!


Title: Audio/Video conversation
Post by: kakaroto on February 13, 2008, 07:39:34 am
Hi,
a bit of news...
1 - I found a way to record live audio using snack, that should help us.. but for playing live audio in snack, that's still a problem... we might have to modify snack.. since snack is an abandonned project, we *might* have to take it to ourselves, fork it, enhance it to support live audio and fix some of the bugs people are complaining about in snack... It would be shipped with amsn on windows and mac, and for linux users, it should be compiled automatically when you compiled amsn.
2 - The WMV3 decoding seems to work great BUT I noticed that it might crash (segfault) on some frames, I already fixed a bug in libavcodec for decoding WMV3, and now, I'll just have to fix yet another one.. hopefully it will become more stable soon.. I'll be in contact with the ffmpeg developers in order to make sure no issues will be coming from there..
I don't have a lot of time available lately, but hopefully, I should be able to finish this pretty soon.. hopefully before FOSDEM! :)
I'll keep you posted! :D


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on February 18, 2008, 06:51:07 pm
I was wondering why I couldn't get an audiovisual session in aMSN... so that is why.
I had set the webcam for audio... but wondered why I couldn't get or send audio... heheh... I see now that sound clips must be used.

Let me tell you, aMSN developers, that I'm really looking forward for this feature (you see... there's this girl...  :roll:  :oops: ) and I really appreciate your work on this! kudos!
I'll be following this thread closely.
Having bidirectional audiovisual session + yahoo contacts working on aMSN... will be pure bliss!  :D

I'm curious, will bi-directional audio + video work with yahoo contacts as well, or would that require reverse engineering yahoo's protocol as well  :x And I perfectly understand if the developers don't even think about it.

microsoft is such a nuissance... Communication protocols and data formats should be open sourced or well documented... Hey that's a good topic for FOSDEM :)
There should be a lobby to force the opening of protocols and data formats which when closed limit the choice of freedom.

BTW (a bit off topic here and maybe silly but), have you ever proposed aMSN improvements to Google's Summer Of Code?

One last thing, you said "I don't have a lot of time available lately, but hopefully, I should be able to finish this pretty soon.. hopefully before FOSDEM!"
You mean item number 2, or the whole bidirectional audiovisual session? That WMV3 audio and video encoder sure seems a real pain :(


Thank you for everything!

I'll see if I can test it and post some results


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on February 18, 2008, 07:29:13 pm
hmm... I was wondering... is this by any chance usefull for WMV3 encoder purposes: http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2007-June/031699.html


Title: Audio/Video conversation
Post by: Kalinda on February 18, 2008, 10:04:08 pm
Quote from: "why.arent.guests.allowed"
I'm curious, will bi-directional audio + video work with yahoo contacts as well, or would that require reverse engineering yahoo's protocol as well  :x And I perfectly understand if the developers don't even think about it.

No, it won't. I mean, as I understand it, the audio chat/webcam doesn't even work with Yahoo contacts in the official client. All you can do is chat with them through text, that's the extent of the interoperability.


Title: Audio/Video conversation
Post by: kakaroto on February 18, 2008, 10:29:43 pm
yep, as Kalinda said, you can only chat with yahoo contacts, you can't do webcam/audio, you can't even do a file transfer as far as I know, and that's a limitation of the protocol and interop decisions, same applies for the official client.


Title: Audio/Video conversation
Post by: kakaroto on February 27, 2008, 11:19:50 am
Hi again,
Just want to post up some news here...
We've worked on this A/V thing a bit at FOSDEM and we got it to work a bit better... the audio is still not being processed, but the video is...
so now, you can go grab the latest SVN version of the video_conf branch, then recompile it (along with ffmpeg...) and see for yourself.. in theory, it should work fine for receiving... for sending, I'm sending an ML20 encoded stream, but I don't know yet if it works or not (my WLM blue screens whenever I try something with it other than typing text...).
So, can you guys test it and see how it goes ? you're now allowed to complain about stuff (video related, and NOT audio related please!). I would really like to know if sending video works or not. Most of the stuff should now work, but it's not yet completed... for example, don't try the 'pause', or don't expect it to work if you don't have a webcam (in theory it should just work without sending the webcam), etc...
You can also review the A/V session in the webcam log viewer...


Title: Audio/Video conversation
Post by: Kalinda on February 27, 2008, 11:03:53 pm
Alright, I tried it (just the wbecam bit) with one of my guinea pigs and he said that after I had accepted the invite, his window changed to receive webcam but he never got anything.

On my end, it said he rejected the webcam session and then I saw my own webcam window appear, but it never sent to him. He's using the latest WLM (2008 (Build 8.5.1302.1018))

Now, I'm assuming I don't need to use the console commands anymore (for webcam, anyway), since it looks like you took out the dump.av thing. Am I correct?

Keep up the good work, though :) I figured you guys would've worked on it at FOSDEM.


Title: Audio/Video conversation
Post by: kakaroto on February 28, 2008, 03:22:03 am
Hello Kalinda.
I finally got my PC to stop BSOD-ing on the video call. I was able to test and I see that it doesn't work with mimic codec. There's something wrong in there because it also doesn't work with WMV3 (I send a WMV3 webcam log as if it was my own webcam).. but it's really weird because it shows the 'no webcam' icon, but if I make it go into full screen mode, it then shows the webcam correctly, if I go back to non-fullscreen mode, it shows the 'no webcam' icon again...
anyways, so that's that... by the way, you didn't say if you were able to receive webcam from your friend correctly or not. I suppose you did.. if you didn't, make sure you compiled ffmpeg correctly. When doing ./configure, it should tell you if ffmpeg was found or not (if not, it will tell you how to compile it).


Title: Audio/Video conversation
Post by: Kalinda on February 28, 2008, 06:39:30 pm
Hello :)

Yeah, I didn't receive from my friend. Sorry, I forgot to mention that. I did compile ffmpeg using the command provided and I got no errors, so I assume it worked out alright. Maybe you and I should test it.

I'll have to see about this fullscreen mode thing. It's unfortunate because I don't have two webcams, but I do have a few computers.


Title: Audio/Video conversation
Post by: kakaroto on February 28, 2008, 11:10:08 pm
install webcammax  : virtual webcam for windows..
maybe you didn't receive your friend's webcam because he didn't have a webcam or was not sending it ?


Title: Audio/Video conversation
Post by: Kalinda on February 29, 2008, 03:00:58 am
Alright, I tried it with myself and it worked better. I was able to receive the WLM webcam, but I still couldn't send the aMSN one to WLM and when I full screened it in WLM, I just got a black screen.

I tried it again with my friend and was able to successfully receive his webcam but, like before, he couldn't receive mine. I dunno what I was doing wrong before, but part of it works, so hooray!

Thanks for the webcammax thing, too, it's really cool and helpful.


Title: Audio/Video conversation
Post by: kakaroto on February 29, 2008, 06:29:18 pm
Cool! :)
Yes, webcammax is awesome! :D Too bad it's not free/open source, but it's good..
anyways, about the full screen trick, it only works if you send a WMV3 video. Currently, aMSN sends a mimic video (so it should work between two aMSN instances, but not with WLM). There is no WMV3 encoder available anywhere (open source), so we can't encode in WMV3 right now, even if we wanted to. What I did to get it to send in WMV3 and have it work in full screen was to send a WMV3 webcam log as my own. It's an old trick existing since we first implemented webcam support :p just open the console and type :
Code:
set ::test_webcam_send_log /path/to/webcam.log


Title: Audio/Video conversation
Post by: Kalinda on March 01, 2008, 05:20:06 pm
Ahh, so are you just going to leave it alone for now and do the audio part instead?

Actually, above someone posted a link about a patch to ffmpeg for a wmv3 encoder. Would that work? I wonder if it's still in their SVN or if it was abandoned...


Title: Audio/Video conversation
Post by: kakaroto on March 01, 2008, 10:58:39 pm
Yes, for now, I'll leave it alone.. well, It does send using mimic, so I'll leave it sending with mimic so that at least it will work between two aMSN clients (since aMSN is not as stupid as WLM and it does check for the FCC and chooses the right decoder depending on how it got encoded, while WLM always uses the WMV3 decoder). There are still some tweaks that need to be fixed though.. for instance, if you don't have a webcam, it will cancel the A/V, and that's because we use the same code for capturing/encoding/sending as for the webcam feature, and it's normal that for the webcam feature, if you are sending and you don't have a webcam, then there's no reason not to cancel the whole thing. So I'll just have to make sure it doesn't do that and have it keep the A/V open but just disable sending webcam. I also need to tweak it to have the long-awaited 'webcam image in the chat window' feature (for both webcam and A/V). This will be useful because of the two way webcam. + we'll need to add a change volume thing to the UI. Also allow for sending webcam+audio or webcam only or audio only or none... etc... all those tweaks to make the feature 'finalized'.
But first, I'll have to work on the audio stuff.. I might have a solution for getting the audio to play correctly so I'll have to try that.. then there's capturing which is also possible as far as I know, but we'll have to wait and see how it works out.

About the wmv3 encoder, it was never committed to SVN, so the code is nowhere to be found apart from the attached patch to the mailing list.. but of course, the attached files are not saved for the web archive of the mailing list... I wrote to the author of the encoder as soon as I read about it but he hasn't answered me yet... I'll wait and maybe try to contact some other ffmpeg developer to see if they still have the patch somewhere on their system or saved in their mail client...

EDIT Ahhhh, found it!! I just asked and they made me realize on how stupid I am.. the attachment IS saved even in the web archive and the link to the patch was in the mail, I just didn't see it!!!  
So here it is : http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/attachments/20070625/bbadb401/attachment.bin :D


Title: Audio/Video conversation
Post by: kakaroto on March 02, 2008, 08:01:39 am
ok.. now I got the encoder in there... if you look at the SVN log... it says :
Quote
 encoder works BUT only for a main profile with sequence layer 0x40010005. the msn profile is a simple profile with sequence layer 0x0ff18001.. This makes it use sequence layer 0x0ff18001 but it doesn't work yet...

In other words.. it works if I encode it, then decode it manually using a specific 'sequence layer' (kind of 'encoding settings' for bitrate, compression used, etc...). If I use the same sequence layer as what WLM uses, it gives us garbage image... which means something is wrong (maybe wrong compression algorithm used or something...).
Other news... now sound should work.. so if someone could test this, update the SVN and try to see if the sound works fine for you... In theory it shouldn't.. here's the svn commit log message. :
Quote
Audio reception should now work BUT with stupid snack lib, audio is play a bit faster than it should so previous buffer finishes playing too early, so we get a small lag between two buffers.. the bigger the buffering, the bigger the gap, the smaller the buffering, the more annoying the gap is... I tried putting a median value... until we wrote our own extension...

In other words, you shouldn't really have any problems.. sound should work fine.. BUT if you test by trying to say a long sentence with no interruptions, you'll notice that there is a small silence every 2 or 3 words, but it's "ok" for now...
To really fix audio we'll have to write our own playing/recording extension and drop snack because it is just too crappy... :(
I'll wait and see what you think.. if it worked for you or not.. if you notice the lag in the sound or not, if it's big or small, etc... (I know snack behaves differently from one system to another...)


Title: Audio/Video conversation
Post by: Kalinda on March 02, 2008, 06:57:26 pm
Ok, I tested it...

AUDIO:
It worked for receiving (it made weird noises and occasionally cut out, but I know MSN's audio isn't perfect). It was generally constant, with only a few pauses. So, good going :)

However, I couldn't send. Or rather, my friend heard nothing... I dunno if that's my mic or the program, but when I ran it through the audio/video wizard and tested it, the mic worked fine and I know it's on and it works for other things. I installed libsnack, but is there anything special I have to do with it, like when I had to set up ALSA for my mic?

WEBCAM:
Since you haven't quite got the encoder down yet, it didn't work.... and my friend has no webcam so I couldn't test that part. I'll test it with my other friend, though, when he gets on.

I got this terminal output during the whole thing:
Code:
vid-probe: trying: v4l2...
ioctl VIDIOC_QUERYCAP: Invalid argument
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found Creative Nx Pro at /dev/video0


Hope that helps :)


Title: Audio/Video conversation
Post by: kakaroto on March 03, 2008, 03:24:30 am
Cool, thanks for the feedback, appreciated :)
About audio, yes, it's normal, the weird noises are caused by the codec I think, but the 'few pauses' are the big problem caused by libsnack... which is why we'll need to rewrite our own sound extension instead of using snack.. sucks :(
About sending, don't bother, it's not there yet, I didn't do anything for sending audio (yet), so just wait...
I might do it with snack now, but I think it will be 10 times easier to just wait until we write our sound extension instead of doing the work twice...
About webcam. yeah, it still wouldn't work sending, but I think I temporarly disabled it for now so we never send anything... (I don't have my webcam anymore so I disabled sending to avoid the auto-cancellation when no webcam is found). Also, the encoder isn't fully working yet, so...
About those messages, it's normal, you should get the same when you do a normal 'webcam' session...


Title: Audio/Video conversation
Post by: kakaroto on March 06, 2008, 03:32:39 pm
news news news!!!
Audio : We have a new tcl extension.. this one uses libao, so it should work MUCH better with it than with snack.. you will now need libao and libao-dev packages installed on your system in order to compile amsn. Once you do that, you will get the sound working a lot better... depending on your connection of course... I tried it on local network and it works extremely well.. I was able to watch a movie on my desktop but listen to it on my laptop.. the sound quality is excellent (even with non-voice/music), the lag is 50ms, there is no gap in the sound, it is very smooth, looks like a steady stream of audio, you don't notice any gaps between the packets and everything is fine! :)
Also, you should notice that those 'noises' that we heard before disappeared.. which means that the noises in the audio were caused by snack and not by the codec...

Video : I just fixed the wmv3 encoder so that it can work with the msn profile (using implicit quantization instead of explicit which was hardcoded before). This means that it's good news because I can now work on integrating the encoder into aMSN and be able to encode the webcam in WMV3 so we can send it.. it's not done yet, but should be coming pretty soon.

TODO :
1 - write a library/extension for capturing audio and for the mixer.
2 - record audio and send it correctly
3 - Try to find out why the WMV3 encoder lets me encode only one frame
4 - integrate the WMV3 encoding into webcamsn extension so it can be access with aMSN
5 - integrate the WMV3 encoding into aMSN so we can use the correct encoder depending on the session and fix the FCC in the header.
6 - try to find out why WMV3 stream doesn't work with WLM unless you go full screen... is it because we send only the webcam and no audio packets ?

Once it's done, I will merge the branch into the svn trunk so that others can try it and work on it. This doesn't mean the job is done, this is merely the beginning.. we will also need to build an appropriate UI for it. Apart from the menu button to start the call, we should adapt the current audio system (we're dropping snack) and update the assistant, have the webcam windows show up in the chat window, show a 'connecting' when we are connecting, have sliders for the volume, allow to pause/mute the webcam/sound. etc...

Any help will be appreciated of course! Feedback is also helpful :)
Enjoy! :)


Title: Audio/Video conversation
Post by: strickerkr on March 09, 2008, 04:50:43 pm
That sounds very good. I would like to tes the feature when's it's integrated.

mfg

Stephan


Title: Audio/Video conversation
Post by: Fabioamd87 on March 09, 2008, 08:08:06 pm
Quote from: "strickerkr"
That sounds very good. I would like to tes the feature when's it's integrated.

mfg

Stephan

me too


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on March 09, 2008, 09:06:24 pm
Thank you very much for your hard work on this kakaroto. That sure is impressive.
You sure hack and slash that code like... mmmh.... *looks at Fabioamd87's avatars* .... like Link through hordes of evil monsters.
I'm wondering... is aMSN a hobby or a day time job? :p

Sorry for not giving feedback as I should... Even though this is really one the features I'm looking forward to.
But to give you an idea, I've not used msn for more than a week now.
Nonetheless, I hope to be able to test the SVN version when things are integrated in trunk.

Once again thanks!


Title: Audio/Video conversation
Post by: kakaroto on March 10, 2008, 12:57:40 am
Hey there why.arent.guests.allowed (your nickname is quite complicated.. if u want to change it PM me with the new username! :p)
Thanks for the appreciation!! hehe, comparing me to Link.. quite nice! You could have compared me to how kakaroto fights against hordes of evil monsters too, no? :p
Anyways, no this is a hobby.. and it's taking a lot of my time, but thankfully I have some free time lately.. some other days I have no time at all...

Right now, I'm working on a new audio output library.. I found libao2 within mplayer's source code, it's like libao but heavily modified (supports volume mixer and has a dsound module for windows!) but it's crappy and needs to be rewritten.. this is going to take a lot of time unfortunately :( But hopefully, once I'm done we'll have a really good audio processing library! I also found some audio input code for recording, but it only supports alsa and OSS... I'll have to look into this..
Hopefully, you'll soon see this merged into the trunk! It can already be merged as it is safe enough already.. but I want to finish it (new audio lib + sending of voice+video) before I merge it into the trunk...
Unfortunately, I found some bugs in the wmv3 encoder for ffmpeg, so I'll have to fix those bugs first...


Title: Audio/Video conversation
Post by: lucianolev on March 10, 2008, 05:10:56 am
You're doing an awesome and revolutionary work here!  :D
aMSN will become the first open source application in supporting voice/video conversation for the MSN protocol as far as i know. This is great news for the open source community. 8)

Thanks for your time and keep up the good work!


Title: Audio/Video conversation
Post by: kakaroto on March 10, 2008, 06:57:04 am
thanks! and yes, we'll be the first :)
For now, in the branch, reception works pretty good, the only problematic thing is the sending...


Title: Audio/Video conversation
Post by: flomar34 on April 05, 2008, 08:24:30 am
Hello,

Let me try it to say in english.  :oops:

I wil have said it exactly as lucianolev :)

It's a really very good  news for linux users and hope you will succeed with it.

I think that every body is going to follow your hard job now.

Thanks for all you are doing

 :)


Title: Audio/Video conversation
Post by: kakaroto on April 05, 2008, 10:50:10 am
Hi flomar34 and welcome to the forums!
Thanks for the nice comments (and for registering just to say it!).

I'm trying to finish this but it takes more time than what I thought... we'll be having a Google Summer Of Code for the audio input/output library and hopefully we'll soon have a nice library for that... the WMV3 encoder has some stuff missing (p-frames are not encoded, only I-frames, but we need P-frames...), and once those two big tasks are done, we can move forward, but it might take a lot of time... :(

in the meantime, I just started yesterday on a different 'project'... I'm currently working on a Tcl SIP client that will work with WLM's SIP server... I was able to have quite easily a working audio conference using farsight/gstreamer in less than a day of work, so that's good news.. a lot of work will still remain there but it's going pretty good so far! The big issue is :
1 - add farsight/gstreamer/glib as a new dependency
2 - farsight needs glib's main loop which is incompatible with tcl's main loop.. so either use farsight2, which is a complete rewrite and still new (not much tested.. not very sure of its stability).. but it also uses the main loop although work is currently being done to remove that... or use a separate process for the streaming...  in any case, we'll move to farsight 2 at some point...
We also have another problem.. WLM doesn't play our sound if it doesn't receive RTCP... unfortunately, RTCP support in farsight is poor or requires jrtplib which is a crappy library... but farsight 2 has great RTCP support... so that should be the way to go now, tomorrow I'll try using farsight 2 and enable RTCP to see if it can play my audio!!

Right now, my work is already in amsn's SVN (trunk, not a separate branch) and we'll hopefully soon be able to demo it!

In the meantime, you can receive audio and video in the video_conf branch without any problems, you just can't send...!


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on April 05, 2008, 09:16:36 pm
Thank you for giving us an update on the status of things kakaroto, or should I say かかろと ;-)
(my big achievement for the day... getting japanese input on Linux... hehe)

Awesome news on the Google SoC!!! But I checked SoC webpage and didn't see aMSN there...
It's just too bad it is more troublesome than it looked like... But we'll wait :D
BTW, I remember seeing a possible SoC project by FFMPEG on WMV3. Maybe this is of interest to you. (don't know if there is any time to take an action given the deadline)

I'm not sure if I quite understand what you said about your other project...
Is it to be integrated in aMSN because it currently doesn't support audio conference? Will bidirectional audio-video conference be implemented through farsight2/gstreamer instead?

Thanks


Title: Audio/Video conversation
Post by: kakaroto on April 06, 2008, 12:51:02 am
Quote
Thank you for giving us an update on the status of things kakaroto, or should I say かかろと  
(my big achievement for the day... getting japanese input on Linux... hehe)

lol, thanks and congrats on japanese input :p

Quote
Awesome news on the Google SoC!!! But I checked SoC webpage and didn't see aMSN there...

it's because it's under the Tcl/Tk community, you can see the 'ideas' here : http://wiki.tcl.tk/20832
the "Audio input/output library and extension" is the one posted by me... I have a friend who applied for it, and I'll be mentoring.
About WMV3 on FFMPEG, yeah, I know and talked to them, but I can't do it since I'm already a mentor, and any other person I know who could do it are too busy and can't do it...
 
Quote

I'm not sure if I quite understand what you said about your other project...
Is it to be integrated in aMSN because it currently doesn't support audio conference? Will bidirectional audio-video conference be implemented through farsight2/gstreamer instead?

No.. let me explain it to you.. WLM has 5 different A/V features.. you can see a summary that a wrote here : http://imfreedom.org/wiki/index.php/MSN:AV
Here's a summary of that :
- The 'Video Call' feature is a bidirectional Audio and Video conversation which relies on the MSNP2P protocol for the signaling.
- The 'Webcam' feature is a unidirectional Video only conversation which relies on MSNP2P for the signaling.
- The 'Computer Call' feature is a bidirectional Audio only feature which uses SIP and RTP for the signaling.
(the other two being the voice clips and an old A/V system that aMSN can already use if you have the linphone extension (read the FAQ, we don't give support for it))

aMSN already supports "Webcam" as you probably know.. the remaining features is the 'video call' and the 'computer call'... the video call is MSNP2P and is bidirectional audio and video... while the computer call is SIP with RTP and is for audio only (bidirectional).
What I was working on from the start (in this thread) was the video call.. now, I put that on hold, and now I'm working on the computer call feature... I already have the SIP working, and farsight/gstreamer takes care of the RTP... I'll be writing soon a farsight 2 app that does the RTP and I'll be porting libnice to implement ICE draft 6 and make it compatible with MSN's ICE implementation in order to be able to do NAT traversal...

Once I complete these.. We'll have both features in aMSN... if you have glib/gstreamer/gst-plugins-base/gst-plugins-good/gst-plugins-bad/gst-plugins-farsight/farsight2 installed on your system as well as the compiled aMSN code for it, you will be able to do an audio only conversation using SIP and RTP (much better performance, quality, speed, nat traversal, etc...)... Or if you don't, then you can still do the Audio/Video call which doesn't have any dependency (apart from ffmpeg and the new libao/libai which will be shipped with aMSN).
Hope it's clear now!


Title: Audio/Video conversation
Post by: kakaroto on April 06, 2008, 10:56:40 am
Hello.. so again with some news.. I've put a lot of work on this today and I just sent my results (instructions on how to use this) to the amsn-devel mailing list...
For those of you who aren't subscribed to the mailing list, here's a copy/paste of the email...
If you don't understand something.. don't ask, it just means you're not qualified to be my tester :p

Quote

Hi,
I just 'finished' writing the first part of my SIP client.. this is useful for the audio conference that uses SIP and RTP (not the one with audio and video which uses msnp2p and is being developped on a separate branch).

So here's the deal.. I have two phases for the client, first phase is being able to make a call (send an invite) and it already works pretty good! second phase is to be able to receive a call (receive an invite) and that won't be too difficult...
I consider the first phase complete, and I like the API and the design a lot! :) What I would need now is to have you guys test it.. so here's how you can do a voice call with aMSN :

- first, you need to update to SVN r9697 or later..
- Download the latest version of farsight and compile it.. (with gst-plugins-farsight of course)
- Open the amsn console and type 'source sip.tcl' Now you have everything you need to make a voice call..
- Enable MSNP15 support and connect so you can have your authentification token...

The easy part :
- type in the console : 'createSIP' which will create a new SIP connection object called 'sip'
- then type 'inviteSIP $email' to invite someone from WLM.
You should then see status messages on the console, once it gets accepted on the other side, you'll see a status OK with the list of codecs and candidates (ip/port combinations).
For example :
Quote
Invite SIP Response : OK -- {{{} 1 {} UDP 1 192.168.1.100 123456} {{} 2 {} UDP 1 192.168.1.100 123457}} {{PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000}}

The first list is the candidate list, you take the first one ("the one for RTP has a '1' as second element, RTCP has '2') and you have your ip and port..
- Go to farsight's directory and type : ./tests/test-rtp-7 192.168.1.100 123456 1
That's it.. you should start hearing the WLM audio in your PC..

The complicated part :
In theory, you should create a new SIP connection with :
Code:
SIPConnection create %AUTO% -user $your_email -password [[$::sso GetSecurityTokenByName Voice] cget -ticket]

The returned object is your SIP connection.. it's not conneced yet.. you could do a :
Code:
$sip Invite $email $codec_list $candidate_list

The codec list is a list containing codecs.. each codec is itself a list of the form
Code:
[list $encoding_name $payload_type $bitrate]

The encoding name and payload types should be following the RTP RFCs.. to make it easy you can use PCMA which has a static payload type of 8 and PCMU with payload type 0.
The condidate list is also a list containing candidates (connection info like ip/port), each candidate is a list of the form :
Code:
[list $candidate_id $component_id $password $transport $qvalue $ip $port]

those are ICE candidates, WLM uses ICE draft 6, so the candidate id and password are randomly generated and used to authenticate once the connection is established... you don't need that.. it can work even without ICE, so you can give an empty string as candidate id and password and it will skip ice candidates... the qvalue is just a 'weight'.. the higher the value (between 0.00 and 1.00) the better it is to connect on it (ip/port priority). transport is either UDP or TCP and that's it...

So let's say we want to create an object called 'sip' it can be done like this :
Code:
SIPConnection create sip -user $your_email -password [[$::sso GetSecurityTokenByName Voice] cget -ticket] -error_handler myErrorHandler -request_handler myRequestHandler

The error and request handlers should be procs that handle errors and new requests.. you can skip the -request_handler for now (since we don't handle incoming requests for now) but the error_handler should have a prototype of the form :
Code:
proc myErrorHandler { reason }

anyways, once the object is created, it's not connected yet, so you can do it with :
Code:
sip Connect

but you don't really need to do that.. you mainly need to Register.. if you try to register and it's not connected, it will connect automatically.. so just do a :
Code:
sip Register

it will then register itself on the SIP server...
If you want to invite someone, you can also skip that and just send the invite, it will automatically connect and register before sending out the invite... So with everything I said so far, here's how you can send an invite :
Code:
sip Invite $email [list [list "PCMA" 8 8000] [list "PCMU" 0 8000]] [list [list "" 1 "" UDP 1 $my_ip $my_port]]

Since we don't use ICE, if you're behind a firewall or a NAT, then you just won't be able to make the connection work, so I was testing between WLM and aMSN inside my local network, so the $my_ip was actually my local ip.. which I was getting with :
Code:
[::abook::getDemographicField localip]

The $port is special though, since we're using a 'test' application from farsight, and that test only takes the destination ip/port, it has a hardcoded value for the port to listen to and it's 7078... so until we write our own application that uses farsight, we have to use the port 7078..
One last thing.. if you want to know what is happening (you must!) then you need to specify one last argument : a callback function with the protocol :
Code:
proc InviteCallback { status detail }

The status tells you what's happening and the detail is an extra detail per status.. for example "ERROR" status will have the reason in the detail.. a "OK" will have a list containing the remote user's candidates and codecs in the $detail variable...
So you end up doing this :
Code:
sip Invite $email [list [list "PCMA" 8 8000] [list "PCMU" 0 8000]] [list [list "" 1 "" UDP 1 [::abook::getDemographicField localip] 7078 InviteCallback

Once you do that, the invite gets sent, the other user's WLM starts ringing.. he can accept/decline/not answer/etc... the Tcl client will tell you exactly what happens.
The rest is the same as the 'easy method', you just look at what you receive in your callback, when you get the OK, you parse the remote candidates, get the ip/port and use that with farsight :
Code:
./tests/test-rtp-7 $ip $port 1


Now To summarize everything I've just explained.. let's look at the last few functions in sip.tcl which serve as test and are what are being used for the 'easy method' :
Code:

proc createSIP { {host "vp.sip.messenger.msn.com"} } {
        global sso
        set token [$sso GetSecurityTokenByName Voice]
        catch {sip destroy}
        return [SIPConnection create sip -user [::config::getKey login] -password [$token cget -ticket] -error_handler errorSIP -request_handler requestSIP -host $host]
}


This helper function just creates the 'sip' object for you just like I said before...
Code:

proc inviteSIP { email } {
        sip Invite $email [list [list "PCMA" 8 8000] [list "PCMU" 0 8000]] [list [list "" 1 "" UDP 1 [::abook::getDemographicField localip] 7078]] inviteSIPCB
}

and this one invites the user without having you type in the list of codecs/candidates everytime...

Code:
proc inviteSIPCB { status detail} {
        puts "Invite SIP Response : $status -- $detail"
}


invite callback which tells you what's happening...

Code:

proc requestSIP { request headers body } {
        puts "Received Request : $request"
}


Request handler..
This will tell you if you receive a request from the other user (assuming you are connected to the same SIP proxy but it does nothing else...)

Code:

proc errorSIP { reason } {
        puts "Error in SIP : $reason"
}


Error handler which will tell you what happened when something goes wrong...


That's about it.. if you follow these instructions, you should *in theory* be able to do a voice call with WLM... problem now is that WLM will not play your sound, and that's because farsight doesn't send RTCP and without it, WLM doesn't know if your SSRC (and id in the rtp header) is yours... with farsight 2 (which we'll use), we'll have RTCP so WLM will play our sound too... yeay!

If you have any questions.. you will find me tomorrow after I wake up on IRC... or write to the ML...

Feedback appreciated!!!
Thanks
KaKaRoTo




EDIT :
Oups... use svn revision 9698 instead..
and here's how it looks like for different scenarios... : http://pastebin.com/m52e0c270


Title: Audio/Video conversation
Post by: kakaroto on April 09, 2008, 01:31:23 am
Quote from: "from email"
Hi,
A bit more news.. I have just tested out the Audio stuff between my aMSN and Tom's aMSN and even though we're both firewalled (and obviously not in the same network) it worked! I could hear him very well..
It is currently using PCMA/PCMU (almost uncompressed audio) which uses about 10KB/s of your bandwidth.. I'll soon try to write siren gst elements so we should get less than 1.6KB/s of bandwidth with that codec...

With my latest commit (r9707) things have changed a little.. here's how you can do it :
First, you must open the console and type 'source sip.tcl' then ...
1 - sending an invite :
type : createSip; inviteSip $email
2 - receiving an invite :
One you receive the invite, you should see the invite printed in there, like this :
Received INVITE : 6935BADC0FC1B544C33B62B465969E09 - {{{}..(removed the rest of the line).. all you have to do is type :
acceptSIP 6935BADC0FC1B544C33B62B465969E09
and that's it... (notice that the 'id' 6935BADC0FC1B544C33B62B465969E09 was taken from the 'Received INVITE' line...)

Notice though that to be able to do this you will first need this :
1 - gstreamer
2 - gst-plugins-base
3 - gst-plugins-good
4 - gst-plugins-bad
5 - gst-plugins-farsight
6 - farsight2 with ./configure --disable-python --with-transmitter-plugins=rawudp
7 - go to amsn/utils/farsight and type 'make'
Make sure you have the latest version of all these components, you can try disabling some stuff if you don't need them (like for farsight2)

This will work bidirectional between two aMSNs but with WLM you can only receive and not send.. That should be fixed soon..

TODO :
1 - use of ICE for having WLM receive our stream
2 - add a Siren encoder/decode/payloader/depayloader for gstreamer
3 - clean the ugly hacky functions I used for testing...
4 - a proper configure/makefile/runtime dependency check...
5 - a proper UI

That's it! have fun!

KaKaRoTo


Title: Audio/Video conversation
Post by: kakaroto on April 10, 2008, 06:10:30 am
Hi Again!
For those who read this thread (doesn't look like many of you are...), I have some really great news!
1 - I was able to write Siren encoder. decoder. RTP payloader and depayloader elements for gstreamer...
What this means is simply that we can now use a more 'advanced' codec which uses only 1.9KB/s !!!
This also means :
2 - It looks like WLM now accepts the sound I'm sending it, it actually didn't want to play PCMA/PCMU codecs, But with the SIREN codec, it now works!
This simply means that you can now audio chat with any amsn OR WLM users without problems!

Now, the next step is what will take a bit more time, it's about cleaning the code, fixing some stuff, making it 'UI-wise' (you click accept/refuse instead of typing commands), etc... but for those who want to try it out, the previous instructions still apply..
I'll keep you posted on when/where you can find the siren gstreamer elements so you can make it use less bandwidth and work with WLM.


Title: Audio/Video conversation
Post by: rowanparker on April 10, 2008, 07:38:10 am
Quote from: "kakaroto"
For those who read this thread (doesn't look like many of you are...)


Some of us do man.
I think its very good what you're doing.
This is one of the features people want baddly.
I'm just not replying because I don't understand all of it :)


Title: Audio/Video conversation
Post by: flomar34 on April 10, 2008, 07:52:56 am
Sure, i read this post everyday, and of course ...........   i 'm not able to  understand it

I just understand that what you are doing is great and very exciting for us.  :)


Title: Audio/Video conversation
Post by: Fabioamd87 on April 10, 2008, 09:39:49 am
I read this post every time, but im still waiting for a friendly gui, because i dont wont to broken my amsn :p

anyway GOOD JOB!


Title: Audio/Video conversation
Post by: kakaroto on April 10, 2008, 04:49:50 pm
hehe, hi,
thanks for the answers :p I got tired of answering myself, lol...
And sorry if you don't understand it, it's pretty technical, but I guess more info is better than not enough info.
Anyways, it should soon be integrated with aMSN so you can use it with the UI. There isn't much work left to do :)


Title: Audio/Video conversation
Post by: rowanparker on April 10, 2008, 06:12:02 pm
One quick question: Does this have to be using version 15 of the protocol?


Title: Audio/Video conversation
Post by: kakaroto on April 10, 2008, 09:39:51 pm
for sending a request, no, for receiving one, yes...  (although you'll need to authenticate first with msnp15, then your authentication token can last a couple of days, even if you disconnect or use msnp11 afterwards)


Title: Audio/Video conversation
Post by: rowanparker on April 10, 2008, 10:09:24 pm
Quote from: "kakaroto"
for sending a request, no, for receiving one, yes...  (although you'll need to authenticate first with msnp15, then your authentication token can last a couple of days, even if you disconnect or use msnp11 afterwards)


Oh right, I see.
Because for some reason I cannot connect using msnp15 (it just timesout, no matter what I try).
And I would like to try this out (once it gets a bit simpler to do so obviously).


Title: Audio/Video conversation
Post by: kakaroto on April 11, 2008, 12:34:21 am
hummm... go to the MSNP15 thread and report the problem there, the status and protocol logs are important... thx


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on April 12, 2008, 05:12:48 pm
Hi かかろと :)
Yes, I also do read this thread... I check it (usually) at least once a week. But like most users here... most technicalities are out of my grasp especially because I'm not familiar with MSN protocol... and don't have much free time to study it :(
But I learn a bit more everytime I read your posts.

Those are great news about Siren... < 2KB/s for audio chatting. :D

BTW, thank you very much for your explanation (1st post of this page... post number 30?) on the MSN A/V features. It did clear things up for me!  :D


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 01:22:28 am
hi (don't expect to write your nick in jap),
thx all for reading :p and sorry for all the technical stuff that makes you feel even n00b-er :p hehe.. well, anyways, I'm working on integrating this into the UI and it's almost done now.

arg.. seems I forgot to finish/send this post.. anyways, now I have already finishes the integration lol...
so... starting revision 9729.. here are the new instructions :

1 - get the dependencies you need (http://amsn-project.net/wiki/Farsight)
2 - type ./configure && make
3 - run amsn and open a chat window..
4 - menu Actions->Start Audio call...

as simple as that! :)

You don't need to be using MSNP15 anymore because it now automatically authentifies you if you're not using msnp15... BUT if you don't use MSNP15 you will not be able to *receive* invites, but you can still call someone.
The current issues are that if you receive a call from WLM, he doesn't seem to work because he expects ICE connections (which will be fixed someday), you we answer him that we're temporarly unavailable and we give you the choice to call him back (if you call WLM, it works, if you get a call from WLM, it doesn't).
If you call a WLM user, you should be able to receive him, but he will not be able to receive your sound *unless you're in the same network* (it disables ICE so it can't find its external IP and gives you only the internal IP).. note though that if the WLM user is not inside a NAT, it should work.
The best compatibility now is between two aMSN users... simply because we send our external ip instead of the internal ip.
If you call someone (amsn or WLM) from the same NAT (local network), it will not work because we send the external ip...

that's the only issues I found so far.. they should *all* be fixed (apart from the 'you have to use msnp15 to receive a call') once I fix libnice and make aMSN use ICE connectivity to get through firewalls.


The configure will tell you whether it found the dependencies or not and will compile the farsight utility if it finds them.
aMSN will also check this at startup and will tell you if a runtime dependency is missing...

Here are the use cases I found and tested to be behaving correctly (use case - message amsn prints in the chat window):
1 - send an invite - Calling $user
2 - receive an invite - Received Audio Call invitation from $1
3 - accept an invite - Audio Call Accepted
4 - your invite gets accepted - User accepted your Audio Call
5 - decline an invite - Audio Call Rejected
6 - your invite gets declined - User Declined your Audio Call invitation
7 - you receive an invite while being already in a call
8 - you send an invite to someone who is already in a call - User is currently Busy
9 - You cancel an invitation - Your Audio Call has ended
10 - The invitation you received was canceled - User has canceled the Audio call Invitation
11 - You don't accept/reject a call for 1 minute - You missed an Audio Call from $user - Call Back
12 - the other contact hangs up - User has ended your Audio Call
13 - You hang up a call - Your Audio Call has ended
14 - your invitation did not get a reply in 1 minute - User did not answer your Call
15 - invite sent to an incompatible or busy client  - User is not available right now
16 - farsight 2 is not installed or crashes on init - You cannot make an Audio Call. Please install Farsight2 and try again.
17 - other client doesn't support SIP or is not using MSNP15 - User does not support Audio Calls.
18 - receive an invite from WLM with ICE connectivity which is not currently working : You have received an Audio Call invitation that is not currently supported. You may call back with a new invitation. The Audio Call will then work
19 - you try to call multiple people at the same time - You can only have one Audio Call at a time.


That's it.. have fun trying it!!!


Title: Audio/Video conversation
Post by: flomar34 on April 13, 2008, 10:23:54 am
Hello, this is a great news!

I had no problem to compile amsn and i have the menu for the audi conversation

the only problem i have is with farsight2

I tried to compile it but on the ./configre  it asked for gstreamer 0.10.17 and with Gutsy we have the 0.10.14 release :(


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 11:21:49 am
Quote from: "flomar34"

the only problem i have is with farsight2

I tried to compile it but on the ./configre  it asked for gstreamer 0.10.17 and with Gutsy we have the 0.10.14 release :(


You can either upgrade to hardy (it will be released untill the end of the month anyway), or compile it by hand (not recommended), or what I recommend is to download the 'source' package of hardy, and make a custom package for you to use it in gutsy (this is called backporting).

The main idea is to change the lines of deb-src in your sources.list file (to make it 'hardy' instead of 'gutsy'), the apt-get update to fetch the updated lists.

then you should do

apt-get -b source gstreamer0.10
and
apt-get -b source gst-plugins-base0.10

This will download these two source packages from hardy, and build them (-b) in your gutsy environment.

Of course theremay be some dependency problems (so you may have to do the same in other packages too, to upgrade them to hardy versions).

Unfortunatelly I have upgraded all my ubuntu systems to gutsy, so the only way i can test the procedure is with a gutsy live cd, but i cant find any right now.. :(


Title: Audio/Video conversation
Post by: bouriquo on April 13, 2008, 02:05:21 pm
Hello All,

I have the same problem, I can't compile farsight2.0.0.2, I have always this error :

fs-rtp-session.c: In function 'fs_rtp_session_class_init':
fs-rtp-session.c:217: erreur: 'GObjectClass' has no member named 'constructed'
make[3]: *** [libfsrtpconference_la-fs-rtp-session.lo] Erreur 1
make[3]: quittant le répertoire « /home/data/farsight2-0.0.1/gst/fsrtpconference »
make[2]: *** [all-recursive] Erreur 1

And like the farsight wiki of amsn, I installe gstream, gst-plugins-base, gst-plugin-good, gst-plugins-bad

Please can you help me :)


Title: Audio/Video conversation
Post by: Kalinda on April 13, 2008, 04:32:04 pm
Woohoo! Thanks KKRT for all your hard work :)

I, too, got as far as compiling Farsight2 when it yelled at me about not having the proper version of GStreamer. So, if anyone manages to backport the Hardy Gstreamer packages (yeah, you'll probably have to do all the plugin ones, too), could you post them for the rest of us?

I could try it, I spose; I might do that later.


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 04:37:52 pm
i can say that gstreamer can be easily backported looking at it's dependencies, but the same does not apply to the plugins package..

that package depends on a newer libc version that exists on hardy, and we would have to go all the way to backport the whole libc, and i dont know how easy that could be!

untill i find a gutsy livecd to test this, i would recommend to only backport gstreamer, then install / compile farsight2, and then maybe install the gutsy version of gstreamer again... :)

the other option would be to upgrade to hardy!


Title: Audio/Video conversation
Post by: bouriquo on April 13, 2008, 05:01:43 pm
I compile the newer glib2

but now I have a new error

odec_discovery-fs-rtp-discover-codecs.o: In function `create_codec_cap_list':
/home/data/farsight2-0.0.2/tests/rtp/../../gst/fsrtpconference/fs-rtp-discover-codecs.c:1164: undefined reference to `g_assertion_message'
codec_discovery-fs-rtp-discover-codecs.o: In function `debug_codec_cap':
/home/data/farsight2-0.0.2/tests/rtp/../../gst/fsrtpconference/fs-rtp-discover-codecs.c:131: undefined reference to `g_assertion_message_expr'
../../gst-libs/gst/farsight/.libs/libgstfarsight-0.10.so: undefined reference to `g_once_init_enter_impl'
../../gst-libs/gst/farsight/.libs/libgstfarsight-0.10.so: undefined reference to `g_once_init_leave'
collect2: ld returned 1 exit status
make[3]: *** [codec-discovery] Erreur 1
make[3]: quittant le répertoire « /home/data/farsight2-0.0.2/tests/rtp »
make[2]: *** [all-recursive] Erreur 1
make[2]: quittant le répertoire « /home/data/farsight2-0.0.2/tests »
make[1]: *** [all-recursive] Erreur 1
make[1]: quittant le répertoire « /home/data/farsight2-0.0.2 »
make: *** [all] Erreur 2


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 07:21:00 pm
Hello all! :D
yeah.. I didn't think that all you guys would need packages prebuilt for you.. all my stuff is custom built, so I never had those issues...
so.. first!
@trv : good advice :) if you could make a .deb and share it for others, maybe that would help them all out! :) I might do it later though...
@bouriquo : your first error was because you need at least glib 2.16 and you seemed to have an older version!
I don't know about your last problem but make sure you have at least glib2 version 2.16 or newer (with the -dev package too!)

EDIT: humm.. it seems your second problem is because you have both versions of the lib installed! please make sure you have only one!


Title: Audio/Video conversation
Post by: Quetzal on April 13, 2008, 07:33:40 pm
I've a problem when i've installed Farsight and all who is needed:

Quote

quetzal@helios:~/svn/amsn$ ./utils/farsight/farsight 1 2
**
** ERROR:(utils/farsight/farsight.c:273):main: assertion failed: (conference != NULL)
Abandon (core dumped)


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 08:15:08 pm
you didn't install farsight2 with --prefix /usr... read the wiki please!
you can fix this by doing this before launching amsn :
Code:
export GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10
./amsn


Title: Audio/Video conversation
Post by: flomar34 on April 13, 2008, 08:23:21 pm
Quote
@trv : good advice Smile if you could make a .deb and share it for others, maybe that would help them all out! Smile I might do it later though...


It should be great  :)

Thanks a lot for your help


Title: Audio/Video conversation
Post by: Quetzal on April 13, 2008, 08:45:43 pm
I've read the wiki but i've not see the faq in the wiki (i've had some more other problems before getting this problem and I forget to re-read the wiki for the last problem).
But all work now, thanks you ! (first i've not see this topic, i've only see that the svn had been updated and the link to the wiki)


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 08:55:17 pm
@Quetzal : ouhh ouhhh! would you be the first to have it working? :p (apart from me obviously:p)
what do you mean when you say "it all works now"... is it just farsight that works ? or you were able to make an audio call one of your contacts and it worked? :D
and I'd like to know if you see the SIREN codec in the farsight output..

@flomar34 : I just tried to make debs but as I'm in gutsy, it was crap.. I apparently need a 'lot of dependencies' that I don't have... and I don't want to update to hardy for it.. especially since it will try to update my libc6 which is not a good idea! :p
will edit the control file and try again


Title: Audio/Video conversation
Post by: Fabioamd87 on April 13, 2008, 09:08:26 pm
i'm very noob maybe... but im on hardy and I still can compile aMsn...
he tell me that gsteramer or farsight ... but i've installed the packages with apt...


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 09:11:40 pm
I'm trying to get this working in hardy...

make deb produces an error and stops.

i have installed gstreamer-plugin-farsight-siren (made a deb for hardy actually)

and i have installed farsight2 (couldn't make a deb, checkinstall refused to function correctly, so i installed the 'make install' way successfully.

now i configured amsn, found farsight, make was successfull, but make deb produces this error in the dh_shlibdeps part:

dpkg-shlibdeps: failure: no dependency information found for /usr/lib/libgstfarsight-0.10.so.0 (used by debian/amsn/usr/local/amsn/share/amsn/utils/farsight/farsight).
dh_shlibdeps: command returned error code 512

i'm looking into it as we speak..


Title: Audio/Video conversation
Post by: Fabioamd87 on April 13, 2008, 09:20:13 pm
i don't know but maybe this can help us?
http://ppa.launchpad.net/telepathy/ubuntu/pool/main/f/farsight2/


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 09:36:54 pm
that's very interesting... :)) i'll test it asap


Title: Audio/Video conversation
Post by: Quetzal on April 13, 2008, 10:00:02 pm
I can see SIREN output, but call don't really work (unfortunatly), i had not test before my last post but now, testing, I find an error in the debug log:

Quote

[22:47:06] <--SIP (207.46.112.32) SIP/2.0 400 Bad Ticket policy=PPM_?id=69264;lifetime=86400
Content-Length: 0
Via: SIP/2.0/TLS 192.168.10.1:51989;received=XXX.XXX.XXX.XXX;ms-received-port=51989;ms-received-cid=91d2b00
From: <sip:MYMSNADDRESS@HOTMAIL.COM>;tag=E548FC17DC;epid=8C7303F8D4
To: <sip:MYMSNADRESS@HOTMAIL.COM>;tag=865EE25B31BD0062DD8886823B82CC58
Call-ID: 1609ECB4E053A71FF19D294397C5BBDE
CSeq: 1 REGISTER



I've tried both msnp12 and 15.
Is it normal that "To:" indicate my msn adress and not the adress of the person that I call ?
With msnp 15, I can't see when my contact send me an invitation (and I can see the invitation with msnp12)


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 10:06:46 pm
@trv, it failed to find farsight because it installed it in /usr/local/lib instead of /usr/lib ... this error is in the 'FAQ' of the wiki/Farsight.. read that :p
@Quetzal : it's normal that it sends your email in the 'To'.. it's because you REGISTER first.. then in the INVITE, it will put the email of your contact...
I don't know why it says bad token..  could you PM me the REGISTER message in full ? (the one that you send).. thanks
also.. svn update and you'll get some more status log messages for debug..

p.s.: I have build gstreamer, gst-plugins-base and now building gst plugins good and bad... for gutsy!


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 10:15:24 pm
Quote from: "kakaroto"
@trv, it failed to find farsight because it installed it in /usr/local/lib instead of /usr/lib ... this error is in the 'FAQ' of the wiki/Farsight.. read that :p


nop, thats not the case, i have had manually installed it in /usr/lib :) the files are there.. the error is there too !

anyway, the ppa repository files worked excelent, and i compiled / made deb of amsn, and now i am running it, and ready to test it


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 10:27:33 pm
cool!! paste me the output of ./utils/farsight/farsight 1 2 (you can remove your ip from there)


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 10:38:43 pm
hmmmm thats not very good

LOCAL_CODEC: 96 AMR-WB 16000
LOCAL_CODEC: 97 AMR 8000
LOCAL_CODEC: 3 GSM 8000
LOCAL_CODEC: 98 MPA 90000
LOCAL_CODEC: 8 PCMA 8000
LOCAL_CODEC: 0 PCMU 8000
LOCAL_CODEC: 99 SIREN 16000
LOCAL_CODECS_DONE
**
** ERROR:(utils/farsight/farsight.c:370):main: assertion failed: (gst_element_set_state (pipeline, GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE)
Aborted (core dumped)


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2008, 10:41:36 pm
humm... ok, it finds SIREN codec which is good.. but it doesn't change the state.. I wonder why!
do you think you can join on #amsn IRC channel on irc.freenode.net ? or send me by PM your ip with an ssh account so I can debug this..
thanks!


Title: Audio/Video conversation
Post by: trv on April 13, 2008, 11:21:47 pm
the previous error was because of alsa misconfiguration (thx kakaroto), now all is fine!

the correct output is:

LOCAL_CODEC: 96 AMR-WB 16000
LOCAL_CODEC: 97 AMR 8000
LOCAL_CODEC: 3 GSM 8000
LOCAL_CODEC: 98 MPA 90000
LOCAL_CODEC: 8 PCMA 8000
LOCAL_CODEC: 0 PCMU 8000
LOCAL_CODEC: 99 SIREN 16000
LOCAL_CODECS_DONE
LOCAL_CANDIDATE: L0 2  UDP 0 123.456.789.100 7079
LOCAL_CANDIDATE: L0 1  UDP 0 123.456.789.100 7078
LOCAL_CANDIDATES_DONE


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 12:06:32 am
Hi all,
for all you ubuntu gutsy users.. you can now go grab the debs that I build from here : http://people.collabora.co.uk/~kakaroto/gutsy-debs/
you should install them all with :
Code:
sudo dpkg -i *.deb

then try configuring amsn again... it should detect farsight2 and compile farsight.. you can then try the audio call...

some users (kr0n1x and trv... the only ones able to compile all that so far) reported that aMSN freezes when trying to make a call.. and they weren't able to make a call.. hopefully, i'll soon find out why this happens and fix it.. for me, it all works correctly (with the debs I provided).


EDIT : The gst-plugins-farsight-siren.tar.gz is there only for those who would want the source, but it is already compiled and provided in one of the debs.. the reason I put the source for that package is because it's a modified version from the latest release (contains siren encoder/decoder/payloader/depayloader)... so you don't need to download/extract that file! just to make things clear..
Good luck.. I can't wait to hear the results of your experiments! :)

EDIT 2 : ok I found what the problem was for trv and the others would have their aMSN freezing and the call not working!!!! it's because of tcl/tk 8.5!! the tls extension is weird in 8.5 so if you use tcl/tk 8.4 it works! that's why I wasn't able to reproduce

EDIT 3 : more good news.. I fixed the bug with an ugly hack (but that should work just fine)  to workaround the bug in tcltls... now, starting revision 9736.. the freezing/not working/ bugs you have should not happen anymore!
@Quetzal, your registration/authentification error you posted before should also be fixed in the latest SVN version. others have reported the same problem (it only worked for me! lol) but I fixed it by modifying the authentication string, just try it again now!


Title: Audio/Video conversation
Post by: flomar34 on April 14, 2008, 07:27:46 am
Thanks, i hope i could test it this evening

So all i have to do now (i'm a bit long to understand sometimes :) )

1) update amsn to revision 9736

2) uninstall gstreamer and plugins from gustsy

3) install your debs

4) compil fasight2 with then option -prefix=/usr

5) Enjoy it

Am i right ?


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 07:52:52 am
no... 1-2-3 are right.. but the 4th no.. you don't need to compile farsight2, it's already compiled and available in those .deb files ... everything you need is there... apart from amsn...
so :
1) update amsn to latest svn
2) install all the .deb files in that link (the old ones will get removed automatically by the newer version)
3) cd amsn && ./configure && make
4) enjoy


Title: Audio/Video conversation
Post by: Quetzal on April 14, 2008, 08:00:39 am
Quote from: "kakaroto"

@Quetzal, your registration/authentification error you posted before should also be fixed in the latest SVN version. others have reported the same problem (it only worked for me! lol) but I fixed it by modifying the authentication string, just try it again now!


I look fixed but I can't test now (not in my home, and nobody connected in my contactlist)
You are too fast for me !! (you have fix the bug before I can send you the debug log ^^)


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 08:13:25 am
lol.. sorry about that :p it just said bad ticket, and a pure coincidence yesterday I was reverse engineering some more for the ICE stuff and I realized that the username/password it sends are not your real username and password.. it's actually :
username : "msmsgs"
password : "PRT_" + the password...
don't ask me why.. but for me it worked with username password.. then with kr1n0x he had the bad ticket problem and I thought I might try the 'msmsgs' username instead.. tried it, worked for me, worked for him.. so that's it! hehe :)
anyways..
a bit more news... in the latest SVN version, I just committed some code that uses STUN over TCP/TLS to authenticate to the relay server (TURN) with a shared secret request/response algorithm.. and it works, so I get a username/password/ip to use for an Allocate request/response on the TURN server (over udp this time). This will allow people who can't connect between them because of a firewall and a network topography too complex for STUN to work (right now STUN is enough to pass through most firewalls, but maybe not huge corporate setups with NAT inside NAT inside NAT and proxies and stuff)... so with this (kind of like the reflector for the webcam where you can do webcam even if both are firewalled), we should be able to work around those kind of network topographies!
Note though that this is not in use yet, it is just a preliminary work for later when we will have ICE introduced... once ICE works, *then* we'll use this piece of code to make aMSN use the TURN server...
Don't expect this code to be used until maybe 2 weeks from now.


Title: Audio/Video conversation
Post by: kr0n1x on April 14, 2008, 08:53:15 am
tonight... 3.30 AM in italy...i heard kakaroto's sexy voice  :oops:  :D

congratz man :D

edit: i'm in my home now, i downgraded tcl/tk to 8.4 and tried a call with a WLM user (she has Windows XP). It works too :)


Title: Audio/Video conversation
Post by: trv on April 14, 2008, 12:05:51 pm
i don't know if its a sideeffect of the tcltls bug that was fixed last night, but i can't login anymore with msnp15, only 11. using 15 it just hangs/freezes at the login, for ever!

edit: hmm ok now it connected with 15, don't know why it froze before..

edit2: i must say i'm experiencing sever 'freezes for ever' but i cannto identify when this happens, it seems to be random.

One thing i noticed is that when it stucks like that, the cpu is 100% becaulse of farsight2 executable running with some localhost arguments.. I kill amsn and must kill farsight too, and then launch amsn again.

this is a ps output

trv       2646 89.8  0.6  83832  6772 ?        Rl   01:31 881:00 ./utils/farsight/farsight user@localhost remote@remotehost

Now i was able to log in with msnp15, after a minute or more frozen (probably because of 100% cpu usage by farsight, tcl is slower.....)


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 03:54:08 pm
hummm... trv, you have the latest SVN ?
vivia just downloaded my .deb files, installed, compiled amsn and ran it and we had a voice call, no problems...
I did notice the 100% cpu for farsight, but it only happened when we had the freezes in amsn with the bug in tls...
I'm wondering if you did update amsn...
anyways, when you say 'random freezes'.. is it when you do a voice call or just when chatting normally ?
afaik, none of my changes should affect anything outside the voice call feature...


Title: Audio/Video conversation
Post by: trv on April 14, 2008, 04:05:16 pm
xm i have the latest svn yes, and the debs for hardy from ppa.

i'll test more in a while.

The freezes are only on the login stage, not everywhere, and i have yet to test if voice conversation actually works :)


Title: Audio/Video conversation
Post by: Quetzal on April 14, 2008, 05:22:57 pm
All seem work but the other person listen nothing.
It's very strange because sound device is busy while farsight run and farsight seem running correctly.
But I don't use .debs, I use my own farsight with tcl/tk 8.5 on ubuntu 8.04...


Title: Audio/Video conversation
Post by: flomar34 on April 14, 2008, 06:23:19 pm
Hello,

We have no problem with the installation using your debs.

I can send an audio call invitation.

MSN user can accept it but he didn't hear anything.

Is there something to verify (like a log file) (With simply words please i'm very new with this :) )

Thanks


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 06:37:13 pm
ahhhh, why is everyone asking the same thing!
someone just asked me over MSN and now.. 3 people asking the same thing here...
let me make it clear to you (you should all read and understood this very already  with all the details I gave before) :
Quote

The current issues are that if you receive a call from WLM, he doesn't seem to work because he expects ICE connections (which will be fixed someday), we answer him that we're temporarly unavailable and we give you the choice to call him back (if you call WLM, it works, if you get a call from WLM, it doesn't).
If you call a WLM user, you should be able to receive him, but he will not be able to receive your sound *unless you're in the same network* (it disables ICE so it can't find its external IP and gives you only the internal IP).. note though that if the WLM user is not inside a NAT, it should work.
The best compatibility now is between two aMSN users... simply because we send our external ip instead of the internal ip.
If you call someone (amsn or WLM) from the same NAT (local network), it will not work because we send the external ip...

so just to make it clear again...

if a user on WLM can't hear you, it's because WLM is stupid.. a WLM user can only hear you if he's not in a NAT (connected directly to internet) but it should work correctly between two amsn users.
And to fix this issue, I'll need to fix this 'ICE' thing I've been talking about from the start that noone understood..

and to simplify even more :
1 - call an aMSN user if you want to both hear each other
2 - call a WLM user if you want to hear him, but he won't hear you


EDIT oh and btw, congrats on quetzal and flomar for making it work! :) now everyone seems to have it working... apart from trv but I'm sure it's because he did something wrong :p


Title: Audio/Video conversation
Post by: flomar34 on April 14, 2008, 06:49:54 pm
Quote
and to simplify even more :
1 - call an aMSN user if you want to both hear each other
2 - call a WLM user if you want to hear him, but he won't hear you


This is clear enought for me now :)

Thanks a lot for the job you made i will continue to follow this post every day because i'm sure  that  ...

[FRENCH] ...je suis sur que quelque chose d'important se passe ici [/FRENCH]


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 07:05:19 pm
hehe, I really had to dumb it down :p
and yes, something important is happening in here! :)
for now, everyone who tried it, worked for them (apart from trv but he didn't test the latest svn yet).
What you'll want to be looking for next in this thread is whether or not I finished implementing ICE ... which will allow you to call a WLM user and both hear each other and have it work if you're in the same network or on different networks, firewalled, etc...


Title: Audio/Video conversation
Post by: Quetzal on April 14, 2008, 10:46:08 pm
Quote from: "kakaroto"

The current issues are that if you receive a call from WLM, he doesn't seem to work because he expects ICE connections (which will be fixed someday), we answer him that we're temporarly unavailable and we give you the choice to call him back (if you call WLM, it works, if you get a call from WLM, it doesn't).
If you call a WLM user, you should be able to receive him, but he will not be able to receive your sound *unless you're in the same network* (it disables ICE so it can't find its external IP and gives you only the internal IP).. note though that if the WLM user is not inside a NAT, it should work.
The best compatibility now is between two aMSN users... simply because we send our external ip instead of the internal ip.
If you call someone (amsn or WLM) from the same NAT (local network), it will not work because we send the external ip...


Rhaaaaa, it seem me that i've read this before, but I was unable to find it and I know that I make a mistake but I post it (traduction in french (parce que je ne suis pas sur d'avoir bien dit ce que je voulais: il me semblait bien que j'avais vu ca qque part mais comme je n'arrivais pas a le retrouver j'ai posté quand meme en sachant que je fesais une betise ^^) bon je ne posterai plus en francais ici promis ^^)


Title: Audio/Video conversation
Post by: kakaroto on April 14, 2008, 11:19:03 pm
lol, ok, c pas grave! :p

btw, I think I found a solution for making it work with WLM without ICE :p


Title: Audio/Video conversation
Post by: rowanparker on April 15, 2008, 08:26:49 am
Quote from: "kakaroto"
btw, I think I found a solution for making it work with WLM without ICE :p


Go on, try us out ;)
We might understand :D


Title: Audio/Video conversation
Post by: kakaroto on April 15, 2008, 10:47:38 am
LOL.. you might.. but you still won't :p ok maybe you can, assuming you know a bit about stun/ice/turn (you can read the rfcs).. basically, the current problem is that we have two types of 'invite', one which uses ICE and the other which doesn't...
if we don't use ICE (raw udp), it means we just use a normal UDP socket an we send/receive with that.. we use STUN to allow the traversal of the NAT by mapping our internally allocated port to an external port opened by the router for our UDP 'answers' to reach us.. (it thinks we sent a message and we expect a response, so it forwards the response to us, while in fact we are receiving the actual data.. that's what stun is used for, you send an outgoing message, the router remembers your internal ip/port and when it receives an answer on that port, it redirects it to you thinking it's an answer.. so you send a STUN (udp) message, and wait for the reply).
With raw udp, we can only send/receive to one ip/port.. with ICE you have multiple 'candidates' (which are just ip/port combinations)... so if you're both in separate locations, you send your external ip/port, and it works, but if you're in the same local network, it doesn't work because you only have the external ip, and the routers usually don't allow a packet going out and coming back in.
The thing is that if we use ICE, we will send multiple candidates, like candidate 1 : internal ip/port, candidate 2 : external ip/port, candidate 3 : a relay server in case internal AND external failed to work (rare). That relay server which is used only when both users are in symmetric NATs where the router is smart enough (5% of the cases?) to blablabla read STUN and NAT traversal to know why it would block those packets :p
anyways... the thing is that during the negociation you give it these 3 (or more depending on how many internal interfaces you have) ip/port combinations... and ICE will try to connect to each one (depending on priority, local network has higher priority than going through the external network) and it will fallback on the relay server (called a TURN server). There's also an interesting fact, when you send your ICE candidates during the negociation, you also send a single 'ip/port' (usually also found in the ICE candidates) for those clients who do not support ICE so they can fallback on it.. and MSN uses the TURN server as the fallback.. and the reason is... I just found out about it in the TURN draft 2 that M$ might be implementing :
TURN specs say that you must request from the TURN server an 'allocate' which simply means you tell it to allocate an ip/port for you on the relay server, it will give you that ip/port and you use that as you ICE candidate... what happens is that the first person to send a packet to that ip/port will be remembered by the server and it will send any data it receives from it back to you (it remembers your ip/port when send the request to allocate the resource)... and in the same way, whenever you send something to it, it will send it back to that person that connected to it...
This means that what we could do is that for WLM users, we could 'accept' the ICE request instead of rejecting it, then just use the relay server which will guarantee that it will work.. problem is, not sure about the bandwidth... maybe the sound will be a bit less 'smooth'... but anyways, at 2KB/s I don't think it will be much of a problem! :)
so that's my idea.. to summarize it.. always use the TURN (relay) server when talking to WLM so it can always get our data :)

oh.. and by the way, if you're wondering why we can't just use the ICE candidates we received, since we receive all 3 ip/port addresses (internal, external, relay).. it's simply because if we use the internal or external ip, then it's not TURN, it's just normal ICE, and that needs authentification.. you see an ice candidate is not only an ip and a port, it's actually a username (random), a password (random), a media type (rtp or rtcp), a transport (udp or tcp), a priority, then the ip and port.. that's for each candidate (so you get 6 (rtp+rtcp) candidates for your internal/external/relay and each one has a different username and password).
ICE is made to receive a lot of candidates (all internal, external (server reflexive and peer reflexive)) and only choose one as the 'selected pair'.. so since it's UDP, you have no way of knowing if the packet arrived or not, so it uses STUN to authenticate and make sure we are talking to the righ person (it doesn't want you to send your video/sound to someone other than the person you want to send to) so it sends a STUN request with *our* username and an sha1 hmac hash using password, so we know that it's the right person sending the request, and we answer it likewise (using their username/password), once and only once it receives an authenticated response to the STUN request, will it then 'elect' that candidate pair (our candidate + the remote's candidate) and start streaming to it... since we don't reply to the STUN message, it thinks that ip/port combination gets bumped on a firewall, so that candidate (or that ip/port) is useless..
it will only elect one candidate pair because if you have many candidates (in my case, I get 12 candidates), you don't want to send the same data to all of them (12 times the bandwidth usage)...
so we either have to answer to the STUN request, or use the TURN server which doesn't need STUN (I think)....
and why don't we just answer the STUN request instead of all that? simple.. we use farsight, we use gstreamer, we use a raw udp transmitter.. first, that raw udp transmitter can only send to one ip/port, so we can't just give it all the candidates we received, and assume it will work... also, because we have no access to the data unless we start playing with the gstreamer pipeline.. and I don't want to do that!!! and also because a 'ice transmitter' will very soon be ready which will take care of all that for us!!!

I hoe this explanation was good enough, I tried to make it as detailed and simple as possible :p


Title: Audio/Video conversation
Post by: rowanparker on April 15, 2008, 03:59:40 pm
Thank you for that explanation.
It made quite a lot of sense actually :)
I wish you luck with what you are doing.
Can't wait for it to be a proper feature. :D


Title: Whaoo !!
Post by: tofg on April 16, 2008, 11:10:40 am
I'm really astonished by your work.
But as i'm a noob on debian, i can't feel it to do it myself.
I hope it will be implement on the standard version soon.
And as i'm french, i will just say "Bravo"


Title: Audio/Video conversation
Post by: lucianolev on April 16, 2008, 03:13:41 pm
I've been reading this thread beginning (this is indeed not my first post) and I'm very impressed by the speed you manage to solve every obstacle in such a complicated job. Providing new support for compatibility of a closed source application (and made by Microsoft!!) like WLM is a very difficult task as far as i'm concerned, so congratulations! Thanks again for your work.

Hope to see this implemented nicely for non-technical people in next stable version.  :D


Title: Audio/Video conversation
Post by: kakaroto on April 16, 2008, 05:50:33 pm
Hi all,
@rowanparker : glad you understood it, I tried my best to make it easy :p
@tofg: thanks for taking the time to register jus to give me that "Bravo".. so.. "Merci" :)
@lucianolev : thanks for the nice comments :) It's a bit complicated, yes (as you might understand from all the technical details I gave), but honestly, it wasn't that difficult simply because WLM is using all standard protocols for this, SIP/SDP/RTP/ICE/TURN/STUN.. all those are standard and the specs are available on the net.. and there is no difference between the official specs and what Microsoft does, so this is very helpful! The difficult part would be to understand all that, but I'm ok since I already knew pretty much all of that because that's my field of work so I already knew all those standards from work so I just had to implement it.


Title: Audio/Video conversation
Post by: Auria on April 16, 2008, 07:48:17 pm
Quote from: "kakaroto"
Hi all,
@rowanparker : glad you understood it, I tried my best to make it easy :p
@tofg: thanks for taking the time to register jus to give me that "Bravo".. so.. "Merci" :)
@lucianolev : thanks for the nice comments :) It's a bit complicated, yes (as you might understand from all the technical details I gave), but honestly, it wasn't that difficult simply because WLM is using all standard protocols for this, SIP/SDP/RTP/ICE/TURN/STUN.. all those are standard and the specs are available on the net.. and there is no difference between the official specs and what Microsoft does, so this is very helpful! The difficult part would be to understand all that, but I'm ok since I already knew pretty much all of that because that's my field of work so I already knew all those standards from work so I just had to implement it.


microsoft using standards? I'm impressed  :lol:


Title: Audio/Video conversation
Post by: H@t Trick on April 17, 2008, 03:32:19 am
Wow, I sure missed a lot with a mini vacation and some family issues, great work  kakaroto! Are all the needed libraries compiled and included for Windows on the SVN? If so I will try and test this for you to let you know about Windows compatibility.


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2008, 06:06:41 am
hehe, welcome back H@t Trick. Yeah, I've advanced quite a bit in this thing! (final exam in a few days, so I had to do something to take it off my mind :p).. About windows, I have just commited the binaries for it, so in theory, as soon as you upgrade the SVN, you should be able to make a call.. but farsight was never tested on windows, and while I quickly tested it, I found a bug, something about glib recusive mutexes not working correctly on windows, so we get a 'freeze' (farsight 2 tries to take a lock/mutex that was already taken but never released, it was ok on linux since it's a recursive lock which means you can lock it multiple times from the same thread, but on windows it doesn't seem to work). End result, you will not be able to hear the other person but I think it would work fine for sending only... hopefully, I'll find and fix this bug sometime next week, in the meantime, use it as is.
Hopefully, farsight 0.0.3 will soon be finalized and we can move to that and hope it doesn't have that bug anymore... (many API changes in the current git trunk, so we have to port the code I wrote that uses farsight 2 to the new API).
Hopefully, tomorrow or in a few days, mac users will also get their binaries in SVN and everyone can enjoy the feature! :)


Title: Audio/Video conversation
Post by: bouriquo on April 17, 2008, 06:37:40 am
Hi kakaroto,

Ok I have just compile glib2.16.3 but where I can find the glib2-dev ?

Thanks

Quote from: "kakaroto"
Hello all! :D
yeah.. I didn't think that all you guys would need packages prebuilt for you.. all my stuff is custom built, so I never had those issues...
so.. first!
@trv : good advice :) if you could make a .deb and share it for others, maybe that would help them all out! :) I might do it later though...
@bouriquo : your first error was because you need at least glib 2.16 and you seemed to have an older version!
I don't know about your last problem but make sure you have at least glib2 version 2.16 or newer (with the -dev package too!)

EDIT: humm.. it seems your second problem is because you have both versions of the lib installed! please make sure you have only one!


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2008, 07:00:50 am
Hi,
if you follow the link I posted previously (also on the first post), you'll find the link to debs for all the dependencies, I think you don't need glib2-dev to get it working...
btw, if you compiled glib yourself, then it's already a "dev" package.. (dev just needs it contains the include files, necessary for compiling other programs who depend on it, if you install it from the source, it will automatically install the include files with it).


Title: Audio/Video conversation
Post by: De Baimbo on April 17, 2008, 05:57:49 pm
Hello everyone, just dropping by to show my appreciation to kakaroto and the AMAZING job he's doing.
If you manage to enable audio-video conversation in AMSN thousands of people will thank you for ever.
I'm following the evolution of this thread since the beginning. I'm compiling the new realases once a week, it's unbelievable how fast you program and fix bugs.
Unfortunately I can't test this new feature because I use linux Fedora, so I would need .rpm packages instead of .deb ones but don't worry 'cause a friend of mine is creating them and when they're ready I'll share them with all the Fedora users here (hehe is there any?).

Kakaroto, you rule! keep up the good work!

cheers from italy


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2008, 08:01:12 pm
Hi,
thanks for your appreciation!
About linux fedora rpms, we have the fedora maintainer on board with aMSN and he's working on providing the rpms for you, it should be ready soon!


Title: Audio/Video conversation
Post by: PjhN on April 17, 2008, 10:52:36 pm
Hey there! This sounds like a really great feature! I know it's still very much in beta but I thought you'd rather know it's unfortunately not working at all over here.

since I last updated aMSN to revision 9742 (running on Vista Ultimate x64), as soon as I load aMSN I am presented with an error box telling me "farsight2.exe has stopped working". The status log has this error: message:
 
Quote
[22:43:03] Farsight : Pipe is now readable
[22:43:03] Farsight answering :
[22:43:03] Farsight : Pipe is now readable
[22:43:03] Farsight : got eof
[22:43:03] Farsight : Closed
[22:43:03] Closed pipe : **
** ERROR:(..\..\farsight.c:382):main: assertion failed: (gst_element_set_state (pipeline, GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE)

This application has requested the Runtime to terminate it in an unusual way.
Please contact the application's support team for more information.
The same happens if I try to use it after aMSN has started. I've tried switching both farsight.exe and aMSN.exe into XP SP2 Compatibility Mode and adding them both to the DEP whitelist too, with no success. It's not bothering me although I'd of course like to play, but I thought I'd let you know.

Thanks for all your hard work!


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2008, 11:34:41 pm
Hi,
Thanks PjhN for your feedback. I wonder why it fails to start, I have no idea, but I'll test it monday when I'll be at work (we have a vista over there). I'm guessing it might be because of some binary incompatibility or a CRT incompatibility... I'll let you know what I find... I only tried it on windows XP...
There's also currently a bug in farsight 2 that makes it not work for sending and receiving... but I have a patch to make it send (but not receive)... monday (after my exams), I'll track down the bug and fix it and provide you with new binaries that will hopefully work correctly on windows...
the error box is because I'm using g_assert to exit in case there's a problem, on linux, it's ok, amsn just gets notified that farsight closed and disabled audio conversation support.. on windows, stupid windows shows an error box... the reason why you get that on startup is because aMSN tries to test/detect whether farsight is correctly installed in order to enable the audio conferencing 'flag' that allows others to send you an invite. I'll fix that by making it just exit normally without having the error box...
so.. windows users, please be patient!


Title: Audio/Video conversation
Post by: kakaroto on April 18, 2008, 02:20:12 am
(I should edit, but I send new post so those who watch this topic get a new notification).
ok, latest SVN version (r9748) is now fixed for windows.. I tracked down the bug and fixed it in farsight 2 and recompiled the binaries.. also made a few fixes to the farsight utility so it works better with the windows gstreamer elements.. so now my friends, it should work fine on windows just as it does on linux ! :D
still remaining : remove the g_assert stuff to avoid the windows error_box and look at whether it runs on vista or not.


Title: Audio/Video conversation
Post by: luismanson on April 18, 2008, 02:28:48 am
Hi i just discoered this under SVN, i can not send any invites, but i have this info:

luis@Vger ~/variado/downloads/amsnsvn/amsn $ ./utils/farsight/farsight 1 2
Error while creating new session (0): Could not create GstRtpBin**
** ERROR:(utils/farsight/farsight.c:283):main: assertion failed: (0)
Aborted


Title: Audio/Video conversation
Post by: kakaroto on April 18, 2008, 03:15:50 am
basically, it means you didn't install gst-plugins-bad... make sure you read http://amsn-project.net/wiki/Farsight and install all the dependencies!


Title: Audio/Video conversation
Post by: Fenix-TX on April 18, 2008, 04:20:29 pm
Hi! I'm on hardy heron, i've used this repository deb http://ppa.launchpad.net/telepathy/ubuntu hardy main universe and i've installed gstreamer packages (there are farsight package for gstreamer too) and farsight package. I've compiled amsn, and i've executed  amsn/utils/farsight/farsight and i have this output:

Code:

./farsight 1 2

(farsight:29414): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstAlsaSrc' has no property named `\x83\xc4\u0010[]Ð\x90\x90\x90\x90\x90\x90\x90\x90\x90\x90\x90U\x89\xe5S\x83\xec\u0010\x8dE\u0010\x89E\xf8\x89D$\u0008\x8bE\u000c\xe8G\xee\xfc\xff\x81ë;\u0010'
LOCAL_CODEC: 96 AMR-WB 16000
LOCAL_CODEC: 97 AMR 8000
LOCAL_CODEC: 3 GSM 8000
LOCAL_CODEC: 98 MPA 90000
LOCAL_CODEC: 8 PCMA 8000
LOCAL_CODEC: 0 PCMU 8000
LOCAL_CODECS_DONE
LOCAL_CANDIDATE: L0 2  UDP 0 *.*.*.* 62277
LOCAL_CANDIDATE: L0 1  UDP 0 *.*.*.* 62276
LOCAL_CANDIDATES_DONE

(I've changed my ip with *)

And it never finishes the program.

Perhaps i need to compile your http://people.collabora.co.uk/~kakaroto/gst-plugins-farsight-siren.tar.gz package?


Title: Audio/Video conversation
Post by: trv on April 18, 2008, 07:02:31 pm
yes you must compile that package.

just extract it, run ./configure and then run checkinstall

it will take care of everythinh and produce you a nice .deb package for your system, then install that with dpkg -i package.deb


Title: Audio/Video conversation
Post by: kakaroto on April 18, 2008, 07:38:30 pm
yeah you'll need that package for the SIREN codec. also, that warning is weird, I also got that, but I have no idea what's doing that.. anyways, it's not an issue it doesn't affect the execution of the thing... and it's normal the program never finishes, it's waiting for you to give the remote candidates and codecs... just type EXIT at the end.
once you have SIREN listed in your codec list from farsight, you should be able to do an audio session and it should just work.


Title: Audio/Video conversation
Post by: Fenix-TX on April 18, 2008, 09:58:54 pm
Ok now i have siren, i'll try to talk with someone. Thanks.


Title: Audio/Video conversation
Post by: luismanson on April 18, 2008, 11:51:55 pm
ok, i have al the deps now, readed all the thread and i still have no luck, i have a new challenge for you ppl :P

amsn $ ./utils/farsight/farsight 1 2
**
** ERROR:(utils/farsight/farsight.c:303):main: assertion failed: (src != NULL)
Aborted


Title: Audio/Video conversation
Post by: kakaroto on April 19, 2008, 12:34:52 am
humm.. this means that you don't have an audio source, in other words, you don't have alsasrc or osssrc, did you install gst-plugins-base and gst-plugins-good correctly ?
type 'gst-inspect alsasrc' it should show you some info about alsasrc (which should be in /usr/lib/gstreamer-0.10/libgstalsa.so) or 'gst-inspect osssrc' which should give you info about osssrc also.. the file should be in /usr/lib/gstreamer-0.10/libgstossaudio.so


Title: Audio/Video conversation
Post by: H@t Trick on April 19, 2008, 08:42:43 am
Quote from: "kakaroto"
hehe, welcome back H@t Trick. Yeah, I've advanced quite a bit in this thing! (final exam in a few days, so I had to do something to take it off my mind :p).. About windows, I have just commited the binaries for it, so in theory, as soon as you upgrade the SVN, you should be able to make a call.. but farsight was never tested on windows, and while I quickly tested it, I found a bug, something about glib recusive mutexes not working correctly on windows, so we get a 'freeze' (farsight 2 tries to take a lock/mutex that was already taken but never released, it was ok on linux since it's a recursive lock which means you can lock it multiple times from the same thread, but on windows it doesn't seem to work). End result, you will not be able to hear the other person but I think it would work fine for sending only... hopefully, I'll find and fix this bug sometime next week, in the meantime, use it as is.
Hopefully, farsight 0.0.3 will soon be finalized and we can move to that and hope it doesn't have that bug anymore... (many API changes in the current git trunk, so we have to port the code I wrote that uses farsight 2 to the new API).
Hopefully, tomorrow or in a few days, mac users will also get their binaries in SVN and everyone can enjoy the feature! :)


Thanks, it's good to be back! You sure have done some great work! I will be sure to update my svn for the 2nd time this week to test this out, I almost did tonight before I read this but I didnt get a chance to test the feature out, I will make a point to try it.

keep it up kakaroto!

<off-topic>man I can't believe I am approaching 200 posts in just under a year, wow! That means I made the switch to aMSN about a year ago, time sure flies eh?</off-topic>


Title: Audio/Video conversation
Post by: kakaroto on April 19, 2008, 10:07:18 am
hehe, ok cool, let me know the results, as i've been the only one testing on windows (apart from someone else on vista where it didn't work, i'll have a look at it monday).
<offtopic>and yeah, 200 posts is near.. you registered the 24th of may, so it's been almost a year (since you registered at least)... for me in 2.5 years, I'm way over 5000... man, I need to do something with my life.. lol :p</offtopic>


Title: Audio/Video conversation
Post by: luismanson on April 20, 2008, 07:16:11 am
Quote from: "kakaroto"
humm.. this means that you don't have an audio source, in other words, you don't have alsasrc or osssrc, did you install gst-plugins-base and gst-plugins-good correctly ?
type 'gst-inspect alsasrc' it should show you some info about alsasrc (which should be in /usr/lib/gstreamer-0.10/libgstalsa.so) or 'gst-inspect osssrc' which should give you info about osssrc also.. the file should be in /usr/lib/gstreamer-0.10/libgstossaudio.so


weird, on gentoo there are separate packages :|

my new present is:
Code:


luis@Vger ~ $ /usr/share/amsn/utils/farsight/farsight 1 2
LOCAL_CODEC: 96 MPA 90000
LOCAL_CODEC: 8 PCMA 8000
LOCAL_CODEC: 0 PCMU 8000
LOCAL_CODEC: 97 SIREN 16000
LOCAL_CODECS_DONE
**
** ERROR:(utils/farsight/farsight.c:370):main: assertion failed: (gst_element_set_state (pipeline, GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE)
Aborted


...i probably should get a life too :P


Title: Audio/Video conversation
Post by: billiob on April 20, 2008, 09:10:20 am
If you're using oss, remove the following lines from farsight.c:
Code:
 if (src == NULL)
     src = gst_element_factory_make ("alsasrc", NULL);


Title: Audio/Video conversation
Post by: kakaroto on April 20, 2008, 07:23:51 pm
right... I'll have to fix that of course, riht now it checks whether or not you have the element, but not if it works...


Title: Audio/Video conversation
Post by: luismanson on April 20, 2008, 08:37:21 pm
i do use alsa :S


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 12:44:33 am
then maybe you don't have a card named '0' or whatever it is by default... try launching it with  an export GST_DEBUG=3
you can also see the default card name used by looking at the output of 'gst-inspect alsasrc'


Title: Audio/Video conversation
Post by: luismanson on April 21, 2008, 02:45:25 am
I dont see anything wrong:

(lots of output striped)
Code:

  device              : ALSA device, as defined in an asound configuration file
                        flags: readable, writable
                        String. Default: "default" Current: "default"
  device-name         : Human-readable name of the sound device
                        flags: readable
                        String. Default: "" Current: null


i could try OSS, but i dont know how, is there any setup GUI or config file?


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 05:57:54 am
just like what billiob told you.. it will use oss then.


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 07:44:48 pm
Hi, pjhn, I just tested it on vista and it works, so I don't know what your problem is.. can you retry with the latest SVN please?
Also, if it still doesn't work, can you run cmd.exe and type :
Code:
cd utils\windows\gstreamer
set GST_PLUGIN_PATH=.
set FS_PLUGIN_PATH=.
set GST_DEBUG=3
farsight.exe x y 2>log

wait until it crashes then send me the log file...


Title: Audio/Video conversation
Post by: PjhN on April 21, 2008, 08:23:43 pm
@kakaroto:  Well I followed your instructions and PMed you the logfile, hope it helps in general. However...

I was just playing with the compatibility settings, and realised I'd forgotten to try running aMSN with admin privileges (had only tried farsight.exe). It turns out that if I run aMSN elevated, then no error message comes up, therefore hopefully all's well.

To clarify, that's ONLY aMSN set to 'Run as Admin', not farsight.exe, and neither set to 'XP SP2' (or any other) Compatibility mode.

Hopefully, since it's not crashing on load it'll work fine in conversation, but there's no one in my friends list online and using aMSN right now, so I can't actually test whether it works. Will do ASAP, and get back to you. Thanks for looking into this, hope it hasn't wasted your time.


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 09:14:30 pm
Hi,
thanks for the fast answer..
first, if you get no error, maybe it means it didn't crash, which is a good thing.. best way to check would be to actually do what I told you to do above, from your log, it looks like you didn't do it correctly, or more like there's something wrong in there, not sure what! It can't find fsrtpconference (which should be in libfsrtpconference.dll in that directory).. no idea why it wouldn't find it... anyways, it did find it before (your first message in page 6 or 7) so it's probably a different issue, or you did something wrong...
I'm not sure about the error box if it's an 'elevated bullshit' thingy or not, but for sure vista is crap :p
About 64bits, I know the binaries are for x86 and not sure if it will work on 64bits.. hopefully windows is able to run compatible 32bit binaries without the need of a 64bits version..
anyways, we'll see when you do a conf and we'll see how it goes.. in theory, you could just open amsn twice and do a call from one account to the other (assuming one of them is configured to use MSNP15) or you can do a call from your amsn to windows live messenger from the same PC... doing like that would make it work at least one way. if aMSN says 'calling ...' then farsight worked, otherwise it will tell you that you need to install farsight2.


Title: Audio/Video conversation
Post by: PjhN on April 21, 2008, 09:38:48 pm
Ooops, I think I missed those decimal points after the 'PATH=' lines, sorry, so tried again and have PMed new log (much longer now!!).

Just tried aMSN -> WLM 8.5; aMSN announced it was 'Calling' WLM, and WLM allowed me to pick-up, so I take it it's working? I couldn't hear anything coming down the line though... would you expect me to be able to receive audio in WLM? If a successful connection is all that's needed to prove it works, then all's good. I'll have to figure out how to log-in to two copies of aMSN on the same computer before I can test otherwise.

I did find that, after I tried the call, aMSN started to interfere with my audio (Winamp output), making it hitch a lot and sound 'staticy'.

EDIT: Although I can't hear anything (i.e. WLM isn't outputting the sound I'm inputting via aMSN), it does look like the volume indicator in WLM is reacting to my voice (with a slight delay), so I'm hopeful that means it will work aMSN -> aMSN!


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 09:52:40 pm
Hi,
so yes, it works! :)
When you do a call, aMSN will launch farsight, and it will wait for it to finish detecting all your codecs and your local candidates and start the pipeline, once it's done, it will make the call.. if it's not able to do so, then it will tell you that farsight needs to be installed.. so this means that it all went well, cool :)
now.. you should be able to hear sound from one side at least.. as explained before, WLM sends its internal ip, so if you're on two separate networks, you won't be able to send to WLM because you don't know his IP, but you should be able to receive from WLM since it knows your IP... if you're on the same network, then you should be able to send to WLM since you know it's internal ip, but WLM won't send to you because it only knows the external and not the internal ip... either way, one side should work...
it's not a good idea to keep winamp playing when testing because you might not hear the sound (you don't know at what volume you receive the sound)... easy way to test is putting the volume high and yelling/scratching the microphone on the other end  :p


Title: Audio/Video conversation
Post by: PjhN on April 21, 2008, 09:59:03 pm
Hey. Should have been clearer about Winamp; I stopped playing for testing, only resuming playback after I'd ended the call, but it was THEN that sound quality was reduced. Closing aMSN immediately fixed it.

Can I run two versions of aMSN from the same install, at the same time? I tried, since I have two users I can log into MSN with, but when I do this they each tell me 'User does not support Audio Calls'. So how do I run two versions of aMSN on one PC, to test? Would a copy -> paste of the install be sufficient?


Title: Audio/Video conversation
Post by: kakaroto on April 21, 2008, 10:28:15 pm
Hi,
yes, you can just launch as many aMSNs as you want and they will all work, the problem with 'user does not support audio calls' is as I explained before, you need to be logged in with MSNP15 to be able to 'receive' calls.. it's because you have a 'client id' which contains 'flags' of what your capabilities are (like, if you support winks or if you have a webcam, etc... in this case if you support audio calls).. in order to receive a call you need to be using MSNP15.. you can read more about MSNP15 here : http://www.amsn-project.net/forums/viewtopic.php?t=4666
to enable it, just do ctrl-shift-C from amsn's main window, in the window that pops up, type :
Code:
::config::setKey protocol 15

then reconnect... note that this is experimental and you will not be able to add/remove/rename/copy/move contacts or groups if you use MSNP15.. everything else should work just fine...


Title: Audio/Video conversation
Post by: H@t Trick on April 22, 2008, 06:21:48 am
Well I just did a little test, as I have mentioned before I run 4 msn accounts, 3 with default protocol and one with MSNP15,  I tried to start an audio call from one account to my MSNP15 account and initially I got "User does not support Audio Calls.", I added myself as a contact on each list, had to log out and change protocols...Kakaroto I think MSNP15 is mature enough for contact list management functionality now eh? :P...anyways, once the contact management stuff was done, change back to P15, log in, send from P12 (aka 11 in aMSN) to P15, it rings, it connects, it disconnects but no sound, but not sure if this is normal.

I guess I will pull my defective laptop out of it's warranty return box (it goes back tomorrow) and if I can get it booted I will load WLM and try it.
I'll keep you posted, oh and this is on XP Pro SP2 btw.


Title: Audio/Video conversation
Post by: H@t Trick on April 22, 2008, 07:39:06 am
Ok now some WLM<<-->>aMSN P15 results, screenshots and logs
So I logged in from one account on my laptop with WLM 8.5 send the request to my P15 account, which tells me it receieved a voice call request that is not supported and a link to return the call, so I click the link to return the call, it rings on both computers, I accept on WLM, and WLM has no errors, says it connects successfully, but I get a pop up window with an error, I think from farsight (screenshot included), and there was a weird random CW opened in a  new tab the first time but not again, and not updated after that. Sound does not seem to be transfered though. For some reason my main msn account which I was not using crashed during this process, and I have included the screenshot form the error but obviously no logs
Here are the logs:

EDIT: I am using SVN r9756

WLM-->aMSN (with call back, is that normal?) Status log

Code:
[01:52:39] MSNSIP : Received SIP invite on 207.46.112.144
[01:52:39] MSNSIP : Creating SIP connection to 207.46.112.144
[01:52:39] MSNSIP : SIP connection created : ::MSNSIP::SIPConnection13
[01:52:39] SIP : Registering :
[01:52:40] SIP : Registered
[01:52:40] Going to Read : 1959
[01:52:40] MSNSIP : requestSIP : ::MSNSIP::SIPConnection13 f62b95252f264ea88482c6b081a91a20 INVITE
[01:52:40] Received unsupported SIP call from user@domain.com
[01:52:40] SIP : Answering Invite with status 480
[01:52:40] MSNSIP : Destroying ::MSNSIP::SIPConnection13
[01:52:40] SIP : Unregistering
[01:52:40] SIP : InviteRequestHandler called
[01:52:40] MSNSIP : requestSIP : ::MSNSIP::SIPConnection13 f62b95252f264ea88482c6b081a91a20 ACK
[01:52:40] Got Disconnected from SIP
[01:52:40] MSNSIP: Got an error
[01:52:40] MSNSIP : Destroying ::MSNSIP::SIPConnection13
[01:52:44] CallInviteUser user@domain.com
[01:52:44] User supports SIP
[01:52:44] MSNSIP : Inviting user user@domain.com to a SIP call
[01:52:44] MSNSIP : Creating SIP connection to vp.sip.messenger.msn.com
[01:52:44] MSNSIP : SIP connection created : ::MSNSIP::SIPConnection15
[01:52:44] Farsight : Preparing
[01:52:45] Farsight answering : LOCAL_CODEC: 8 PCMA 8000
[01:52:45] Farsight answering : LOCAL_CODEC: 0 PCMU 8000
[01:52:45] Farsight answering : LOCAL_CODEC: 96 SIREN 16000
[01:52:45] Farsight answering : LOCAL_CODECS_DONE
[01:52:45] Farsight answering : LOCAL_CANDIDATE: L0 1  UDP 0 X.X.X.X 7078
[01:52:45] Farsight answering : LOCAL_CANDIDATE: L0 2  UDP 0 X.X.X.X 7079
[01:52:45] Farsight answering : LOCAL_CANDIDATES_DONE
[01:52:45] Farsight : Farsight is now prepared!
[01:52:45] SIP : Registering :
[01:52:45] SIP invite sent
[01:52:45] SIP : Registered
[01:52:45] SIP : Sending Invite
[01:52:45] Received INVITE response
[01:52:45] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection15 user@domain.com 788ed7475a536820c70f3e8c3260abaf TRYING
[01:52:47] Received INVITE response
[01:52:47] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection15 user@domain.com 788ed7475a536820c70f3e8c3260abaf RINGING
[01:52:50] Got Disconnected from SIP
[01:52:54] Going to Read : 293
[01:52:54] Received INVITE response
[01:52:54] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection15 user@domain.com 788ed7475a536820c70f3e8c3260abaf OK
[01:52:54] SIP callee accepted our call
[01:52:54] Farsight starting : {SIREN 111 16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000} - {{} 1 {} UDP 1 X.X.X.X 34531} {{} 2 {} UDP 1 X.X.X.X 34532}
[01:53:05] SIP Keepalive
[01:53:25] SIP Keepalive
[01:53:43] Received INVITE response
[01:53:43] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection15 user@domain.com 788ed7475a536820c70f3e8c3260abaf CLOSED REMOTE_BYE
[01:53:43] SIP callee closed the call
[01:53:43] MSNSIP : Destroying ::MSNSIP::SIPConnection15
[01:53:43] SIP : Unregistering
[01:53:52] Closed pipe :
(farsight.exe:2500): GLib-GObject-WARNING **: g_object_set_valist: object class `GstDshowAudioSrc' has no property named `@9\xb3'

This application has requested the Runtime to terminate it in an unusual way.
Please contact the application's support team for more information.
[01:53:52] Got Disconnected from SIP
[01:53:52] MSNSIP: Got an error
[01:53:52] MSNSIP : Destroying ::MSNSIP::SIPConnection15
[01:53:53] Got Disconnected from SIP

WLM-->aMSN Protocol Log
Code:
[01:52:35] <-ns-sock8084 UBN user@domain.com 11 28
[01:52:35] Message Contents:
1 1 65541 134546710 54000000
[01:52:39] <-ns-sock8084 UBN user@domain.com 2 54
[01:52:39] Message Contents:
INVITE 207.46.112.144 by2msg2vp6.sip.messenger.msn.com
[01:52:39] -->SIP (207.46.112.144) REGISTER sip:rogers.com SIP/2.0

v: SIP/2.0/TLS 0.0.0.0:3476

Max-Forwards: 70

f: <sip:user2@domain2.com>;tag=9f2b865858;epid=300959dd11

t: <sip:user2@domain2.com>

i: 604f92ce8d0e21809f1c916a51e8a514

CSeq: 1 REGISTER

m: <sip:0.0.0.0:3476;transport=tls>;proxy=replace

User-Agent: aTSC/0.1

ms-keep-alive: UAC;hop-hop=yes

o: registration

Authorization: Basic <censored>

l: 0




[01:52:40] <--SIP (207.46.112.144) SIP/2.0 200 OK 207.46.112.144
Content-Length: 0
Via: SIP/2.0/TLS 0.0.0.0:3476;received=X.X.X.X;ms-received-port=3476;ms-received-cid=4bd3cf00
From: <sip:user2@domain2.com>;tag=9f2b865858;epid=300959dd11
To: <sip:user2@domain2.com>;tag=05E1D5C089ADD95B289C63C587E40731
Call-ID: 604f92ce8d0e21809f1c916a51e8a514
CSeq: 1 REGISTER
Expires: 600
ms-keep-alive: UAS; tcp=no; hop-hop=yes; end-end=no; timeout=20
Contact: <sip:X.X.X.X:3476;transport=tls;ms-received-cid=4BD3CF00>;Expires=600




[01:52:40] <--SIP (207.46.112.144) INVITE sip:X.X.X.X:3476;transport=tls;ms-received-cid=4BD3CF00 SIP/2.0
Record-Route: <sip:BY2MSG2132906.phx.gbl:443;transport=tls;lr;ms-route-sig=aa57eD4EYS5iNfHAxk2USviSx8v0EA>;tag=05E1D5C089ADD95B289C63C587E40731
Via: SIP/2.0/TLS 207.46.112.144:443;branch=z9hG4bKF5087241.C4B20558;branched=TRUE;ms-internal-info="bah3O6_vbcI99PXX2820DN2Ugxn-AA"
Max-Forwards: 69
Content-Length: 1959
Via: SIP/2.0/TLS 192.168.0.120:1210;received=207.46.112.236;ms-received-port=1210;ms-received-cid=4bd3c800
Contact: <sip:user@domain.com:1210;maddr=207.46.112.236;transport=tls;ms-received-cid=4BD3C800>
From: "0" <sip:user@domain.com;wl-type=1>;tag=3cee529b29;epid=61fddf42eb
To: <sip:user2@domain2.com>
Call-ID: f62b95252f264ea88482c6b081a91a20
CSeq: 1 INVITE
User-Agent: LCC/1.7
Ms-Conversation-ID: f=0
Content-Type: application/sdp



v=0

o=- 0 0 IN IP4 207.46.112.165

s=session

c=IN IP4 207.46.112.165

b=CT:100

t=0 0

m=audio 46257 RTP/AVP 114 111 112 115 116 4 8 0 97 101

a=candidate:hHKFfQITSOHjq28FED0o8PHt+jBru2ukbkI9qCpK5/g= 1 atWZFnu2G0uR1aFbGP4Pqw== UDP 0.850 192.168.0.120 12286

a=candidate:hHKFfQITSOHjq28FED0o8PHt+jBru2ukbkI9qCpK5/g= 2 atWZFnu2G0uR1aFbGP4Pqw== UDP 0.850 192.168.0.120 63162

a=candidate:C8j4kS35gZh/FPVB8a9nTt629jWxkKSlfBzxqyLHV7Q= 1 y/z6vFHwx9RISd2zyJBhtg== UDP 0.450 207.46.112.165 46257

a=candidate:C8j4kS35gZh/FPVB8a9nTt629jWxkKSlfBzxqyLHV7Q= 2 y/z6vFHwx9RISd2zyJBhtg== UDP 0.450 207.46.112.165 39527

a=candidate:GroupW/W+N6mOY/DwDhGao3852XV+d0VF6PTuz4bBRI= 1 jyV8XGw5QkZUIj8wG/sJlw== TCP 0.150 207.46.112.165 51010

a=candidate:GroupW/W+N6mOY/DwDhGao3852XV+d0VF6PTuz4bBRI= 2 jyV8XGw5QkZUIj8wG/sJlw== TCP 0.150 207.46.112.165 47377

a=candidate:irBxm+T/9rcpfUZVKzhHy9RaxSJucB2xZ7/UUqboxMs= 1 TjUrKHh8bG4zUoK3sGZ1tg== UDP 0.550 X.X.X.X 37332

a=candidate:irBxm+T/9rcpfUZVKzhHy9RaxSJucB2xZ7/UUqboxMs= 2 TjUrKHh8bG4zUoK3sGZ1tg== UDP 0.550 X.X.X.X 43450

a=candidate:iSGBkh15vY5E9+Ss3VrMJuYr2+bAfsg3U6J4QD4nzQE= 1 8WgFezIRwVyybTd1xBNrZg== TCP 0.250 X.X.X.X 1212

a=candidate:iSGBkh15vY5E9+Ss3VrMJuYr2+bAfsg3U6J4QD4nzQE= 2 8WgFezIRwVyybTd1xBNrZg== TCP 0.250 X.X.X.X 1213

a=candidate:EpaHfy/Q9XQmWUvMXnJtv2JPGbGiR9zcR3tI1Yw4vYY= 1 v6DKM4qaRsbsbY0E8kfNNg== UDP 0.670 X.X.X.X 12014

a=candidate:EpaHfy/Q9XQmWUvMXnJtv2JPGbGiR9zcR3tI1Yw4vYY= 2 v6DKM4qaRsbsbY0E8kfNNg== UDP 0.670 X.X.X.X 12015

a=rtcp:39527

a=rtpmap:114 x-msrta/16000

a=fmtp:114 bitrate=12000

a=rtpmap:111 SIREN/16000

a=fmtp:111 bitrate=16000

a=rtpmap:112 G7221/16000

a=fmtp:112 bitrate=24000

a=rtpmap:115 x-msrta/8000

a=fmtp:115 bitrate=12000

a=rtpmap:116 AAL2-G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:rejected


[01:52:40] -->SIP (207.46.112.144) SIP/2.0 100 Trying

v: SIP/2.0/TLS 207.46.112.144:443;branch=z9hG4bKF5087241.C4B20558;branched=TRUE;ms-internal-info="bah3O6_vbcI99PXX2820DN2Ugxn-AA"

v: SIP/2.0/TLS 192.168.0.120:1210;received=207.46.112.236;ms-received-port=1210;ms-received-cid=4bd3c800

Max-Forwards: 70

f: "0" <sip:user@domain.com;wl-type=1>;tag=3cee529b29;epid=61fddf42eb

t: <sip:user2@domain2.com>

i: f62b95252f264ea88482c6b081a91a20

CSeq: 1 INVITE

User-Agent: aTSC/0.1

l: 0




[01:52:40] -->SIP (207.46.112.144) SIP/2.0 480 Temporarily Unavailable

v: SIP/2.0/TLS 207.46.112.144:443;branch=z9hG4bKF5087241.C4B20558;branched=TRUE;ms-internal-info="bah3O6_vbcI99PXX2820DN2Ugxn-AA"

v: SIP/2.0/TLS 192.168.0.120:1210;received=207.46.112.236;ms-received-port=1210;ms-received-cid=4bd3c800

Record-Route: <sip:BY2MSG2132906.phx.gbl:443;transport=tls;lr;ms-route-sig=aa57eD4EYS5iNfHAxk2USviSx8v0EA>;tag=05E1D5C089ADD95B289C63C587E40731

Max-Forwards: 70

f: "0" <sip:user@domain.com;wl-type=1>;tag=3cee529b29;epid=61fddf42eb

t: "0" <sip:user2@domain2.com>;tag=3b27177e7f

i: f62b95252f264ea88482c6b081a91a20

CSeq: 1 INVITE

User-Agent: aTSC/0.1

l: 0




[01:52:40] -->SIP (207.46.112.144) REGISTER sip:rogers.com SIP/2.0

v: SIP/2.0/TLS 192.168.0.108:3476

Max-Forwards: 70

f: <sip:user2@domain2.com>;tag=e1d8a4bddf;epid=9f5dc552e7

t: <sip:user0@domain2.com>

i: afacfbeb37c93356e820ef595982751f

CSeq: 2 REGISTER

m: <sip:192.168.0.108:3476;transport=tls>;proxy=replace

User-Agent: aTSC/0.1

ms-keep-alive: UAC;hop-hop=yes

Expires: 0

Authorization: Basic  <censored>

l: 0




[01:52:40] <--SIP (207.46.112.144) ACK sip:X.X.X.X:3476;transport=tls;ms-received-cid=4BD3CF00 SIP/2.0
Via: SIP/2.0/TLS 207.46.112.144:443;branch=z9hG4bKF5087241.C4B20558;branched=FALSE
Max-Forwards: 70
CSeq: 1 ACK
Call-ID: f62b95252f264ea88482c6b081a91a20
To: "0" <sip:user2@domain2.com>;tag=3b27177e7f
From: "0" <sip:user@domain.com;wl-type=1>;tag=3cee529b29;epid=61fddf42eb
Content-Length: 0




[01:52:40] <--SIP (207.46.112.144) SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/TLS 192.168.0.108:3476;received=X.X.X.X;ms-received-port=3476;ms-received-cid=4bd3cf00
From: <sip:user2@domain2.com>;tag=e1d8a4bddf;epid=9f5dc552e7
To: <sip:user2@domain2.com>;tag=05E1D5C089ADD95B289C63C587E40731
Call-ID: afacfbeb37c93356e820ef595982751f
CSeq: 2 REGISTER




[01:52:45] -->SIP (vp.sip.messenger.msn.com) REGISTER sip:rogers.com SIP/2.0

v: SIP/2.0/TLS 0.0.0.0:3477

Max-Forwards: 70

f: <sip:user2@domain2.com>;tag=503373efd4;epid=0bbb02f53a

t: <sip:user2@domain2.com>

i: e1fc008319ec87b63301cfb520ce8e41

CSeq: 1 REGISTER

m: <sip:0.0.0.0:3477;transport=tls>;proxy=replace

User-Agent: aTSC/0.1

ms-keep-alive: UAC;hop-hop=yes

o: registration

Authorization: Basic  <censored>


l: 0




[01:52:45] <--SIP (vp.sip.messenger.msn.com) SIP/2.0 200 OK 207.46.112.142
Content-Length: 0
Via: SIP/2.0/TLS 0.0.0.0:3477;received=207.46.112.242;ms-received-port=3477;ms-received-cid=41082800
From: <sip:user2@domain2.com>;tag=503373efd4;epid=0bbb02f53a
To: <sip:user2@domain2.com>;tag=B7081A6C0685B5FE82B8C756046DA4A6
Call-ID: e1fc008319ec87b63301cfb520ce8e41
CSeq: 1 REGISTER
Expires: 600
ms-keep-alive: UAS; tcp=no; hop-hop=yes; end-end=no; timeout=20
Contact: <sip:207.46.112.242:3477;transport=tls;ms-received-cid=41082800>;Expires=600




[01:52:45] -->SIP (vp.sip.messenger.msn.com) INVITE sip:user@domain.com SIP/2.0

v: SIP/2.0/TLS 0.0.0.0:3477

Max-Forwards: 70

f: "0" <sip:user2@domain2.com>;tag=33bc484715;epid=207c4ff8ab

t: <sip:user@domain.com>

i: 788ed7475a536820c70f3e8c3260abaf

CSeq: 1 INVITE

m: "0" <sip:user2@domain2.com:3477;maddr=0.0.0.0;transport=tls>;proxy=replace

User-Agent: aTSC/0.1

Ms-Conversation-ID: f=0

c: application/sdp

l: 245



v=0
o=- 0 0 IN IP4 X.X.X.X

s=session

c=IN IP4 X.X.X.X

b=CT:100

t=0 0

m=audio 7078 RTP/AVP 96 8 0

a=rtcp:7079

a=rtpmap:96 SIREN/16000

a=fmtp:96 bitrate=16000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=encryption:rejected


[01:52:45] <--SIP (vp.sip.messenger.msn.com) SIP/2.0 100 Trying
Via: SIP/2.0/TLS 0.0.0.0:3477;received=207.46.112.242;ms-received-port=3477;ms-received-cid=41082800
From: "0" <sip:user2@domain2.com>;tag=33bc484715;epid=207c4ff8ab
To: <sip:user@domain.com>
Call-ID: 788ed7475a536820c70f3e8c3260abaf
CSeq: 1 INVITE
Content-Length: 0




[01:52:47] <--SIP (vp.sip.messenger.msn.com) SIP/2.0 180 Ringing
v: SIP/2.0/TLS 0.0.0.0:3477;received=207.46.112.242;ms-received-port=3477;ms-received-cid=41082800
l: 0
From: "0" <sip:user2@domain2.com;wl-type=1>;tag=33bc484715;epid=207c4ff8ab
To: "0" <sip:user@domain.com>;tag=3afea4582a
Call-ID: 788ed7475a536820c70f3e8c3260abaf
CSeq: 1 INVITE
User-Agent: LCC/1.7




[01:52:54] <--SIP (vp.sip.messenger.msn.com) SIP/2.0 200 OK
v: SIP/2.0/TLS 0.0.0.0:3477;received=207.46.112.242;ms-received-port=3477;ms-received-cid=41082800
l: 293
Contact: <sip:user@domain.com:1214;maddr=X.X.X.X;transport=tls;ms-received-cid=41082B00>
From: "0" <sip:user2@domain2.com;wl-type=1>;tag=33bc484715;epid=207c4ff8ab
To: "0" <sip:user@domain.com>;tag=3afea4582a
Call-ID: 788ed7475a536820c70f3e8c3260abaf
CSeq: 1 INVITE
Record-Route: <sip:BY2MSG2132904.phx.gbl:444;transport=tls;lr;ms-route-sig=dafymVG_Rt_R7afW_0o9znhQ_vhVz_>
User-Agent: LCC/1.7
Content-Type: application/sdp



v=0

o=- 0 0 IN IP4 X.X.X.X

s=session

c=IN IP4 X.X.X.X

b=CT:100

t=0 0

m=audio 34531 RTP/AVP 111 8 0 101

a=rtpmap:111 SIREN/16000

a=fmtp:111 bitrate=16000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:rejected


[01:52:54] -->SIP (vp.sip.messenger.msn.com) ACK sip:user@domain.com SIP/2.0

v: SIP/2.0/TLS 192.168.0.108:3477

Max-Forwards: 70

f: "0" <sip:user2@domain2.com;wl-type=1>;tag=33bc484715;epid=207c4ff8ab

t: "0" <sip:user@domain.com>;tag=3afea4582a

i: 788ed7475a536820c70f3e8c3260abaf

CSeq: 1 ACK

User-Agent: aTSC/0.1

l: 0




[01:52:58] <-::MSN::SB23-sock8348 BYE user@domain.com
[01:53:14] ->ns-sock8084 PNG


[01:53:14] <-ns-sock8084 QNG 40
[01:53:43] <--SIP (vp.sip.messenger.msn.com) BYE sip:207.46.112.242:3477;transport=tls;ms-received-cid=41082800 SIP/2.0
Via: SIP/2.0/TLS 207.46.112.142:444;branch=z9hG4bKF4786E55.D5A3C060;branched=TRUE;ms-internal-info="daDIVpQOV0v1ANHqe63pn2c_7nkbkA"
Max-Forwards: 69
Content-Length: 0
Via: SIP/2.0/TLS 192.168.0.120:1214;received=X.X.X.X;ms-received-port=1214;ms-received-cid=41082b00
From: "0" <sip:user@domain.com;wl-type=1>;tag=3afea4582a
To: "0" <sip:user2@domain2.com;wl-type=1>;tag=33bc484715;epid=207c4ff8ab
Call-ID: 788ed7475a536820c70f3e8c3260abaf
CSeq: 1 BYE
User-Agent: LCC/1.7




[01:53:43] -->SIP (vp.sip.messenger.msn.com) SIP/2.0 200 OK

v: SIP/2.0/TLS 207.46.112.142:444;branch=z9hG4bKF4786E55.D5A3C060;branched=TRUE;ms-internal-info="daDIVpQOV0v1ANHqe63pn2c_7nkbkA"

v: SIP/2.0/TLS 192.168.0.120:1214;received=X.X.X.X;ms-received-port=1214;ms-received-cid=41082b00

Max-Forwards: 70

f: "0" <sip:user2@domain2.com;wl-type=1>;tag=33bc484715;epid=207c4ff8ab

t: "0" <sip:user@domain.com>;tag=3afea4582a

i: 788ed7475a536820c70f3e8c3260abaf

CSeq: 1 BYE

User-Agent: aTSC/0.1

l: 0




[01:53:43] -->SIP (vp.sip.messenger.msn.com) REGISTER sip:rogers.com SIP/2.0

v: SIP/2.0/TLS 192.168.0.108:3477

Max-Forwards: 70

f: <sip:user2@domain2.com>;tag=8d7bc006a9;epid=b6103aeeb7

t: <sip:user2@domain2.com>

i: fa0fba26e438ef3a2f6ce83a1b7ed477

CSeq: 2 REGISTER

m: <sip:192.168.0.108:3477;transport=tls>;proxy=replace

User-Agent: aTSC/0.1

ms-keep-alive: UAC;hop-hop=yes

Expires: 0

Authorization: Basic  <censored>

l: 0




[01:53:52] <--SIP (vp.sip.messenger.msn.com) SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/TLS 192.168.0.108:3477;received=207.46.112.242;ms-received-port=3477;ms-received-cid=41082800
From: <sip:user2@domain2.com>;tag=8d7bc006a9;epid=b6103aeeb7
To: <sip:user2@domain2.com>;tag=B7081A6C0685B5FE82B8C756046DA4A6
Call-ID: fa0fba26e438ef3a2f6ce83a1b7ed477
CSeq: 2 REGISTER




[01:54:14] ->ns-sock8084 PNG


[01:54:14] <-ns-sock8084 QNG 46
[01:55:14] ->ns-sock8084 PNG


[01:55:14] <-ns-sock8084 QNG 47
[01:56:14] ->ns-sock8084 PNG


[01:56:14] <-ns-sock8084 QNG 49


Screenshots:

Farsight error:
(http://www.postimage.org/aV2TuS89-00a382c559c6df454ceddc9d90969815.png)

Random CW tab opened:
(http://www.postimage.org/aV2TtjRA-00a382c559c6df454ceddc9d90969815.png)

Error on crash of main account:
(http://www.postimage.org/aV2TtVgJ-00a382c559c6df454ceddc9d90969815.png)

I am not sure if this helps but My Event log has 3 Identical entries for farsight.exe:
Event Type:   Error
Event Source:   Application Hang
Event Category:   (101)
Event ID:   1002
Date:      4/22/08
Time:      1:11:34 AM
User:      N/A
Computer:   DANIEL2
Description:
Hanging application farsight.exe, version 0.0.0.0, hang module hungapp, version 0.0.0.0, hang address 0x00000000.

For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp.
Data:
0000: 41 70 70 6c 69 63 61 74   Applicat
0008: 69 6f 6e 20 48 61 6e 67   ion Hang
0010: 20 20 66 61 72 73 69 67     farsig
0018: 68 74 2e 65 78 65 20 30   ht.exe 0
0020: 2e 30 2e 30 2e 30 20 69   .0.0.0 i
0028: 6e 20 68 75 6e 67 61 70   n hungap
0030: 70 20 30 2e 30 2e 30 2e   p 0.0.0.
0038: 30 20 61 74 20 6f 66 66   0 at off
0040: 73 65 74 20 30 30 30 30   set 0000
0048: 30 30 30 30               0000    

And the wish.exe crash:
Event Type:   Error
Event Source:   Application Error
Event Category:   None
Event ID:   1000
Date:      4/22/08
Time:      1:43:36 AM
User:      N/A
Computer:   DANIEL2
Description:
Faulting application wish.exe, version 8.4.2.14, faulting module ntdll.dll, version 5.1.2600.2180, fault address 0x00001230.

For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp.
Data:
0000: 41 70 70 6c 69 63 61 74   Applicat
0008: 69 6f 6e 20 46 61 69 6c   ion Fail
0010: 75 72 65 20 20 77 69 73   ure  wis
0018: 68 2e 65 78 65 20 38 2e   h.exe 8.
0020: 34 2e 32 2e 31 34 20 69   4.2.14 i
0028: 6e 20 6e 74 64 6c 6c 2e   n ntdll.
0030: 64 6c 6c 20 35 2e 31 2e   dll 5.1.
0038: 32 36 30 30 2e 32 31 38   2600.218
0040: 30 20 61 74 20 6f 66 66   0 at off
0048: 73 65 74 20 30 30 30 30   set 0000
0050: 31 32 33 30 0d 0a         1230..  


I hope I have not provided useless irrelevant information, but I have tried to be as thorough as I can be. I hope it helps. Did I do something wrong when I installed farsight? I just updated my SVN as normal with TortoiseSVN using the context menu for SVN update in explorer. I had an idea of what could be wrong as I was typing this but I just forgot it, so any ideas?

<off-topic> Yes I know this was a long post but I wanted to wish you a congratulations (or as we say in Hebrew Mazal Tov!) on passing your last and final exam and on becoming an Engineer kakaroto!! And we all really appreciate the work you and the rest of the dev team put into this project!</off-topic>

EDIT: Cleaned logs of personal info, external IP =X.X.X.X, Internal IP's modified, p15 account=user2@domain2.com, and P12 (default P11) account=user@domain.com


Title: Audio/Video conversation
Post by: kakaroto on April 22, 2008, 07:36:49 pm
Hi,
ok, first, you forgot to clean the Authorization: Basic ...... that contains your password... :p ok, not your 'password' but your ticket token that can be used to connect to the SIP server and make calls (it's a password, but with something like 24h or 48h expiration).
anyways, the wish.exe crash is probably totally unrelated (especially if you say it's another account).
the farsight 'crash' is not actually a crash.. it should just exit but windows (or glib) is stupid... I do a g_error() which should assert() and make amsn realize that farsight quit and that's it, you get the error in status log.. but the assert seems to trigger windows to consider it a crash.. have it show you an error box and add a crash report to the event viewer...
the most important thing is the error message you screenshoted.. stupid thing... also a windows compatibility issue.. when sending a UDP packet to somewhere it's not listening on, we get an ICMP response telling us it's unreachable, which triggers that 'error'.. I should just make the utility ignore that error instead of saying "humm. I got an unknown error, ok, I'll just cancel everything".


Title: issues compiling amsn
Post by: MastaG on April 22, 2008, 08:47:22 pm
Ive installed the following things with ./configure --prefix=/usr, make, make install
1.  gstreamer-0.10.19/
2.  gst-plugins-base-0.10.19/          
3.  gst-plugins-good-0.10.7/
4.  gst-plugins-bad-0.10.6/              
5.  gst-python-0.10.11/
6.  farsight2/ (git pulled at 22-04-2008)
7.  gst-plugins-farsight-0.12.7/ (link from the wiki)
No problems at all with the above.
Now I configure amsn with:
**************************************************************************************************
[mastag@amd3000 amsn]$ ./configure --prefix=/usr --with-tcl=/usr/local/lib --with-tk=/usr/local/lib
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking for g++... g++
checking whether we are using the GNU C++ compiler... yes
checking whether g++ accepts -g... yes
checking tcl build dir... using tcl library in /usr/local/lib
checking tk build dir... using tk library in /usr/local/lib
checking for main in -lstdc++... yes
checking how to run the C preprocessor... gcc -E
checking for X... libraries , headers
checking for gethostbyname... yes
checking for connect... yes
checking for remove... yes
checking for shmat... yes
checking for IceConnectionNumber in -lICE... yes
checking for png_read_info in -lpng... yes
checking png.h usability... yes
checking png.h presence... yes
checking for png.h... yes
checking for jpeg_CreateDecompress in -ljpeg... yes
checking jpeglib.h usability... yes
checking jpeglib.h presence... yes
checking for jpeglib.h... yes
checking jerror.h usability... yes
checking jerror.h presence... yes
checking for jerror.h... yes
checking for ftello... yes
checking for fseeko... yes
checking for getpt... yes
checking for strcasestr... yes
checking for memmem... yes
checking for dlopen... no
checking for pthread_create in -lpthread... yes
checking if mmx should be used... yes
checking for pkg-config... yes
checking for pkg-config... /usr/bin/pkg-config
checking pkg-config is at least version 0.9.0... yes
checking for GLIB... yes
checking for GST... yes
checking for FARSIGHT2... yes
configure: creating ./config.status
config.status: creating Makefile
config.status: creating utils/linux/capture/config.h
config.status: utils/linux/capture/config.h is unchanged

compile time options summary
============================

    X11          : yes
    Tcl          : 8.5
    TK           : 8.5
    DEBUG        : no
    STATIC       : no
    FARSIGHT     : yes

[mastag@amd3000 amsn]$
**************************************************************************************************
And then make:
**************************************************************************************************
[mastag@amd3000 amsn]$ make
  CXX     utils/TkCximage/src/TkCximage.cpp.o
  CXX     utils/TkCximage/src/PhotoFormat.cpp.o
  CXX     utils/TkCximage/src/procs.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximadsp.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaexif.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximagif.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximainfo.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximajpg.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximalyr.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximapng.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximatga.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximatran.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximabmp.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaenc.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximage.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximahist.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaint.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximalpha.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximapal.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximasel.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximath.cpp.o
  CXX     utils/TkCximage/src/CxImage/xmemfile.cpp.o
  AR      utils/TkCximage/src/CxImage/libCxImage.a
ar: creating utils/TkCximage/src/CxImage/libCxImage.a
  LDX     utils/TkCximage/src/TkCximage.cpp.so
  LDX     utils/TkCximage/src/TkCximage.cpp.so
cp utils/TkCximage/src/TkCximage.cpp.so utils/TkCximage/TkCximage.so
  CC      utils/webcamsn/src/webcamsn.o
  CC      utils/webcamsn/src/kidhash.o
  CC      utils/webcamsn/src/bitstring.o
  CC      utils/webcamsn/src/deblock.o
  CC      utils/webcamsn/src/encode.o
  CC      utils/webcamsn/src/idct_dequant.o
  CC      utils/webcamsn/src/mimic.o
  CC      utils/webcamsn/src/vlc_decode.o
  CC      utils/webcamsn/src/colorspace.o
  CC      utils/webcamsn/src/decode.o
  CC      utils/webcamsn/src/fdct_quant.o
  CC      utils/webcamsn/src/vlc_common.o
  CC      utils/webcamsn/src/vlc_encode.o
  AR      utils/webcamsn/src/libmimic.a
ar: creating utils/webcamsn/src/libmimic.a
  LD      utils/webcamsn/src/webcamsn.so
cp utils/webcamsn/src/webcamsn.so utils/webcamsn/webcamsn.so
  CC      utils/tcl_siren/src/tcl_siren.o
  CC      utils/tcl_siren/src/common.o
  CC      utils/tcl_siren/src/dct4.o
  CC      utils/tcl_siren/src/encoder.o
  CC      utils/tcl_siren/src/decoder.o
  CC      utils/tcl_siren/src/huffman.o
  CC      utils/tcl_siren/src/rmlt.o
  AR      utils/tcl_siren/src/libsiren.a
ar: creating utils/tcl_siren/src/libsiren.a
  LD      utils/tcl_siren/src/tcl_siren.so
cp utils/tcl_siren/src/tcl_siren.so utils/tcl_siren/tcl_siren.so
  CC      utils/tclISF/src/tclISF.o
  CC      utils/tclISF/src/libISF/compression.o
  CC      utils/tclISF/src/libISF/createTags.o
  CC      utils/tclISF/src/libISF/decodeTags.o
  CC      utils/tclISF/src/libISF/decompression.o
  CC      utils/tclISF/src/libISF/decProperty.o
  CC      utils/tclISF/src/libISF/encoding.o
  CC      utils/tclISF/src/libISF/libISF.o
  CC      utils/tclISF/src/libISF/read.o
  AR      utils/tclISF/src/libISF/libISF.a
ar: creating utils/tclISF/src/libISF/libISF.a
  LD      utils/tclISF/src/tclISF.so
  LD      utils/tclISF/src/tclISF.so
cp utils/tclISF/src/tclISF.so utils/tclISF/tclISF.so
  CC      utils/farsight/farsight.o
utils/farsight/farsight.c: In function ‘_new_local_candidate’:
utils/farsight/farsight.c:19: fout: ‘FsCandidate’ has no member named ‘candidate_id’
utils/farsight/farsight.c:19: fout: ‘FsCandidate’ has no member named ‘candidate_id’
utils/farsight/farsight.c: In function ‘stdin_io_cb’:
utils/farsight/farsight.c:125: fout: ‘FsCandidate’ has no member named ‘candidate_id’
make: *** [utils/farsight/farsight.o] Fout 1
**************************************************************************************************
Whats wrong?


Title: Audio/Video conversation
Post by: H@t Trick on April 22, 2008, 09:38:05 pm
Quote from: "kakaroto"
Hi,
ok, first, you forgot to clean the Authorization: Basic ...... that contains your password... :p ok, not your 'password' but your ticket token that can be used to connect to the SIP server and make calls (it's a password, but with something like 24h or 48h expiration).
anyways, the wish.exe crash is probably totally unrelated (especially if you say it's another account).
the farsight 'crash' is not actually a crash.. it should just exit but windows (or glib) is stupid... I do a g_error() which should assert() and make amsn realize that farsight quit and that's it, you get the error in status log.. but the assert seems to trigger windows to consider it a crash.. have it show you an error box and add a crash report to the event viewer...
the most important thing is the error message you screenshoted.. stupid thing... also a windows compatibility issue.. when sending a UDP packet to somewhere it's not listening on, we get an ICMP response telling us it's unreachable, which triggers that 'error'.. I should just make the utility ignore that error instead of saying "humm. I got an unknown error, ok, I'll just cancel everything".


Lol well I didn't know what the authorization stuff was plus I thought it might be relevant. Plus I am not sure what exactly is the authorization info in the logs.
So basically all is working as it should on Windows, based on how it is coded right now? Should I get sound or not on Windows?
Should I wait for some updates before I try testing again?


Title: Audio/Video conversation
Post by: kakaroto on April 23, 2008, 12:12:16 am
@MastaG : the issue is this :  
6. farsight2/ (git pulled at 22-04-2008)
as is said in the wiki :
Quote
 8 - farsight2 - 0.0.2 (for big endian systems, there is a bug fixed in git, but you'll have to wait until the API gets stable)

so.. the git version has an incompatible API, which means that it will not work (and will not even compile).. you should just go get the 0.0.2 version and install that...
the current git is moving a lot, once the API gets more stable and stops changing, we'll release 0.0.3 and I'll update farsight to use the new API.

@H@t Trick : ok, latest revision now fixes this bug.. it was a bug in gstreamer's udpsrc. I sent the patch to gstreamer upstream and updated the binary, you shouldn't get that error anymore, and hopefully it will work now.


Title: Audio/Video conversation
Post by: H@t Trick on April 24, 2008, 08:53:26 am
Well I can finally give you some testing results on XP. While testing I went to plug my mic in and cause a reboot of my system (I think there is a loose standoff under the mobo that when jostled shorts/grounds something), this happened twice and SeaMonkey decided it didnt want to to read my profile files, lost access to over 2GBs of emails too, and the normal process I have done in the past to restore my profile manually (it used to happen on a crash every few months but not in over a year) failed. I have finally rebuilt my profile as best as I could, got my emails back so thats good. Anyways enough off topic rambling.
Well theres good news and bad news. Kakaroto's fix works for the error I had. So aMSN to aMSN (both p15) on two pc's only works 1 way it seems, not sure what is happening. IT connects no problem no matter which pc initiates the request. When I initiate from my desktop to my laptop I can send sound from my laptop through the mic to the desktop, but not from the desktop to the laptop using either the mic or routing wav output from WMP (yeah I know more stupid Voleware but I don't have any other music files other than the samples so VLC and MPC weren't options) using the Creative driver app. When I initiate the call from my laptop to my desktop no sound either way. When I log in using WLM 8.5 on my laptop, and connect, sound works bi-directionally no problems and my mic input from my laptop is outputted on the desktop louder than aMSN to aMSN when I get one way sound working. I also had some gstreamer/farsight error windows come up after prolonged SIP sessions, I didnt get a chance to take screenshots, but I think I have a status log with the error, can I PM you with that log Kakaroto instead of posting and editing out all the personal info, man that was pain.

I hope this info helps, do you want logs of my sessions so you can see why I get the behaviour I am getting with the sound?
Anyways it works perfectly with WLM so thats good.


Title: Audio/Video conversation
Post by: kakaroto on April 25, 2008, 12:49:39 am
Hello,
Finally, H@t Trick was able to test :p I also tried it with Tjikkun today, he managed to build rpms for fedora core (should be ready soon for public download).
anyways, I don't care about your rambling, so let's skip to your second paragraph :p
so if I summarize :
1 - amsn -> amsn (1 way...)
2 - desktop <- laptop
3 - laptop >< desktop
4 - WLM8.5 on laptop <--> desktop

ok... so, first case, you say it works 1 way whoever initiates the call.. but you don't say who can hear who? the caller of the callee ?
second case, you don't specify what is used on each pc.. is it amsn to amsn ? in that case, case 2 = case 1, but you just specified that the caller can hear the callee, but callee can't hear the caller.
third case, same as above, you don't specify the client.. if it's amsn to amsn, then it contradicts case 1 where you said it connects no matter who is calling... if it's wlm, then it contracts case 4 where you specify the client (as if other cases were not using WLM8.5)
fourth case works... with the desktop as the callee.

ok, so my analysis with the so bad explanation you just gave is that you're doing this from the same network with one pc having some problems accessing the stun server...
as I said before, amsn to amsn will only work if you're on separate networks, and wlm to amsn will only work one way (wlm user hears nothing), unless you're on the networks in which case it also works one way but the other way (amsn user hears nothing). and that's all because of the external vs. internal ip problem.

1 - one amsn timed out on the stun server and falled back on using its internal ip. the other sent his external ip.. so it worked one way only
2 - same here
3 - the stun request didn't time out and they both sent their external ips so they couldn't connect to each other
4 - the stun request in amsn timed out again and amsn sent his internal ip, while WLM always sends his internal IP, so it worked both ways.


you can PM me the logs for each session if you have them, if you only have the log with the crash, then PM me only the status log for that one.. but I think that the crash error won't show up in the status log (not sure but it seems that it gets eaten by the error box).

can you just confirm whether your 4 cases happen 100% of the time or not (stun timeout would make the results change from time to time), and if you understand the issue, look at the status logs for remote and local candidates, they should contain the remote and local ips you have, if it's internal/external, it will help you know what's happening...
then I'll be waiting for a more detailed report explaining which client to which client (amsn protocol version is useless info in this case because it works the same, it's just a matter of being able to receive a request or not) and what the outcome is.. a useful thing would be for example : case X happened 3/10 of the tests.... and of course, before reporting a case, make sure it's reproducable so I can judge if it's a possible stun timeout or a consistent bug...
if you could get whether the internal or external ip was used in each case (for amsn only, since WLM always uses internal ip), that would certainly help too (status log, line like "Farsight answered : LOCAL_CANDIDATE: ..... ip here ....")


Title: Audio/Video conversation
Post by: H@t Trick on April 25, 2008, 02:36:43 am
Hi,
Sorry for not being so clear, I was stressed from my SeaMonkey profile issues, sorry about the rambling :P
Anyways quick summary:

- Both PCs on my internal network.
- Making a successful connection happens each attempt in all situations.

PC1=Desktop with aMSN P15
PC2=Laptop with aMSN P15
WLM=Laptop with WLM8.5

Cases:
1) PC1 calls PC2, PC1 hears PC2, but PC2 does not hear PC1
2) PC2 calls PC1, no sound either direction
3) PC1 calls WLM, bi-directional sound
4) WLM calls PC1, which causes aMSN to do a call back, bi-directional sound.
Case 4=Case3

I just reconfirmed all cases quickly but I am just on my way out and will re-test these cases and get the status logs for all of them and do 10 tests for each, do you want the log files to show all 10 tests?

i.e. 6 or 8 log files (depending on if case 4 is considered different from case 3) containing 10 tests each?
Hopefully you respond before I get back so i get you the right log files as I wont have use of my laptop after tonight for a few days.

I hope I have clarified my results for you.


Title: Audio/Video conversation
Post by: kakaroto on April 25, 2008, 07:45:50 am
hummm... thanks for the clarified answer!
well, if you can reproduce the cases on each attempt, then only one log for one test per case would be enough, and yes, case3=case4 so no need for that log...
and only status log would be enough so no need for the protocol log (has too much data and all the important bits are outputted in status log anyways).

I'm guessing PC1 for some reason doesn't get the external ip.. no idea why! or... hummm... ok, maybe an idea :
you have forwarded the UDP port 7078 to the desktop... so when you PC1 calls PC2, it listens on 7078 (always), gives external IP and port 7078 since it's forwarded, STUN says the external port is 7078 too... PC2 sends to external IP and port 7078 which gets to PC1 (so PC1 hears PC2), but the opposite doesn't work....
on the second case, I have no idea why it doesn't work :p
third case would be the same reason case 1 works... I see no reason why the case 1 and 2 would give different results...
the logs will help me know what ips are getting sent... but important, please give me also any routing information you have (port forwarding, etc...) from your router.

thx

edit
here's from your log above :
Quote

[01:52:45] Farsight answering : LOCAL_CANDIDATE: L0 1  UDP 0 X.X.X.X 7078
[01:52:45] Farsight answering : LOCAL_CANDIDATE: L0 2  UDP 0 X.X.X.X 7079

you removed the ip.. so I'm guessing it's the external ip... but you see the port is 7078 (+1 for rtcp).. so I guess I was right (7078 is the internal port used), your router is doing it, it maps the ports 1 to 1... or maybe you have the port forwarding or DMZ or something.. anyways, the port would be random otherwise...


Title: Audio/Video conversation
Post by: H@t Trick on April 25, 2008, 08:14:01 pm
You're welcome!
As I was reading your reply I was thinking it was my forwardings and DMZ settings and then you were thinking the same thing. Yes the X.X.X.X is my external IP and PC1 is in the DMZ, so thats my problem eh? My desktop is on the DMZ because using my webcam on aMSN kept giving me a solid grey image and no one saw the picture, enabling the DMZ solved this. I guess I could just forward to correct ports for for the webcam. I will get you the logs as soon as I can, hopefully this afternoon, thats if I can get approved to leave work early lol. Now I am going to have to search the forums for those port numbers :P lol, but how do i ensure those forwardings don't affect my brother on WLM on his pc's?


Title: Audio/Video conversation
Post by: kakaroto on April 25, 2008, 09:19:33 pm
hehe, I knew it!!! :p
ok, so I guess case 1 is because you're in DMZ so the router lets the data through, case 2 might be because PC2 requests port 7078 internally for himself, so the port 7078 is not 'DMZ' anymore... so in that case, noone receives anything... it all makes sense :p
no more need for the logs now since we figured it out...
about port forwarding, it's just the ports specified in the 'connection' tab of preferences (+ a few more for multiple FT/webcam)... that should do it...
you don't need to forward any port for the audio conf, it will use STUN and do firewall punching, so it will work without having to configure anything... I don't even know if it will cause problems if you forward ports or not...


Title: Audio/Video conversation
Post by: H@t Trick on April 26, 2008, 02:04:07 am
oh ok cool, I will still get you that gstreamer/farsight error from the status log when I have a chance, but I will try tomorrow night or sunday and confirm proper behavior with the DMZ off, this is fun to actually be testing an experimental feature :P


Title: Audio/Video conversation
Post by: square87 on April 28, 2008, 08:09:56 am
Hi.

When i give ./utils/farsight/farsight 1 2
i get:

Code:
(farsight:9216): GLib-GObject-WARNING **: specified class size for type `FsRtpSession' is smaller than the parent type's `FsSession' class size

(farsight:9216): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed

(farsight:9216): GLib-GObject-CRITICAL **: g_param_spec_object: assertion `g_type_is_a (object_type, G_TYPE_OBJECT)' failed

(farsight:9216): GLib-GObject-CRITICAL **: g_object_class_install_property: assertion `G_IS_PARAM_SPEC (pspec)' failed


Title: Audio/Video conversation
Post by: kakaroto on April 28, 2008, 08:02:45 pm
humm.. apparently your bug (square87) is a 1 change in 100 of getting it.. it shouldn't happen all the time.. it's a bug in glib, if you update to glib 2.16, it shouldn't happen, but since it's a race condition and would rarely happen, I don't think you would get it if you retry...


Title: Audio/Video conversation
Post by: square87 on April 28, 2008, 08:09:29 pm
I already tried variuos time before posting... :(
I have libglib 2.16.3-1 (the hardy version...)
Btw i also think that's a my problem... i'll try to reinstall/compile

Thanks, byez :)


Title: Audio/Video conversation
Post by: mgomes on April 29, 2008, 01:43:57 pm
Hi everybody,

First of all, i would like to congratulate amsn development team for this project, i use amsn since 0.95 version, and i like so much to use this software! ;)

Well, i've read that amsn svn has the new feature of audio call, and check the farsight wiki page to use it. I'm on ubuntu hardy, and installed the following packages from ubuntu repository:

gstreamer0.10-alsa 0.10.18-3
gstreamer0.10-plugins-base 0.10.18-3
gstreamer0.10-plugins-good 0.10.7-3
gstreamer0.10-plugins-bad 0.10.6-5
gstreamer0.10-plugins-farsight 0.12.5-2ubuntu1

And compile farsight2 - 0.0.2 using ./configure --prefix=/usr

I didn't use that packages supplied at wiki, because them all are for i386 machines, and i'll compile amsn at a amd64 environment, so i guess that's could be a problem to use i386 libraries with amd64 binary.

Well, after compiling amsn (compiled amsn using: ./configure --with-tk=/usr/local/lib/ --with-tcl=/usr/local/lib/) tcl and tk 8.5.2, i executed ./utils/farsight/farsight 1 2, and got the following:

user@machine:/usr/local/src/amsn$ ./utils/farsight/farsight 1 2
Error while creating new session (0): Could not create the fsvalve element**
** ERROR:(utils/farsight/farsight.c:305):main: assertion failed: (0)
Aborted

I don't know what's happening, i read this thread entirely, and didn't see this error before, them someone could help me?

Sorry for my bad english, and thanks in advance for the help :)


Title: Audio/Video conversation
Post by: trv on April 29, 2008, 03:17:41 pm
why did you use /usr/local/lib in the amsn's configure options? Hardy has tcl8.5 in the repositories, for 64bit systems too. For starters, you should install tcl8.5 and tk8.5 from the repositories, and configure amsn with with-tcl=/usr/lib/tcl8.5 and with-tk=/usr/lib/tk8.5

About the farsight error, i don't know if farsight2 has been tested in 64bit pcs, maybe someone with 64bit can share his experience :)


Title: Audio/Video conversation
Post by: kakaroto on April 29, 2008, 03:19:17 pm
Hi mgomes, Thanks for the nice comments and welcome to the forums.

You never saw the error before because noone ever did what you did before :p
in the wiki, it does say that you need to install the gst-plugins-farsight version that I linked.. that's a darcs version (like svn) of gst-plugins-farsight which is between 0.12.7 and 0.12.8... you installed 0.12.5 which is pretty old and doesn't have the necessary gstreamer plugins (like the fsvalve it's complaining about).
So please, just download the gst-plugins-farsight-siren.tar.gz from the link provided in the wiki, and compile that instead, that should fix your issue.
have fun!


Title: Audio/Video conversation
Post by: mgomes on April 30, 2008, 03:50:36 am
Hi trv, thanks for your reply! :) i compiled tcl and tk because i would enable the antialiasing on the fonts in amsn. I didn't know if the ubuntu packages comes with this feature. At 8.4 version didn't, and in earlier versions of amsn i compiled it to enable antialiasing.

Hi Kakaroto, thanks for your reply too! Sorry, but i didn't know that gst-plugins-farsight at ubuntu repositories is an older version, but i removed it, downloaded the tar.gz at wiki, compiled and succesfull run ./utils/farsight/farsight 1 2

user@machine:/usr/local/src/amsn$ utils/farsight/farsight 1 2
LOCAL_CODEC: 96 SIREN 16000
LOCAL_CODEC: 0 PCMU 8000
LOCAL_CODEC: 8 PCMA 8000
LOCAL_CODEC: 97 MPA 90000
LOCAL_CODEC: 3 GSM 8000
LOCAL_CODECS_DONE
LOCAL_CANDIDATE: L0 1  UDP 0 127.0.0.1 7078
LOCAL_CANDIDATE: L0 2  UDP 0 127.0.0.1 7079
LOCAL_CANDIDATES_DONE

Well, after this program working, i logged in at aMSN and called a friend that uses WLM, and i could hear him, but he didn't, then i thought this problem could be that issue you mentioned about the WLM user be on a subnet, and i tested with another friend, and works fine at WLM, i tested only to sendo audio call invitation, not receiving! Unfortunately, i still didn't test with another amsn user, but when i found one, i'll test :)

Once more, thank for the help, and congratulations for the project!

One more question, Kakaroto, i saw that the packages you supplied at wiki are for i386 machines, if needed i could make amd64 packages to improve the support :)


Title: Audio/Video conversation
Post by: kakaroto on April 30, 2008, 05:51:39 am
Hi mgomes,
yes, gst-plugins-farsight in ubuntu is not the latest, but also, you needed a 'special' release of gst-plugins-farsight with the support for the Siren codec, and there's no official release with that yet.
Anyways, thanks for taking the time to read the thread.. when I say you report that sound worked one way only, I thought "oh no, do I have to repeat myself", but then you read it :p
About receiving audio call invitation, you can only receive one if you use MSNP15, so if you don't, you can't receive it anyway, sending or receiving will work the same way, so it won't make much of a difference to test one or the other...
Yes, the packages are i386, I don't think we need amd64 unless someone requests it... anyways it's svn, an experimental feature, so they should get the right packages (by the distro) once we release amsn 0.98.. in the meantime, it was just to help more people try it faster...
thanks anyway! :)

p.s: if you still do it, you can send me the packages.


Title: Audio/Video conversation
Post by: H@t Trick on April 30, 2008, 07:12:32 am
Well I finally go the chance to test with the DMZ disabled and well funny, no combination seems to give any audio now, re-enable the DMZ, no audio still, this is getting frustrating now when it was excting before, but hopefully tomorrow I can retest and get you some status logs and PM them to you.


Title: Audio/Video conversation
Post by: Yannick on April 30, 2008, 03:58:09 pm
Hello,

I'm out of topic, so please forgive me.

Reading the page 9 I saw some protocol logs used by WLM.

It seems to be SIP.

kakaroto, do you know if it is real SIP compliant or some custom variant? Is there documentation available somewhere about it if it is custom?

Best regards,
Yannick


Title: Audio/Video conversation
Post by: kakaroto on April 30, 2008, 07:23:11 pm
Hi Yannick.. I think I pretty much answered that question in my previous posts, I've given very detailed description of what happens...  so if you need any technical info, just read the whole thread (it's a lot I know.. I wrote most of it :p)
anyways, to summarize, yes, it's 100% SIP compliant, nothing is different, everything is as is.. (I also was able to make it work using telepathy-sofiasip..) note that the abbreviated header fields ("t" = "To", "v" = Via", "m" = "Contact", etc...) are also SIP compliant (they are called "Compact headers" : RFC 3261).
A few notable differences from 'standard' SIP (but not non-rfc compliant afaik) is that the From field needs a parameter "epid" randomly generated.. and it all goes through TLS (BUT the sip URI is 'sip:" not "sips:") and the authentication used is Basic with a fixed username and the password being the MessengerSecure token received by SSO.
as I said before.. SIP/SDP/RTP/ICE/STUN/TURN/etc.. all those protocols used by this feature are 100% RFC compliant (apart from the stun message-integrity which appends some null characters before doing the hmac.. but that's just an authentification issue to make sure not anyone uses their servers.. but it still uses the same md5+hmac algorithm for authentication as described in the RFC... it's just that the message being checksummed has 60 null bytes appended to it).. note that ICE is a bit different, but that's normal (even gtalk has a different ICE implementation), that's because ICE is not yet an RFC, it's still a draft, gtalk and MSN both implement the 6th draft of ICE (there are 19 so far), and they each have some differences from the draft since the draft doesn't work perfectly. WLM changes are mainly (afaik) about the username/message-integrity authentication parameters.


Title: Audio/Video conversation
Post by: Yannick on May 05, 2008, 06:29:02 pm
Hello,

Thank you kakaroto. I'm part of the Ekiga team (well, I'm not a coder myself). Damien Sandras is interested ;) It seems, after this "discovery" for us, compatibility with WLM is on his TODO. (using SIP)

Best regards,
Keep up the good work :)
Yannick


Title: Audio/Video conversation
Post by: kakaroto on May 05, 2008, 07:11:13 pm
hi Yannick,
I'm glad this discovery interests the ekiga team! I just hope you won't release your compatible version before ours :p If you need help porting Ekiga to use farsight, I'd gladly help (ok, just teasing :p).
Anyways, you will need SIP + RTP + ICE, and probably TURN.. I didn't release my RE work on the ICE+TURN but I will soon (once I fix up libnice which will need refactoring) implement that with farsight too..
If you need help, don't hesitate to ping me on #amsn channel at irc.freenode.net


Title: Audio/Video conversation
Post by: Fabioamd87 on May 13, 2008, 02:03:13 pm
http://forums.microsoft.com/MSDN/ShowPost.aspx?PostID=833356&SiteID=1

i think they ignore us :P


Title: Audio/Video conversation
Post by: Fabioamd87 on May 17, 2008, 06:00:09 pm
is possible to split this support posts and leave this tread for devs communication?


Title: Audio/Video conversation
Post by: kakaroto on May 17, 2008, 07:06:15 pm
As per fabio's suggestion, the thread was split, for support questions, use this thread instead :
http://www.amsn-project.net/forums/viewtopic.php?t=5175
thx


Title: Audio/Video conversation
Post by: kakaroto on July 31, 2008, 07:29:23 am
Hello hello everyone..
long time no see!
Well, I'm posting here to give you a little update on my progress...
good news! I have just finished making libnice MSN compatible... This basically means that we can now have ICE support with aMSN so you can always have the audio conversation working, whether it is with another WLM user, or with aMSN, whether the user is behind a NAT/firewall or not...
you will need the 'nice-kakaroto' branch of libnice from GIT : http://git.collabora.co.uk/?p=user/kakaroto/nice.git;a=shortlog;h=refs/heads/nice-kakaroto
then you will need the 'nice' branch of farsight 2 from GIT : http://git.collabora.co.uk/?p=user/tester/farsight2.git;a=shortlog;h=refs/heads/nice
Then you will need the changes I have locally on my laptop.. LOL.. yeah, I wanted to commit that, but SVN is down for today, so tomorrow it should be fixed, and I'll commit my changes...

Anyways, my previous TODO (on page 3:p) was :
Quote
TODO :
1 - use of ICE for having WLM receive our stream
2 - add a Siren encoder/decode/payloader/depayloader for gstreamer
3 - clean the ugly hacky functions I used for testing...
4 - a proper configure/makefile/runtime dependency check...
5 - a proper UI

So I'm glad to say that '1' is finally DONE!!
the 2, 3, 4 and 5 were already done... so the TODO is finished... oh well, not exactly..

here's the new TODO :
1 - Handle the SIP re-invite with the remote-candidate chosen by the controlling endpoint
2 - send the SIP re-invite when we are in controlling mode and we elect a candidate pair
3 - add support for TURN
4 - make the 'farsight' application currently used, as a Tcl extension (we don't need glib main loop anymore with farsight) and use a tcl API to get farsight running
5 - Remove debug output from libnice and libstun
6 - release libnice
7 - release farsight2

OK, so 1 and 2 should be done tomorrow, pretty simple, I just need to do it...  
3 will be hard, but I reverse engineered most of it, and the only remaining part is implementing it in libnice.. unfortunately, it will require a lot of refactoring since libnice wasn't written for that... thankfully though, it's the same protocol specifications as the google relay, and I am paid to implement google relay support in libnice for my job, so it will be done eventually (hopefully in the next 2 weeks).
4 will be the tricky one since farsight2/gstreamer uses a huge amount of threads and I will need to make the extension thread-safe to avoid crashing Tcl.
5 is simple, in my TODO list, might be done tomorrow
6 and 7 were planned for this week, so expect them soon.. (maybe for next week or in two weeks, depending on some other stuff...)

Bad news (for you guys :p). I will be leaving town/country/continent/not-planet in about 10 days.. and I'll be gone for one month, so I will most probably (for sure) not do any code in that time... You should get a working thingy before then, but the TODO might not be finished by then...
Once this TODO is finished, we can put the "audio conference" feature as 100% complete.. all that will remain is testing it, using it, bugfixing it.

I'll let you know when everything is in SVN and when you can start trying it out!
see you!


Title: Audio/Video conversation
Post by: muratasenel on July 31, 2008, 08:11:48 am
Kakaroto, you are GREAT!

Uhm, before start testing, I just want to make sure that those git repo addresses are correct?

git://git.collabora.co.uk/git/user/kakaroto/nice.git for nice
git://git.collabora.co.uk/git/user/tester/farsight2.git for farsight2 with nice

Murat


Title: Audio/Video conversation
Post by: kakaroto on July 31, 2008, 04:42:34 pm
thx!
yes, once you do the 'git clone', you must do a 'git checkout <branch>' to change to the branch that contains the code you want..
so..
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git checkout nice-kakaroto
cd ..
git clone git://git.collabora.co.uk/git/user/tester/farsight2.git  farsight2
cd farsight2
git checkout nice


btw, I also committed my changes to SVN... I'm working on making this more complete and a real tcl extension!


Title: Audio/Video conversation
Post by: muratasenel on July 31, 2008, 07:51:48 pm
git checkout nice-kakaroto doesn't work
murat@infidel libnice $ git checkout nice-kakaroto
error: pathspec 'nice-kakaroto' did not match any file(s) known to git.

Also, same for farsight2. :\


Title: Audio/Video conversation
Post by: kakaroto on July 31, 2008, 10:01:26 pm
humm.. it's actually 'git checkout origin/nice-kakaroto'
and for farsight2, check the git website, the branch name is now 'nice-rebased-30-07-2008'
be a little bit more patient, I'm working on the tcl extension for this...


Title: Audio/Video conversation
Post by: bigbadben on August 01, 2008, 01:14:58 am
Quote from: "kakaroto"
humm.. it's actually 'git checkout origin/nice-kakaroto'
and for farsight2, check the git website, the branch name is now 'nice-rebased-30-07-2008'
be a little bit more patient, I'm working on the tcl extension for this...


merçi beaucoup pour ton dur travail accomplie j'usqu'a présent.


Title: Audio/Video conversation
Post by: kakaroto on August 01, 2008, 08:41:38 am
Hello friends...
Quote
here's the new TODO :
1 - Handle the SIP re-invite with the remote-candidate chosen by the controlling endpoint
2 - send the SIP re-invite when we are in controlling mode and we elect a candidate pair
3 - add support for TURN
4 - make the 'farsight' application currently used, as a Tcl extension (we don't need glib main loop anymore with farsight) and use a tcl API to get farsight running
5 - Remove debug output from libnice and libstun
6 - release libnice
7 - release farsight2


1 - DONE
2 - DONE
3 - nooooo, not yet (but mostly not needed)
4 - DONE
5 - half done
6 - next week
7 - probably next week too...

so, basically, it will now all work correctly.. it's not a half-finished job anymore (it's not well tested though.. it's almost 4AM now!!!)
so... just follow the instructions and try it out!

Audio conferences should now be stable and should work correctly between amsn<->amsn and amsn<->WLM... whether you are firewalled or not, behind a NAT or not, on the same network or not, etc...
I'll be waiting for your reviews/comments/test results!

p.s.: I'm updating http://amsn-project.net/wiki/Farsight so make sure to check the new requirements!!!!


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on August 01, 2008, 09:41:52 am
so that means.... that means... that means the work on bi-directional audio-video sessions can start after your vacations!!!! :D
That is my #1 wished feature.


Have fun during your vacations!
Where are you heading?


Title: Audio/Video conversation
Post by: muratasenel on August 01, 2008, 09:54:20 am
Hi,

I just checked the wiki page and I think some corrections should be made.

The clone and checkout process should be like this:
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git branch kakaroto origin/nice-kakaroto
git checkout -f kakaroto

cd ..

git clone git://git.collabora.co.uk/git/user/tester/farsight2.git  farsight2
cd farsight2
git branch tester origin/nice-rebased-30-07-2008
git checkout -f nice-rebased-30-07-2008


I didn't do any changes on the wiki since I could be wrong or even I don't have enough permission to do any changes on the wiki.  : )


Title: Audio/Video conversation
Post by: kakaroto on August 01, 2008, 05:28:26 pm
@why.arent.guests.allowed : congratulations on becoming power user!
and no, the bidirectional audio-video will not be for now.. it is also a VERY BAD design (not ours, the protocol).. so the best you could every do is do this audio conversation then do a separate webcam session!

and thanks, I'll be in Morocco for a month.

@muratasenel : humm.. maybe, but no need to make a local branch, I just updated the wiki, instead of "git checkout nice-kakaroto" it should be "git checkout origin/nice-kakaroto". Thanks for noticing!
anyone had a chance to test this yet (I can't anymore, my laptop has 1% battery left and forgot the charger at work... that's why I stopped coding yesterday, lol :p) ?


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on August 01, 2008, 08:15:26 pm
Quote from: "kakaroto"
@why.arent.guests.allowed : congratulations on becoming power user!
I'm more of a power spammer or annoyer...  :roll: heheh

Quote from: "kakaroto"
and no, the bidirectional audio-video will not be for now.. it is also a VERY BAD design (not ours, the protocol)..
Not sure if I understood you. I wasn't expecting it (as you said) "for now", but I also said  "the work ... can start *after* your vacations".
So does that mean you will never work on bidirectional audio-video **or** that you will work on it (even if not immediatly after your vacations)???

Quote from: "kakaroto"
so the best you could every do is do this audio conversation then do a separate webcam session!
So I can start and record a bi-directional audio session, while simultaneously starting 2 webcam sessions (one receiving and the other sending)???

Quote from: "kakaroto"
and thanks, I'll be in Morocco for a month.
Watch out there... I heard they sell some funny-smoking stuff there... bahhh... they sell it everywhere now...  :lol:


Title: Audio/Video conversation
Post by: kakaroto on August 01, 2008, 08:33:42 pm
Quote
Not sure if I understood you. I wasn't expecting it (as you said) "for now", but I also said "the work ... can start *after* your vacations".
So does that mean you will never work on bidirectional audio-video **or** that you will work on it (even if not immediatly after your vacations)???

yeah.. I don't know if I'll continue working on it.. once the libao project (GSoc) is done, we can hopefully implement the send/receive for audio with reception of video.. I was able to make ffmpeg encode video for sending correctly but the quality is REALLY bad, so I don't know about that.. maybe some ffmpeg guru can fix that quite easily (the keyframe is good, subsequent frames not that much, so we need to make it write a keyframe for each frame... bad for bandwidth but good for quality and no other choice right now... )

Quote
So I can start and record a bi-directional audio session, while simultaneously starting 2 webcam sessions (one receiving and the other sending)???

sure! there's nothing preventing you from doing that...

Quote
Watch out there... I heard they sell some funny-smoking stuff there... bahhh... they sell it everywhere now...

lol, no worries.. I'm Moroccan btw :p and yeah, we do have the best 'funny-smoking stuff' in the world :)


Title: Audio/Video conversation
Post by: muratasenel on August 01, 2008, 10:04:50 pm
Quote from: "kakaroto"

anyone had a chance to test this yet (I can't anymore, my laptop has 1% battery left and forgot the charger at work... that's why I stopped coding yesterday, lol :p) ?


Yep, I tried but after all, even amsn can detect farsight in ./configure and creates necessary libs as intended, it cannot detect farsight in amsn settings menu.
I can't figure out what is wrong and I beleive I meet all requirements mentioned in the wiki.


Title: Audio/Video conversation
Post by: Montblanc on August 02, 2008, 01:58:58 am
@ kakaroto: Thank you for redirecting me to the right post, but I still can't build libnice on Ubuntu Hardy i386 :P! Here's the output log:

Code:
cc1: warnings being treated as errors
usages/ice.c: In function ‘stun_usage_ice_conncheck_process’:
usages/ice.c:156: warning: implicit declaration of function ‘htonl’
usages/ice.c:156: warning: nested extern declaration of ‘htonl’
make[3]: *** [ice.lo] Error 1
make[3]: *** Waiting for unfinished jobs....
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -g -O2 -MT utils.lo -MD -MP -MF .deps/utils.Tpo -c utils.c -o utils.o >/dev/null 2>&1
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -g -O2 -MT bind.lo -MD -MP -MF .deps/bind.Tpo -c usages/bind.c  -fPIC -DPIC -o .libs/bind.o
mv -f .deps/utils.Tpo .deps/utils.Plo
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -g -O2 -MT bind.lo -MD -MP -MF .deps/bind.Tpo -c usages/bind.c -o bind.o >/dev/null 2>&1
mv -f .deps/bind.Tpo .deps/bind.Plo
make[3]: Leaving directory `~/Packages/aMSN/Farsight2/libnice/stun'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `~/Packages/aMSN/Farsight2/libnice/stun'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `~/Packages/aMSN/Farsight2/libnice'
make: *** [all] Error 2


Let me just say I don't really need farsight, I'm reporting this bug to help your development. If you need other info or help just tell me.

Keep up the good work and thank you again!


Title: Audio/Video conversation
Post by: kakaroto on August 02, 2008, 05:04:18 am
@muratasenel : I don't understand, aMSN detected farsight and compiled the library, or it didn't ? if it did compile it (and you have utils/farsight/tcl_farsight.so ) and aMSN can't detect farsight in the audio/video assistant... then open amsn's console with ctrl-shift-C and type "package require Farsight".. if it answers with some error copy/paste it here.. if it answers with "0.1", then launch aMSN from an X terminal, and copy/paste the output from the terminal in here...

@Montblanc.. ah sorry, someone reported that already and I answered him here : http://www.amsn-project.net/forums/viewtopic.php?t=5248
to everyone else and to keep all answers in this thread, here's my answer :
Quote
ah.. right.. I fixed that but forgot to push my changes to the server (and I can't do it right now since I have less than 2% battery on the laptop, and I forgot the charger at work...).
To fix it, open the file libnice/stun/usages/ice.c and add at the top of the file below to the other include lines :
Code:

   #include <arpa/inet.h>



I hope someone can get this working... I'll see how it works on monday, now I'll be busy this week-end.


Title: Audio/Video conversation
Post by: kakaroto on August 02, 2008, 08:06:16 am
UPDATE : I pushed the changes to fix the include problem with libnice... and added autogen.sh to libnice... guide updated!
UPDATE2: Changed the git repository URL and changed the branch name to get the latest changes... guide updated!

Hi again...
so here is my experience after trying to test this on a 'virgin' PC... I hope my experience will help others fix any problem they might have...
I was doing all this on a debian/unstable machine, so any ubuntu/other distro users might have a different experience...
First, the easy part... I did a simple 'apt-get update' to update my debian unstable package list, then, at my big surprise, I found that debian had all the LATEST versions of gstreamer available, which is just great! so I did :
Code:
sudo apt-get update
sudo apt-get install libgstreamer0.10-0 gstreamer0.10-plugins-base gstreamer0.10-plugins-good gstreamer0.10-plugins-bad gstreamer0.10-tools gstreamer0.10-alsa

This installed for me gstreamer 0.10.20, gst-plugins-base 0.10.20, gst-plugins-good 0.10.8 and gst-plugins-bad, 0.10.7
Now I wanted to install gstreamer0.10-plugins-farsight, but we need 0.12.9 and they only have 0.12.8.. so here's how I got it to install...
Code:
sudo apt-get install debhelper
sudo apt-get build-dep gstreamer0.10-plugins-farsight
apt-get source gstreamer0.10-plugins-farsight
wget http://farsight.freedesktop.org/releases/gst-plugins-farsight/gst-plugins-farsight-0.12.9.tar.gz
tar -xzvf gst-plugins-farsight-0.12.9.tar.gz
cp -r gst-plugins-farsight-0.12.8/debian gst-plugins-farsight-0.12.9/
cd gst-plugins-farsight-0.12.9/debian
vi changelog

Here, I modified the changelog so that I can get the correct version in the generated deb file...
so I just added at the top of the file :      
Code:
gst-plugins-farsight (0.12.9-1) unstable; urgency=low
  * New upstream release

 -- Youness Alaoui <CENSORED@CENSORED.COM> Sat, 02 Aug 2008 01:53:00 -0500

 

and then ... ":wq" to quit vi, and :
Code:

cd ..
dpkg-buildpackage -rfakeroot
cd ..
sudo dpkg -i gstreamer0.10-plugins-farsight_0.12.9-1_i386.deb

and voila!
most dependencies are installed.. now I just need libnice and farsight...
so I use the commands on the wiki to download them...
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git checkout origin/nice-kakaroto
cd ..
git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git  farsight2
cd farsight2
git checkout origin/nice
cd ..

Then I go to the libnice directory (cd libnice)...and I do :
Code:
./autogen.sh --prefix=/usr
make
sudo make install

(yeah, I didn't do the --prefix=/usr at first, then I realized it was needed otherwise farsight won't find libnice at runtime)
Then I go to the farsight2 directory.. and after some problems with the ./autogen.sh which gave me "automake failed" I found the solution, so :
Code:
sudo apt-get install gtk-doc-tools
./autogen.sh --prefix=/usr --disable-python --disable-gtk-doc
make
sudo make install

yes.. here again, the --prefix=/usr is needed as well as --disable-python otherwise you would need a lot more dependencies (gst-python, etc...) and the 'sudo apt-get install gtk-doc-tools' will allow you to do the ./autogen.sh without the annoying "automake failed" error.. The --disable-gtk-doc is because the gtk doc will currently not build correctly, so use that configure option until it gets fixed.
and now finally.. the most important step otherwise it won't work... you have to do :
Code:
sudo ldconfig

yep.. that might be why muratasenel couldn't make it work (or because of the --prefix=/usr)
Once you do that, in theory, everything should be great!
Now, you go over to amsn.. and you do :
Code:
svn update
./configure
make clean
make
./amsn &

amsn loads.. you press 'sign in' and you should see a lot of output in the terminal with the last three being :
Code:
** (<unknown>:17803): DEBUG: CANDIDATES ARE PREPARED
** (<unknown>:17803): DEBUG: Detach source 0x8fca228 (stream 1).
** (<unknown>:17803): DEBUG: Detach source 0x8fb18a8 (stream 1).

This means all went well...
You could also press Ctrl-N from the main aMSN window to see the audio/video assistant which will test if farsight is correctly installed...

In theory, it should all be working correctly at this point!!!!

If it doesn't.. follow these steps carefully!
1 - read this post
2 - read this post again
3 - Make sure you didn't forget anything... (--prefix=/usr or ldconfig..)
4 - press Ctrl-Shift-C from the main window
5 - type "package require Farsight"
6 - the error should tell you what's wrong, if it's something like "can't find shared library : libgstfarsight.so" it means you either didn't install farsight2 or you forgot 'ldconfig'
7 - if it returned "0.1", then type "farsight Prepare 1"
8 - the error again should tell you what's wrong... if it says "couldn't create fsrtpconference", then you might have forgotten the --prefix=/usr, if it says "can't find nicesrc element", then it means you didn't install libnice correctly (or forgot --prefix=/usr for it), other kind of errors might be because you forgot to install gst-plugins-bad or gst-plugins-farsight, etc...
9 - if it doesn't give any error there, but the audio/video assistant still says that it's not installed.. then the error might be coming from farsight after it's been prepared correctly.. so look at the error messages on your terminal (the xterm terminal, not amsn's console), it should again point you in the right direction as to what you did wrong..
10 - if you get no output at whatever you wrote in amsn's console, then TYPE THE COMMANDS in the console, DO NOT copy/paste them please...
11 - make sure you installed the CORRECT VERSIONS!!!! for example... if you installed gst-plugins-farsight 0.12.8 instead of 0.12.9, IT WILL CRASH...
11 - if you REALLY don't get it.. post here the output of either the "package require Farsight" (5th step) or the "::farsight Prepare 1" (7th step), or the output on the terminal (9th step).. and tell me exactly what you did to get that error message...

That's it...
I think this will serve as a good guide for anyone having trouble with this...
Please note that :
1 - hopefully debian will soon put the gst-plugins-farsight 0.12.9 in their repositories you won't need the 'apt-get source' + 'apt-get build-dep' + 'cp debian' + 'vi changelog' + 'dpkg-buildpackage' + 'dpkg -i' trick to get the 0.12.9 debian file into your system...
2 - hopefully, other distributions will follow in debian's footsteps in order to provide packages from the latest versions of gstreamer, etc...

In theory, when aMSN 0.98 will be released, this whole thing should become as simple as :
Code:
sudo apt-get install libgstfarsight2-dev



p.s.: When farsight2 is working and you test the call... you should see ALOT of output on your X terminal.. something like this :
Code:
** (<unknown>:25932): DEBUG: Agent 0x8f16680 : s1:1: sending 52 bytes to [192.168.1.101]:62878
** (<unknown>:25932): DEBUG: Agent 0x8f16680 : Packet received on local socket 30 from [192.168.1.101]:59636 (80 octets).
STUN error: RTP or other non-protocol packet!

This shows your connection succeeded and that you are sending and receiving data... yeay, great! :)

NOTE: IF YOU DO NOT USE DEBIAN UNSTABLE, DO NOT BOTHER ME, TRY TO FIGURE THINGS OUT ON YOUR OWN, USING THIS GUIDE AS A REFERENCE!!!!

thanks all! :)


Title: Audio/Video conversation
Post by: dave_p_b on August 02, 2008, 11:39:09 am
Hi

In order to get libnice to compile, after I included the line #include <arpa/inet.h> to the libnice/stun/usages/ice.c file, it crashes again.
Now I had to open the file libnice/agent/conncheck.c and change the line 472 from agent, p, buf_len); to agent, p, (int)buf_len);
It now compiles fine.

Hope this helps someone.
All the best

Dave


Title: Audio/Video conversation
Post by: muratasenel on August 02, 2008, 12:47:40 pm
Quote from: "kakaroto"
@muratasenel : I don't understand, aMSN detected farsight and compiled the library, or it didn't ? if it did compile it (and you have utils/farsight/tcl_farsight.so ) and aMSN can't detect farsight in the audio/video assistant... then open amsn's console with ctrl-shift-C and type "package require Farsight".. if it answers with some error copy/paste it here.. if it answers with "0.1", then launch aMSN from an X terminal, and copy/paste the output from the terminal in here...


aMSN detected farsight and compiled the library tcl_farsight.so but aMSN still can't detect farsight in the audio/video assistant. Then I opened amsn's console with ctrl-shift-C and typed "package require Farsight" and it complained it "can't find package Farsight". Then I opened the audio/video assistant with Ctrl+N and still it can't detect farsight. Below is the output from the terminal:

Code:
vid-probe: trying: v4l2...
ioctl VIDIOC_QUERYCAP: Invalid argument
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found OV519 USB Camera at /dev/video0

Input File     : '/usr/share/amsn/skins/Sapphire-2.0/sounds/newemail.wav'
Sample Size    : 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels       : 2
Sample Rate    : 44100

Time: 00:00.76 [00:00.00] of 00:00.76 (100% ) Samples out: 33.7k Clips: 0
Done.

Input File     : '/usr/share/amsn/skins/Sapphire-2.0/sounds/online.wav'
Sample Size    : 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels       : 2
Sample Rate    : 44100

Time: 00:00.81 [00:00.00] of 00:00.81 (100% ) Samples out: 35.9k Clips: 0
Done.
vid-probe: trying: v4l2...
ioctl VIDIOC_QUERYCAP: Invalid argument
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found OV519 USB Camera at /dev/video0
vid-probe: trying: v4l2...
ioctl VIDIOC_QUERYCAP: Invalid argument
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found OV519 USB Camera at /dev/video0
vid-probe: trying: v4l2...
ioctl VIDIOC_QUERYCAP: Invalid argument
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found OV519 USB Camera at /dev/video0
vid-open: trying: v4l2...
v4l2: open
vid-open: failed: v4l2
vid-open: trying: v4l...
v4l: open
  mbuf: size=1843216 frames=2
  v4l: using mapped buffers for capture
v4l: init: /dev/video0 (OV519 USB Camera)
  capabilities:  capture
  size    : 64x48 => 640x480
  channels: 1
    Camera: 0  camera
  audios  : 0
 PAL NTSC SECAM AUTO
  fbuffer : base=0x(nil) size=0x0 depth=0 bpl=0
  picture : brightness=32768 hue=38912 colour=28672 contrast=19456
  picture : whiteness=26880 depth=12 palette=yuv420
v4l: close
vid-open: ok: v4l
vid-open: flags: 2
v4l: close
vid-open: trying: v4l2...
v4l2: open
vid-open: failed: v4l2
vid-open: trying: v4l...
v4l: open
  mbuf: size=1843216 frames=2
  v4l: using mapped buffers for capture
v4l: init: /dev/video0 (OV519 USB Camera)
  capabilities:  capture
  size    : 64x48 => 640x480
  channels: 1
    Camera: 0  camera
  audios  : 0
 PAL NTSC SECAM AUTO
  fbuffer : base=0x(nil) size=0x0 depth=0 bpl=0
  picture : brightness=32768 hue=38912 colour=28672 contrast=19456
  picture : whiteness=26880 depth=12 palette=yuv420
v4l: close
vid-open: ok: v4l
vid-open: flags: 2
v4l: open
  mbuf: size=1843216 frames=2
  v4l: using mapped buffers for capture
ng_dev_open: opened OV519 USB Camera [refcount 1]
v4l: setformat
v4l: capture probe 24 bit TrueColor (BE: rgb)...        failed
v4l: setformat
v4l: capture probe 24 bit TrueColor (LE: bgr)...        ok
v4l: startvideo
v4l: stopvideo
v4l: close
ng_dev_close: closed OV519 USB Camera [refcount 0]
v4l: close
vid-open: trying: v4l2...
v4l2: open
vid-open: failed: v4l2
vid-open: trying: v4l...
v4l: open
  mbuf: size=1843216 frames=2
  v4l: using mapped buffers for capture
v4l: init: /dev/video0 (OV519 USB Camera)
  capabilities:  capture
  size    : 64x48 => 640x480
  channels: 1
    Camera: 0  camera
  audios  : 0
 PAL NTSC SECAM AUTO
  fbuffer : base=0x(nil) size=0x0 depth=0 bpl=0
  picture : brightness=32768 hue=38912 colour=28672 contrast=27648
  picture : whiteness=26880 depth=12 palette=yuv420
v4l: close
vid-open: ok: v4l
vid-open: flags: 2
v4l: open
  mbuf: size=1843216 frames=2
  v4l: using mapped buffers for capture
ng_dev_open: opened OV519 USB Camera [refcount 1]
v4l: setformat
v4l: capture probe 24 bit TrueColor (BE: rgb)...        failed
v4l: setformat
v4l: capture probe 24 bit TrueColor (LE: bgr)...        ok
v4l: startvideo
v4l: stopvideo
v4l: close
ng_dev_close: closed OV519 USB Camera [refcount 0]
v4l: close


I can't see anything about farsight here, I hope it helps.
And in your another post, you mentioned that this problem might be occuring because of not doing ldconfig or of not giving --prefix=/usr in configure but I did all of them and still have the problem.
To be honest, I assume something is wrong with my configuration or the way I installed dependencies and aMsn. So, I'm willing to spend my weekend to find out where I did wrong :)


Title: Audio/Video conversation
Post by: muratasenel on August 02, 2008, 01:07:18 pm
Quote from: "dave_p_b"
Hi

In order to get libnice to compile, after I included the line #include <arpa/inet.h> to the libnice/stun/usages/ice.c file, it crashes again.
Now I had to open the file libnice/agent/conncheck.c and change the line 472 from agent, p, buf_len); to agent, p, (int)buf_len);
It now compiles fine.

Hope this helps someone.
All the best

Dave


I didn't encounter the second problem you had but for anyone who uses gcc43, if you get INT_MAX error, add #include <limits.h>
 in file stun/tools/stund.c.
The patch is also here http://svn.pardus.org.tr/pardus/playground/murat/programming/libs/libnice/files/gcc43.patch


Title: Audio/Video conversation
Post by: kakaroto on August 02, 2008, 07:35:52 pm
thanks dave_p_b and muratasenel for the info. I will commit those patches too on git on monday...


Title: Audio/Video conversation
Post by: fcastillo on August 02, 2008, 08:39:28 pm
After trying everything, I can't make it work, I can't build the package of gstreamer farsight plugin. I don't know why things can be a little bit easier, i just going to give up... I really want to use aMSN with voice, that was the only thing stopping me to move completly to linux from windows, but I guess I'll have to get stuck with windows since linux things are too complicated. Thanks for everybodies help by posting this help, but things work different in evey linux distro, so what one did doesn't work in another, but thanks again...


Title: Audio/Video conversation
Post by: kakaroto on August 02, 2008, 09:47:49 pm
@fcastillo, it works the same pretty much everywhere, but this isn't because linux is complicated, it's just because this is in development, and no distribution has the packages..
as I said :
Quote
In theory, when aMSN 0.98 will be released, this whole thing should become as simple as :
Code:

sudo apt-get install libgstfarsight2-dev

or actually.. to be more precise, just doing 'apt-get install amsn' and that's it... for any distribution actually you would just need to click on 'amsn' from the install package system interface... (synaptic or whatever on the other distro).
You can try to get it to compile (make sure you have gstreamer -dev packages to compile gst-plugins-farsight) or just be patient and wait for the release.


Title: Audio/Video conversation
Post by: Montblanc on August 03, 2008, 02:18:02 am
@ kakaroto: Thank you! Everything is fine, now! :D Sorry about not noticing that post earlier... Keep up the good work!


Title: Audio/Video conversation
Post by: kakaroto on August 03, 2008, 02:51:17 am
cool! Does that mean that you were able to compile everything and that amsn reported farsight as working, or it means that you tried to make a call with a WLM user and that it succeeded ?


Title: Audio/Video conversation
Post by: fcastillo on August 03, 2008, 02:57:08 am
I really appriciaty what you do, I think I'll just have to figure out how to use voice calls on MSN, compiling aMSN it's not difficult, only farsight it's the problem. Well, I'll keep trying but for some weird reason farsight doesn't want to be compiled  :cry: I don't know if any of the people in this forum can post the .deb package generated for the gstreamer farsight plugin. After that, the rest it's easier, I don't know why buy my computer doesn't want to generate the .deb package with the necesary version. Ubuntu only hast version number 5, not even 8.


Title: Audio/Video conversation
Post by: muratasenel on August 03, 2008, 09:49:24 am
Quote from: "fcastillo"
I really appriciaty what you do, I think I'll just have to figure out how to use voice calls on MSN, compiling aMSN it's not difficult, only farsight it's the problem. Well, I'll keep trying but for some weird reason farsight doesn't want to be compiled  :cry: I don't know if any of the people in this forum can post the .deb package generated for the gstreamer farsight plugin. After that, the rest it's easier, I don't know why buy my computer doesn't want to generate the .deb package with the necesary version. Ubuntu only hast version number 5, not even 8.


fcastillo, please don't think about giving up. More testers mean better aMSN. So, please just copy whole compiling output of farsight and we will figure out what's wrong. I'm sure it's not a big problem since most of us are able to compile it.


Title: Audio/Video conversation
Post by: fcastillo on August 04, 2008, 12:09:41 am
it's not farsight itself it the gstreamer farsight plugin.... Since I need it to compile farsight... The plugin is the one that's giving me more trouble... Well, I'll try one more time, later on and I'll post the result as soon as I can... Thanks again, and you're right that if more people test aMSN, it'll be better. I hope you can help me...


Title: Audio/Video conversation
Post by: kakaroto on August 04, 2008, 08:34:12 pm
@fcastillo : just give us the output you get from gst-plugins-farsight when you try to compile it.. as long as you have the necessary dependencies, it should compile without any problems.. so make sure you did the "apt-get build-dep gstreamer0.10-plugins-farsight" if you are on a debian-based distro (ubuntu ?). You can also tell us which distro you are using, it might help us tell you which commands to execute.

@everyone else: I updated libnice with some changes that should make it easier for you to compile (autogen.sh added, and fixed the include problem). It should also fix some issues that we were able to find with muratasenel, where it didn't work with people not on the same local network (that's the only thing I tested.. lol). We still have problems with people on asymetric NATs, and I'm still reverse engineering the issue with peer-reflexive candidates... although I finished reverse engineering it, and it should now work correctly... it appears to still not want to work.. more investigation that way...


Title: Audio/Video conversation
Post by: kanondgeminis on August 05, 2008, 12:13:39 pm
Sorry for bother you... but, i can fix a "make" error for libnice.... i make de ./config without errors... but, when i try to "make" I receive this:

Code:
make[3]: Entering directory `/installing/libnice/agent'
if /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I..    -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -pthread -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include   -I .. -I ../address -I ../random -I ../udp -I ../stun -g -O2 -MT discovery.lo -MD -MP -MF ".deps/discovery.Tpo" -c -o discovery.lo discovery.c; \
then mv -f ".deps/discovery.Tpo" ".deps/discovery.Plo"; else rm -f ".deps/discovery.Tpo"; exit 1; fi
 gcc -DHAVE_CONFIG_H -I. -I. -I.. -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -pthread -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I .. -I ../address -I ../random -I ../udp -I ../stun -g -O2 -MT discovery.lo -MD -MP -MF .deps/discovery.Tpo -c discovery.c  -fPIC -DPIC -o .libs/discovery.o
cc1: warnings being treated as errors
discovery.c: In function ‘priv_discovery_tick_unlocked’:
discovery.c:608: warning: implicit declaration of function ‘stun_usage_turn_create’
discovery.c:608: warning: nested extern declaration of ‘stun_usage_turn_create’
make[3]: *** [discovery.lo] Error 1
make[3]: Leaving directory `/installing/libnice/agent'
make[2]: *** [all] Error 2
make[2]: Leaving directory `/installing/libnice/agent'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/installing/libnice'
make: *** [all] Error 2



Can you helpme please?


Title: Audio/Video conversation
Post by: muratasenel on August 05, 2008, 03:00:00 pm
Do the followings please;

Code:
cd libnice
wget http://svn.pardus.org.tr/pardus/playground/murat/programming/libs/libnice/files/disable-Werror.patch
patch -p0 < disable-Werror.patch
make clean
configure
make
...


it should work...


Title: Audio/Video conversation
Post by: MastaG on August 05, 2008, 05:01:25 pm
Quote from: "muratasenel"
Do the followings please;

Code:
cd libnice
wget http://svn.pardus.org.tr/pardus/playground/murat/programming/libs/libnice/files/disable-Werror.patch
patch -p0 < disable-Werror.patch
make clean
configure
make
...


it should work...


yeah it did for me:)
But when I try to build farsight2 it fails with this error:

gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Werror -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/nice -g -O2 -MT libnice_transmitter_la-fs-nice-agent.lo -MD -MP -MF .deps/libnice_transmitter_la-fs-nice-agent.Tpo -c fs-nice-agent.c  -fPIC -DPIC -o .libs/libnice_transmitter_la-fs-nice-agent.o
cc1: warnings being treated as errors
fs-nice-agent.c: In function 'fs_nice_agent_new':
fs-nice-agent.c:409: error: passing argument 1 of 'nice_agent_new' from incompatible pointer type
fs-nice-agent.c:409: error: incompatible type for argument 2 of 'nice_agent_new'
fs-nice-agent.c:409: error: too many arguments to function 'nice_agent_new'
make[3]: *** [libnice_transmitter_la-fs-nice-agent.lo] Fout 1
make[3]: Map '/home/mastag/src/farsight2/transmitters/nice' wordt verlaten
make[2]: *** [all-recursive] Fout 1
make[2]: Map '/home/mastag/src/farsight2/transmitters' wordt verlaten
make[1]: *** [all-recursive] Fout 1
make[1]: Map '/home/mastag/src/farsight2' wordt verlaten
make: *** [all] Fout 2
[mastag@amd3000 farsight2]$

any ideas?


Title: Audio/Video conversation
Post by: muratasenel on August 05, 2008, 05:51:17 pm
Same problem :)
This time, please use this patch http://svn.pardus.org.tr/pardus/playground/murat/programming/libs/farsight2/files/disable-Werror.patch


Title: Audio/Video conversation
Post by: kakaroto on August 05, 2008, 08:05:49 pm
hi,
sorry, just be patient, I'm still working on libnice and farsight2, the API is now being changed to make TURN support work. So for now, just forget about getting farsight2 to work, etc... and just wait until I finishe what I'm currently doing with the libnice refactoring.


Title: Audio/Video conversation
Post by: fcastillo on August 06, 2008, 04:59:09 am
This post show some errors I was having and trying to fix myself, now I know I have to wait.... Thanks again, you can't imagine how much I appreciate the work you do.


Title: Audio/Video conversation
Post by: kakaroto on August 08, 2008, 03:54:19 pm
hello all,
I have *almost* finished my work... I need one more day and it should be done.. BUT, I don't have that one more day, I'm leaving for the airport in a few hours, and I won't be back until september... hopefully Ollivier (farsight2 maintainer) will try to keep this in a stable form, so you guys can have fun testing this during the month I'll be away...
In theory, the latest GIT now should work fine, make sure you 'git pull' to get the latest changes on both libnice and farsight2.. there might be bugs though, so let's hope it all works fine! :p


Title: Audio/Video conversation
Post by: muratasenel on August 08, 2008, 03:56:16 pm
Quote from: "kakaroto"
hello all,
I have *almost* finished my work... I need one more day and it should be done.. BUT, I don't have that one more day, I'm leaving for the airport in a few hours, and I won't be back until september... hopefully Ollivier (farsight2 maintainer) will try to keep this in a stable form, so you guys can have fun testing this during the month I'll be away...
In theory, the latest GIT now should work fine, make sure you 'git pull' to get the latest changes on both libnice and farsight2.. there might be bugs though, so let's hope it all works fine! :p


Take care and have a nice vacation :)


Title: Audio/Video conversation
Post by: Alfredo on August 08, 2008, 10:34:26 pm
Just had a go at installing farsight2 too, and ran into a number of problems of course. Most of them were due to not having the devel packages, which still confuses me (why aren't the errors understandable?). I encountered one problem while trying to "make" libnice, namely "format not a string literal and no format arguments". This was in upd-client.c, line 73.
I believe this is due to the fact that -Wformat=2 in Ubuntu 8.10? Anyway, I think I fixed it by changing it to g_print "("%s", buf);".

EDIT - Just finished the installation and while the console doesn't give any errors with "package require Farsight", the Wizard complains that Farsight isn't loaded. Not that it really matters, I don't even have a webcam and I can't get my sound/mic to work properly in Ubuntu 8.10.


Title: Libnice compile doesn't work
Post by: db2912 on August 11, 2008, 08:51:26 am
Hello there,

After many researches, I still have problems to compile libnice. So I decided to post in this forum. Thank you in advance for your help.

When I run ./autogen.sh --prefix=/usr I have the following :

Code:
didier@bureau:/opt/libnice$ sudo ./autogen.sh --prefix=/usr
libtoolize: putting macros in AC_CONFIG_MACRO_DIR, `m4'.
libtoolize: You should add the contents of `m4/libtool.m4' to `aclocal.m4'.
libtoolize: You should add the contents of `m4/ltoptions.m4' to `aclocal.m4'.
libtoolize: You should add the contents of `m4/ltsugar.m4' to `aclocal.m4'.
libtoolize: You should add the contents of `m4/ltversion.m4' to `aclocal.m4'.
libtoolize: You should add the contents of `m4/lt~obsolete.m4' to `aclocal.m4'.
libtoolize: Consider adding `-I m4' to ACLOCAL_AMFLAGS in Makefile.am.
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for gawk... gawk
checking whether make sets $(MAKE)... yes
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... gcc3
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for AIX... no
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for a sed that does not truncate output... /bin/sed
checking for ld used by gcc... /usr/bin/ld
checking if the linker (/usr/bin/ld) is GNU ld... yes
checking for /usr/bin/ld option to reload object files... -r
checking for BSD-compatible nm... /usr/bin/nm -B
checking whether ln -s works... yes
checking how to recognize dependent libraries... pass_all
checking dlfcn.h usability... yes
checking dlfcn.h presence... yes
checking for dlfcn.h... yes
checking for g++... g++
checking whether we are using the GNU C++ compiler... yes
checking whether g++ accepts -g... yes
checking dependency style of g++... gcc3
checking how to run the C++ preprocessor... g++ -E
checking for g77... no
checking for xlf... no
checking for f77... no
checking for frt... no
checking for pgf77... no
checking for cf77... no
checking for fort77... no
checking for fl32... no
checking for af77... no
checking for xlf90... no
checking for f90... no
checking for pgf90... no
checking for pghpf... no
checking for epcf90... no
checking for gfortran... no
checking for g95... no
checking for xlf95... no
checking for f95... no
checking for fort... no
checking for ifort... no
checking for ifc... no
checking for efc... no
checking for pgf95... no
checking for lf95... no
checking for ftn... no
checking whether we are using the GNU Fortran 77 compiler... no
checking whether  accepts -g... no
checking the maximum length of command line arguments... 98304
checking command to parse /usr/bin/nm -B output from gcc object... ok
checking for objdir... .libs
checking for ar... ar
checking for ranlib... ranlib
checking for strip... strip
checking if gcc supports -fno-rtti -fno-exceptions... no
checking for gcc option to produce PIC... -fPIC
checking if gcc PIC flag -fPIC works... yes
checking if gcc static flag -static works... yes
checking if gcc supports -c -o file.o... yes
checking whether the gcc linker (/usr/bin/ld) supports shared libraries... yes
checking whether -lc should be explicitly linked in... no
checking dynamic linker characteristics... GNU/Linux ld.so
checking how to hardcode library paths into programs... immediate
checking whether stripping libraries is possible... yes
checking if libtool supports shared libraries... yes
checking whether to build shared libraries... yes
checking whether to build static libraries... yes
configure: creating libtool
appending configuration tag "CXX" to libtool
checking for ld used by g++... /usr/bin/ld
checking if the linker (/usr/bin/ld) is GNU ld... yes
checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes
checking for g++ option to produce PIC... -fPIC
checking if g++ PIC flag -fPIC works... yes
checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes
checking dynamic linker characteristics... GNU/Linux ld.so
(cached) (cached) checking how to hardcode library paths into programs... immediate
appending configuration tag "F77" to libtool
checking for C/C++ restrict keyword... __restrict
checking for variable-length arrays... yes
checking whether to enable assertions... yes
checking for stdbool.h that conforms to C99... yes
checking for _Bool... yes
checking arpa/inet.h usability... yes
checking arpa/inet.h presence... yes
checking for arpa/inet.h... yes
checking net/in.h usability... no
checking net/in.h presence... no
checking for net/in.h... no
checking ifaddrs.h usability... yes
checking ifaddrs.h presence... yes
checking for ifaddrs.h... yes
checking for clock_gettime in -lrt... yes
checking for poll... yes
checking for pkg-config... /usr/bin/pkg-config
checking pkg-config is at least version 0.9.0... yes
checking for OPENSSL... yes
checking for GLIB... yes
checking for GST... yes
configure: creating ./config.status
config.status: creating Makefile
config.status: creating address/Makefile
config.status: creating agent/Makefile
config.status: creating tests/Makefile
config.status: creating stun/Makefile
config.status: creating stun/tests/Makefile
config.status: creating stun/tools/Makefile
config.status: creating local/Makefile
config.status: creating udp/Makefile
config.status: creating nice/Makefile
config.status: creating nice/nice.pc
config.status: creating random/Makefile
config.status: creating gst/Makefile
config.status: creating config.h
config.status: config.h is unchanged
config.status: executing depfiles commands


When I run make, I have the following:

Code:
didier@bureau:/opt/libnice$ sudo make
cd . && /bin/bash /opt/libnice/missing --run autoheader
rm -f stamp-h1
touch config.h.in
cd . && /bin/bash ./config.status config.h
config.status: creating config.h
config.status: config.h is unchanged
make  all-recursive
make[1]: entrant dans le répertoire « /opt/libnice »
Making all in address
make[2]: entrant dans le répertoire « /opt/libnice/address »
if /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I..    -Wall -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -pthread -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include   -g -O2 -MT address.lo -MD -MP -MF ".deps/address.Tpo" -c -o address.lo address.c; \
then mv -f ".deps/address.Tpo" ".deps/address.Plo"; else rm -f ".deps/address.Tpo"; exit 1; fi
../libtool: line 467: CDPATH: command not found
../libtool: line 1145: func_opt_split: command not found
libtool: Version mismatch error.  This is libtool 2.2.4, but the
libtool: definition of this LT_INIT comes from an older release.
libtool: You should recreate aclocal.m4 with macros from libtool 2.2.4
libtool: and run autoconf again.
make[2]: *** [address.lo] Erreur 1
make[2]: quittant le répertoire « /opt/libnice/address »
make[1]: *** [all-recursive] Erreur 1
make[1]: quittant le répertoire « /opt/libnice »
make: *** [all] Erreur 2


What's wrong ?


Title: Audio/Video conversation
Post by: Alfredo on August 11, 2008, 06:35:57 pm
I would assume you don't have libtool 2.2.4? Check your distro's packet manager, Ubuntu has v2.2.4-0ubuntu3 as the latest version.


Title: Audio/Video conversation
Post by: db2912 on August 11, 2008, 08:22:06 pm
Yes I have libtool 2.2.4.
When I run libtool --version, I have :

Code:
didier@bureau:/opt/libnice$ libtool --version
ltmain.sh (GNU libtool) 2.2.4
Written by Gordon Matzigkeit <gord@gnu.ai.mit.edu>, 1996

Copyright (C) 2008 Free Software Foundation, Inc.
This is free software; see the source for copying conditions.  There is NO
warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.


It seems that the problem is due to aclocal.m4 but I have no idea how to fix it.

Code:
libtool: Version mismatch error.  This is libtool 2.2.4, but the
libtool: definition of this LT_INIT comes from an older release.
libtool: You should recreate aclocal.m4 with macros from libtool 2.2.4
libtool: and run autoconf again.


Any suggestion ?


Title: Audio/Video conversation
Post by: Phil on August 11, 2008, 08:56:12 pm
autoreconf -vfi doesn't work ?


Title: Audio/Video conversation
Post by: db2912 on August 12, 2008, 06:52:15 am
Unfortunately, it doesn't.

When I run autoreconf -vfi, I get :

Code:
didier@bureau:/opt/libnice$ sudo autoreconf -vfi
[sudo] password for didier:
autoreconf: Entering directory `.'
autoreconf: configure.ac: not using Gettext
autoreconf: running: aclocal --force
autoreconf: configure.ac: tracing
autoreconf: running: libtoolize --copy --force
libtoolize: putting auxiliary files in `.'.
libtoolize: copying file `./ltmain.sh'
libtoolize: You should add the contents of the following files to `aclocal.m4':
libtoolize:   `/usr/local/share/aclocal/libtool.m4'
libtoolize:   `/usr/local/share/aclocal/ltoptions.m4'
libtoolize:   `/usr/local/share/aclocal/ltversion.m4'
libtoolize:   `/usr/local/share/aclocal/ltsugar.m4'
libtoolize:   `/usr/local/share/aclocal/lt~obsolete.m4'
libtoolize: Consider adding `AC_CONFIG_MACRO_DIR([m4])' to configure.ac and
libtoolize: rerunning libtoolize, to keep the correct libtool macros in-tree.
libtoolize: Consider adding `-I m4' to ACLOCAL_AMFLAGS in Makefile.am.
autoreconf: running: /usr/bin/autoconf --force
autoreconf: running: /usr/bin/autoheader --force
autoreconf: running: automake --add-missing --copy --force-missing
configure.ac:28: installing `./config.guess'
configure.ac:28: installing `./config.sub'
autoreconf: Leaving directory `.'


I'm a little puzzled by "You should add the contents of the following files to aclocal.m4...".
How can I do that ?


Title: Audio/Video conversation
Post by: m33ts4k0z on August 12, 2008, 07:12:28 am
Hello guys,

I'm new to the whole linux thing but with many great tutorials I've managed to get into the point.


I love amsn and the only big drawback it has is the audio calls. I followed the instructions at the wiki on how to implement this feature but it didn't help me that much.

However following kakaroto's instructions on page 12 of this thread made me able to compile everything correctly. amsn recognises the existance of farsight2 when I ./configure but when i start the program it says it's not installed. After shift+ctrl+C this is what I get:

Code:
(amsn) 4 % package require Farsight
0.1
(amsn) 5 % farsight Prepare 1
Error while creating new session (0): Could not create GstRtpBin


I have every dependency installed at at least the minimum version.

If you need any other info let me know, it's just that I'm not such an expert with Linux.


Thanks in advance


Title: Audio/Video conversation
Post by: db2912 on August 12, 2008, 09:53:22 am
Hello,

Eventually I succeded in compiling libnice.
The issue was due to a bad version of libtool : I had to come back to libtool 1.5.26 in order to fix the problem.

I installed farsight2 as well and amsn has recognised farsight2 while compiling.

However, like m33ts4k0z, the audio/video assistant still says that farsight is not correctly installed.

After Ctrl+Shift+C, I obtain :

Code:
(amsn) 1 % package require Farsight
0.1
(amsn) 2 % farsight Prepare 1
Couldn't create fsrtpconference
(amsn) 3 %


I am sure I didn't forget the --prefix=/usr and I have the last version of each dependencies.

What's wrong again ?


Title: Audio/Video conversation
Post by: muratasenel on August 12, 2008, 12:46:30 pm
Quote from: "m33ts4k0z"

Code:
(amsn) 4 % package require Farsight
0.1
(amsn) 5 % farsight Prepare 1
Error while creating new session (0): Could not create GstRtpBin



I assume the reason is that gst-plugins-farsight is not compiled with "--enable-jrtplib".
You can try to re-compile gst-plugins-farsigh with --enable-jrtplib parameter. But not sure if it works...


Title: Audio/Video conversation
Post by: m33ts4k0z on August 12, 2008, 01:12:13 pm
Quote from: "muratasenel"
Quote from: "m33ts4k0z"

Code:
(amsn) 4 % package require Farsight
0.1
(amsn) 5 % farsight Prepare 1
Error while creating new session (0): Could not create GstRtpBin



I assume the reason is that gst-plugins-farsight is not compiled with "--enable-jrtplib".
You can try to re-compile gst-plugins-farsigh with --enable-jrtplib parameter. But not sure if it works...


Thanks for you reply mate but no this didn't do the trick.

Btw, last night I used a deb package with the version 0.12.9 of the farsight plugin and I could see it in the synaptic. Now that I compiled from source myself I can't see it there anymore. Is that a problem meaning that it's not installed?

EDIT: From the terminal I also get the following error:
Code:
(<unknown>:15947): GLib-GObject-CRITICAL **: g_type_class_unref: assertion `g_class != NULL' failed


Title: What is this about? Please help
Post by: jzhou on August 24, 2008, 03:11:45 pm
I am using the last svn amsn and farsight2, and yet I got these outputs when initiating an audio conversation. Surprisingly video conversation is fine.

--------------------------------------------------------
can't read "sdp": no such variable
    while executing
"return $sdp"
    (procedure "::SIPConnection::Snit_methodBuildIceCandidates" line 17)
    invoked from within
"$self BuildIceCandidates $local $remote"
    (procedure "::SIPConnection::Snit_methodBuildSDP" line 34)
    invoked from within
"$self BuildSDP"
    ("OK" arm line 4)
    invoked from within
"switch -- $status {
         "RINGING" {
            set status 180
            set content_type ""
            set content ""
         }
         "UNAVAILABLE" {
            set status 480
            set co..."
    (procedure "::SIPConnection::Snit_methodAnswerInvite" line 6)
    invoked from within
"$sip AnswerInvite $callid OK"
    (procedure "::MSNSIP::AcceptInvite" line 8)
    invoked from within
"::MSNSIP::AcceptInvite $sip $callid"
    (procedure "::amsn::AcceptSIPCall" line 24)
    invoked from within
"::amsn::AcceptSIPCall xxx@sad-amsn-user.com ::MSNSIP::SIPConnection1 ae5eccb5e228452e9580067561ba850d"
    (command bound to event)
-----------------------------------------------
My system is Archlinux x86_64


Title: Audio/Video conversation
Post by: MastaG on August 26, 2008, 12:56:58 pm
When I checkout the latest libnice git tree with:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
I configure it with:
autoreconf -vfi
./configure --prefix=/usr
But when I try to build it, it will fail with:
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include -g -O2 -MT stun-ice.lo -MD -MP -MF .deps/stun-ice.Tpo -c stun-ice.c -o stun-ice.o >/dev/null 2>&1
mv -f .deps/stun-ice.Tpo .deps/stun-ice.Plo
/bin/sh ../libtool --tag=CC   --mode=link gcc -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include   -g -O2   -o libstun.la  stunsend.lo stunrecv.lo utils.lo unknown.lo crc32.lo hmac.lo timer.lo trans.lo bind.lo stun-ice.lo -L/usr/kerberos/lib -lssl -lcrypto -ldl -lz   -lrt
ar cru .libs/libstun.a .libs/stunsend.o .libs/stunrecv.o .libs/utils.o .libs/unknown.o .libs/crc32.o .libs/hmac.o .libs/timer.o .libs/trans.o .libs/bind.o .libs/stun-ice.o
ranlib .libs/libstun.a
creating libstun.la
(cd .libs && rm -f libstun.la && ln -s ../libstun.la libstun.la)
gcc -DHAVE_CONFIG_H -I. -I..  -I..  -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include   -g -O2 -MT stund.o -MD -MP -MF .deps/stund.Tpo -c -o stund.o stund.c
stund.c: In functie ‘listen_socket’:
stund.c:132: fout: ‘INT_MAX’ undeclared (first use in this function)
stund.c:132: fout: (Each undeclared identifier is reported only once
stund.c:132: fout: for each function it appears in.)
make[4]: *** [stund.o] Fout 1
make[4]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[3]: *** [all-recursive] Fout 1
make[3]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[2]: *** [all] Fout 2
make[2]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[1]: *** [all-recursive] Fout 1
make[1]: Map '/home/mastag/src/libnice' wordt verlaten
make: *** [all] Fout 2

I think I have all dependencies installed.
Do I need the latest git? or some branch?


Title: Audio/Video conversation
Post by: MastaG on August 26, 2008, 01:07:12 pm
Quote from: "MastaG"
When I checkout the latest libnice git tree with:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
I configure it with:
autoreconf -vfi
./configure --prefix=/usr
But when I try to build it, it will fail with:
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include -g -O2 -MT stun-ice.lo -MD -MP -MF .deps/stun-ice.Tpo -c stun-ice.c -o stun-ice.o >/dev/null 2>&1
mv -f .deps/stun-ice.Tpo .deps/stun-ice.Plo
/bin/sh ../libtool --tag=CC   --mode=link gcc -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include   -g -O2   -o libstun.la  stunsend.lo stunrecv.lo utils.lo unknown.lo crc32.lo hmac.lo timer.lo trans.lo bind.lo stun-ice.lo -L/usr/kerberos/lib -lssl -lcrypto -ldl -lz   -lrt
ar cru .libs/libstun.a .libs/stunsend.o .libs/stunrecv.o .libs/utils.o .libs/unknown.o .libs/crc32.o .libs/hmac.o .libs/timer.o .libs/trans.o .libs/bind.o .libs/stun-ice.o
ranlib .libs/libstun.a
creating libstun.la
(cd .libs && rm -f libstun.la && ln -s ../libstun.la libstun.la)
gcc -DHAVE_CONFIG_H -I. -I..  -I..  -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -I/usr/kerberos/include   -g -O2 -MT stund.o -MD -MP -MF .deps/stund.Tpo -c -o stund.o stund.c
stund.c: In functie ‘listen_socket’:
stund.c:132: fout: ‘INT_MAX’ undeclared (first use in this function)
stund.c:132: fout: (Each undeclared identifier is reported only once
stund.c:132: fout: for each function it appears in.)
make[4]: *** [stund.o] Fout 1
make[4]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[3]: *** [all-recursive] Fout 1
make[3]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[2]: *** [all] Fout 2
make[2]: Map '/home/mastag/src/libnice/stun' wordt verlaten
make[1]: *** [all-recursive] Fout 1
make[1]: Map '/home/mastag/src/libnice' wordt verlaten
make: *** [all] Fout 2

I think I have all dependencies installed.
Do I need the latest git? or some branch?


Nevermind my stupidity..
Just inserted #include <limits.h> in stun/stund.c and it worked:)


Title: Audio/Video conversation
Post by: MastaG on August 26, 2008, 01:14:40 pm
After I have built and installed the latest farsight2 git.
amsn wont like the new farsight2?
configure:6872: checking for FARSIGHT2
configure:6881: $PKG_CONFIG --exists --print-errors "\
             farsight2-$GST_MAJORMINOR = $FARSIGHT_REQUIRED"
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6885: $? = 1
configure:6902: $PKG_CONFIG --exists --print-errors "\
             farsight2-$GST_MAJORMINOR = $FARSIGHT_REQUIRED"
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6906: $? = 1
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6937: result: no


Title: Audio/Video conversation
Post by: MastaG on August 26, 2008, 01:26:02 pm
Quote from: "MastaG"
After I have built and installed the latest farsight2 git.
amsn wont like the new farsight2?
configure:6872: checking for FARSIGHT2
configure:6881: $PKG_CONFIG --exists --print-errors "\
             farsight2-$GST_MAJORMINOR = $FARSIGHT_REQUIRED"
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6885: $? = 1
configure:6902: $PKG_CONFIG --exists --print-errors "\
             farsight2-$GST_MAJORMINOR = $FARSIGHT_REQUIRED"
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6906: $? = 1
Requested 'farsight2-0.10 = 0.0.2.1' but version of Farsight2 is 0.0.3.1
configure:6937: result: no


Someone should change the: FARSIGHT_REQUIRED line in the configure file to: 0.0.3.1
EDIT: Even though farsight2 git is compiled and installed, there seems to be no farsight utility in /usr/bin ?
aMSN also wont start an audio conversation because of that...


Title: Audio/Video conversation
Post by: Kreuger on August 26, 2008, 07:39:13 pm
When I try to run the autogen for libnice, it says
Quote
autoreconf: failed to run aclocal: no such file or directory

This is on my laptop. It worked fine on my desktop.


When I run make on the farsight, I get this
Quote
ec_discovery-fs-rtp-session.Tpo"; exit 1; fi
../../gst/fsrtpconference/fs-rtp-session.c:40:34: error: gst/rtp/gstrtpbuffer.h: No such file or directory
../../gst/fsrtpconference/fs-rtp-session.c:41:35: error: gst/rtp/gstrtcpbuffer.h: No such file or directory
../../gst/fsrtpconference/fs-rtp-session.c: In function ‘_stream_known_source_packet_received’:
../../gst/fsrtpconference/fs-rtp-session.c:1266: warning: implicit declaration of function ‘gst_rtp_buffer_validate’
../../gst/fsrtpconference/fs-rtp-session.c:1268: warning: implicit declaration of function ‘gst_rtp_buffer_get_ssrc’
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: ‘GstRTCPPacket’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: (Each undeclared identifier is reported only once
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: for each function it appears in.)
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: expected ‘;’ before ‘rtcppacket’
../../gst/fsrtpconference/fs-rtp-session.c:1276: warning: implicit declaration of function ‘gst_rtcp_buffer_validate’
../../gst/fsrtpconference/fs-rtp-session.c:1278: warning: implicit declaration of function ‘gst_rtcp_buffer_get_first_packet’
../../gst/fsrtpconference/fs-rtp-session.c:1278: error: ‘rtcppacket’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1281: warning: implicit declaration of function ‘gst_rtcp_packet_get_type’
../../gst/fsrtpconference/fs-rtp-session.c:1281: error: ‘GST_RTCP_TYPE_SDES’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1283: warning: implicit declaration of function ‘gst_rtcp_packet_sdes_get_ssrc’
../../gst/fsrtpconference/fs-rtp-session.c:1286: warning: implicit declaration of function ‘gst_rtcp_packet_move_to_next’
make[3]: *** [codec_discovery-fs-rtp-session.o] Error 1
make[3]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2/tests/rtp'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2/tests'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2'
make: *** [all] Error 2
kreuger@kreuger-desktop:~/Downloads/Updates/farsight2$


Title: Audio/Video conversation
Post by: billiob on August 26, 2008, 09:30:18 pm
kakaroto is on holiday, wait for him.


Title: Audio/Video conversation
Post by: kakaroto on September 02, 2008, 11:59:11 pm
arg... too many problems.. I'll be back from holidays friday, and will need to rest, so on monday, you should expect some news from me...
1 - the 0.0.3.1 version issue is because you are not using the right branch on git... or because Tester released and rebased everything...
2 - @m33ts4k0z : you are missing gstrtpbin.. so you didn't install gstreamer-plugins-bad correctly.. make sure it's installed correctly and with the /usr prefix...
3 - @db2912 : same as above, but you didn't install the farsight2 plugins correctly...
4 - @Kreuger : you must install the autotools and gstreamer dev packages...


Title: Audio/Video conversation
Post by: Kreuger on September 03, 2008, 05:42:44 am
Quote
@Kreuger : you must install the autotools and gstreamer dev packages...
According to Synaptic, I have autotools-dev (version 20070725.1) and I also have libgstreamer0.10-dev (version 0.10.20-1).


Title: Audio/Video conversation
Post by: kakaroto on September 03, 2008, 02:25:19 pm
try installing aclocal, or typing 'aclocal' to see if it's installed...
you can also try autoreconf -vfi to see the verbose output of what happens...


Title: Audio/Video conversation
Post by: Kreuger on September 03, 2008, 03:24:27 pm
autoreconf -vfi shows

Quote
kreuger@kreuger-desktop:~/Downloads/Updates/farsight2$ autoreconf -vfi
autoreconf: Entering directory `.'
autoreconf: configure.ac: not using Gettext
autoreconf: running: aclocal --force -I common/m4
autoreconf: configure.ac: tracing
autoreconf: running: libtoolize --copy --force
autoreconf: running: /usr/bin/autoconf --force
autoreconf: running: /usr/bin/autoheader --force
autoreconf: running: automake --add-missing --copy --force-missing
autoreconf: Leaving directory `.'


I'm pretty sure I installed aclocal since the autogen works fine.


Title: Audio/Video conversation
Post by: kakaroto on September 04, 2008, 01:03:27 am
@Kreuger :
ok, so your error :
Quote
autoreconf: failed to run aclocal: no such file or directory


is not valid anymore ? it's not showing that anymore ?
did you try to relaunch ./configure ?


Title: Audio/Video conversation
Post by: Kreuger on September 04, 2008, 03:51:14 pm
No, I don't have that error anymore and no I didn't run the configure because after running autogen it says I can just run make.


Title: Audio/Video conversation
Post by: senlegen on September 15, 2008, 07:54:07 pm
A doubt about the length of the siren audio packets: on AV conversations (not SIP/RTP) the packets are always 40 bytes or multiples, but on Computer Call (SIP/RTP) not. How is decoded these audio packets? The RTP payloads are concatenated and decoded every 40 bytes? All the bytes of the RTP payload are sent to the Siren decoder or there are others informations?

Thank you!

Senlegen


Title: Audio/Video conversation
Post by: Montblanc on September 16, 2008, 12:08:46 am
I'm using Hardy and got latest amsn svn revision (10474 at the moment), but configure can't detect latest git Farsight2. I've built everything as I always did, user prefix and dependencies are fine. Can you confirm it?


Title: Audio/Video conversation
Post by: kakaroto on September 16, 2008, 06:31:28 am
Hi,
@senlegen: first, welcome to the forums, but who are you? no introduction, so I don't know, but your post clearly isn't a n00b user asking for help? Are you developing something using the siren codec? or another msn clone ?
Anyways, yes, siren uses frames of 40 bytes which represent 20ms of sound. The packets sent over RTP are concatenated frames, you'll need an RTP depayloader that will depayload the RTP packets to get you the siren data, which will then be a multiple of 40 bytes, and then you decode each frame and play.

@Montblanc : I am still currently working on libnice and farsight2, it should be ready in about two weeks, so just be patient.
My current progress is :
1 - It seems I have fixed libnice to work correctly for accounts on the same network and for networks behind a full-cone NAT
2 - For symetric NAT users (who can't be accessed), it seems to work correctly if they use WLM since we'll fallback to their TURN server
3 - I still need to take a look at peer-reflexive candidates generated by symetric NAT detection, but it *should* work fine.. I just need to test it with symetric NAT users.
4 - I have made huge progress into adding TURN support into libnice, it just needs some tweaking and some if/else cases for handling MSN compatibility.
5 - I would still need to implement channel handling for TURN sockets and auto refreshing the allocated ip/port given by the TURN server
6 - I still need to have aMSN automatically exchange a shared secret short term credential username/password over TLS (it's already there and works, just needs to be done automatically), and give the information to farsight which will relay it to libnice.

So, in short, for 95% of the users, it should now all work correctly, BUT, my main two issues are :
1 - there seems to be some memory corruption somewhere that might cause a segfault
2 - I need to finish support for TURN in order to have a stable release of libnice and farsight2.

So I repeat, be patient!

p.s:
If you still want to try it out, you can get my git repo of libnice at :
git://git.collabora.co.uk/git/user/kakaroto/nice.git
then checkout the 'origin/nice-kakaroto' branch
Then take my git repo of farsight2 at :
git://git.collabora.co.uk/git/user/kakaroto/farsight2.git
then checkout the 'origin/nice' branch, then merge over it the 'master' and the 'relay-info' branches of tester's repo : git://git.collabora.co.uk/git/user/tester/farsight2.git
then compile everything and try it out!
If it doesn't work, then do this : WAIT! :p


Title: Audio/Video conversation
Post by: Montblanc on September 16, 2008, 01:42:40 pm
Thank you very much kakaroto. As i said some posts earlier, I DON'T need audio/video conversation, I just thought I could be of some help with bugtracking, but since you do know it won't compile 'cause you're working on it, it's not a problem :).


Title: Audio/Video conversation
Post by: kakaroto on September 16, 2008, 04:38:18 pm
ok, cool then! :)


Title: Audio/Video conversation
Post by: senlegen on September 16, 2008, 07:15:43 pm
Quote from: "kakaroto"
Hi,
@senlegen: first, welcome to the forums, but who are you? no introduction, so I don't know, but your post clearly isn't a n00b user asking for help? Are you developing something using the siren codec? or another msn clone ?
Anyways, yes, siren uses frames of 40 bytes which represent 20ms of sound. The packets sent over RTP are concatenated frames, you'll need an RTP depayloader that will depayload the RTP packets to get you the siren data, which will then be a multiple of 40 bytes, and then you decode each frame and play.


Sorry, i have entered without nocking :). I'm working on a tool for playing voip calls using FFMpeg, and i am studying aMSN, MSN WebCam Recorder and other tools.
Now i have noted that the RTP packets i was analyzing have payload type 114 (msrta) (111 is siren, isn´t it?). Is there any msrta codec available? Can i force WLM to work with siren on computer call (config ou registry)?

Congratulations for you good job here!

Thanks for your help! (and sorry for my bad english.)

Senlegen


Title: Audio/Video conversation
Post by: kakaroto on September 16, 2008, 07:53:44 pm
hehe, no problem. It's nice to see a new tool being developed for this, but how will it play voip calls? is it like a recorder and you plan on recording the voip calls then replay them later ?
Yes, 114 is msrta, and I don't think there is any way for you to force siren (unless the call is made between WLM and aMSN :p) and there is no free codec available for msrta, but Ole Andre Vadla Ravnas did some work on having access to the msrta codecs for gstreamer, by using the .dll used by WLM directly. You might want to ask him your question about msrta. You'll find him hanging on the #pymsn channel on Freenode IRC.
good luck!


Title: Audio/Video conversation
Post by: farseeing on September 18, 2008, 12:00:59 pm
Got it work on Fedora 9 !
Spent hours with snack2.2 and finally gave up and tried farsight. So for those who may be helped by that, here are some tips I had to do to adapt the procedure on the beginning of this thread to my needs.

I'm using :
- fedora 9 with kernel 1.6.25 (due to graphic driver issues, hum, well...anyway)
-pulse-audio, alsa and all this stuff I don't understand anything in !
-aMSN from svn : v.10479 at this very second
- farsight 2 : 0.0.3 (had to cheat see below) --> http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
- gst-plugins : 0.12.9  --> use yum or add/remove software, all plugins available for F9 in my configured repositories (usuals + livna)
- libnice.


I've just followed the above procedure (obviously using yum instead of apt)  except for several points where I had to struggle with versions problems :

# I followed the procedure using git but the version of farsight2 there didn't seem to work.
so I downloaded farsight2-0.0.3.tar.gz (see link above), unpacked and manually copy files in farsight2-0.0.3/common from the git tree of farsight2 I created : gst-autogen.sh, gst.supp. -I have 4 files & 1 folder in this directory). I'm then using this version of farsight2 instead of the git tree version

# After the ./configure step in farsight2-0.0.3 installation and just before the make, I edited farsight2.pc and farsight2-0.10.pc and change version 0.0.3 by 0.0.3.1. then go back to the procedure and perform make and make install. (otherwise aMSN doesn't recognized farsight.).

#  (yes, don't forget to do the ldconfig as prescribed !)

# then I installed aMSN

AND THAT'S WORKIN DAMN FINE RIGHT NOW !!!!
(http://[url=http://www.postimage.org/image.php?v=aV1LK8wJ][img]http://www.postimage.org/aV1LK8wJ.jpg)[/url]
[/img]


Title: Audio/Video conversation
Post by: MastaG on September 18, 2008, 08:18:40 pm
Quote from: "farseeing"
Got it work on Fedora 9 !
Spent hours with snack2.2 and finally gave up and tried farsight. So for those who may be helped by that, here are some tips I had to do to adapt the procedure on the beginning of this thread to my needs.

I'm using :
- fedora 9 with kernel 1.6.25 (due to graphic driver issues, hum, well...anyway)
-pulse-audio, alsa and all this stuff I don't understand anything in !
-aMSN from svn : v.10479 at this very second
- farsight 2 : 0.0.3 (had to cheat see below) --> http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
- gst-plugins : 0.12.9  --> use yum or add/remove software, all plugins available for F9 in my configured repositories (usuals + livna)
- libnice.


I've just followed the above procedure (obviously using yum instead of apt)  except for several points where I had to struggle with versions problems :

# I followed the procedure using git but the version of farsight2 there didn't seem to work.
so I downloaded farsight2-0.0.3.tar.gz (see link above), unpacked and manually copy files in farsight2-0.0.3/common from the git tree of farsight2 I created : gst-autogen.sh, gst.supp. -I have 4 files & 1 folder in this directory). I'm then using this version of farsight2 instead of the git tree version

# After the ./configure step in farsight2-0.0.3 installation and just before the make, I edited farsight2.pc and farsight2-0.10.pc and change version 0.0.3 by 0.0.3.1. then go back to the procedure and perform make and make install. (otherwise aMSN doesn't recognized farsight.).

#  (yes, don't forget to do the ldconfig as prescribed !)

# then I installed aMSN

AND THAT'S WORKIN DAMN FINE RIGHT NOW !!!!
(http://[url=http://www.postimage.org/image.php?v=aV1LK8wJ][img]http://www.postimage.org/aV1LK8wJ.jpg)[/url]
[/img]


Hell yes!
Same here :)

So to sum things up:

Install the following packages from the repo's:
Code:

gstreamer-plugins-bad-0.10.7-1.lvn9.i386
gstreamer-0.10.19-1.fc9.i386
gstreamer-plugins-good-0.10.8-8.fc9.i386
gstreamer-tools-0.10.19-1.fc9.i386
gstreamer-plugins-pulse-0.9.5-0.5.svn20070924.fc9.i386
gstreamer-plugins-base-devel-0.10.19-2.fc9.i386
gstreamer-plugins-bad-extras-0.10.7-1.lvn9.i386
gstreamer-plugins-base-0.10.19-2.fc9.i386
gstreamer-plugins-bad-devel-0.10.7-1.lvn9.i386
gstreamer-devel-0.10.19-1.fc9.i386
gstreamer-plugins-good-devel-0.10.8-8.fc9.i386
gstreamer-plugins-ugly-0.10.8-1.lvn9.i386


Install the latest gst-plugins-farsight (the one from the repo's is outdated).
Code:

wget http://farsight.freedesktop.org/releases/gst-plugins-farsight/gst-plugins-farsight-0.12.9.tar.gz
tar zxfv gst-plugins-farsight-0.12.9.tar.gz
cd gst-plugins-farsight-0.12.9
./configure --prefix=/usr
make
sudo make install
cd ..


Obtain and install libnice (be sure to have git installed: sudo yum install git)
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git checkout -b nice-kakaroto origin/nice-kakaroto
./autogen.sh --prefix=/usr
make
sudo make install
cd ..


Install farsight2
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
cd farsight2
git checkout -b nice origin/nice
cd ..
wget http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
tar zxfv farsight2-0.0.3.tar.gz
cd farsight2-0.0.3
cp -f ../farsight2/common/check.mak ./common/
cp -f ../farsight2/common/gst.supp ./common/
cp -f ../farsight2/common/gst-autogen.sh ./common/
cp -f ../farsight2/common/gtk-doc.mak ./common/
./autogen --prefix=/usr --disable-python
nano -w farsight2.pc

Now change the version from 0.0.3 to 0.0.3.1
Press CTRL+O to save and CTRL+X to quit nano.
Code:

make
sudo make install
sudo /sbin/ldconfig
cd ..


Voila, check out the latest svn of amsn and you should be able to make voice calls:)


Title: Audio/Video conversation
Post by: Kreuger on September 19, 2008, 12:50:43 am
I just tried MastaG's way of doing it. Got to the part of make for farsight and still getting the error
Quote

   then mv -f ".deps/codec_discovery-fs-rtp-session.Tpo" ".deps/codec_discovery-fs-rtp-session.Po"; else rm -f ".deps/codec_discovery-fs-rtp-session.Tpo"; exit 1; fi
../../gst/fsrtpconference/fs-rtp-session.c:40:34: error: gst/rtp/gstrtpbuffer.h: No such file or directory
../../gst/fsrtpconference/fs-rtp-session.c:41:35: error: gst/rtp/gstrtcpbuffer.h: No such file or directory
../../gst/fsrtpconference/fs-rtp-session.c: In function ‘_stream_known_source_packet_received’:
../../gst/fsrtpconference/fs-rtp-session.c:1266: warning: implicit declaration of function ‘gst_rtp_buffer_validate’
../../gst/fsrtpconference/fs-rtp-session.c:1268: warning: implicit declaration of function ‘gst_rtp_buffer_get_ssrc’
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: ‘GstRTCPPacket’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: (Each undeclared identifier is reported only once
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: for each function it appears in.)
../../gst/fsrtpconference/fs-rtp-session.c:1274: error: expected ‘;’ before ‘rtcppacket’
../../gst/fsrtpconference/fs-rtp-session.c:1276: warning: implicit declaration of function ‘gst_rtcp_buffer_validate’
../../gst/fsrtpconference/fs-rtp-session.c:1278: warning: implicit declaration of function ‘gst_rtcp_buffer_get_first_packet’
../../gst/fsrtpconference/fs-rtp-session.c:1278: error: ‘rtcppacket’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1281: warning: implicit declaration of function ‘gst_rtcp_packet_get_type’
../../gst/fsrtpconference/fs-rtp-session.c:1281: error: ‘GST_RTCP_TYPE_SDES’ undeclared (first use in this function)
../../gst/fsrtpconference/fs-rtp-session.c:1283: warning: implicit declaration of function ‘gst_rtcp_packet_sdes_get_ssrc’
../../gst/fsrtpconference/fs-rtp-session.c:1286: warning: implicit declaration of function ‘gst_rtcp_packet_move_to_next’
make[3]: *** [codec_discovery-fs-rtp-session.o] Error 1
make[3]: Leaving directory `/home/kreuger/farsight2-0.0.3/tests/rtp'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/kreuger/farsight2-0.0.3/tests'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/kreuger/farsight2-0.0.3'
make: *** [all] Error 2
kreuger@kreuger-desktop:~/farsight2-0.0.3$


Title: Audio/Video conversation
Post by: farseeing on September 19, 2008, 08:58:50 am
Quote
I just tried MastaG's way of doing it. Got to the part of make for farsight and still getting the error

Hey Kreuger, are gstreamer pkgs well installed ?
 
when I try :
locate gstrtcpbuffer.h
I get : /usr/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h

Hey aMsn guys,
now that all is workin fine I have some silly subsidiary GUI questions (sorry for that !) :
Why do I have to send my webcam, accept the webcam of the other then start an audio call ? Is there a way to perform all this one shot ?
When I want to close I have a warning msg that says : "close the conversation first" (Hell ! where is it you can close the conversation ? can't find it in the menus !)

My webcam image quality is quite poor (in comparison with the same cam working on one of those commercial OS for cowards who don't dare to spend days compiling and searchin the Internet each time they want to get a new functionality) . Is there a way to improve that ? Are there some farsight2 fine tunes I could use ?


Title: Audio/Video conversation
Post by: kakaroto on September 19, 2008, 09:11:11 am
hey guys calm downnnnnnnnn!
I just said I'll let you know when to try it! lol
but yes, it should work fine now... BUT I'm still modifying stuff, AND you need to use the 'relay-info' branch of farsight2, not the 'nice' one. In any case, we'll soon have a release of libnice and farsight2, so just be a bit more patient...
A bit of news, yesterday, I was able to make a call with aMSN over a restricted network (I had blocked my internal and external ips), the same for WLM, so they both had to use a relay server, and the communication was going like this :
amsn <-> relay server 1 <-> relay server 2 <-> WLM
And the communication was flawless! So it's all good! :)
Anyways, Kreuger, you still have the same problem and it's still because you don't have gstreamer include files installed properly!
and farseeing, this is an audio call, not an audio+video call, so you have to send/receive webcam independently. What you need is the video conference feature, if you read the first page of this thread, I'm talking about it, and it's not using farsight at all, there's a branch of amsn that partially supports it (uses ffmpeg, and you can receive audio and video, but you don't send anything). But I would not suggest you use it because the protocol is soooo crappy, it won't work very good!


Title: Audio/Video conversation
Post by: farseeing on September 19, 2008, 12:14:39 pm
Arghh ! damn ! Not using farsight at all ? Are you kiddin ? you can't imagine how long I spend to get it workin like it is right now ! And how proud I'm right now, chatting with those windows people from my fedora box.
yep I read the firsts posts but my own personal internal processor doesn't have enough neurones to get it all clear. (too much resources allocated to the farsight install)

So, I'll stay like this and try to convince those msn/wlm users that they'll have to have it independently to talk with me !   :D
It was just in case, in fact. It's working very good like this.

Anyway thanks a lot for the answer and for the great job. Sorry for patience issues.


Title: Audio/Video conversation
Post by: kakaroto on September 19, 2008, 08:42:49 pm
hehe, blame microsoft :p


Title: Audio/Video conversation
Post by: kakaroto on September 20, 2008, 02:37:11 am
by the way, for those who have tested aMSN with farsight and libnice... care to comment ? how is it ? is it working correctly for you ? does it work with everyone or it works with 'some' users ? how is the sound quality ? any bugs I should be aware of ? etc...
I know that there seems to be a nasty memory corruption somewhere that I just can't seem to find :@:@:@:@ so if you get segfaults, it's "normal"... if you don't get any, tell me that you don't! apart from that, it should all work just fine now!


Title: Audio/Video conversation
Post by: farseeing on September 20, 2008, 08:25:57 am
comment : AWESOME !
( I didn't try the relay version of aMSN.still usig svn 10479)
audio is just fine : duplex mode even seems less jerked than with WLM.
video reception is perfect.
emission fluidity is correct. quality is not very good but not sure it comes from farsight.
stability : no segfault, no problems. I've stayed more than hour in conversation and it' was all fine.
didn't try it with others aMSN users, just WLM ones.
Kakaroto, if you need some logs or printscreens just tell and provide an email address. would be a pleasure.


Title: Audio/Video conversation
Post by: khaan on September 20, 2008, 11:59:07 am
Hello and thanks for all the efforts and great work!

I just have a problem when compiling farsight2. I am under Debian Testing, but switched to Unstable to download all the needed stuff. I think I've followed the instructions correctly, but I get:
Code:
fs-nice-stream-transmitter.c: In function ‘fs_nice_stream_transmitter_class_init’:
fs-nice-stream-transmitter.c:295: error: ‘NICE_COMPATIBILITY_ID19’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:295: error: (Each undeclared identifier is reported only once
fs-nice-stream-transmitter.c:295: error: for each function it appears in.)
make[3]: *** [libnice_transmitter_la-fs-nice-stream-transmitter.lo] Error 1
make[3]: Leaving directory `/tmp/amsn/farsight2/transmitters/nice'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/tmp/amsn/farsight2/transmitters'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/tmp/amsn/farsight2'
make: *** [all] Error 2


Does it have anything to do with the instruction no more to use the 'nice' branch as said on the wiki, but the 'relay-info'? If so, could you please update the wiki? If not... well any help appreciated!

Thanks in advance


Title: Audio/Video conversation
Post by: billiob on September 20, 2008, 12:56:31 pm
The stuff is still in heavy development, so don't expect it to work. We won't update the wiki until Kakaroto says the development is complete.
Don't read the wiki, it's a bit outdated, read the whole thread.


Title: Audio/Video conversation
Post by: khaan on September 20, 2008, 05:53:07 pm
I understand that full well and don't expect anything.
I'm just trying to do what I can, which might often be limited to point out things that do not work for me, in case it might help.


Title: Error compiling Farsight2
Post by: jamaj on September 20, 2008, 08:49:29 pm
fs-nice-stream-transmitter.c: Na função ‘fs_nice_stream_transmitter_class_init’:
fs-nice-stream-transmitter.c:295: erro: ‘NICE_COMPATIBILITY_ID19’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:295: erro: (Each undeclared identifier is reported only once
fs-nice-stream-transmitter.c:295: erro: for each function it appears in.)
make[3]: ** [libnice_transmitter_la-fs-nice-stream-transmitter.lo] Erro 1
make[3]: Saindo do diretório `/usr/src/farsight2/transmitters/nice'
make[2]: ** [all-recursive] Erro 1
make[2]: Saindo do diretório `/usr/src/farsight2/transmitters'
make[1]: ** [all-recursive] Erro 1
make[1]: Saindo do diretório `/usr/src/farsight2'
make: ** [all] Erro 2


Title: Audio/Video conversation
Post by: kakaroto on September 21, 2008, 05:25:12 am
khan, jamaj, read what I just said, you need a different branch!
instructions have been updated in the wiki. Use my own git repository instead.

Quote from: "farseeing"
comment : AWESOME !
( I didn't try the relay version of aMSN.still usig svn 10479)
audio is just fine : duplex mode even seems less jerked than with WLM.
video reception is perfect.
emission fluidity is correct. quality is not very good but not sure it comes from farsight.
stability : no segfault, no problems. I've stayed more than hour in conversation and it' was all fine.
didn't try it with others aMSN users, just WLM ones.
Kakaroto, if you need some logs or printscreens just tell and provide an email address. would be a pleasure.

hehe, thanks, glad the sound works great and I'm glad the audio is working fine for you! I think the memory corruption happens mostly when you 'hang up'.. but it won't always happen, so sometimes I had to call/cancel/call/cancel like 10 times to get it.. but it's been a while since I saw it, so I'm hoping it got fixed with my last changes!
video reception/sending is a completely different feature (old and known to be stable).
I don't need logs or anything, thanks for suggesting, it seems to be all fine now! If you ever get an error or a segfault though, (can't connect with someone for example) and you can reproduce it everytime, then the output on the terminal when amsn is compiled with --enable-debug would be helpful!

By the way, I have updated the instructions on the wiki and updated my branches for the latest code.. but there's a little bug with farsight2 when compiling, so to get it to compile, just configure it with --disable-gtk-doc.. this hopefully will be fixed and won't be needed starting monday!
have fun with that!


Title: Audio/Video conversation
Post by: Brian on September 21, 2008, 09:32:26 am
kakaroto, glad I found the right thread. The instructions here were different to what I found somewhere else?
I have followed the instructions but unfortunately I still get, when running maake,

cc1: warnings being treated as errors
agent.c: In function 'nice_agent_add_stream':
agent.c:755: warning: implicit declaration of function 'g_warn_if_fail'
agent.c:755: warning: nested extern declaration of 'g_warn_if_fail'
make[2]: *** [agent.lo] Error 1
make[2]: Leaving directory `/Documents/SOURCE/libnice/agent'
make[1]: *** [install] Error 2
make[1]: Leaving directory `/Documents/SOURCE/libnice/agent'
make: *** [install-recursive] Error 1

I am not on a debian system but I ran the git stuff anyway.

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
Initialized empty Git repository in /Documents/SOURCE/libnice/.git/
remote: Counting objects: 4772, done.
remote: Compressing objects: 100% (1533/1533), done.
Indexing 4772 objects...
remote: Total 4772 (delta 3578), reused 4304 (delta 3219)
 100% (4772/4772) done
Resolving 3578 deltas...
 100% (3578/3578) done

libnice #  git checkout origin/nice-kakaroto
Note: moving to "origin/nice-kakaroto" which isn't a local branch
If you want to create a new branch from this checkout, you may do so
(now or later) by using -b with the checkout command again. Example:
  git checkout -b <new_branch_name>
HEAD is now at cdd0db3... use the NO_INDICATION_AUTH flag for msn turn support

I then run
/autogen.sh --prefix=/usr

and get at the start

/usr/share/aclocal/progsreiserfs.m4:13: warning: underquoted definition of AC_CHECK_LIBREISERFS
/usr/share/aclocal/progsreiserfs.m4:13:   run info '(automake)Extending aclocal'
/usr/share/aclocal/progsreiserfs.m4:13:   or see http://sources.redhat.com/automake/automake.html#Extending-aclocal
/usr/share/aclocal/progsreiserfs.m4:13: warning: underquoted definition of AC_CHECK_LIBREISERFS
/usr/share/aclocal/progsreiserfs.m4:13:   run info '(automake)Extending aclocal'
/usr/share/aclocal/progsreiserfs.m4:13:   or see http://sources.redhat.com/automake/automake.html#Extending-aclocal

the rest seems to run ok, but as quoted above the make fails and libnice does not get built.

Is this expected to work on a non debian system?
I cannot find any reference to gtk-doc-tools  in Suse.


Title: Audio/Video conversation
Post by: kakaroto on September 21, 2008, 01:10:38 pm
Hi Brian, yes it should work on non-debian... the only debian specific stuff is my instructions with 'apt-get' that speeds up the process!
Anyways, the error you get is weird.. and seems to be incomplete... or maybe you just didn't follow the wiki page http://www.amsn-project.net/wiki/Farsight#Linux
Make sure you have the minimum required dependencies!!!
I'm guessing the missing 'g_warn_if_fail' is because you have an old version of glib! Can you check which version of glib you are using and tell me what it is ?
Thanks!

EDIT
ok.. my bad, the wiki said that glib required was 2.14, and I just checked : http://library.gnome.org/devel/glib/unstable/glib-Warnings-and-Assertions.html#g-warn-if-fail
which says for "g_warn_if_fail"
Quote
Since 2.16

So, I've now updated the wiki to say that glib 2.16 is the required version.
Please update your glib installation to 2.16...

EDIT2
I have just pushed a 'fix' for this, I removed the g_warn_if_fail, so please update your libnice source
Code:
git pull origin nice-kakaroto

You shouldn't have that warning anymore.. BUT I cannot assure you that everything will work just fine.. it probably will, but all my tests were done on glib 2.16.. so if you encounter any error, please upgrade your glib installation and retry before reporting any problems.
Thanks!


Title: aMSN does not recognize Farsight
Post by: jamaj on September 21, 2008, 09:50:08 pm
Kakaroto,

I did what you said and recompiled farsight2 and libnice. They compiled ok.
I used --prefix=/usr and sudo ldoconfig after
compiling and doing "sudo make install".

aMsn now does not recognize the presence of Farsight and/or libnice.

What could be happening?

Thanks in  advance.

PS: I'm compiling in Ubuntu 8.04.


Title: Audio/Video conversation
Post by: kakaroto on September 21, 2008, 09:55:05 pm
@jamaj : re-read http://www.amsn-project.net/forums/viewtopic.php?p=32255#32255
and especially read ALL the steps you need to do "in case it doesn't work" at the end.
no info = no help.


Title: Audio/Video conversation
Post by: Brian on September 22, 2008, 05:11:19 am
Ok, I can confirm that I am using glib 2.14.
I pulled the fix and the original error is gone but I now get

 cc1: warnings being treated as errors
gstnicesrc.c: In function 'gst_nice_src_get_type':
gstnicesrc.c:103: warning: passing argument 1 of 'g_once_init_enter' from incompatible pointer type
gstnicesrc.c:103: warning: passing argument 1 of 'g_once_init_leave' from incompatible pointer type
gstnicesrc.c:103: warning: passing argument 2 of 'g_once_init_leave' makes pointer from integer without a cast
make[2]: *** [gstnicesrc.lo] Error 1
make[2]: Leaving directory `/Documents/SOURCE/libnice/gst'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/Documents/SOURCE/libnice'
make: *** [all] Error 2

This means we have two ways to go from here.
1) I can stick with 2.14 and keep sending you the issues, if you wish to fix them or
2) I can try and find a glib 2.16 and see how I go.

Happy to go either way.


Title: Audio/Video conversation
Post by: kakaroto on September 22, 2008, 06:52:35 am
Hi again,
yeah, I guessed it was a glib 2.14 issue... and glad it got fixed... about your problem, this seems quite odd as it's not coming from my code, it's coming from gstreamer, AND it should work on 2.14... there's even a check to make sure glib 2.14 is used, otherwise it would call something else to stay compatible...
so I don't know what the problem is...
tomorrow I'll look into this, ask the gstreamer people if they know what that error is about...
I'll keep you informed!

now you have two choices :
1 - you want it to work NOW, so you just upgrade to 2.16 and make it work..
2 - you wait patiently for us to fix us, as it would be nice if the code was 2.14-compatible...


EDIT: I would suggest you try to upgrade your gcc compiler to the latest version you can find, it might be a compiler problem rather than something else... The version I use is gcc 4.1.3


Title: Audio/Video conversation
Post by: Brian on September 22, 2008, 08:04:07 am
Hi, I have managed to find a new version of glib, glib2-2.18.1-29.1.i586.rpm, but I will hold off installing it just in case you would like me to test something.
I am not sure how to communicate with the gstreamer people so I will just wait until you can get back to me. The gcc I am running is
gcc (GCC) 4.2.1 (SUSE Linux) so I guess this is ok.


EDIT: I have just tried to signin again and it now hangs during the signon process. So it looks like I will have to update the glib to proceed.
Just to be clear I am running AMSN 0.98b 2007/12/25 SVN version 10399


Title: Audio/Video conversation
Post by: kakaroto on September 22, 2008, 05:12:02 pm
humm.. ok, well, glad you found a glib 2.18 rpm!
I have just installed glib 2.14 on my system and tried it out, and it did compile correctly with glib-2.14 (at least, libnice did, farsight2 is failing for some other reason).
The only clue I might have is about wrong/corrupted include files.. make sure you have the exact matching glib 2.14 dev package on your system.
Anyways, your case is probably abnormal, since it worked for me, and everything in the code says that it should work just fine.
I think we can just forget this issue unless it pops up again, so just go ahead and install glib 2.18 for your system (along with the glib-dev package)
Have fun!

EDIT : just saw your edit... amsn doesn't use glib, and as long as you couldn't get farsight installed, it won't be able to run, so the fact that it freezes on signon is probably caused by something completely different (snack trying to play a sound but /dev/dsp is occupied?)


Title: Audio/Video conversation
Post by: farseeing on September 22, 2008, 07:11:55 pm
Hi ,
I've got a problem transmitting sound with one of my contact who is currently using WLM 8.5.13.02.10-18
chat and video are ok, however.
I can manage audio calls with my other contacts who use this version of WLM.
This contact (with whom I can't talk) can talk with other WLM users without any pb.
I first thought about firewall issues. What I'd be glad to know is  : Would using farsight2 'relay' branch instead of the one I'm currently using (downloaded from website) would solve this issue ?
(don't want to break all my aMSN setup if I'm on bad tracks)
here below a log of ./amsn --enable-debug during a try :

Quote
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #701: 0 frozen, 6 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #751: 0 frozen, 6 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #801: 0 frozen, 6 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #851: 0 frozen, 6 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #901: 0 frozen, 6 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe3eb70
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe6ed50
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe5ecb0
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe8ee90
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe4ec10
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : Retransmissions failed, giving up on connectivity check 0xbe7edf0
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : timer(0xbdf1800) tick #917: 0 frozen, 0 in-progress, 0 waiting, 0 succeeded, 0 nominated, 0 waiting-for-nom.
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : priv_conn_check_tick_unlocked: stopping conncheck timer
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : stream 1 component 1 STATE-CHANGE 2 -> 5.
** (<unknown>:2996): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:2996): DEBUG: bus message : farsight-error
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : stream 1 component 2 STATE-CHANGE 2 -> 5.
** (<unknown>:2996): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:2996): DEBUG: Agent 0xbdf1800 : changing conncheck state to COMPLETED.
** (<unknown>:2996): DEBUG: Receive bus message from the event proc : farsight-error
** (<unknown>:2996): DEBUG: Error on BUS (11) Could not establish connection .. Could not establish connection on the RTP component

thanks for reading


Title: Audio/Video conversation
Post by: Brian on September 23, 2008, 01:53:55 am
Ok, I agree time to move on I think. Last night I downloaded version 10507 of amsn. Tried to compile, it wouldn't without farsight and a I cannot compile farsight without installing glib 2.18. So give me a few hours and I will have another go:)


Title: Audio/Video conversation
Post by: kakaroto on September 23, 2008, 07:03:17 am
euhhh.. you 'downloaded from website' farsight ? that would not work! amsn will only compile if you are using the latest git version...
Also note that you should follow the instructions on the wiki, you should take MY own repository for farsight, as it contains a merge of many branches (the master merged with nice and relay-info branches) : http://www.amsn-project.net/wiki/Farsight#Linux
And yeah, it might happen that you can't connect if you are both behind a symetric NAT with no relay servers being used! which is why you need to use my branch of farsight2, which contains the necessary code for setting up the relay sever.

@Brian: no.. amsn WILL compile without farsight, it just *warns* you that you don't have farsight, but it's not a mandatory dependency.. it will just not compile the farsight module, everything else will compile. aMSN will run just fine, but if you try to do a voice call, it will tell you "you don't have farsight 2 installed". That's it! :)
Enjoy...


Title: Audio/Video conversation
Post by: Brian on September 23, 2008, 07:27:20 am
kakaroto, I take your point about farsight, not being required. I was under the impression that I 'had' to have it.
Anyway to be clear I downloaded the latest version of amsn from the web site. I have read the wiki several times and DO have your git versions for farsight2 and libnice:).
The purpose of this note is to let you know I am massive dependency problems trying to install glib2 2.16.
At the moment I cannot see a way around it short of installing Suse 11.0? Will let you know how I go.


Title: Audio/Video conversation
Post by: farseeing on September 23, 2008, 08:53:17 am
@kakaroto : ok this is what I needed to know. thank you for telling.
                      Yep I had to use the website version of fs and do my owm cooking because it wasn't working with the git version (I'm not debian user maybe other versions problems there).
                      I've already posted the ugly things I had to do to have it work one or two pages upper in this thread
                      Brian seems to have problems installing, I'll rather stay like that and get my contact open his ports (if he manages !)


Title: Audio/Video conversation
Post by: kakaroto on September 23, 2008, 09:58:30 am
@Brian : ok, good luck... you could still try to make it run with glib 2.14, I've tried on my pc and it works, so I'm guessing you have some faulty installation of the glib headers... try to see what's wrong there, reinstall the glib-dev package... I was guessing that going from glib 2.14 to 2.18 would bring in a lot of dependency issues... I hope you can fix them, otherwise, maybe just try to fix your header issues...
(other solution would be as simple as downloading glib (2.14 just to stay on the safe side with any possible dependency problem) and compile it yourself, this way you can be sure that the headers are correct...

@farseeing : You remember what I answered to your post ?
Quote
hey guys calm downnnnnnnnn!
I just said I'll let you know when to try it! lol

basically, yes, it wouldn't have worked at that time, of course not, because it was not ready! But since then, as I said, it should now be fully working, and you can now go fetch the proper git repositories! Follow the instructions in the wiki : http://www.amsn-project.net/wiki/Farsight#Linux
as they have been updated since then!
I also updated my guide, so if you're having *any* problem, you can probably find your answer in there... and if you still can't fix it, you can ask me in here and I'll answer you.. but I'm 99% sure that you will have no problems getting the latest git to compile!
(note that Brian's problems are not because of git, it's because he has bad header files for glib, and he will have this same problem compiling ANY gstreamer plugin).
My guide is here : http://www.amsn-project.net/forums/viewtopic.php?p=32255#32255
Just follow it, and don't forget to run ./autogen.sh !
Have fun!


Title: Audio/Video conversation
Post by: fcastillo on September 23, 2008, 08:11:37 pm
Hi everybody, after being away travel for such long, now I've tried to compile farsight again, just to see that I'm having the same problems as before. Sorry I couldn't send an output of my error, so here it is:

Code:
dpkg-buildpackage: set CPPFLAGS to default value:
dpkg-buildpackage: set CFLAGS to default value: -g -O2
dpkg-buildpackage: set CXXFLAGS to default value: -g -O2
dpkg-buildpackage: set FFLAGS to default value: -g -O2
dpkg-buildpackage: set LDFLAGS to default value: -Wl,-Bsymbolic-functions
dpkg-buildpackage: source package gst-plugins-farsight
dpkg-buildpackage: source version 0.12.9-1
dpkg-buildpackage: source changed by Youness Alaoui <cens@cens.com>
dpkg-buildpackage: host architecture i386
 fakeroot debian/rules clean
/usr/share/cdbs/1/rules/simple-patchsys.mk:58: WARNING:  The following patches are modifying auto-updated files.  Please exclude the following files from your patch:  debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.guess debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.sub
test -x debian/rules
test "`id -u`" = 0
/usr/bin/make -f debian/rules reverse-config
make[1]: Entering directory `/opt/amsn/gst-plugins-farsight-0.12.9'
/usr/share/cdbs/1/rules/simple-patchsys.mk:58: WARNING:  The following patches are modifying auto-updated files.  Please exclude the following files from your patch:  debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.guess debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.sub
for i in ./config.guess ./config.sub  ; do \
if test -e $i.cdbs-orig ; then \
mv $i.cdbs-orig $i ; \
fi ; \
done
make[1]: Leaving directory `/opt/amsn/gst-plugins-farsight-0.12.9'
if [ "reverse-patches" = "reverse-patches" ]; then rm -f debian/stamp-patched; fi
patches: debian/patches/01quite-debug.patch debian/patches/02_correct_marshaller.patch debian/patches/99_autoreconf.patch
Patch debian/patches/99_autoreconf.patch is not applied.
Patch debian/patches/02_correct_marshaller.patch is not applied.
Patch debian/patches/01quite-debug.patch is not applied.
if [ "reverse-patches" != "reverse-patches" ]; then touch debian/stamp-patched; fi
if [ "reverse-patches" != "reverse-patches" ] ; then \
/usr/bin/make -f debian/rules update-config ; \
fi
for dir in debian/patches ; do \
   rm -f $dir/*.log ; \
done
for i in ./config.guess ./config.sub  ; do \
if test -e $i.cdbs-orig ; then \
mv $i.cdbs-orig $i ; \
fi ; \
done
/usr/bin/make -C . -k distclean
make[1]: Entering directory `/opt/amsn/gst-plugins-farsight-0.12.9'
make[1]: *** No rule to make target `distclean'.
make[1]: Leaving directory `/opt/amsn/gst-plugins-farsight-0.12.9'
make: [makefile-clean] Error 2 (ignored)
rm -f debian/stamp-makefile-build
rm -f debian/stamp-autotools-files
dh_clean
rm -f debian/cdbs-install-list debian/cdbs-package-list
 dpkg-source -b gst-plugins-farsight-0.12.9
dpkg-source: building gst-plugins-farsight in gst-plugins-farsight_0.12.9-1.tar.gz
dpkg-source: building gst-plugins-farsight in gst-plugins-farsight_0.12.9-1.dsc
 debian/rules build
/usr/share/cdbs/1/rules/simple-patchsys.mk:58: WARNING:  The following patches are modifying auto-updated files.  Please exclude the following files from your patch:  debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.guess debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.sub
test -x debian/rules
mkdir -p "."
/usr/bin/make -f debian/rules reverse-config
make[1]: Entering directory `/opt/amsn/gst-plugins-farsight-0.12.9'
/usr/share/cdbs/1/rules/simple-patchsys.mk:58: WARNING:  The following patches are modifying auto-updated files.  Please exclude the following files from your patch:  debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.guess debian/patches/99_autoreconf.patch:gst-plugins-farsight-0.12.5/config.sub
for i in ./config.guess ./config.sub  ; do \
if test -e $i.cdbs-orig ; then \
mv $i.cdbs-orig $i ; \
fi ; \
done
make[1]: Leaving directory `/opt/amsn/gst-plugins-farsight-0.12.9'
if [ "debian/stamp-patched" = "reverse-patches" ]; then rm -f debian/stamp-patched; fi
patches: debian/patches/01quite-debug.patch debian/patches/02_correct_marshaller.patch debian/patches/99_autoreconf.patch
Trying patch debian/patches/01quite-debug.patch at level 1 ... 0 ... 2 ... failure.
make: *** [debian/stamp-patched] Error 1
dpkg-buildpackage: failure: debian/rules build gave error exit status 2

That's what I see everytime I try to compile the new farsight plugin, compiling farsight is not a problem at all, libnice isn't a problem too. Just his plugin. I don't know if it has something to do with the fact that Ubuntu 8.04.1 has and even older plugin in their repositories, the version is 0.12.5
If anybody could help me, i really want to use farsight and this is driving me crazy!!!


Title: Audio/Video conversation
Post by: fcastillo on September 23, 2008, 08:39:23 pm
Ok, the previous error is when i try to make the .deb package. I was reading the whole thread and somebody suggested just to download the latest farsight plugin and compile it, instead of creating a .deb
It works for me now that I compiled, but of course, I'd love it if somebody could give me a .deb package, since it's a nicer and cleaner way of installing...
By the way, somebody should update the wiki to let everybody knows that you need farsight plugin also, and how to get and install it.


Title: Audio/Video conversation
Post by: fcastillo on September 23, 2008, 08:54:51 pm
First, sorry for this many replys, I should have made just one big, but I was doing step by step. Well, now that farsight plugin is installed, libnice and farsight are also installed according to the wiki (since i saw it was updated recently). I run ./configure on the amsn svn 10508 and it detects farsight without problem. I ran
Code:
make deb
and it appears that there isn't any problem until the very end, when I get this error:

Code:
dpkg-shlibdeps: failure: no dependency information found for /usr/lib/libgstfarsight-0.10.so.0 (used by debian/amsn/usr/share/amsn/utils/farsight/tcl_farsight.so).
dh_shlibdeps: command returned error code 512
make[1]: *** [binary-arch] Error 1
make[1]: Leaving directory `/opt/amsn/amsn'
make: *** [deb] Error 2

The error shows only after amsn has been compiled, and the package is trying to be created. I guess I really have to compile farsight plugin as a .deb instead of installing it. Any suggestions or comments?


Title: Audio/Video conversation
Post by: kakaroto on September 23, 2008, 09:25:49 pm
hi,
Quote

By the way, somebody should update the wiki to let everybody knows that you need farsight plugin also, and how to get and install it.

The wiki has always had, since day one, the requirement of the gst-plugins-farsight.. and my guide also contains info about it.
Quote

7 - gst-plugins-farsight - 0.12.9 or newer


About your last error, yes, it won't work creating the .deb because the deb utilities will not be able to match the tcl_farsight.so dependencies since you don't have farsight 2 installed as a deb...
In any case, you don't need amsn to create the deb, as I always said, I never had to install amsn, just run it from source with
Code:
./amsn &
and you're done.

p.s.: next time, just edit your post instead of triple-posting.


Title: Audio/Video conversation
Post by: kakaroto on September 23, 2008, 09:38:31 pm
In private :
Quote from: "Brian"
Hi, I took your tip and decided to try and compile libnice with my existing glib.
I am sending this private message because I got three errors during the compile process which I have documented and "fixed" Now my knowledge of C is pretty minimalm so would you like to take a look.
Did not want to clutter up the main thread.

So now I am happy that I have LIBNICE compiled, it has only taken 4 days:)

Please advise if any of the fixes below are rubbish.
Thanks,
Brian

Trying to compile libnice with the original glib2 214 library.

1) First error I got was
cc1: warnings being treated as errors
agent.c: In function 'nice_agent_add_stream':
agent.c:755: warning: implicit declaration of function 'g_warn_if_fail'
agent.c:755: warning: nested extern declaration of 'g_warn_if_fail'
make[3]: *** [agent.lo] Error 1
make[3]: Leaving directory `/Documents/SOURCE/libnice/agent'
make[2]: *** [all] Error 2
make[2]: Leaving directory `/Documents/SOURCE/libnice/agent'
make[1]: *** [all-recursive] Error 1


commented out line 755 /* g_warn_if_fail(agent->local_addresses); bnc */ in agent.c




2) This piece of code seemed to be calling gstnicesrc.lo when all I could find in the gst directory was gstnicesrc.loT. Copied .loT to .lo

cc1: warnings being treated as errors
gstnicesrc.c: In function 'gst_nice_src_get_type':
gstnicesrc.c:103: warning: passing argument 1 of 'g_once_init_enter' from incompatible pointer type
gstnicesrc.c:103: warning: passing argument 1 of 'g_once_init_leave' from incompatible pointer type
gstnicesrc.c:103: warning: passing argument 2 of 'g_once_init_leave' makes pointer from integer without a cast
make[2]: *** [gstnicesrc.lo] Error 1
make[2]: Leaving directory `/Documents/SOURCE/libnice/gst'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/Documents/SOURCE/libnice'
make: *** [all] Error 2


3) had similar problem with gstnicesink.loT created gstnicesink.lo


cc1: warnings being treated as errors
gstnicesink.c: In function 'gst_nice_sink_get_type':
gstnicesink.c:88: warning: passing argument 1 of 'g_once_init_enter' from incompatible pointer type
gstnicesink.c:88: warning: passing argument 1 of 'g_once_init_leave' from incompatible pointer type
gstnicesink.c:88: warning: passing argument 2 of 'g_once_init_leave' makes pointer from integer without a cast
make[2]: *** [gstnicesink.lo] Error 1
make[2]: Leaving directory `/Documents/SOURCE/libnice/gst'


Quote from: "Brian"
I have had two problems with the build which I have fixed but I am stuck on this one

Brian
Making all in fsrtpconference
make[3]: Entering directory `/Documents/SOURCE/farsight2/gst/fsrtpconference'
echo "#include \"glib-object.h\"" >> fs-rtp-marshal.c.tmp
echo "#include \"fs-rtp-marshal.h\"" >> fs-rtp-marshal.c.tmp
glib-genmarshal --body --prefix=_fs_rtp_marshal ./fs-rtp-marshal.list >> fs-rtp-marshal.c.tmp
mv fs-rtp-marshal.c.tmp fs-rtp-marshal.c
glib-genmarshal --header --prefix=_fs_rtp_marshal ./fs-rtp-marshal.list > fs-rtp-marshal.h.tmp
mv fs-rtp-marshal.h.tmp fs-rtp-marshal.h
make[3]: *** No rule to make target `gstfsrtpconference_doc.c', needed by `all'. Stop.
make[3]: Leaving directory `/Documents/SOURCE/farsight2/gst/fsrtpconference'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/Documents/SOURCE/farsight2/gst'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/Documents/SOURCE/farsight2'
make: *** [all] Error 2


Quote from: "kakaroto"
we've got a rule around here that says that PMs should only be used to send email addresses and passwords.
Also, whatever you posted would be very interesting to other people, so please post in the public forums unless it's a private issue.
I'll paste your messages in the thread and answer them there.


ok, so first, let's talk about your first message..
error 1 : this was fixed already in my libnice branch, did you update your git code ? just do a 'git pull origin nice-kakaroto' from libnice, it should fix it.
error 2+3 : this is really the weird thing and I don't know why it's doing this... and the .loT to .lo change, I don't get it, I don't see why it would fix the problem (maybe .loT is the temporary file and once the compile is good, it gets renamed by libtool.. I don't know)
Anyways, the error is not an actual error, it's just a warning, which is a good thing, the problem is this :
Quote
cc1: warnings being treated as errors

So what you need to do is simply tell it to leave the warnings and not to treat them as errors... to do that, simply edit the file common.mk and remove the line that says "-Werror", then do ./autogen.sh again and recompile!

About your second message, the error is known and has been fixed already in git, so you just need to update your git repository again before compiling, simply do 'git pull origin nice' from farsight2 and it should fix it. (if for some reason you get another error later on, make sure you used ./configure --disable-gtk-doc)

good luck.


Title: Audio/Video conversation
Post by: Brian on September 24, 2008, 03:44:11 am
Firstly, apologies for the PM, got the idea now.

I was under the impression that I did have the latest code so in order to avoid confusion I started again. Deleted libnice and farsight , went back to the wiki and reloaded everything as per instructions. Then I went into common.mk in libnice and removed the Werror. Reran autogen ok. Then tried the make but still ran into the file problems. Specifically in
/libnice/gst
gstnicesink.loT gstnicesrc.loT

Renaming these two files without the T got me a clean make of libnice.

Switched to farsight2 and ran ./autogen.sh --prefix=/usr --disable-python --disable-gtk-doc
Got similar problems with these;
/farsight2/gst/fsrtpconference/
libfsrtpconference_la-fs-rtp-conference.loT

/farsight2/gst-libs/gst/farsight
libgstfarsight_0.10_la-fs-base-conference.loT

/farsight2/gst/funnel
libfsfunnel_la-fs-funnel.loT

In all cases I just created the same file but without the T

Now However, I amstuck on;

make[3]: Entering directory `/Documents/SOURCE/farsight2/gst/funnel'
/bin/sh ../../libtool --tag=CC   --mode=link gcc -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -Wall -Werror  -g   -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -g -O2 -module -avoid-version -export-symbols-regex '^_*gst_plugin_desc$' -no-undefined  -o libfsfunnel.la -rpath /usr/lib/gstreamer-0.10 libfsfunnel_la-fs-funnel.lo -pthread -lgstreamer-0.10 -lgobject-2.0 -lgmodule-2.0 -ldl -lgthread-2.0 -lrt -lxml2 -lglib-2.0    -pthread -lgstbase-0.10 -lgstreamer-0.10 -lgobject-2.0 -lgmodule-2.0 -ldl -lgthread-2.0 -lrt -lxml2 -lglib-2.0   -pthread -lgstreamer-0.10 -lgobject-2.0 -lgmodule-2.0 -ldl -lgthread-2.0 -lrt -lxml2 -lglib-2.0
generating symbol list for `libfsfunnel.la'
/usr/bin/nm -B    | sed -n -e 's/^.*[   ]\([ABCDGIRSTW][ABCDGIRSTW]*\)[         ][      ]*\([_A-Za-z][_A-Za-z0-9]*\)$/\1 \2 \2/p' | /usr/bin/sed 's/.* //' | sort | uniq > .libs/libfsfunnel.exp
/usr/bin/nm: 'a.out': No such file
/usr/bin/grep -E -e "^_*gst_plugin_desc$" ".libs/libfsfunnel.exp" > ".libs/libfsfunnel.expT"
mv -f ".libs/libfsfunnel.expT" ".libs/libfsfunnel.exp"
echo "{ global:" > .libs/libfsfunnel.ver
 cat .libs/libfsfunnel.exp | sed -e "s/\(.*\)/\1;/" >> .libs/libfsfunnel.ver
 echo "local: *; };" >> .libs/libfsfunnel.ver
 gcc -shared    /usr/lib/libgstbase-0.10.so -L/usr/lib /usr/lib/libgstreamer-0.10.so /usr/lib/libgobject-2.0.so /usr/lib/libgmodule-2.0.so -ldl /usr/lib/libgthread-2.0.so -lrt /usr/lib/libxml2.so /usr/lib/libglib-2.0.so  -pthread -pthread -pthread -pthread -pthread -pthread -Wl,-soname -Wl,libfsfunnel.so -Wl,-version-script -Wl,.libs/libfsfunnel.ver -o .libs/libfsfunnel.so
/usr/lib/gcc/i586-suse-linux/4.2.1/../../../../i586-suse-linux/bin/ld:.libs/libfsfunnel.ver:2: syntax error in VERSION script
collect2: ld returned 1 exit status
make[3]: *** [libfsfunnel.la] Error 1
make[3]: Leaving directory `/Documents/SOURCE/farsight2/gst/funnel'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/Documents/SOURCE/farsight2/gst'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/Documents/SOURCE/farsight2'
make: *** [all] Error 2

The ver file gets generated with no parms in it like so;

{ global:
local: *; };

This is probably because libfsfunnel.exp does not get built, it is empty.

Brian


Title: Audio/Video conversation
Post by: kakaroto on September 24, 2008, 05:53:22 am
Hi again..
humm.. this looks to me like a serious problem of dev tools on your system.. looks like your libtool/make/gcc/something is not doing what it should do and is causing all these problems...
anyways, first, you probably had the latest code.. but then yesterday that bug you had got fixed... so you just needed to update :)
about the  libnice problem, this is weird, as long as you remove -Werror it should build fine..
the problem you're having with the .loT into .lo is probably just a hack to fool libtool into thinking the file was compiled while in fact it's not...

You're having the same problem with farsight2.. and your trick of renaming the .loT into .lo is probably not the solution.. I would prefer if you told me exactly what errors you get so we can fix them...
but I think this whole thing is taking way too much time, so the best solution would be to just :
connect on IRC #amsn@irc.freenode.net
talk to me over IRC and provide me with SSH access to your computer, and i'll try to fix all that for you.
I should be online for at least a couple more hours, so if you can be on #amsn @ IRC, and ping me (just say 'kakaroto' in there, and I'll answer you...


Title: Audio/Video conversation
Post by: kakaroto on September 24, 2008, 08:29:48 am
Hello all!
just a bit of news.. I got access to Brian's PC and I took a look, the only problem he had was as I predicted, his glib headers were wrong, and it was causing all these issues. Once we switched the glib headers (glib/gthread.h and gobject/gtype.h) with the correct ones from the official glib source, it all compiled correctly without any error or warning (both libnice and farsight2).
What I would really like to understand though is why there was that change...
here are the diffs : http://pastebin.com/m7895ed33 and http://pastebin.com/mb33220d
as  you can see, the g_once_init_enter/leave functions had a 'gpointer' as argument instead of a 'gsize'... and the weird thing is that the official source code always had it as a 'gsize'.. you can even see the svn log where that code was first written :
http://svn.gnome.org/viewvc/glib/trunk/glib/gthread.h?r1=5616&r2=5615&pathrev=5616
so the question is... why would the opensuse 10.3 package maintainer modify those headers before creating the package???
anyways, just my ranting about this issue...
Have fun guys!


Title: Audio/Video conversation
Post by: Brian on September 24, 2008, 10:08:22 am
kakaroto, thanks for your time today, it is a shame when a 'standard' gets corrupted. One day we may find out why suse changed them. I hope it is not the all too familiar MS ploy:(

As you predicted I had no problems recompiling libnice, farsight or amsn.
However, when I try to run it I get a popup window that says Loading TkCximage failed, please compile aMSN first?
Well I had just done that ;
make clean all
find . -name \*~ -print | xargs rm -f
find . -name \*.o -print | xargs rm -f
find . -name \*.a -print | xargs rm -f
find . -name \*.dep -print | xargs rm -f
rm -f ./utils/TkCximage/src/TkCximage.cpp.so ./utils/TkCximage/src/TkCximage.cpp.o ./utils/TkCximage/src/PhotoFormat.cpp.o ./utils/TkCximage/src/procs.cpp.o ./utils/TkCximage/src/CxImage/libCxImage.a
rm -f ./utils/TkCximage/src/TkCximage.cpp.so ./utils/TkCximage/src/TkCximage.cpp.o ./utils/TkCximage/src/PhotoFormat.cpp.o ./utils/TkCximage/src/procs.cpp.o ./utils/TkCximage/src/CxImage/libCxImage.a
rm -f ./utils/TkCximage/src/TkCximage.cpp.so ./utils/TkCximage/src/TkCximage.cpp.o ./utils/TkCximage/src/PhotoFormat.cpp.o ./utils/TkCximage/src/procs.cpp.o ./utils/TkCximage/src/CxImage/libCxImage.a
rm -f ./utils/TkCximage/src/CxImage/libCxImage.a ./utils/TkCximage/src/CxImage/ximadsp.cpp.o  ./utils/TkCximage/src/CxImage/ximaexif.cpp.o ./utils/TkCximage/src/CxImage/ximagif.cpp.o  ./utils/TkCximage/src/CxImage/ximainfo.cpp.o ./utils/TkCximage/src/CxImage/ximajpg.cpp.o  ./utils/TkCximage/src/CxImage/ximalyr.cpp.o ./utils/TkCximage/src/CxImage/ximapng.cpp.o  ./utils/TkCximage/src/CxImage/ximatga.cpp.o ./utils/TkCximage/src/CxImage/ximatran.cpp.o ./utils/TkCximage/src/CxImage/ximabmp.cpp.o ./utils/TkCximage/src/CxImage/ximaenc.cpp.o  ./utils/TkCximage/src/CxImage/ximage.cpp.o ./utils/TkCximage/src/CxImage/ximahist.cpp.o ./utils/TkCximage/src/CxImage/ximaint.cpp.o ./utils/TkCximage/src/CxImage/ximalpha.cpp.o ./utils/TkCximage/src/CxImage/ximapal.cpp.o ./utils/TkCximage/src/CxImage/ximasel.cpp.o  ./utils/TkCximage/src/CxImage/ximath.cpp.o ./utils/TkCximage/src/CxImage/xmemfile.cpp.o
rm -f ./utils/webcamsn/src/webcamsn.so  ./utils/webcamsn/src/webcamsn.o ./utils/webcamsn/src/kidhash.o ./utils/webcamsn/src/libmimic.a
rm -f ./utils/webcamsn/src/webcamsn.so  ./utils/webcamsn/src/webcamsn.o ./utils/webcamsn/src/kidhash.o ./utils/webcamsn/src/libmimic.a
rm -f ./utils/tcl_siren/src/libsiren.a ./utils/tcl_siren/src/common.o  ./utils/tcl_siren/src/dct4.o  ./utils/tcl_siren/src/encoder.o ./utils/tcl_siren/src/decoder.o ./utils/tcl_siren/src/huffman.o ./utils/tcl_siren/src/rmlt.o
rm -f ./utils/tcl_siren/src/libsiren.a ./utils/tcl_siren/src/common.o  ./utils/tcl_siren/src/dct4.o  ./utils/tcl_siren/src/encoder.o ./utils/tcl_siren/src/decoder.o ./utils/tcl_siren/src/huffman.o ./utils/tcl_siren/src/rmlt.o
rm -f ./utils/tcl_siren/src/tcl_siren.so  ./utils/tcl_siren/src/tcl_siren.o ./utils/tcl_siren/src/libsiren.a
rm -f ./utils/tclISF/src/tclISF.so ./utils/tclISF/src/tclISF.o ./utils/tclISF/src/libISF/libISF.a
rm -f ./utils/tclISF/src/tclISF.so ./utils/tclISF/src/tclISF.o ./utils/tclISF/src/libISF/libISF.a
rm -f ./utils/tclISF/src/tclISF.so ./utils/tclISF/src/tclISF.o ./utils/tclISF/src/libISF/libISF.a
rm -f ./utils/tclISF/src/libISF/libISF.a ./utils/tclISF/src/libISF/compression.o  ./utils/tclISF/src/libISF/createTags.o ./utils/tclISF/src/libISF/decodeTags.o  ./utils/tclISF/src/libISF/decompression.o ./utils/tclISF/src/libISF/decProperty.o  ./utils/tclISF/src/libISF/encoding.o ./utils/tclISF/src/libISF/libISF.o  ./utils/tclISF/src/libISF/read.o
rm -f ./utils/asyncresolver/src/asyncresolver.o ./utils/asyncresolver/src/libasyncresolver.so
rm -f ./utils/asyncresolver/src/asyncresolver.o
rm -f ./utils/farsight/tcl_farsight.so  ./utils/farsight/src/tcl_farsight.so
rm -f ./utils/farsight/src/tcl_farsight.so  ./utils/farsight/src/tcl_farsight.o
rm -f ./utils/linux/capture/capture.so
rm -f ./utils/linux/capture/libng/libng.a ./utils/linux/capture/libng/grab-ng.o ./utils/linux/capture/libng/devices.o ./utils/linux/capture/libng/writefile.o ./utils/linux/capture/libng/parse-mpeg.o ./utils/linux/capture/libng/parse-dvb.o ./utils/linux/capture/libng/color_common.o ./utils/linux/capture/libng/color_packed.o ./utils/linux/capture/libng/color_lut.o ./utils/linux/capture/libng/color_yuv2rgb.o ./utils/linux/capture/libng/convert.o ./utils/linux/capture/libng/misc.o
rm -f ./utils/linux/capture/libng/plugins/conv-mjpeg.so ./utils/linux/capture/libng/plugins/drv0-v4l2.so ./utils/linux/capture/libng/plugins/drv1-v4l.so ./utils/linux/capture/libng/plugins/sn9c10x.so
rm -f ./utils/linux/traydock/libtray.so ./utils/linux/traydock/libtray.o
rm -f ./utils/linux/linflash/flash.so  ./utils/linux/linflash/flash.o
  CXX     utils/TkCximage/src/TkCximage.cpp.o
  CXX     utils/TkCximage/src/PhotoFormat.cpp.o
  CXX     utils/TkCximage/src/procs.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximadsp.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaexif.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximagif.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximainfo.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximajpg.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximalyr.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximapng.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximatga.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximatran.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximabmp.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaenc.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximage.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximahist.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximaint.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximalpha.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximapal.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximasel.cpp.o
  CXX     utils/TkCximage/src/CxImage/ximath.cpp.o
  CXX     utils/TkCximage/src/CxImage/xmemfile.cpp.o
  AR      utils/TkCximage/src/CxImage/libCxImage.a
ar: creating utils/TkCximage/src/CxImage/libCxImage.a
  LDX     utils/TkCximage/src/TkCximage.cpp.so
  LDX     utils/TkCximage/src/TkCximage.cpp.so
cp utils/TkCximage/src/TkCximage.cpp.so utils/TkCximage/TkCximage.so
  CC      utils/webcamsn/src/webcamsn.o
  CC      utils/webcamsn/src/kidhash.o
  CC      utils/webcamsn/src/bitstring.o
  CC      utils/webcamsn/src/deblock.o
  CC      utils/webcamsn/src/encode.o
  CC      utils/webcamsn/src/idct_dequant.o
  CC      utils/webcamsn/src/mimic.o
  CC      utils/webcamsn/src/vlc_decode.o
  CC      utils/webcamsn/src/colorspace.o
  CC      utils/webcamsn/src/decode.o
  CC      utils/webcamsn/src/fdct_quant.o
  CC      utils/webcamsn/src/vlc_common.o
  CC      utils/webcamsn/src/vlc_encode.o
  AR      utils/webcamsn/src/libmimic.a
ar: creating utils/webcamsn/src/libmimic.a
  LD      utils/webcamsn/src/webcamsn.so
cp utils/webcamsn/src/webcamsn.so utils/webcamsn/webcamsn.so
  CC      utils/tcl_siren/src/tcl_siren.o
  CC      utils/tcl_siren/src/common.o
  CC      utils/tcl_siren/src/dct4.o
  CC      utils/tcl_siren/src/encoder.o
  CC      utils/tcl_siren/src/decoder.o
  CC      utils/tcl_siren/src/huffman.o
  CC      utils/tcl_siren/src/rmlt.o
  AR      utils/tcl_siren/src/libsiren.a
ar: creating utils/tcl_siren/src/libsiren.a
  LD      utils/tcl_siren/src/tcl_siren.so
cp utils/tcl_siren/src/tcl_siren.so utils/tcl_siren/tcl_siren.so
g++ -g -O2 -O2 -w -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/local/include/tcl8.5 -I/usr/local/include/tcl8.5 -I./utils/TkCximage/src -I./utils/TkCximage/src/CxImage -DPACKAGE='"amsn"' -DVERSION='"0.98b-svn`which svnversion > /dev/null && svnversion`"' -DLIBDIR='"/Documents/SOURCE/asmn/amsn"' -fPIC   -c -o utils/tclISF/src/tclISF.o utils/tclISF/src/tclISF.cpp
  CC      utils/tclISF/src/libISF/compression.o
  CC      utils/tclISF/src/libISF/createTags.o
  CC      utils/tclISF/src/libISF/decodeTags.o
  CC      utils/tclISF/src/libISF/decompression.o
  CC      utils/tclISF/src/libISF/decProperty.o
  CC      utils/tclISF/src/libISF/encoding.o
  CC      utils/tclISF/src/libISF/libISF.o
  CC      utils/tclISF/src/libISF/read.o
  AR      utils/tclISF/src/libISF/libISF.a
ar: creating utils/tclISF/src/libISF/libISF.a
  LD      utils/tclISF/src/tclISF.so
  LD      utils/tclISF/src/tclISF.so
cp utils/tclISF/src/tclISF.so utils/tclISF/tclISF.so
  CC      utils/asyncresolver/src/asyncresolver.o
  LD      utils/asyncresolver/src/libasyncresolver.so
cp utils/asyncresolver/src/libasyncresolver.so utils/asyncresolver/libasyncresolver.so
  CC      utils/farsight/src/tcl_farsight.o
  LD      utils/farsight/src/tcl_farsight.so
cp utils/farsight/src/tcl_farsight.so utils/farsight/tcl_farsight.so
  CC      utils/linux/capture/capture.o
  CC      utils/linux/capture/libng/grab-ng.o
  CC      utils/linux/capture/libng/devices.o
  CC      utils/linux/capture/libng/writefile.o
  CC      utils/linux/capture/libng/parse-mpeg.o
  CC      utils/linux/capture/libng/parse-dvb.o
  CC      utils/linux/capture/libng/color_common.o
  CC      utils/linux/capture/libng/color_packed.o
  CC      utils/linux/capture/libng/color_lut.o
  CC      utils/linux/capture/libng/color_yuv2rgb.o
  CC      utils/linux/capture/libng/convert.o
  CC      utils/linux/capture/libng/misc.o
  AR      utils/linux/capture/libng/libng.a
ar: creating utils/linux/capture/libng/libng.a
  LD      utils/linux/capture/capture.so
  CC      utils/linux/capture/libng/plugins/conv-mjpeg.o
  LD      utils/linux/capture/libng/plugins/conv-mjpeg.so
  CC      utils/linux/capture/libng/plugins/drv0-v4l2.o
  CC      utils/linux/capture/libng/plugins/struct-v4l2.o
  CC      utils/linux/capture/libng/plugins/struct-dump.o
  LD      utils/linux/capture/libng/plugins/drv0-v4l2.so
  CC      utils/linux/capture/libng/plugins/drv1-v4l.o
  CC      utils/linux/capture/libng/plugins/struct-v4l.o
  LD      utils/linux/capture/libng/plugins/drv1-v4l.so
  CC      utils/linux/capture/libng/plugins/sn9c10x.o
  LD      utils/linux/capture/libng/plugins/sn9c10x.so
  CC      utils/linux/traydock/libtray.o
  LD      utils/linux/traydock/libtray.so
  CC      utils/linux/linflash/flash.o
  LD      utils/linux/linflash/flash.so
L2:/Documents/SOURCE/asmn/amsn # make install
rm -Rf /usr/share/amsn
mkdir -p /usr/share/amsn
mkdir -p /usr/bin
mkdir -p /usr/share/applications/
mkdir -p /usr/share/pixmaps/
find /usr/share/amsn -name '.svn' -print | xargs rm -Rf
ln -sf /usr/share/amsn/amsn /usr/bin/amsn
ln -sf /usr/share/amsn/amsn-remote /usr/bin/amsn-remote
ln -sf /usr/share/amsn/amsn-remote-CLI /usr/bin/amsn-remote-CLI
cp ./amsn.desktop /usr/share/applications/
ln -sf /usr/share/amsn/desktop-icons/48x48/apps/amsn.png /usr/share/pixmaps/

I have a ~/amsn/utils/TkCximage/ directory with what looks like the correct stuff in it?


Title: Audio/Video conversation
Post by: kakaroto on September 24, 2008, 10:41:37 am
humm.. that's not very nice! could you try running amsn from the source directly, with "./amsn &" ? if still nothing, then try doing this :
Code:
bash > tclsh
> load ./utils/TkCximage/TkCximage.so
...
> exit
bash >

it should give us why it refuses to load it... but if it was working correctly before, I see no reason why it would fail now! all you did was compile tcl_farsight, there shouldn't be anything related to TkCximage...
in any case, if you can't fix it with what I just said, please post in a separate thread, so we can keep this one farsight-related. Use the new thread if you want to post the output of tclsh...

p.s.: oh and by the way, you could try running the command 'wish' and in the console type 'info patchlevel' and see which version of tcl/tk is used, and make sure it's the same one as what the ./configure gave you at the end...


Title: Audio/Video conversation
Post by: Brian on September 24, 2008, 12:25:27 pm
As requested started a new thread


Title: Audio/Video conversation
Post by: farseeing on September 24, 2008, 11:02:19 pm
Hi,
switched to git versions of farsight2, libnice and aMSN svn 10515.
Everything went right compiling and installing throughout kakaroto's guide.
but ...
I've had a problem with one of my contact for speech (everything else was okay video and text chat).
The speech problem may be solved (I could hear him, he could hear me for several minutes until it freezes !) but we made things on our respective firewalls related to port forwarding in the mean time so can't say where the solution comes from.
I can still switch back to the stack I was formerly using, concentrating on the firewalls configuration while waiting for your next release. So, help would be welcome but I can go back to a stable state by myself, I think.
 
Below main troubles :
- Farsight2 (git version) isn't recognized all the time in the audio/video wizard. didn't figure out why. sometimes I try to reboot with webcam unplugged, then hot-plug it and it seems to re-enable farsight2 in aMSN.
- The conversation hanged up abruptely and freezed (see log screen attached, I could just capture this, each amsn window was freezed. I had to ***-9 it)
- The Preferences/Conections/IP-Restrict NAT section doesn't look reliable : it opens on a green 'Your ports look well configured' and when I test the port 6891 both msgs (main one and the one related to port 6891) become red and says I'm firewalled. However, audio video and chat went okay (until it freezes) during the session (didn"t try file transfert, think I read somewhere it wasn't working, anyway...).
- while trying to reconnect, don't really know in which state the application was but : When I started back an audio conversation, when my peer accepted it opened a new chat window ! Logged off then on my session (I mean my Gnome session). tried to reconnect and invited for audio : 'No Farsight2 module available' (I then gave up without even telling my contact he was polluting my proud & gorgeous linux box with his crappy commercial windowed box ! text was still workin however...what a lack of reflex...I'll surely pay for this !)
 - contact and me were both 'online'. I had no other conversation running at this moment, no tabbed window (any ghost session here ? don't think so : Yes ,I was with someone else just before but everything was fine, the session ended up cleanly, I think). About firewalling and NAT : it looks wide open on both side now !
I was launching through a desktop launcher so I don't have console logs. .
 
(http://)
(http://www.postimage.org/gx9er0.jpg) (http://www.postimage.org/image.php?v=gx9er0)


Title: Audio/Video conversation
Post by: kakaroto on September 25, 2008, 12:54:38 am
Hi farseeing,
as I said, video and chat are completely independant of audio call, so it's normal that they might work and audio not (although with the latest versions, it should always work...)
unfortunately, I didn't understand much of what you said... so let's try little by little...
Quote

 
I've had a problem with one of my contact for speech (everything else was okay video and text chat).

did you mean that you had this problem before updating, or that you had this problem even after updating? or that you didn't have a problem before, and then there was this problem after you updated ?

Quote
The speech problem may be solved (I could hear him, he could hear me for several minutes until it freezes !) but we made things on our respective firewalls related to port forwarding in the mean time so can't say where the solution comes from.

ok.. it may be solved.. so I'm guessing you had a problem before updating...or maybe your problem was that it froze... in any case, the port forwarding means nothing because you can't forward a port for amsn for audio chat since farsight will always choose a random port to listen on at every execution... so opening your firewall ports will only help for the webcam feature, not for audio chat.

Quote
I can still switch back to the stack I was formerly using, concentrating on the firewalls configuration while waiting for your next release. So, help would be welcome but I can go back to a stable state by myself, I think.

I didn't understand this at all.. sorry.

Quote
- Farsight2 (git version) isn't recognized all the time in the audio/video wizard. didn't figure out why. sometimes I try to reboot with webcam unplugged, then hot-plug it and it seems to re-enable farsight2 in aMSN.

humm.. that's interesting, I wonder why... You might want to give me the output on the terminal, and what amsn's console (ctrl-shift-C) outputs when you do :
Code:
proc tstFS { args } {puts $args } ; ::MSNSIP::TestFarsight tstFS


Quote
- The conversation hanged up abruptely and freezed (see log screen attached, I could just capture this, each amsn window was freezed. I had to ***-9 it)

humm.. the conversation hanged.. actually the status log says that the conversation was ended, maybe your friend canceled or something else caused it to abort (maybe a farsight error).. the question though is why it would freeze aMSN.. I haven't experienced the freeze, so I don't know...

Quote
- The Preferences/Conections/IP-Restrict NAT section doesn't look reliable : it opens on a green 'Your ports look well configured' and when I test the port 6891 both msgs (main one and the one related to port 6891) become red and says I'm firewalled. However, audio video and chat went okay (until it freezes) during the session (didn"t try file transfert, think I read somewhere it wasn't working, anyway...).

humm... you're right, there seems to be a small bug in that code where it might timeout after less than a second (instead of the 6 second timeout it should have), so I guess we'll have to fix that.. but notice that if you click on "test port" multiple times, you should get the 'your ports are well configured' a few times...
In any case, as I explained before, this is completely unrelated to how farsight works, so it doesn't matter.. and no, file transfers should work just fine... oh and video, chat and filetransfers will work EVEN if you are firewalled.. but when you are firewalled (AND the other contact is also firewalled), the file transfer and webcam's speed will be slower, that's it.

Quote
- while trying to reconnect, don't really know in which state the application was but : When I started back an audio conversation, when my peer accepted it opened a new chat window ! Logged off then on my session (I mean my Gnome session). tried to reconnect and invited for audio : 'No Farsight2 module available' (I then gave up without even telling my contact he was polluting my proud & gorgeous linux box with his crappy commercial windowed box ! text was still workin however...what a lack of reflex...I'll surely pay for this !)

humm.. yeah, the new chat window is a bug that happened to me a few times, it happens when the audio call gets canceled..
about the 'no farsight2 module available', this is probably because amsn checks for farsight when you signin, and in the same way that the assistant sometimes seems to fail to find farsight, it probably failed to find it when you signed in (I still don't know why though) and thought that it wasn't there... if you launched the assistant and it found it, then it would have worked...
and.. I didn't understand the rest about polluting your proud linux box, etc...

Quote
- contact and me were both 'online'. I had no other conversation running at this moment, no tabbed window (any ghost session here ? don't think so : Yes ,I was with someone else just before but everything was fine, the session ended up cleanly, I think). About firewalling and NAT : it looks wide open on both side now !
I was launching through a desktop launcher so I don't have console logs. .

ok... well as I said, I would like to have the terminal logs and the amsn console output when you try
Code:
proc tstFS { args } {puts $args } ; ::MSNSIP::TestFarsight tstFS

but only when it returns '0', so I can see what goes wrong in farsight when it says that it failed to find farsight...
Hopefully, we'll get more details on what goes wrong with your contact and why it freezes...
thanks for testing and the feedback!


Title: Audio/Video conversation
Post by: farseeing on September 25, 2008, 06:09:45 pm
Hi Kakaroto,
thanks to you for your patience and availability. Here below the outputs.

Yep I was quite unclear, trying to sum up :

-I was using the svn version of aMSN before this one, farsight2.0.0.3 (added0.0.3.1 to have it recognized in*.pc files) gstreamer0.19 libnice from git.
-Everything was quite fine except with one WLM contact (could only get video and chat no audio).
-I've updated to last versions of farsight2, svn amsn libnice (from your folder on collabora.co.uk, kakaroto. just followed the guide).
-I've opened up UDP/TCP ranges on my router (Internet box fro my ISP). Idem on my contact router (D-Link Wi-fi, but Ethernet in use on the PC he tries to chat with me). enabled forwarding etc.... (<--seen what you said about the uselessness of this for audio chat)
-We managed a 'full-power' call with video, audio, text,...(silly emoticons, etc.....!) for a while then the freeze problems told upper.
-it's still okay with other WLM contacts (they have ISP boxes too as far as I know. Don't know if you see what I'm talkin about).



Quote
I didn't understand this at all.. sorry.

<--just to say that if you have better to do than answering my whining it's okay because I kept the old sources I was using and was wondering about rolling things back and trying to solve my pb with firewall configuration : doesn't look applicable anymore due to what you said about audio and port forwarding.

So tried what you last said and got these outputs:
They're quite clear about the audio device being busy. Can't understand why as it works ! but well, I'll try to figure it out.
amsn console output :

Quote
(amsn) 1 % proc tstFS { args } {puts $args } ; ::MSNSIP::TestFarsight tstFS
0
(amsn) 2 %


console output:
Quote

** (<unknown>:8314): DEBUG: CODECS ARE READY

(<unknown>:8314): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:8314): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:8314): DEBUG: stun ip : xxx.xxx.xxx.xxx : 3478 (<---who is this ? a device, the webcam ???? Anyway, taking care of his privacy ! )
** (<unknown>:8314): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:8314): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:8314): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:8314): DEBUG: Got an error on the BUS (4): Could not open audio device for recording. Device is being used by another application. (gstalsasrc.c(635): gst_alsasrc_open (): /pipeline/gconfaudiosrc0/bin2/alsasrc0:
Device 'hw:1,0' is busy)
** (<unknown>:8314): DEBUG: An error occured : Gstreamer error
** (<unknown>:8314): DEBUG: CODECS ARE READY

(<unknown>:8314): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:8314): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:8314): DEBUG: FS: relay info = 0x9950950 - 2
** (<unknown>:8314): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:8314): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:8314): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:8314): DEBUG: Got an error on the BUS (4): Could not open audio device for recording. Device is being used by another application. (gstalsasrc.c(635): gst_alsasrc_open (): /pipeline/gconfaudiosrc1/bin5/alsasrc1:
Device 'hw:1,0' is busy)
** (<unknown>:8314): DEBUG: An error occured : Gstreamer error


Quote
I didn't understand the rest about polluting your proud linux box, etc...

<--Well, okay nevermind....:)


Title: Audio/Video conversation
Post by: kakaroto on September 25, 2008, 11:08:35 pm
ah ok, I get it.. so it's all working nice now... the update fixes the issue you had with your friend, so that's good news... and the 'farsight is not installed' problem was causing because your microphone was being used by another application (all that stuff will be fixed in the future, when we add a mute/unmute/volume controls for the audio call, and allow being able to make a call even if you have no microphone).
The only question remaining now is "why did it freeze"... and I can't answer that... maybe it's because the call was ended for some reason (like we saw in your status log), and it caused the memory corruption I was talking about earlier, and that made it crash/freeze...
can you retry to call that friend of yours and see if you can get it to work, or will it freeze again ? if you are able to reproduce the freezing everytime, then it would be nice if you could provide me with the console log (launch amsn and redirect stdout and stderr into a file). Make sure you recompile amsn with ./configure --enable-debug when you do that... then launch amsn, do the call, and wait until it freezes again. put the log somewhere and send me the link.
thx


Title: Audio/Video conversation
Post by: farseeing on September 26, 2008, 07:25:19 am
okay, I'll send you that as soon as possible. I think it's related to network communication as I don't think my contact or I did something to end up the session.


Title: Audio/Video conversation
Post by: kakaroto on September 26, 2008, 06:43:42 pm
could be that, or a gstreamer/farsight error that caused it to cancel itself...
btw, how long were you in a call before it froze ? 1 minute? 5 minutes? 1 hour ? that might help...


Title: Audio/Video conversation
Post by: kakaroto on September 26, 2008, 07:01:57 pm
Quote from: "Brian"
Turns out I was a bit premature. I am now getting segmentation faults whenever it goes near the audio.

When I start amsn I get

(<unknown>:4886): GLib-GObject-WARNING **: gsignal.c:1669: signal `on-bye-ssrc' is invalid for instance `
0x984f9c8'

(<unknown>:4886): GLib-GObject-WARNING **: IA__g_object_set_property: object class `GstRTPBin' has no pro
perty named `sdes-cname'
** (<unknown>:4886): DEBUG: CODECS ARE READY

(<unknown>:4886): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has n
o property named `blocksize'

(<unknown>:4886): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has n
o property named `buffer-time'

Then as soon as I try to test the sound from the OTHERS panel or try to sign in it gets
Segmentation fault
Nothing else.
Just tried to play the sound again not using snack and I get
ioctl: VIDIOC_QUERYCTRL(id=9963777;type=unknown;name="";minimum=0;maximum=0;step=0;default_value=0;flags=0): Input/output error
Segmentation fault

**************************************************************************************
I also had amsn crash with
E: socket-client.c: socket(): Address family not supported by protocol
** (<unknown>:5306): DEBUG: bus message : farsight-recv-codecs-changed
** (<unknown>:5306): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:5306): DEBUG: Got an error on the BUS (1): Internal data flow error. (gstbasesrc.c(2240): gst_base_src_loop (): /pipeline/fsrtpconference3/bin11/nicesrc7:
streaming task paused, reason error (-5))
** (<unknown>:5306): DEBUG: An error occured : Gstreamer error
Corrupt JPEG data: premature end of data segment
Invalid JPEG file structure: two SOI markers
Oops: malloc_video_bufs is 1 (expected 0) [utils/linux/capture/libng/grab-ng.c:malloc_bufs_check:235]
Oops: processes is 1 (expected 0) [utils/linux/capture/libng/convert.c:process_check:179]

Please advise if there is a better way to report this sort of bug:)

*************************************************************************************************
Also I can report that as soon as I try to use SNACK and then try tu use the audio I get a segmentation fault.


Title: Audio/Video conversation
Post by: kakaroto on September 26, 2008, 10:02:10 pm
ok, so here's my answer to Brian :
It looks to me like you have an old version of the gst-plugins-bad package... Make sure that you have the appropriate dependencies! the segfault is normal if you don't use gst-plugins-bad-0.10.6.
check again all your dependencies : http://www.amsn-project.net/wiki/Farsight


Title: Audio/Video conversation
Post by: Brian on September 27, 2008, 08:59:44 am
Ok, just for the record I am running with the plugins below.
they "appear" to be ok.
   
# Status             Package                                          |       Size | Avail. Ver.              | Inst. Ver.      
[Keep]               gstreamer-0_10-plugins-bad           |    1.8 M   | 0.10.8-42.pm.1     | 0.10.8-42.pm.1  
[Keep]               gstreamer-0_10-plugins-bad-devel |  180.9 K   | 0.10.8-42.pm.1   | 0.10.8-42.pm.1

Keep]               gstreamer-0_10-plugins-base          |    1.9 M   | 0.10.20-42.pm.1  | 0.10.20-42.pm.1
[Keep]               gstreamer-0_10-plugins-base-devel |    2.4 M   | 0.10.20-42.pm.1  | 0.10.20-42.pm.1
 
[Do Not Install]     gstreamer-0_10-plugins-base-doc |    2.6 M   | 0.10.20-42.pm.1  | ---            
[Keep]               gstreamer-0_10-plugins-base-lang   |  202.8 K   | 0.10.20-42.pm.1  | 0.10.20-42.pm.1

[Keep]               gstreamer-0_10-plugins-farsight       |  583.8 K   | 0.12.9-0.pm.2    | 0.12.9-0.pm.2  
[Keep]               gstreamer-0_10-plugins-farsight-devel |   14.6 K   | 0.12.9-0.pm.2    | 0.12.9-0.pm.2
 
[Keep]               gstreamer-0_10-plugins-good            |    2.4 M   | 0.10.10-42.pm.2  | 0.10.10-42.pm.2
{Keep]               gstreamer-0_10-plugins-good-extra   |  364.8 K   | 0.10.10-42.pm.2  | 0.10.10-42.pm.2

[Keep]               gstreamer-0_10-plugins-ugly             |  740.8 K   | 0.10.9-42.pm.1   | 0.10.9-42.pm.1


Title: Audio/Video conversation
Post by: Kreuger on September 28, 2008, 03:34:35 pm
Quote
Hey Kreuger, are gstreamer pkgs well installed ?

when I try :
locate gstrtcpbuffer.h
I get : /usr/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h
Well it shows this for me
Quote
kreuger@kreuger-desktop:~$ locate gstrtcpbuffer.h
/usr/local/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h
/usr/local/share/gtk-doc/html/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtcpbuffer.html
kreuger@kreuger-desktop:~$

Quote
Anyways, Kreuger, you still have the same problem and it's still because you don't have gstreamer include files installed properly!
Okay but I have no idea what's wrong and what to do about it. Should I reinstall the gstreamer files? Is there a way to determine which are causing the problem or should I reinstall them all or what?


Title: Audio/Video conversation
Post by: kakaroto on September 28, 2008, 07:28:40 pm
@Brian : well.. you have a newer version from what I have, so maybe they've changed something in gstreamer which broke everything, I'll have to upgrade gstreamer and test to see if it's not broken. thx, I'll let you know how it goes...

@Kreuger :
well, compare your result and the one from farseeing :
Quote from: "farseeing"
/usr/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h

Quote from: "Kreuger"
 /usr/local/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h


This basically means that you did not follow my guide correctly, you didn't follow the instructions and you installes gstreamer without the --prefix=/usr so it got installed to /usr/local/ and that's why it can't find it...


Title: Audio/Video conversation
Post by: Kreuger on September 29, 2008, 10:14:37 pm
But I've erased it and done it a few times with and without the prefix so I'm a bit confused. I will try it again though


Title: Audio/Video conversation
Post by: kakaroto on September 29, 2008, 10:52:49 pm
yeah, well.. I don't know, but make sure you do use the --prefix=/usr when compiling gstreamer!


Title: Audio/Video conversation
Post by: kakaroto on September 30, 2008, 01:59:08 am
@farseeing :
ok, i got some news for you :
1 - the freezing seems to happen when someone sends you an invite and cancels it right away, so I'm guessing that while you were having that audio call, someone sent you an invite and canceled it (or something similar happened) and that made amsn freeze... that issue is now fixed in the latest SVN version of amsn.
2 - The problem where amsn says you don't have farsight because it reports :
 
Quote
** (<unknown>:8314): DEBUG: Got an error on the BUS (4): Could not open audio device for recording. Device is being used by another application. (gstalsasrc.c(635): gst_alsasrc_open (): /pipeline/gconfaudiosrc1/bin5/alsasrc1:
Device 'hw:1,0' is busy)

this happens when amsn tries to launch farsight at the same time as it's playing a sound, so if for example, you start a call, cancel it right away, then try to start it again (before the 'ringing' sound has finished playing), then it will have this error..
If you try to launch amsn at the same time as someone connects/changes state or whatever that causes amsn to play a sound, then it will fail... so that's the trick! Hopefully, we'll have sometime in the future a better system for all of this, don't worry.

p.s.: I still get the memory corruption, but it seems much harder to reproduce...


Title: Audio/Video conversation
Post by: seal20 on September 30, 2008, 03:12:15 pm
I register to this forum just to thank you for this great work.

  !! Thank you !! I can now throw the last windows that I use to keep only to talk with my old parents, who didn't like wengophone and were bored when I asked them to try ekiga!... generation gap...

Here is my success strory:
First I try to compile everything on a debian stable, around 2 weeks ago. A first time it went well, but the assistant didn't detect farsight... the amsn console also didn't give any good info when checking for farsight, only not present. So I tried again all compilation steps but it didn't detect anymore farsight when ./configure of aMSN. Anyway this week I upgraded my debian to testing, recompile everything following your howto, and soooo easily, everything worked ! I made 2 conversation of more than 1h each ! ,
 Only minor problem : during an audio/video conversation if you receive a file, first it will be very slow to chose the destination folder, hanging a lot, then it will cut audio (video still working) without hanging up during the transfer. To restart audio I tried to hang up but then it crashed. The send bug report went out so I filled it but I will try to debug it and post the debug here soon.

Once more thank you kakarot,  you're really the strongest saiyajin !
Also thanks to all developpers of libnice, farsight and  amsn.


Title: Audio/Video conversation
Post by: kakaroto on September 30, 2008, 08:45:45 pm
Thanks a lot for your appreciation! And welcome to the forums :)
2 weeks ago, the whole git thingy was not working because I was away on holidays and I left it in a 'bad state'.  I'm glad you took the time to give us your success story! And I'm happy to see it all worked out good for you! :)
Concerning your minor problem : the slowness when trying to send a file is probably normal and would be caused by the webcam you were doing rather than the audio call, the webcam feature is mostly written in Tcl, so it's pretty slow. In any case, when you do stuff, if the CPU is being used for something, anything else will be slow. Anyways.. the sound probably started to get cut because your bandwidth was completely used for the webcam and file transfer.
The bug report though is an interesting thing, next time, please click on 'details' and copy/paste the content of the bug so I can know how to fix it.. even if you reported it automatically to the bug report system, we have too many bugs in there so I can't find wich one was yours.

once more dou itashi mashite :)


Title: Audio/Video conversation
Post by: Kreuger on September 30, 2008, 11:29:50 pm
Quote
yeah, well.. I don't know, but make sure you do use the --prefix=/usr when compiling gstreamer!
I've done it now, again making sure to use the prefix (copy and paste MastaG's version) and it worked just fine but the configure script for amsn still can't pick it up.

Quote

*** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
*** Read this for more information : http://amsn-project.net/wiki/Farsight


Quote
kreuger@kreuger-desktop:~/Downloads/Updates/msn$ locate gstrtcpbuffer.h
/usr/include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h
kreuger@kreuger-desktop:~/Downloads/Updates/msn$


Title: Audio/Video conversation
Post by: kjir on September 30, 2008, 11:32:57 pm
Quote from: "Kreuger"

Quote

*** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
*** Read this for more information : http://amsn-project.net/wiki/Farsight



I might be wrong, but is the correct version of farsight installed? Maybe the configure script does find gstreamer, but not farsight and the error message is the same for both cases. Just guessing, I didn't check!


Title: Audio/Video conversation
Post by: kakaroto on October 01, 2008, 12:06:26 am
kjir is right, the warning is if either gstreamer or farsight is not found.. make sure you have the correct farsight2 and libnice branches compiled and installed correctly in /usr too... go back to the wiki and follow it exactly to make sure you use the correct repository and branch with the latest changes...
Glad you got your gstreamer problem fixed btw!


Title: Audio/Video conversation
Post by: Brian on October 01, 2008, 12:46:53 am
kakaroto, Given seal20's success I am contemplating starting again, going through the wiki and recompiling everything. Has there been any major changes over the last two weeks? Should I wait a bit?


Title: Audio/Video conversation
Post by: kakaroto on October 01, 2008, 02:07:45 am
there were indeed major changes over the last two weeks (added relay support, and it went from 'not working' into 'working' :p). So yes, you should fetch the whole thing again as if you never downloaded it...
and there's no need to wait for now, as it seems to be working quite good and I don't see any bug that needs fixing (apart from the memory corruption that I can't find the source and might take a while to fix)... anyways, the current work I'm doing on libnice is about adding TCP Relay support for google talk (for users who have a firewall blocking ALL UDP packets), so it doesn't concern us at all here...


Title: Audio/Video conversation
Post by: kakaroto on October 02, 2008, 11:14:04 pm
yeay!! good news! the memory corruption problem is now gone!!!! The problem was in aMSN, not farsight, and not libnice, so you don't need to update farsight/libnice, you just need to update amsn and recompile it... no more crashes! :)
If you still have a crash or any other kind of bug, please report it... I want as many people testing this in order to fix anything right now... I finally consider this feature 'complete'
(ok, apart from the fine choosing of the source, and the volume changing, etc... )


Title: Audio/Video conversation
Post by: Kreuger on October 04, 2008, 01:57:12 pm
Quote
kjir is right, the warning is if either gstreamer or farsight is not found.. make sure you have the correct farsight2 and libnice branches compiled and installed correctly in /usr too... go back to the wiki and follow it exactly to make sure you use the correct repository and branch with the latest changes...
Glad you got your gstreamer problem fixed btw!
Alright, I'll go through it all again.

Edit: Got a problem already, compiling gst-plugins-farsight.

Quote
kreuger@kreuger-desktop:~/Downloads/gst-plugins-farsight-0.12.9$ make
make  all-recursive
make[1]: Entering directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9'
Making all in ext
make[2]: Entering directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9/ext'
Making all in jrtp
make[3]: Entering directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9/ext/jrtp'
/bin/bash ../../libtool --tag=CC   --mode=compile gcc -std=gnu99 -DHAVE_CONFIG_H -I. -I../..    -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -pthread -I/usr/local/include/gstreamer-0.10 -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -Wall -Wdeclaration-after-statement -g -O2 -MT libgstjrtp_la-gstjrtp.lo -MD -MP -MF .deps/libgstjrtp_la-gstjrtp.Tpo -c -o libgstjrtp_la-gstjrtp.lo `test -f 'gstjrtp.c' || echo './'`gstjrtp.c
mkdir .libs
 gcc -std=gnu99 -DHAVE_CONFIG_H -I. -I../.. -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/local/include/gstreamer-0.10 -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Wdeclaration-after-statement -g -O2 -MT libgstjrtp_la-gstjrtp.lo -MD -MP -MF .deps/libgstjrtp_la-gstjrtp.Tpo -c gstjrtp.c  -fPIC -DPIC -o .libs/libgstjrtp_la-gstjrtp.o
mv -f .deps/libgstjrtp_la-gstjrtp.Tpo .deps/libgstjrtp_la-gstjrtp.Plo
/bin/bash ../../libtool --tag=CC   --mode=compile gcc -std=gnu99 -DHAVE_CONFIG_H -I. -I../..    -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -pthread -I/usr/local/include/gstreamer-0.10 -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2   -Wall -Wdeclaration-after-statement -g -O2 -MT libgstjrtp_la-gstrtpsend.lo -MD -MP -MF .deps/libgstjrtp_la-gstrtpsend.Tpo -c -o libgstjrtp_la-gstrtpsend.lo `test -f 'gstrtpsend.c' || echo './'`gstrtpsend.c
 gcc -std=gnu99 -DHAVE_CONFIG_H -I. -I../.. -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/local/include/gstreamer-0.10 -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Wdeclaration-after-statement -g -O2 -MT libgstjrtp_la-gstrtpsend.lo -MD -MP -MF .deps/libgstjrtp_la-gstrtpsend.Tpo -c gstrtpsend.c  -fPIC -DPIC -o .libs/libgstjrtp_la-gstrtpsend.o
gstrtpsend.c:26:40: error: gst/netbuffer/gstnetbuffer.h: No such file or directory
gstrtpsend.c: In function ‘gst_rtpsend_datasink_chain’:
gstrtpsend.c:280: error: ‘GstNetBuffer’ undeclared (first use in this function)
gstrtpsend.c:280: error: (Each undeclared identifier is reported only once
gstrtpsend.c:280: error: for each function it appears in.)
gstrtpsend.c:280: error: ‘out_buf’ undeclared (first use in this function)
gstrtpsend.c:281: warning: ISO C90 forbids mixed declarations and code
gstrtpsend.c:290: warning: implicit declaration of function ‘gst_netbuffer_new’
gstrtpsend.c:296: warning: implicit declaration of function ‘gst_netaddress_set_ip4_address’
make[3]: *** [libgstjrtp_la-gstrtpsend.lo] Error 1
make[3]: Leaving directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9/ext/jrtp'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9/ext'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/kreuger/Downloads/gst-plugins-farsight-0.12.9'
make: *** [all] Error 2
kreuger@kreuger-desktop:~/Downloads/gst-plugins-farsight-0.12.9$


Title: Audio/Video conversation
Post by: Fabioamd87 on October 04, 2008, 01:59:00 pm
what about make a topic with all the news and updates about this?


Title: Audio/Video conversation
Post by: kakaroto on October 04, 2008, 07:34:06 pm
@Kreuger :
Quote
gstrtpsend.c:26:40: error: gst/netbuffer/gstnetbuffer.h: No such file or directory

this looks like you are missing gstnetbuffer... it's part of gstreamer-plugins-base so make sure you have that installed, and that you have the appropriate version of it AND that it got installed with --prefix=/usr
You should then be able to compile gst-plugins-farsight (make sure you install everything with the same order as in the wiki).

@Fabio: it will be done once libnice and farsight are released.


Title: Audio/Video conversation
Post by: Brian on October 06, 2008, 07:10:22 am
1) managed to fix my snack problem, in Suse, there is no /dev/sound/dsp.
changed ~/snack2.2.10/unix/jkAudIO_oss.c /dev/sound/dsp to /dev/dsp and recompiled.
I can now use Others/VideoAudio to test and set my audio without crashing

2) I can now establish a webcam session but get these errors on startup

(<unknown>:12586): GLib-GObject-WARNING **: gsignal.c:1669: signal `on-bye-ssrc'
is invalid for instance `0x974d000'

(<unknown>:12586): GLib-GObject-WARNING **: IA__g_object_set_property: object cla
ss `GstRTPBin' has no property named `sdes-cname'
** (<unknown>:12586): DEBUG: CODECS ARE READY

(<unknown>:12586): GLib-GObject-WARNING **: IA__g_object_set_valist: object class
 `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:12586): GLib-GObject-WARNING **: IA__g_object_set_valist: object class
 `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:12586): DEBUG: stun ip : 64.14.48.28 : 3478
** (<unknown>:12586): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:12586): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:12586): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:12586): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:12586): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:12586): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:12586): DEBUG: bus message : farsight-local-candidates-prepared
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-n
ew-local-candidate
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-n
ew-local-candidate
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-n
ew-local-candidate
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-n
ew-local-candidate
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-l
ocal-candidates-prepared
** (<unknown>:12586): DEBUG: CANDIDATES ARE PREPARED

(<unknown>:12586): GLib-GObject-WARNING **: gsignal.c:1669: signal `on-bye-ssrc'
is invalid for instance `0x974d1b0'

(<unknown>:12586): GLib-GObject-WARNING **: IA__g_object_set_property: object cla
ss `GstRTPBin' has no property named `sdes-cname'
** (<unknown>:12586): DEBUG: CODECS ARE READY


Not sure what this effects or if I should compile gstreamer stuff?


3) Refer NOTE: The line marked NOTE occurs when I try to establish an audio call.
The sound can be heard for a second but then stops. There was approx. a minute between the call attempt and the crash

** (<unknown>:12586): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:12586): DEBUG: bus message : farsight-new-active-candidate-pair
** (<unknown>:12586): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : farsight-new-active-candidate-pair
** (<unknown>:12586): DEBUG: New active candidate pair :
** (<unknown>:12586): DEBUG: Local candidate: eRseEe0jjjLjl7KtY67afyj665xVbP6K21BVgfcVw64= 2 9g3xgHsUl8/o8lJ9LsNk/w== UDP 829 192.168.1.5 1049

** (<unknown>:12586): DEBUG: Remote candidate: NJEsLfVxH0a2zSxYMtLRUBNoOWkLYicBA5EDZxG3BLA= 2 svf6rynUbbvQPL18EQajGQ== UDP 550 219.90.185.152 17300

====> NOTE: E: socket-client.c: socket(): Address family not supported by protocol
** (<unknown>:12586): DEBUG: bus message : farsight-recv-codecs-changed
** (<unknown>:12586): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:12586): DEBUG: Got an error on the BUS (1): Internal data flow error. (gstbasesrc.c(2240): gst_base_src_loop (): /pipeline/fsrtpconference4/bin12/nicesrc9:


4) Session was up for about 4 mins when I got a crash.
streaming task paused, reason error (-5))
** (<unknown>:12586): DEBUG: An error occured : Gstreamer error
Corrupt JPEG data: bad Huffman code
mjpg: panic: no more input data
Oops: malloc_video_bufs is 1 (expected 0) [utils/linux/capture/libng/grab-ng.c:malloc_bufs_check:242]
Oops: processes is 1 (expected 0) [utils/linux/capture/libng/convert.c:process_check:179]

The link appeared to be slow, if that makes a difference?

Thanks, Brian


Title: Audio/Video conversation
Post by: kakaroto on October 06, 2008, 07:07:24 pm
Hi Brian...
I see that you still have that error with gstrtpbin missing some properties, which they should have with 0.10.8 (the version you said you installed). Can you please type in a terminal :
Code:
gst-inspect-0.10 gstrtpbin

And paste here the output of that command... this may be helpful!
thanks


Title: Audio/Video conversation
Post by: Brian on October 07, 2008, 12:57:41 am
gst-inspect-0.10 gstrtpbin
Factory Details:
  Long name:    RTP Bin
  Class:        Filter/Network/RTP
  Description:  Implement an RTP bin
  Author(s):    Wim Taymans <wim@fluendo.com>
  Rank:         none (0)

Plugin Details:
  Name:                 gstrtpmanager
  Description:          RTP session management plugin library
  Filename:             /usr/lib/gstreamer-0.10/libgstrtpmanager.so
  Version:              0.10.5
  License:              LGPL
  Source module:        gst-plugins-bad
  Binary package:       GStreamer Bad Plug-ins source release
  Origin URL:           Unknown package origin

GObject
 +----GstObject
       +----GstElement
             +----GstBin
                   +----GstRTPBin

Implemented Interfaces:
  GstChildProxy

Pad Templates:
  SINK template: 'recv_rtp_sink_%d'
    Availability: On request
      Has request_new_pad() function: gst_rtp_bin_request_new_pad
    Capabilities:
      application/x-rtp

  SINK template: 'recv_rtcp_sink_%d'
    Availability: On request
      Has request_new_pad() function: gst_rtp_bin_request_new_pad
    Capabilities:
      application/x-rtcp

  SINK template: 'send_rtp_sink_%d'
    Availability: On request
      Has request_new_pad() function: gst_rtp_bin_request_new_pad
    Capabilities:
      application/x-rtp

  SRC template: 'recv_rtp_src_%d_%d_%d'
    Availability: Sometimes
    Capabilities:
      application/x-rtp

  SRC template: 'send_rtcp_src_%d'
    Availability: On request
      Has request_new_pad() function: gst_rtp_bin_request_new_pad
    Capabilities:
      application/x-rtcp

  SRC template: 'send_rtp_src_%d'
    Availability: Sometimes
    Capabilities:
      application/x-rtp


Element Flags:
  no flags set

Bin Flags:
  no flags set

Element Implementation:
  Has change_state() function: gst_rtp_bin_change_state
  Has custom save_thyself() function: gst_bin_save_thyself
  Has custom restore_thyself() function: gst_bin_restore_thyself

Clocking Interaction:
  element requires a clock
  element is supposed to provide a clock but returned NULL

Indexing capabilities:
  element can do indexing

Pads:
  none

Element Properties:
  name                : The name of the object
                        flags: readable, writable
                        String. Default: null Current: "rtpbin0"
  async-handling      : The bin will handle Asynchronous state changes
                        flags: readable, writable
                        Boolean. Default: false Current: false
  latency             : Default amount of ms to buffer in the jitterbuffers
                        flags: readable, writable
                        Unsigned Integer. Range: 0 - 4294967295 Default: 200 Current: 0

Element Signals:
  "request-pt-map" :  GstCaps user_function (GstElement* object,
                                             guint arg0,
                                             guint arg1,
                                             gpointer user_data);

Element Actions:
  "clear-pt-map" :  void user_function (GstElement* object);


Title: Audio/Video conversation
Post by: kakaroto on October 07, 2008, 02:25:46 am
well.. that's what I thought :
Code:
Version: 0.10.5
License: LGPL
Source module: gst-plugins-bad

You have gst-plugins-bad 0.10.5, while you need at least 0.10.6, that's what I told you before but you said you had 0.10.8... if you have both versions installed, make sure you uninstall the old version, and make sure the new version is installed in /usr not /usr/local...


Title: Audio/Video conversation
Post by: Brian on October 07, 2008, 06:11:45 am
kakaroto,  ", that's what I told you before but you said you had 0.10.8.." You did and I did:) But who knows the vagaries of yast?
It turns out, I think, that I had BOTH loaded, probably because the packman ones had a different package name! Anyway I deleted the old ones and rebooted. They were still there and the gst command still showed 10.5, go figure. So I went back to packman and reinstalled the lot. I now get
Plugin Details:
  Name:                 gstrtpmanager
  Description:          RTP session management plugin library
  Filename:             /usr/lib/gstreamer-0.10/libgstrtpmanager.so
  Version:              0.10.8
  License:              LGPL
  Source module:        gst-plugins-bad
  Binary package:       GStreamer Bad Plug-ins source release
  Origin URL:           Unknown package origin

and no gst errors on the way up:)

Part of the problem was that I did not know how to check the versions. So if it is ok with you I will highlight the command here.


gst-inspect-0.10 gstrtpbin

So below is the sum total of all my messages at startup. Not had a chance to test a connection yet but it looks good.

FYI I support quite a few people here on Suse so I will now be able to get them going too.

Thanks for your support.

 amsn

ioctl: VIDIOC_QUERYCTRL(id=9963777;type=unknown;name="";minimum=0;maximum=0;step=0;default_value=0;flags=0): Input/output error
ioctl: VIDIOC_S_CTRL(id=9963776;value=178): Input/output error
** (<unknown>:6167): DEBUG: CODECS ARE READY

(<unknown>:6167): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:6167): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:6167): DEBUG: stun ip : x
** (<unknown>:6167): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:6167): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:6167): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: bus message : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: bus message : farsight-local-candidates-prepared
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-new-local-candidate
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-local-candidates-prepared
** (<unknown>:6167): DEBUG: CANDIDATES ARE PREPARED
                                                                                         



[/i]


Title: Audio/Video conversation
Post by: kakaroto on October 07, 2008, 10:40:50 am
Hey Brian, this is good news. I'm glad you got it figured out. Yes, the output from amsn does seem to be correct (yeah, those two warnings are my fault, sorry :p)
I'll wait for your feedback once you get a real call tested, but it should work fine now (unless you have some other old gst package)! :)


Title: Audio/Video conversation
Post by: Brian on October 08, 2008, 04:31:20 am
kakaroto, just tried to establish an audio call but got the errors below. Looks like I still may not have something quite right?

** (<unknown>:6167): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:6167): DEBUG: bus message : farsight-new-active-candidate-pair
** (<unknown>:6167): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : farsight-new-active-candidate-pair
** (<unknown>:6167): DEBUG: New active candidate pair :
** (<unknown>:6167): DEBUG: Local candidate: AkOChbq14oUjphND/KMkFO6Cdyw65TlpagEukJXij0I= 2 mPYiLe+PoEG1HTqQHh0           dDA== UDP 829 192.168.1.5 1050

** (<unknown>:6167): DEBUG: Remote candidate: e0ipTbd6Fs7hfiN9MqFH23cIixEzJkXfmzIQA+36Jeo= 2 F+VsPmIWXiLby3Ouh4           xr0Q== UDP 550 122.49.135.16 22094

E: socket-client.c: socket(): Address family not supported by protocol
** (<unknown>:6167): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:6167): DEBUG: Got an error on the BUS (1): Failed to connect: Connection refused (pulsesink.c(394           ): gst_pulsesink_open (): /GstPulseSink:autoaudiosink0-actual-sink-pulse)
** (<unknown>:6167): DEBUG: An error occured : Gstreamer error
** (<unknown>:6167): DEBUG: bus message : farsight-recv-codecs-changed
Segmentation fault


Title: Audio/Video conversation
Post by: kakaroto on October 08, 2008, 04:55:34 am
yeah, it looks like an error with pulseaudio, if you have pulse audio, try to disable it....
I'm interested though in that segmentation fault you're having.. could you give me a gdb backtrace of that ? (http://amsn-project.net/faq.php search for 'gdb')


Title: Audio/Video conversation
Post by: Brian on October 10, 2008, 01:01:15 pm
kakaroto, Murphy has been here! I ran  ulimit -c unlimited and then ran stuff to set up for a core dump. I then started amsn  --g-fatal-warnings. Then started an audio call with my brother and it WORKED:) We had a chat for about 15mins, sorted out some mixer settings no problem. I used to get the seg fault as soon as I started the audio call? So I guess we will just have to wait until it happens again, unless you thing setting the ulimit made a difference?
Anyway I am happy that I finally got to make my first real live call.
Many thanks for your support.

During the call I got the messages below.
1) I do not understand how the conversation worked if I got

E: socket-client.c: socket(): Address family not supported by protocol

2) I do not understand why I should get this  There is no particpant with cname Peters@home for ssrc 592249241 .

3) I did get lots of messages like
ioctl: VIDIOC_S_FMT(type=VIDEO_CAPTURE;fmt.pix.width=320;fmt.pix.height=240;fmt.pix.pixelformat=0x31384142 [BA81];fmt.pix.field=ANY;fmt.pix.bytesperline=0;fmt.pix.sizeimage=0;fmt.pix.colorspace=unknown;fmt.pix.priv=0): Device or resource busy





** (<unknown>:22939): DEBUG: Remote candidate: nmczbstwt0ZZmqVE+LZoeVFsQSN2aOzAYZAIJWRigAU= 1 JVM0TcN1bLxvlqT50jdtgA== UDP 550 115.166.24.247 10738

** (<unknown>:22939): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:22939): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:22939): DEBUG: bus message : farsight-new-active-candidate-pair
** (<unknown>:22939): DEBUG: Receive bus message from the event proc : farsight-new-active-candidate-pair
** (<unknown>:22939): DEBUG: New active candidate pair :
** (<unknown>:22939): DEBUG: Local candidate: eAKNmb3nHnlY2EN2/Il4SKR2GMZhNH0NGkvAN2quSN4= 2 RNx37A8K4gFwGgRdrkUh0g== UDP 829 192.168.1.5 1109

** (<unknown>:22939): DEBUG: Remote candidate: nmczbstwt0ZZmqVE+LZoeVFsQSN2aOzAYZAIJWRigAU= 2 JVM0TcN1bLxvlqT50jdtgA== UDP 550 115.166.24.247 46200

** (<unknown>:22939): DEBUG: bus message : farsight-component-state-changed
E: socket-client.c: socket(): Address family not supported by protocol
** (<unknown>:22939): DEBUG: bus message : farsight-recv-codecs-changed
** (<unknown>:22939): DEBUG: bus message : farsight-error
** (<unknown>:22939): DEBUG: bus message : GstRTPBinSDES
** (<unknown>:22939): DEBUG: Receive bus message from the event proc : farsight-error
** (<unknown>:22939): DEBUG: Error on BUS (11) There is no particpant with cname Peters@home for ssrc 592249241 .. There is no particpant with cname Peters@home for ssrc 592249241


Title: Audio/Video conversation
Post by: kakaroto on October 10, 2008, 07:35:48 pm
ok cool! I'm glad it worked!
I don't think the ulimit caused it to work, as it's a kernel thing, I think it was just murphy doing his job :p
In any case, I'm sure the segmentation fault was caused by the pulseaudio gstreamer element which was not working correctly
1 - it's a message by pulse audio, I don't know why but it seems it's a bug somewhere in there
2 - That's perfectly normal, because with RTP (the protocol used for real time communications), there are messages giving 'who' you are talking to, and that is what we received (The "Peters@home").. the problem is that we have no way of knowing that 'name' (cname is 'canonical name') of the person you are talking to since it's his windows username @ his pc's hostname.. but farsight expects to know that information, so that's what the error is all about.. But it's ok, because aMSN ignores that error, so it should just be considered as a warning instead of an actual error...
3 - that's not the audio call, it's the webcam capture because you were also sending your webcam.. that message might happen if we are trying to send faster than the camera is able to capture....


Title: Audio/Video conversation
Post by: Brian on October 11, 2008, 01:07:35 pm
kakaroto, Ok, I beat Murphy at his own game. Just tried to do an audio call and got a segfault and a core dump..

** (<unknown>:27162): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:27162): DEBUG: bus message : farsight-component-state-changed
** (<unknown>:27162): DEBUG: bus message : farsight-new-active-candidate-pair
** (<unknown>:27162): DEBUG: Receive bus message from the event proc : farsight-new-active-candidate-pair
** (<unknown>:27162): DEBUG: New active candidate pair :
** (<unknown>:27162): DEBUG: Local candidate: wPYmmjOOnGaZxZ00BXX3iGYXy1FaON7A42Rt8eg8rek= 2 9X1ZrON+htqk3+2ZVZxfXQ== UDP 829 192.168.1.5 1122

** (<unknown>:27162): DEBUG: Remote candidate: X+6S9W1mzcMOex0ql1xFrvVE9T+oaKaYm7SIrKxxVAQ= 2 3nZ4kmd9Glrg/OuHEwgX8A== UDP 550 122.49.163.150 44092

** (<unknown>:27162): DEBUG: bus message : farsight-component-state-changed
E: socket-client.c: socket(): Address family not supported by protocol
** (<unknown>:27162): DEBUG: Receive bus message from the event proc : GstMessageError
** (<unknown>:27162): DEBUG: Got an error on the BUS (1): Failed to connect: Connection refused (pulsesink.c(342): gst_pulsesink_open (): /GstPulseSink:autoaudiosink0-actual-sink-pulse)
** (<unknown>:27162): DEBUG: An error occured : Gstreamer error
** (<unknown>:27162): DEBUG: bus message : farsight-recv-codecs-changed
Segmentation fault (core dumped)

This is my initial go at getting something out of the dump. If there is a better command/s let me know.

gdb --core=/var/local/dumps/core.wish8.5.27162 file=/usr/bin/amsn
GNU gdb 6.6.50.20070726-cvs
Copyright (C) 2007 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "i586-suse-linux"...
file=/usr/bin/amsn: No such file or directory.
(no debugging symbols found)
Using host libthread_db library "/lib/libthread_db.so.1".
Core was generated by `wish8.5 /usr/bin/amsn --g-fatal-warnings'.
---Type <return> to continue, or q <return> to quit---
Program terminated with signal 11, Segmentation fault.
#0  0xb6524caa in ?? ()
(gdb) bt
#0  0xb6524caa in ?? ()
#1  0x00000000 in ?? ()


Title: Audio/Video conversation
Post by: kakaroto on October 11, 2008, 07:57:14 pm
humm... you couldn't get the backtrace because you don't have the debug symbols installed... I suggest you compile amsn with ./configure --enable-debug then recompile...
You might also need to install the glib dbg package..
then retry... and this time type :
thread all apply bt


Title: Audio/Video conversation
Post by: Brian on October 12, 2008, 05:03:04 pm
kakaroto, Ok done that. Took me a while because I ran out of space on the disk and had to fix that first. Tried running the command against the old dump, but got the same result, so will have to generate the error again.
Will get back when I can


Title: Audio/Video conversation
Post by: Brian on October 14, 2008, 02:27:46 pm
kakaroto, I have just run again, this time for about 10mins then got the crash. See output below. However, I did not get a dump. I recompiled no problem this time with
./configure --with-tcl=/usr/local/lib --with-tk=/usr/local/lib  --prefix=/usr --enable-debug  
but I did not get a core dump, just the output. My ulimit said unlimited, but do I have to run ulimit -c unlimited before each run of amsn? Cannot see why I did not get a core dump?

TUN error: RTP or other non-protocol packet!
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : Incorrectly multiplexed STUN message ignored.
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : s1:1: sending 52 bytes to [115.166.6.214]:42174
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : Packet received on local socket 32 from [115.166.6.214]:42174 (52 octets).
STUN error: RTP or other non-protocol packet!
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8: inbound STUN packet for 1/1 (stream/component) from [115.166.6.214]:42174 (52 octets) :
STUN error: RTP or other non-protocol packet!
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : Incorrectly multiplexed STUN message ignored.
Unsupported marker type 0x1b
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : s1:1: sending 52 bytes to [115.166.6.214]:42174
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : Packet received on local socket 32 from [115.166.6.214]:42174 (52 octets).
STUN error: RTP or other non-protocol packet!
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8: inbound STUN packet for 1/1 (stream/component) from [115.166.6.214]:42174 (52 octets) :
STUN error: RTP or other non-protocol packet!
** (<unknown>:21720): DEBUG: Agent 0xa07d9d8 : Incorrectly multiplexed STUN message ignored.
Oops: malloc_video_bufs is 1 (expected 0) [utils/linux/capture/libng/grab-ng.c:malloc_bufs_check:242]
Oops: processes is 1 (expected 0) [utils/linux/capture/libng/convert.c:process_check:179]


Title: Audio/Video conversation
Post by: kakaroto on October 14, 2008, 04:58:02 pm
humm.. you have to run ulimit -c unlimited every time you open the terminal... and the output doesn't show anything...
If you don't want to have to wait for the core dump, just run amsn directly from gdb with :
Code:
gdb --args wish amsn

then in gdb type 'run' and wait until it crashes.. then you can just do the 'bt' from gdb at that point.


Title: Audio/Video conversation
Post by: Brian on October 15, 2008, 12:01:50 am
I decide to recompile amsn and did an svn update and got revision 10578.
I then did a configure and got

compile time options summary
============================

    X11          : yes
    Tcl          : 8.5
    TK           : 8.5
    DEBUG        : yes
    STATIC       : no
    FARSIGHT     : yes
    LIBV4L       : no

I thendid a make clean all and make install.
Fired up amsn and about shows version 10577?
I also get

gdb --args wish8.5 /usr/bin/amsn
GNU gdb 6.6.50.20070726-cvs
Copyright (C) 2007 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "i586-suse-linux"...
(no debugging symbols found)
Using host libthread_db library "/lib/libthread_db.so.1".
(gdb) run
Starting program: /usr/local/bin/wish8.5 /usr/bin/amsn
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
[Thread debugging using libthread_db enabled]
[New Thread 0xb796b8d0 (LWP 1262)]
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
---Type <return> to continue, or q <return> to quit---
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
---Type <return> to continue, or q <return> to quit---
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
[New Thread 0xb792bb90 (LWP 1265)]
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
(no debugging symbols found)
Error while mapping shared library sections:
utils/TkCximage/TkCximage.so: No such file or directory.
Error while mapping shared library sections:
utils/linux/traydock/libtray.so: No such file or directory.
[Thread 0xb792bb90 (LWP 1265) exited]
Program exited normally.


Title: Audio/Video conversation
Post by: kakaroto on October 15, 2008, 07:55:25 am
it says "Program exited normally", so it didn't crash, it just quit... you also don't seem to have launched an audio call since I don't see any farsight messages... so this doesn't help.
Anyways, like I said, if you want it to stop crashing, just disable pulseaudio... it just looks like a bug in the pulseaudio code...


Title: Audio/Video conversation
Post by: Kalinda on October 15, 2008, 11:49:56 pm
Hey, uh, I'm wondering.. for all us who don't wanna compile all this stuff, could we get some packages, perhaps for Ubuntu Hardy?

Or is the voice chat not ready for that yet?

Thanks :)


Title: Audio/Video conversation
Post by: kakaroto on October 16, 2008, 12:10:17 am
Hi Kalinda,
the voice chat is ready, but I haven't released libnice yet, so I'm waiting until we release the 0.0.2 version of libnice and 0.0.4 version of farsight2 (which will support libnice). It should be done sometime next week.
Once it's all done, then you can probably request ubuntu to make a package out of those... if they are too slow, I can make one...
But note that ubuntu hardy has old dependencies, so even if they create libnice and farsight2 packages, there dependencies will be too old to work, so you will need to change your apt source to Intrepid instead of Hardy... or just use the Intrepid packages as they should work on Hardy...
either way, packaging is not my concern :p


Title: Audio/Video conversation
Post by: trv on October 16, 2008, 03:11:38 pm
or just upgrade to Intrepid, it's pretty stable even now. Or just way 2 weeks until it's released. Once you're on intrepid, then you will need just the libnice and farsight2 debs (made from the packages kakaroto will release next week)


Title: Audio/Video conversation
Post by: farseeing on October 16, 2008, 03:48:48 pm
Hi Kakaroto,
(been busy for a while sorry for answering 2 pages late !). Thanks you for this, that sounds useful. I will try new svn version and be careful with sound mixing.

 

Quote
@farseeing :
ok, i got some news for you :
1 - the freezing seems to happen when someone sends you an invite and cancels it right away, so I'm guessing that while you were having that audio call, someone sent you an invite and canceled it (or something similar happened) and that made amsn freeze... that issue is now fixed in the latest SVN version of amsn.
2 - The problem where amsn says you don't have farsight because it reports :

Quote:
** (<unknown>:8314): DEBUG: Got an error on the BUS (4): Could not open audio device for recording. Device is being used by another application. (gstalsasrc.c(635): gst_alsasrc_open (): /pipeline/gconfaudiosrc1/bin5/alsasrc1:
Device 'hw:1,0' is busy)

this happens when amsn tries to launch farsight at the same time as it's playing a sound, so if for example, you start a call, cancel it right away, then try to start it again (before the 'ringing' sound has finished playing), then it will have this error..
If you try to launch amsn at the same time as someone connects/changes state or whatever that causes amsn to play a sound, then it will fail... so that's the trick! Hopefully, we'll have sometime in the future a better system for all of this, don't worry.

p.s.: I still get the memory corruption, but it seems much harder to reproduce...


Title: Audio/Video conversation
Post by: Kalinda on October 16, 2008, 03:59:19 pm
Quote from: "kakaroto"
Hi Kalinda,
the voice chat is ready, but I haven't released libnice yet, so I'm waiting until we release the 0.0.2 version of libnice and 0.0.4 version of farsight2 (which will support libnice). It should be done sometime next week.
Once it's all done, then you can probably request ubuntu to make a package out of those... if they are too slow, I can make one...
But note that ubuntu hardy has old dependencies, so even if they create libnice and farsight2 packages, there dependencies will be too old to work, so you will need to change your apt source to Intrepid instead of Hardy... or just use the Intrepid packages as they should work on Hardy...
either way, packaging is not my concern :p

Alrighty, sounds good :) Thanks

Quote from: "trv"
or just upgrade to Intrepid, it's pretty stable even now. Or just way 2 weeks until it's released. Once you're on intrepid, then you will need just the libnice and farsight2 debs (made from the packages kakaroto will release next week)

Yeeeah... I could... buuuut... I use Kubuntu... and KDE 4 is not ready for me yet... perhaps in January when they release 4.2. I'm glad the release I'm stuck with is LTS, though. That doesn't matter too much anyhow, as Intrepid packages will work in Hardy.


Title: Audio/Video conversation
Post by: lordamus on October 17, 2008, 04:44:43 am
I did everything in your post but amsn svn cant find farsight2 to compile sadly :(   I`m on F9 x86_64    Any clue ?       (I did ldconfig thing as writed)

checking for GLIB... yes                                        
checking for GST... yes                                        
checking for FARSIGHT2... no                                    
checking for LIBV4L... no                                      
configure: creating ./config.status                            
config.status: creating Makefile                                
config.status: creating utils/linux/capture/config.h            
config.status: utils/linux/capture/config.h is unchanged        

compile time options summary
============================

    X11          : yes
    Tcl          : 8.5
    TK           : 8.5
    DEBUG        : no
    STATIC       : no
    FARSIGHT     : no
    LIBV4L       : no




Quote from: "MastaG"
Quote from: "farseeing"
Got it work on Fedora 9 !
Spent hours with snack2.2 and finally gave up and tried farsight. So for those who may be helped by that, here are some tips I had to do to adapt the procedure on the beginning of this thread to my needs.

I'm using :
- fedora 9 with kernel 1.6.25 (due to graphic driver issues, hum, well...anyway)
-pulse-audio, alsa and all this stuff I don't understand anything in !
-aMSN from svn : v.10479 at this very second
- farsight 2 : 0.0.3 (had to cheat see below) --> http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
- gst-plugins : 0.12.9  --> use yum or add/remove software, all plugins available for F9 in my configured repositories (usuals + livna)
- libnice.


I've just followed the above procedure (obviously using yum instead of apt)  except for several points where I had to struggle with versions problems :

# I followed the procedure using git but the version of farsight2 there didn't seem to work.
so I downloaded farsight2-0.0.3.tar.gz (see link above), unpacked and manually copy files in farsight2-0.0.3/common from the git tree of farsight2 I created : gst-autogen.sh, gst.supp. -I have 4 files & 1 folder in this directory). I'm then using this version of farsight2 instead of the git tree version

# After the ./configure step in farsight2-0.0.3 installation and just before the make, I edited farsight2.pc and farsight2-0.10.pc and change version 0.0.3 by 0.0.3.1. then go back to the procedure and perform make and make install. (otherwise aMSN doesn't recognized farsight.).

#  (yes, don't forget to do the ldconfig as prescribed !)

# then I installed aMSN

AND THAT'S WORKIN DAMN FINE RIGHT NOW !!!!
(http://[url=http://www.postimage.org/image.php?v=aV1LK8wJ][img]http://www.postimage.org/aV1LK8wJ.jpg)[/url]
[/img]


Hell yes!
Same here :)

So to sum things up:

Install the following packages from the repo's:
Code:

gstreamer-plugins-bad-0.10.7-1.lvn9.i386
gstreamer-0.10.19-1.fc9.i386
gstreamer-plugins-good-0.10.8-8.fc9.i386
gstreamer-tools-0.10.19-1.fc9.i386
gstreamer-plugins-pulse-0.9.5-0.5.svn20070924.fc9.i386
gstreamer-plugins-base-devel-0.10.19-2.fc9.i386
gstreamer-plugins-bad-extras-0.10.7-1.lvn9.i386
gstreamer-plugins-base-0.10.19-2.fc9.i386
gstreamer-plugins-bad-devel-0.10.7-1.lvn9.i386
gstreamer-devel-0.10.19-1.fc9.i386
gstreamer-plugins-good-devel-0.10.8-8.fc9.i386
gstreamer-plugins-ugly-0.10.8-1.lvn9.i386


Install the latest gst-plugins-farsight (the one from the repo's is outdated).
Code:

wget http://farsight.freedesktop.org/releases/gst-plugins-farsight/gst-plugins-farsight-0.12.9.tar.gz
tar zxfv gst-plugins-farsight-0.12.9.tar.gz
cd gst-plugins-farsight-0.12.9
./configure --prefix=/usr
make
sudo make install
cd ..


Obtain and install libnice (be sure to have git installed: sudo yum install git)
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git checkout -b nice-kakaroto origin/nice-kakaroto
./autogen.sh --prefix=/usr
make
sudo make install
cd ..


Install farsight2
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
cd farsight2
git checkout -b nice origin/nice
cd ..
wget http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
tar zxfv farsight2-0.0.3.tar.gz
cd farsight2-0.0.3
cp -f ../farsight2/common/check.mak ./common/
cp -f ../farsight2/common/gst.supp ./common/
cp -f ../farsight2/common/gst-autogen.sh ./common/
cp -f ../farsight2/common/gtk-doc.mak ./common/
./autogen --prefix=/usr --disable-python
nano -w farsight2.pc

Now change the version from 0.0.3 to 0.0.3.1
Press CTRL+O to save and CTRL+X to quit nano.
Code:

make
sudo make install
sudo /sbin/ldconfig
cd ..


Voila, check out the latest svn of amsn and you should be able to make voice calls:)


Title: Audio/Video conversation
Post by: kakaroto on October 17, 2008, 07:07:36 am
make sure you used the GIT repository of farsight2, that you did the checkout of the appropriate branch, and that you configured it with --prefix=/usr...
Many people seem to screw it up, but then when they retry everything and make sure they follow my step by step guide, it works fine for them in the end...


Title: Audio/Video conversation
Post by: lordamus on October 17, 2008, 05:22:34 pm
yeah First I tryed farsight2.0.0.3 then your GIT repository but farsight2 cant find libnice to compile error..LIbnice compiled well from your repository(I use ./autogen.sh --prefix=/usr )

here is libnice compile info    http://pastebin.com/d477f45bb     some incompatible warnings there..

 farsight2]# ./autogen.sh --prefix=/usr --disable-python
+ passing argument --prefix=/usr to configure                      
+ passing argument --disable-python to configure                    
+ options passed to configure:  --prefix=/usr --disable-python      
+ check for build tools                                            
  checking for autoconf >= 2.52 ... found 2.61, ok.                
  checking for automake >= 1.7 ... found 1.10.1, ok.                
  checking for libtoolize >= 1.5.0 ... found 1.5.24, ok.            
  checking for pkg-config >= 0.8.0 ... found 0.23, ok.              
+ running aclocal -I common/m4 ...                                  
+ running libtoolize --copy --force...                              
+ running autoheader ...                                            
+ running autoconf ...                                              
+ running automake -a -c -Wno-portability...                        
+ running configure ...                                            
  ./configure default flags: --enable-gtk-doc --enable-plugin-docs  
  ./configure external flags:  --prefix=/usr --disable-python      

checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes          
checking for a thread-safe mkdir -p... /bin/mkdir -p        
checking for gawk... gawk                                  
checking whether make sets $(MAKE)... yes                  
checking nano version... 1                                  
checking build system type... x86_64-unknown-linux-gnu      
checking host system type... x86_64-unknown-linux-gnu      
checking for style of include used by make... GNU          
checking for gcc... gcc                                    
checking for C compiler default output file name... a.out  
checking whether the C compiler works... yes                
checking whether we are cross compiling... no              
checking for suffix of executables...                                                                          
checking for suffix of object files... o                                                                        
checking whether we are using the GNU C compiler... yes                                                        
checking whether gcc accepts -g... yes                                                                          
checking for gcc option to accept ISO C89... none needed                                                        
checking dependency style of gcc... gcc3                                                                        
checking for a sed that does not truncate output... /bin/sed                                                    
checking for grep that handles long lines and -e... /bin/grep                                                  
checking for egrep... /bin/grep -E                                                                              
checking for ld used by gcc... /usr/bin/ld                                                                      
checking if the linker (/usr/bin/ld) is GNU ld... yes                                                          
checking for /usr/bin/ld option to reload object files... -r                                                    
checking for BSD-compatible nm... /usr/bin/nm -B                                                                
checking whether ln -s works... yes                                                                            
checking how to recognize dependent libraries... pass_all                                                      
checking how to run the C preprocessor... gcc -E                                                                
checking for ANSI C header files... yes                                                                        
checking for sys/types.h... yes                                                                                
checking for sys/stat.h... yes                                                                                  
checking for stdlib.h... yes                                                                                    
checking for string.h... yes                                                                                    
checking for memory.h... yes                                                                                    
checking for strings.h... yes                                                                                  
checking for inttypes.h... yes                                                                                  
checking for stdint.h... yes                                                                                    
checking for unistd.h... yes                                                                                    
checking dlfcn.h usability... yes                                                                              
checking dlfcn.h presence... yes                                                                                
checking for dlfcn.h... yes                                                                                    
checking for g++... g++                                                                                        
checking whether we are using the GNU C++ compiler... yes                                                      
checking whether g++ accepts -g... yes                                                                          
checking dependency style of g++... gcc3                                                                        
checking how to run the C++ preprocessor... g++ -E                                                              
checking for g77... no                                                                                          
checking for xlf... no                                                                                          
checking for f77... no                                                                                          
checking for frt... no                                                                                          
checking for pgf77... no                                                                                        
checking for cf77... no                                                                                        
checking for fort77... no                                                                                      
checking for fl32... no                                                                                        
checking for af77... no                                                                                        
checking for xlf90... no                                                                                        
checking for f90... no                                                                                          
checking for pgf90... no                                                                                        
checking for pghpf... no                                                                                        
checking for epcf90... no                                                                                      
checking for gfortran... gfortran                                                                              
checking whether we are using the GNU Fortran 77 compiler... yes                                                
checking whether gfortran accepts -g... yes                                                                    
checking the maximum length of command line arguments... 1966080                                                
checking command to parse /usr/bin/nm -B output from gcc object... ok                                          
checking for objdir... .libs                                                                                    
checking for ar... ar                                                                                          
checking for ranlib... ranlib                                                                                  
checking for strip... strip                                                                                    
checking if gcc supports -fno-rtti -fno-exceptions... no                                                        
checking for gcc option to produce PIC... -fPIC                                                                
checking if gcc PIC flag -fPIC works... yes                                                                    
checking if gcc static flag -static works... yes                                                                
checking if gcc supports -c -o file.o... yes                                                                    
checking whether the gcc linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes                    
checking whether -lc should be explicitly linked in... no                                                      
checking dynamic linker characteristics... GNU/Linux ld.so                                                      
checking how to hardcode library paths into programs... immediate                                              
checking whether stripping libraries is possible... yes                                                        
checking for shl_load... no                                                                                    
checking for shl_load in -ldld... no
checking for dlopen... no
checking for dlopen in -ldl... yes
checking whether a program can dlopen itself... yes
checking whether a statically linked program can dlopen itself... no
checking if libtool supports shared libraries... yes
checking whether to build shared libraries... yes
checking whether to build static libraries... yes
configure: creating libtool
appending configuration tag "CXX" to libtool
checking for ld used by g++... /usr/bin/ld -m elf_x86_64
checking if the linker (/usr/bin/ld -m elf_x86_64) is GNU ld... yes
checking whether the g++ linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes
checking for g++ option to produce PIC... -fPIC
checking if g++ PIC flag -fPIC works... yes
checking if g++ static flag -static works... yes
checking if g++ supports -c -o file.o... yes
checking whether the g++ linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes
checking dynamic linker characteristics... GNU/Linux ld.so
checking how to hardcode library paths into programs... immediate
appending configuration tag "F77" to libtool
checking if libtool supports shared libraries... yes
checking whether to build shared libraries... yes
checking whether to build static libraries... yes
checking for gfortran option to produce PIC... -fPIC
checking if gfortran PIC flag -fPIC works... yes
checking if gfortran static flag -static works... yes
checking if gfortran supports -c -o file.o... yes
checking whether the gfortran linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes
checking dynamic linker characteristics... GNU/Linux ld.so
checking how to hardcode library paths into programs... immediate
checking for pkg-config... /usr/bin/pkg-config
checking pkg-config is at least version 0.9.0... yes
checking for VALGRIND... no
no
configure: Using Farsight2 source release as package name
configure: Using Unknown package origin as package origin
configure: Using /usr/libexec/gst-install-plugins-helper as plugin install helper

configure: *** checking feature: enable building of plug-ins with external deps ***
configure: building external plug-ins
checking for NICE... no
configure: error: Could not build libnice plugin because libnice is not installed
  configure failed


Quote from: "kakaroto"
make sure you used the GIT repository of farsight2, that you did the checkout of the appropriate branch, and that you configured it with --prefix=/usr...
Many people seem to screw it up, but then when they retry everything and make sure they follow my step by step guide, it works fine for them in the end...


Title: Audio/Video conversation
Post by: kakaroto on October 17, 2008, 06:03:21 pm
hummm... libnice seems to install correctly.
Can you please see if there's a file  /usr/lib/pkgconfig/nice.pc on your system.
and type
Code:
pkg-config nice --modversion
and paste here the output...
this should tell us if pkg-config can find libnice or not.
thanks!


Title: Audio/Video conversation
Post by: lordamus on October 17, 2008, 09:40:44 pm
farsight2]# pkg-config nice --modversion
Package nice was not found in the pkg-config search path.
Perhaps you should add the directory containing `nice.pc'
to the PKG_CONFIG_PATH environment variable
No package 'nice' found

yea there is a file nice.pc  in  /usr/lib/pkgconfig/    

prefix=/usr
exec_prefix=${prefix}
libdir=${exec_prefix}/lib
includedir=${prefix}/include

Name: libnice
Description: ICE library
Requires: glib-2.0 >= 2.10.0
Version: 0.0.1
Libs: -L${libdir} -lnice -pthread -lgobject-2.0 -lgthread-2.0 -lrt -lglib-2.0
Cflags: -pthread -I/usr/include/glib-2.0 -I/usr/lib64/glib-2.0/include   -I${includedir} -I${includedir}/nice

Maybe some shared lib problem on my x86_64 fedora  ?


Quote from: "kakaroto"
hummm... libnice seems to install correctly.
Can you please see if there's a file  /usr/lib/pkgconfig/nice.pc on your system.
and type
Code:
pkg-config nice --modversion
and paste here the output...
this should tell us if pkg-config can find libnice or not.
thanks!


Title: Audio/Video conversation
Post by: kakaroto on October 18, 2008, 06:05:29 pm
humm... :
Code:
echo $PKG_CONFIG_PATH

? in theory /usr/lib/pkgconfig should be the default.. maybe something is different on your system.


Title: Audio/Video conversation
Post by: lordamus on October 19, 2008, 12:51:59 pm
that command shows nothing... I`ve  /usr/lib/pkgconfig  and /usr/lib64/pkgconfig   /usr/share/pkgconfig...   also  /usr/local/lib/pkgconfig/nice.pc     Any clue ?

Quote from: "kakaroto"
humm... :
Code:
echo $PKG_CONFIG_PATH

? in theory /usr/lib/pkgconfig should be the default.. maybe something is different on your system.


Title: Audio/Video conversation
Post by: kakaroto on October 19, 2008, 07:06:17 pm
nope!, no clue at all! try pasting here the output of
Code:
pkg-config --debug --print-errors nice --modversion
with this --debug option, it should give you the list of all files it scans! if there's anything wrong, then you'll find it with it.. maybe try to see if it searches for .pc files in /usr/lib/pkgconfig or if it looks in another directory...
either way, in theory if you do "export PKG_CONFIG_PATH=/usr/lib/pkgconfig" (since you said the file /usr/lib/pkgconfig/nice.pc exists there) then it should work....
make sure btw that the file is readable!


Title: Audio/Video conversation
Post by: lordamus on October 19, 2008, 09:55:49 pm
here is the output   http://pastebin.com/d2302dd38      package nice not found...it looks directly to /usr/lib64/pkgconfig   so I`m on 64bit system... Maybe if I change pkg_config_path to /usr/lib/pkgconfig  there will be problems

Edit:  I did  export PKG_CONFIG_PATH=/usr/lib64/pkgconfig:/usr/lib/pkgconfig/  and then farsight2 compiled well with no errors and I did ldconfig then compiled amsn svn 10606 without errors it found farsight well..But when I open amsn audio/video config it cant find farsight module ?  I did package require Farsight in amsn console it returns "0.1" and then farsight Prepare 1 it returns "Couldn't create fsrtpconference"   what can I do ?   I`m using gst-plugins-farsight-0.12.9  (compiled)


Quote from: "kakaroto"
nope!, no clue at all! try pasting here the output of
Code:
pkg-config --debug --print-errors nice --modversion
with this --debug option, it should give you the list of all files it scans! if there's anything wrong, then you'll find it with it.. maybe try to see if it searches for .pc files in /usr/lib/pkgconfig or if it looks in another directory...
either way, in theory if you do "export PKG_CONFIG_PATH=/usr/lib/pkgconfig" (since you said the file /usr/lib/pkgconfig/nice.pc exists there) then it should work....
make sure btw that the file is readable!


Title: Audio/Video conversation
Post by: kakaroto on October 19, 2008, 11:48:56 pm
you're missing gstreamer-plugins-bad.
Again.. like I said, make sure you follow the wiki and don't forget any dependency!!!!

EDIT: it looks like it actually can't find the farsight2 plugin.. make sure you installed farsight2 with --prefix=/usr... again, read the guide correctly and do everything as it tells you to...


Title: Audio/Video conversation
Post by: lordamus on October 20, 2008, 03:15:00 am
that is allready installed gstreamer-plugins-bad-0.10.7-1.lvn9.x86_64    I compiled farsight2 with ./autogen.sh --prefix=/usr --disable-python and then I did ldconfig

rpm -qa |grep gstreamer
gstreamer-0.10.21-1.fc10.x86_64
gstreamer-devel-0.10.21-1.fc10.i386
gstreamer-plugins-good-0.10.8-8.fc9.x86_64
gstreamer-plugins-base-0.10.21-2.fc10.x86_64
gstreamer-plugins-base-devel-0.10.21-2.fc10.i386
gstreamer-plugins-base-0.10.21-2.fc10.i386
gstreamer-plugins-pulse-0.9.5-0.5.svn20070924.fc9.x86_64
gstreamer-plugins-flumpegdemux-0.10.15-2.fc9.x86_64
gstreamer-plugins-good-devel-0.10.8-8.fc9.x86_64
totem-gstreamer-2.23.2-7.fc9.x86_64
gstreamer-plugins-bad-devel-0.10.7-1.lvn9.x86_64
gstreamer-python-0.10.11-2.fc9.x86_64
gstreamer-plugins-good-devel-0.10.8-8.fc9.i386
gstreamer-devel-0.10.21-1.fc10.x86_64
gstreamer-plugins-bad-0.10.7-1.lvn9.x86_64
gstreamer-tools-0.10.21-1.fc10.x86_64
gstreamer-plugins-bad-extras-0.10.7-1.lvn9.x86_64
gstreamer-plugins-ugly-0.10.8-1.lvn9.x86_64
gstreamer-0.10.21-1.fc10.i386
gstreamer-plugins-base-devel-0.10.21-2.fc10.x86_64

Quote from: "kakaroto"
you're missing gstreamer-plugins-bad.
Again.. like I said, make sure you follow the wiki and don't forget any dependency!!!!

EDIT: it looks like it actually can't find the farsight2 plugin.. make sure you installed farsight2 with --prefix=/usr... again, read the guide correctly and do everything as it tells you to...


Title: Audio/Video conversation
Post by: kakaroto on October 20, 2008, 07:11:19 am
ok, but for some reason it can't find farsight2...  maybe I know why though... I think when installing farsight2, it installed to /usr/lib/gstreamer-0.10 while gstreamer tries to find farsight's plugins in /usr/lib64/gstreamer-0.10...
Maybe the fix would be to recompile libnice and farsight2 with :
Code:
./configure --prefix=/usr --libdir=/usr/lib64

try that and see if it works.


Title: Audio/Video conversation
Post by: lordamus on October 20, 2008, 07:43:34 am
No luck :(    libnice compiled but farsight2 gived errors with "./configure --prefix=/usr --libdir=/usr/lib64"     yea files in /usr/lib64/gstreamer-0.10

here is farsight2 compile errors  http://amsn.pastebin.com/d7d3ba5cc

Quote from: "kakaroto"
ok, but for some reason it can't find farsight2...  maybe I know why though... I think when installing farsight2, it installed to /usr/lib/gstreamer-0.10 while gstreamer tries to find farsight's plugins in /usr/lib64/gstreamer-0.10...
Maybe the fix would be to recompile libnice and farsight2 with :
Code:
./configure --prefix=/usr --libdir=/usr/lib64

try that and see if it works.


Title: Audio/Video conversation
Post by: farseeing on October 20, 2008, 10:17:47 am
Hi,
Tried out svn version10588. Looks more stable relating to audio calls. It doesn't seem to freeze anymore for me. But I still have an issue with a contact (the same WLM user I've problem with since the beginning) when making audio.  The voice stream I receive from him is interrupted in a strange manner : I can get approximately 3 secondes of voice then it fades intto half a second silence and get back. It does that every 2-3 seconds (maybe less). If he speaks very slowly marking a pause beetween each two words he says, I can get everything he says, otherwise I miss some words (What that makes us do !!!). Looks like buffering problems, don't know. My contact receives my voice stream correctly (continuous).
He has audio calls working fine with other MS users, I've them working fine with other MS users too. That's why I post.
Ctrl-S doesn't display anything useful, just 'keeping alive'.  
Forgot to compile with debug option this time, tell me if it can be of some use.


Title: Audio/Video conversation
Post by: kakaroto on October 20, 2008, 04:24:03 pm
@lordamus: it should work if you recompile farsight2 with ./configure --prefix=/usr --libdir=/usr/lib64 --disable-python
@farseeing : humm.. I would guess that the problem is caused by either one of these :
1 - he has a low bandwidth (or his bandwidth is being used by something else like bittorrent) which is why he can't send you data fast enough
2 - his microphone's volume is too low, when MSN detects that there is 'no sound' (or too low so it thinks it's just background noise), it doesn't send anything.
3 - you're firewalled or both behind symetric NATs, so it has to use the relay TURN server, which itself sets a bandwidth limitation on your stream (but it should not affect a low bandwidth audio call).

the other possibilities is that farsight has a bug, but that's very very not probably (it does work fine with everyone else, right?). If we would want to 'debug' this, it would be really really complicated, so I don't think it's useful to tell you how... but you could try to use wireshark and see the data that you receive... once it's sniffed, just filter by UDP, find his own ip address and filter to see only the packets that you've received from him. You can right click a packet and do 'decode as' and select RTP. You should then look at the timestamps of each packet.. you should be receiving one packet every 40ms I think.. so look at the timestamps to see if you do receive packets every 40ms.. and if there is a gap in the timestamps... You can also look at the RTP timestamps if you want.


Title: Audio/Video conversation
Post by: lordamus on October 20, 2008, 09:35:04 pm
farsight2 installed well with "./configure --prefix=/usr --libdir=/usr/lib64 --disable-python "  after I did ldconfig too but amsn still cant find farsight when ./configure  also my /etc/ld.so.conf  nothing related to gstreamer or farsight..

here is the files in  /usr/lib64/gstreamer-0.10      http://amsn.pastebin.com/d7d3ba5cc


Title: Audio/Video conversation
Post by: kakaroto on October 20, 2008, 09:39:18 pm
humm.. well, did you install farsight2 correctly? what does amsn say? it can't find it in the ./configure? what does pkg-config farsight2-0.10 --modversion gives you?
your pastebin is wrong, but I'm more interested in knowing whether amsn can detect farsight in the ./configure...


Title: Audio/Video conversation
Post by: lordamus on October 20, 2008, 09:58:33 pm
yea farsight2 installed well without errors.. amsn cant find farsight in the ./configure (checking for FARSIGHT2... no You do not seem to have gstreamer and farsight2 installed)

pkg-config farsight2-0.10 --modversion gives =  0.0.2.1

Quote from: "kakaroto"
humm.. well, did you install farsight2 correctly? what does amsn say? it can't find it in the ./configure? what does pkg-config farsight2-0.10 --modversion gives you?
your pastebin is wrong, but I'm more interested in knowing whether amsn can detect farsight in the ./configure...


Title: Audio/Video conversation
Post by: kakaroto on October 20, 2008, 11:27:49 pm
ok, well, the version is 0.0.2.1, which means you didn't use the correct branch.. make sure you do the 'git checkout origin/nice' as instructed.


Title: Audio/Video conversation
Post by: lordamus on October 20, 2008, 11:33:44 pm
I allready did git checkout -b origin/nice when I was first compile session..(fatal: A branch named 'origin/nice' already exists.
) do I need to re-checkout every compile try ?

Edit:  I deleted every folder and re-get via git  there is no autogen.sh file in libnice/ folder ?  


Quote from: "kakaroto"
ok, well, the version is 0.0.2.1, which means you didn't use the correct branch.. make sure you do the 'git checkout origin/nice' as instructed.


Title: Audio/Video conversation
Post by: kakaroto on October 21, 2008, 03:12:21 am
I never said 'git checkout -b origin/nice', I said 'git checkout origin/nice' in farsight2 and 'git checkout origin/nice-kakaroto' on libnice. You don't have to do it everytime, but you clearly failed to do it the first time, or something else happened that caused it to switch to master (deleted and restarted from scratch and forgot to do it the second time ?). In any case, if there is no autogen.sh in libnice then that's because you didn't switch to the appropriate branch!!!! Once again, I repeat, follow the step by step guide correctly!


Title: Audio/Video conversation
Post by: lordamus on October 21, 2008, 06:50:11 am
You cant do `git checkout origin/nice`  in fedora system the only correct command to switch branch 'git checkout -b origin/nice'  I didnt failed to do something

-----------------------------------------------------------------
libnice]# git checkout -b origin/nice
Switched to a new branch "origin/nice"
-----------------------------------------------------------------

If I use git checkout origin/nice   ---------->    error: pathspec 'origin/nice' did not match any file(s) known to git.

I`m using these branch`s

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
 cd libnice
 git checkout origin/nice-kakaroto
 cd ..
 git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
 cd farsight2
 git checkout origin/nice


Quote from: "kakaroto"
I never said 'git checkout -b origin/nice', I said 'git checkout origin/nice' in farsight2 and 'git checkout origin/nice-kakaroto' on libnice. You don't have to do it everytime, but you clearly failed to do it the first time, or something else happened that caused it to switch to master (deleted and restarted from scratch and forgot to do it the second time ?). In any case, if there is no autogen.sh in libnice then that's because you didn't switch to the appropriate branch!!!! Once again, I repeat, follow the step by step guide correctly!


Title: Audio/Video conversation
Post by: kakaroto on October 21, 2008, 05:10:21 pm
ok, well, you did fail to do it because you didn't switch to the branch!!!! I don't know why your git checkout wouldn't work (for sure it's not fedora's fault, maybe it's just your version of git that is too old), but anyways, the "git checkout -b origin/nice-kakaroto" is NOT the solution.. all it does is create a new branch without using my own branch, so you don't get any of the necessary changes... didn't you notice that when you typed it :
Code:
kakaroto% git checkout -b origin/nice-kakaroto
Switched to a new branch "origin/nice-kakaroto"

anyways, here are the solutions :
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git fetch origin/nice-kakaroto
git checkout origin/nice-kakaroto

or
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git pull origin nice-kakaroto

or
Code:
wget http://git.collabora.co.uk/?p=user/kakaroto/nice.git;a=snapshot;h=refs/heads/nice-kakaroto;sf=tgz
tar -xzvf nice-kakaroto.tar.gz
cd nice


The same goes for farsight2. You can either do a 'fetch' before the checkout, or do a 'git pull' instead of checkout (but use a space between origin and nice-kakaroto, not a '/' ) or use this link for a snapshot : http://git.collabora.co.uk/?p=user/kakaroto/farsight2.git;a=snapshot;h=refs/heads/nice;sf=tgz


Title: Audio/Video conversation
Post by: farseeing on October 21, 2008, 08:57:25 pm
Quote
@farseeing : humm.. I would guess that the problem is caused by either one of these :
1 - he has a low bandwidth (or his bandwidth is being used by something else like bittorrent) which is why he can't send you data fast enough
2 - his microphone's volume is too low, when MSN detects that there is 'no sound' (or too low so it thinks it's just background noise), it doesn't send anything.
3 - you're firewalled or both behind symetric NATs, so it has to use the relay TURN server, which itself sets a bandwidth limitation on your stream (but it should not affect a low bandwidth audio call).

the other possibilities is that farsight has a bug, but that's very very not probably (it does work fine with everyone else, right?). If we would want to 'debug' this, it would be really really complicated, so I don't think it's useful to tell you how... but you could try to use wireshark and see the data that you receive... once it's sniffed, just filter by UDP, find his own ip address and filter to see only the packets that you've received from him. You can right click a packet and do 'decode as' and select RTP. You should then look at the timestamps of each packet.. you should be receiving one packet every 40ms I think.. so look at the timestamps to see if you do receive packets every 40ms.. and if there is a gap in the timestamps... You can also look at the RTP timestamps if you want.


yep works fine with everyone else( but for both of us, however).
thanks, I will try to find. thought about second hypothesis too. rejected first (at least for bittorrent !). third one drives me mad so I hope it's not !


Title: Audio/Video conversation
Post by: kakaroto on October 21, 2008, 09:45:06 pm
hehe, I don't think it's the 3rd one, because the server is made to work with audio calls correctly.. the bandwidth limitation is usually just for video transfers (which is not even used on those servers) so it's superior to what an audio call can use (even uncompressed, the bandwidth is lower than the server's limit).
I would still go for the volume problem or he has a low bandwidth...


Title: Audio/Video conversation
Post by: Brian on October 26, 2008, 02:33:35 pm
Last time we spoke I was having problems with audio and you suggested removing pulse.
Well I did that and have recompiled a couple of times and it only seemed to get worse.
I eventually got to the point where is said it could not find farsight. Anyway today   I found your statements where you PULL both libnice and farsight so I decided to redo everything. Just to complete the doco I used the --disable-alsa switch as well this time. The recompile of libnicce/farsight went well then I pulled amsn again and
 Updated to revision 10641

I have just completed a 30min call with my father with no hitches at all. So Suse 10.3 plus pkg-config farsight2-0.10 --modversion  0.0.3.1 plus amsn 10641 is looking good.

Hope to do another long test shortly.

The only thing I found on the log was
W: client-conf.c: Failed to open configuration file '/etc/pulse/client.conf': No such file or directory
E: socket-client.c: socket(): Address family not supported by protocol

but as I said it did not seem to effect anything.


Title: Audio/Video conversation
Post by: lordamus on October 26, 2008, 03:04:30 pm
Ok I did that libnice compiled well but now farsight2 cant find libnice  ?

Quote from: "kakaroto"
ok, well, you did fail to do it because you didn't switch to the branch!!!! I don't know why your git checkout wouldn't work (for sure it's not fedora's fault, maybe it's just your version of git that is too old), but anyways, the "git checkout -b origin/nice-kakaroto" is NOT the solution.. all it does is create a new branch without using my own branch, so you don't get any of the necessary changes... didn't you notice that when you typed it :
Code:
kakaroto% git checkout -b origin/nice-kakaroto
Switched to a new branch "origin/nice-kakaroto"

anyways, here are the solutions :
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git fetch origin/nice-kakaroto
git checkout origin/nice-kakaroto

or
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git pull origin nice-kakaroto

or
Code:
wget http://git.collabora.co.uk/?p=user/kakaroto/nice.git;a=snapshot;h=refs/heads/nice-kakaroto;sf=tgz
tar -xzvf nice-kakaroto.tar.gz
cd nice


The same goes for farsight2. You can either do a 'fetch' before the checkout, or do a 'git pull' instead of checkout (but use a space between origin and nice-kakaroto, not a '/' ) or use this link for a snapshot : http://git.collabora.co.uk/?p=user/kakaroto/farsight2.git;a=snapshot;h=refs/heads/nice;sf=tgz


Title: Audio/Video conversation
Post by: kakaroto on October 27, 2008, 01:17:55 am
@Brian : well.. I'm wondering whether that warning is because pulse is causing problems or if it's just the "autoaudiosink" that tries pulse, gets the error, so it then tries something else (alsa, oss, whatever) then it works.. so maybe that error should just be ignored for now!
Glad it works fine for you! :)

@lordamus : which method did you use ? If it can't find libnice, maybe you didn't install it correctly! Anyways, the fact that farsight says it can't find libnice means that you have the correct branch of farsight (the one that actually needs libnice).. so just make sure you compile it and install it all correctly.


P.s: Expect a release of libnice and farsight2 tomorrow, so this should all become much easier!


Title: Audio/Video conversation
Post by: lordamus on October 27, 2008, 02:13:12 pm
this one...
git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git fetch origin/nice-kakaroto
git checkout origin/nice-kakaroto

I get allready up to date message then I compiled with no errors  thats all..


Quote from: "kakaroto"
@Brian : well.. I'm wondering whether that warning is because pulse is causing problems or if it's just the "autoaudiosink" that tries pulse, gets the error, so it then tries something else (alsa, oss, whatever) then it works.. so maybe that error should just be ignored for now!
Glad it works fine for you! :)

@lordamus : which method did you use ? If it can't find libnice, maybe you didn't install it correctly! Anyways, the fact that farsight says it can't find libnice means that you have the correct branch of farsight (the one that actually needs libnice).. so just make sure you compile it and install it all correctly.


P.s: Expect a release of libnice and farsight2 tomorrow, so this should all become much easier!


Title: Audio/Video conversation
Post by: kakaroto on October 27, 2008, 04:48:06 pm
so... I repeat.. follow the guide step by step!!! just make sure you do the 'git fetch origin/nice-kakroto' before the checkout, but redo everything correctly from the step by step guide and it should work...


Title: Audio/Video conversation
Post by: Brian on October 28, 2008, 01:08:49 pm
kakaroto, this is just for future reference. I got the bug below when I was trying to start a voice call. I have been having trouble with my sound after a reboot so I went to check it. Sure enough my sound card was out. ran alsaconf which reloaded the driver. Went back to amsn and started the voice call no problem. Was on for about 20mins.
Thanks,
Brian
* (<unknown>:6017): DEBUG: Got an error on the BUS (1): Failed to connect: Connection refused (pulsesink.c(342): gst_pulsesink_open (): /GstPulseSink:autoaudiosink0-actual-sink-pulse)
** (<unknown>:6017): DEBUG: An error occured : Gstreamer error
** (<unknown>:6017): DEBUG: bus message : farsight-recv-codecs-changed

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0xb235cb90 (LWP 6285)]
0xb657acaa in ?? () from /usr/lib/libgstbase-0.10.so.0
(gdb) bt
#0  0xb657acaa in ?? () from /usr/lib/libgstbase-0.10.so.0
#1  0x00000000 in ?? ()
(gdb) run


Title: Audio/Video conversation
Post by: trv on October 28, 2008, 02:05:14 pm
seems like your pulseaudio daemon was dead or crashed, so you could not connect to it.


Title: Audio/Video conversation
Post by: kakaroto on October 28, 2008, 05:16:41 pm
yeah... it looks like it.. the problem is that gstreamer should not crash the application in such a case...
Brian, make sure your sound card (and all other cards) are correctly screwed in your pc, if they get disconnected while your PC is on, it might damage the card or the PC...

p.s.: btw, I am currently in the process of releasing libnice and farsight2 into a stable release.. so if you can all just wait for a little while, because the current git is not synced with farsight and may not compile.


Title: Audio/Video conversation
Post by: microcris on October 30, 2008, 01:47:23 pm
Hi there :)

I did this
Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/nice.git libnice
cd libnice
git pull origin nice-kakaroto
./autogen.sh --prefix=/usr  (OK)
make    (OK)
make install    (OK)



Code:

git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
cd farsight2
git pull origin nice
./autogen.sh --prefix=/usr  (OK)
make    (NOT OK :( )


When I try to compile farsight2 I get this error :
Code:

.
.
.
 gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../gst-libs -I../../gst-libs -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Werror -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/nice -g -O2 -MT libnice_transmitter_la-fs-nice-stream-transmitter.lo -MD -MP -MF .deps/libnice_transmitter_la-fs-nice-stream-transmitter.Tpo -c fs-nice-stream-transmitter.c  -fPIC -DPIC -o .libs/libnice_transmitter_la-fs-nice-stream-transmitter.o
fs-nice-stream-transmitter.c: In function ‘fs_nice_stream_transmitter_build’:
fs-nice-stream-transmitter.c:1038: error: too few arguments to function ‘nice_agent_set_relay_info’
make[3]: *** [libnice_transmitter_la-fs-nice-stream-transmitter.lo] Error 1
make[3]: Leaving directory `/home/cris/farsight2/transmitters/nice'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/cris/farsight2/transmitters'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/cris/farsight2'
make: *** [all] Error 2


What can I do?


Title: Audio/Video conversation
Post by: billiob on October 30, 2008, 02:22:59 pm
microcris, just read the message above yours.


Title: Audio/Video conversation
Post by: microcris on October 31, 2008, 10:52:50 am
Quote from: "billiob"
microcris, just read the message above yours.


Ups, the answer to my question was right there and I didn't saw that... Ok, I will wait :)

I have it working. I compiled it about one month ago. Now, I just tried to update it but as said, I will wait.

Thanks :)


Title: Audio/Video conversation
Post by: marc2009 on October 31, 2008, 11:05:11 am
Hi all,

I've installed farsight, and amsn, and it worked ! But, when I want to make the audio test with ctrl+N in amsn, it tells me that farsight is not loaded ... whereas it has been successfully compiled with amsn !

What could I do ?

Thanks

ps : here is a part of the debug ( in the terminal ) :
Quote
Using package Farsight version 0.1

Farsight : Preparing
** (<unknown>:16136): DEBUG: CODECS ARE READY

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:16136): DEBUG: stun ip : 64.14.48.28 : 3478
** (<unknown>:16136): DEBUG: CODECS ARE READY

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:16136): DEBUG: stun ip : 64.14.48.28 : 3478

(<unknown>:16136): GLib-GObject-CRITICAL **: g_object_new_valist: assertion `G_TYPE_IS_OBJECT (object_type)' failed
** (<unknown>:16136): DEBUG: CODECS ARE READY

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:16136): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:16136): DEBUG: FS: relay info = 0x3c30600 - 2


Title: Audio/Video conversation
Post by: kakaroto on October 31, 2008, 09:26:53 pm
what does amsn tell you in the 'details' log of the audio/video assistant.. where it says it can't find farsight, there should be a details button... if you don't find it, update amsn, try again and paste here the output of that window...
note also that, as said before... you should all just be patient until we make a stable release of libnice and farsight2....
I just need to make sure libnice compiles on windows then I'll release it.


Title: Audio/Video conversation
Post by: marc2009 on November 01, 2008, 11:06:14 am
Quote from: "kakaroto"
what does amsn tell you in the 'details' log of the audio/video assistant.. where it says it can't find farsight, there should be a details button... if you don't find it, update amsn, try again and paste here the output of that window...
note also that, as said before... you should all just be patient until we make a stable release of libnice and farsight2....
I just need to make sure libnice compiles on windows then I'll release it.


here is what I obtain during the audio test when clicking on "show details" :
Quote
Using package Farsight version 0.1
Farsight : Preparing


I followed all the instructions given on http://www.amsn-project.net/forums/viewtopic.php?p=33631#33631

I'm on ubuntu 8.10 64bits


Title: Audio/Video conversation
Post by: F3d3r1k0 on November 01, 2008, 01:47:41 pm
Hi all, I'm Italian and i apologize me for my English :D
So i try to follow all the instructions in this post and i've installed successfully Farsight and Libnice but the result that i have in linux console when i run amsn are this:
Quote

Using package Farsight version 0.1

Farsight : Preparing
** (<unknown>:28928): DEBUG: CODECS ARE READY

(<unknown>:28928): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:28928): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:28928): DEBUG: stun ip : 64.14.48.28 : 3478

(<unknown>:28928): GLib-GObject-CRITICAL **: g_object_new_valist: assertion `G_TYPE_IS_OBJECT (object_type)' failed

While the result that i have in Amsn console are this
Quote
(amsn) 1 % package require Farsight
0.1
(amsn) 2 % farsight Prepare 1
Couldn't create new stream


What can I do to ??
Thanks;)


Title: Audio/Video conversation
Post by: kakaroto on November 01, 2008, 04:30:38 pm
ok... everyone... STOP TRYING!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
@marc2009, the problem is simply because of the change in API of libnice and farsight hasn't been updated.. so just like I said, be a little be patent, we're in the process of releasng libnice and farsight.
@F3d3r1k0, same answer!


Title: Audio/Video conversation
Post by: F3d3r1k0 on November 01, 2008, 04:40:31 pm
So Thank, we waiting your news.
Byeeeeeeeeee


Title: Audio/Video conversation
Post by: marc2009 on November 01, 2008, 05:31:12 pm
Quote from: "kakaroto"
ok... everyone... STOP TRYING!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
@marc2009, the problem is simply because of the change in API of libnice and farsight hasn't been updated.. so just like I said, be a little be patent, we're in the process of releasng libnice and farsight.
@F3d3r1k0, same answer!


Roger Admin  8)

but which API ? It works fine with others ! :?: ( maybe because I'm on ubuntu 8.10 64bits !? )

Thanks


Title: Audio/Video conversation
Post by: kakaroto on November 01, 2008, 06:44:53 pm
I'm talking about the API of libnice.. and farsight 2 which is using libnice needs to be updated to the new API...  The API change was done 2 days ago, so anyone who got libnice/farsight before that, it works for them, if they downloaded it after, then it won't work for them.,


Title: Audio/Video conversation
Post by: marc2009 on November 03, 2008, 09:20:27 am
Quote from: "kakaroto"
I'm talking about the API of libnice.. and farsight 2 which is using libnice needs to be updated to the new API...  The API change was done 2 days ago, so anyone who got libnice/farsight before that, it works for them, if they downloaded it after, then it won't work for them.,


Thank you for your explanation


Title: Audio/Video conversation
Post by: marc2009 on November 03, 2008, 10:59:05 am
a question please : If we are on a 64bits OS ( fedora 9 64bits ), when we compile gst-plugins-farsight, libnice and farsight2, should we add to the configure a parameter to tell them where is the libdir ? ( because otherwise, it will be installed in /usr/lib which is on 64bits /usr/lib64 ) ? so it would be
Code:
./configure --prefix=/usr --libdir=/usr/lib64


thanks


Title: Audio/Video conversation
Post by: kakaroto on November 03, 2008, 05:25:40 pm
Hi marc2009. Yes I think you're right.. someone else had this problem, I can't remember if it was discussed in this thread or on IRC... but anyways, yes, you should add --libdir=/usr/lib64... the thing is, I think it should automatically find it all by itself without having the need for you to specify it manually.. but some systems are broken and they don't detect the libdir path automatically, so you need to add the --libdir option yourself.


Title: Audio/Video conversation
Post by: marc2009 on November 03, 2008, 07:16:14 pm
Quote from: "kakaroto"
Hi marc2009. Yes I think you're right.. someone else had this problem, I can't remember if it was discussed in this thread or on IRC... but anyways, yes, you should add --libdir=/usr/lib64... the thing is, I think it should automatically find it all by itself without having the need for you to specify it manually.. but some systems are broken and they don't detect the libdir path automatically, so you need to add the --libdir option yourself.


Thanks for your answer, but I wonder one thing more : should we add the same parameters to the configure of amsn or not ?


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 02:29:02 am
no need for it for amsn...
and btw.. the git versions should now work again! So you can all start to download (as specified in the previous guide/wiki) and get it working...
the libnice/farsight2 release is still scheduled for sometime this week...


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 09:42:58 am
Quote from: "kakaroto"
no need for it for amsn...
and btw.. the git versions should now work again! So you can all start to download (as specified in the previous guide/wiki) and get it working...
the libnice/farsight2 release is still scheduled for sometime this week...


I've just tried your method http://www.amsn-project.net/forums/viewtopic.php?p=32255#32255 ( and so I tried your farsight2, not the other one ), and it worked now !!!  :P

Good work

ps : I'm on Fedora 9 64 bits


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 04:03:57 pm
cool..
and by the way.. windows users *should* now be able to make audio calls since farsight2 and libnice binaries are included in the SVN version... that's of course if it doesn't crash on login....


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 06:27:32 pm
Hi,

In my case i can't compile it correctly. I have all dependencies. Libnice compiled, installed... but farsight2 is the problem:

Code:

...
libtool: compile:  i486-linux-gnu-gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../gst-libs -I../../gst-libs -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Werror -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -g -O2 -MT libfsrtpconference_la-fs-rtp-codec-negotiation.lo -MD -MP -MF .deps/libfsrtpconference_la-fs-rtp-codec-negotiation.Tpo -c fs-rtp-codec-negotiation.c  -fPIC -DPIC -o .libs/libfsrtpconference_la-fs-rtp-codec-negotiation.o                                                                          
cc1: warnings being treated as errors                                                                                          
fs-rtp-codec-negotiation.c: En la función ‘link_unlinked_pads’:                                                                
fs-rtp-codec-negotiation.c:71: error: el formato no es una cadena literal y no tiene argumentos de formato
make[4]: *** [libfsrtpconference_la-fs-rtp-codec-negotiation.lo] Error 1
make[4]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst/fsrtpconference'
make[3]: *** [all] Error 2
make[3]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst/fsrtpconference'
make[2]: *** [all-recursive] Error 1
make[2]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst'
make[1]: *** [all-recursive] Error 1
make[1]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2'
make: *** [all] Error 2


Using Ubuntu Intrepid (which has all dependencies necesary from gstreamer).
Any Tip?

BTW, autogen always gives me error, even when i followed the instruction here:

Code:

./autogen.sh --prefix=/usr --disable-python --disable-gtk-doc                                                                                                                                
+ passing argument --prefix=/usr to configure                                                                                  
+ passing argument --disable-python to configure                                                                                
+ passing argument --disable-gtk-doc to configure                                                                              
+ options passed to configure:  --prefix=/usr --disable-python --disable-gtk-doc                                                
+ check for build tools                                                                                                        
  checking for autoconf >= 2.52 ... found 2.61, ok.                                                                            
  checking for automake >= 1.7 ... found 1.7.9, ok.                                                                            
  checking for libtoolize >= 1.5.0 ... found 2.2.4, ok.                                                                        
  checking for pkg-config >= 0.8.0 ... found 0.22, ok.                                                                          
+ running aclocal -I common/m4 ...                                                                                              
+ running libtoolize --copy --force...                                                                                          
libtoolize: putting auxiliary files in `.'.                                                                                    
libtoolize: copying file `./ltmain.sh'                                                                                          
libtoolize: putting macros in `common/m4'.                                                                                      
libtoolize: copying file `common/m4/libtool.m4'                                                                                
libtoolize: copying file `common/m4/ltoptions.m4'                                                                              
libtoolize: copying file `common/m4/ltsugar.m4'                                                                                
libtoolize: copying file `common/m4/ltversion.m4'                                                                              
libtoolize: copying file `common/m4/lt~obsolete.m4'                                                                            
libtoolize: Consider adding `AC_CONFIG_MACRO_DIR([m4])' to configure.ac and
libtoolize: rerunning libtoolize, to keep the correct libtool macros in-tree.
+ running autoheader ...
+ running autoconf ...
+ running automake -a -c -Wno-portability...
configure.ac:30: installing `./config.guess'
configure.ac:30: installing `./config.sub'
EXTRA_DIST: variable `content_files' is used but `content_files' is undefined

automake failed


Thanks.

Cheers.


Title: Audio/Video conversation
Post by: F3d3r1k0 on November 04, 2008, 07:19:26 pm
Uhmmm, I compile correctly all dependencies, but when i do the Audio/Video Confiiguration by amsn (ctrl+n) i have this error:
Quote
Farsight : Preparing
Farsight Prepare error : Couldn't create new stream

Anyone know why???
Thanks


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 08:39:53 pm
Quote from: "F3d3r1k0"
Uhmmm, I compile correctly all dependencies, but when i do the Audio/Video Confiiguration by amsn (ctrl+n) i have this error:
Quote
Farsight : Preparing
Farsight Prepare error : Couldn't create new stream

Anyone know why???
Thanks


I had the same problem, it's because you used the method with the replacement of the SVN version farsight files with the farsight of the other site... try the farsight2 from kakaroto

so follow his instructions : http://www.amsn-project.net/forums/viewtopic.php?p=32255#32255


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 08:41:37 pm
Quote from: "GuS-Arg"
Hi,

In my case i can't compile it correctly. I have all dependencies. Libnice compiled, installed... but farsight2 is the problem:

Code:

...
libtool: compile:  i486-linux-gnu-gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../gst-libs -I../../gst-libs -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Werror -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -g -O2 -MT libfsrtpconference_la-fs-rtp-codec-negotiation.lo -MD -MP -MF .deps/libfsrtpconference_la-fs-rtp-codec-negotiation.Tpo -c fs-rtp-codec-negotiation.c  -fPIC -DPIC -o .libs/libfsrtpconference_la-fs-rtp-codec-negotiation.o                                                                          
cc1: warnings being treated as errors                                                                                          
fs-rtp-codec-negotiation.c: En la función ‘link_unlinked_pads’:                                                                
fs-rtp-codec-negotiation.c:71: error: el formato no es una cadena literal y no tiene argumentos de formato
make[4]: *** [libfsrtpconference_la-fs-rtp-codec-negotiation.lo] Error 1
make[4]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst/fsrtpconference'
make[3]: *** [all] Error 2
make[3]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst/fsrtpconference'
make[2]: *** [all-recursive] Error 1
make[2]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2/gst'
make[1]: *** [all-recursive] Error 1
make[1]: se sale del directorio `/home/gustavo/Descargas/SVN/Packages/deps/new/farsight/farsight2'
make: *** [all] Error 2


Using Ubuntu Intrepid (which has all dependencies necesary from gstreamer).
Any Tip?

BTW, autogen always gives me error, even when i followed the instruction here:

Code:

./autogen.sh --prefix=/usr --disable-python --disable-gtk-doc                                                                                                                                
+ passing argument --prefix=/usr to configure                                                                                  
+ passing argument --disable-python to configure                                                                                
+ passing argument --disable-gtk-doc to configure                                                                              
+ options passed to configure:  --prefix=/usr --disable-python --disable-gtk-doc                                                
+ check for build tools                                                                                                        
  checking for autoconf >= 2.52 ... found 2.61, ok.                                                                            
  checking for automake >= 1.7 ... found 1.7.9, ok.                                                                            
  checking for libtoolize >= 1.5.0 ... found 2.2.4, ok.                                                                        
  checking for pkg-config >= 0.8.0 ... found 0.22, ok.                                                                          
+ running aclocal -I common/m4 ...                                                                                              
+ running libtoolize --copy --force...                                                                                          
libtoolize: putting auxiliary files in `.'.                                                                                    
libtoolize: copying file `./ltmain.sh'                                                                                          
libtoolize: putting macros in `common/m4'.                                                                                      
libtoolize: copying file `common/m4/libtool.m4'                                                                                
libtoolize: copying file `common/m4/ltoptions.m4'                                                                              
libtoolize: copying file `common/m4/ltsugar.m4'                                                                                
libtoolize: copying file `common/m4/ltversion.m4'                                                                              
libtoolize: copying file `common/m4/lt~obsolete.m4'                                                                            
libtoolize: Consider adding `AC_CONFIG_MACRO_DIR([m4])' to configure.ac and
libtoolize: rerunning libtoolize, to keep the correct libtool macros in-tree.
+ running autoheader ...
+ running autoconf ...
+ running automake -a -c -Wno-portability...
configure.ac:30: installing `./config.guess'
configure.ac:30: installing `./config.sub'
EXTRA_DIST: variable `content_files' is used but `content_files' is undefined

automake failed


Thanks.

Cheers.


did you follow all the instructions http://www.amsn-project.net/forums/viewtopic.php?p=32255#32255 ?


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 09:13:00 pm
Yes, of course i did. And i re-check the farsight2 repository. Still have compilation problems. And maybe is cause the autogen failure.


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 09:18:35 pm
Quote from: "GuS-Arg"
Yes, of course i did. And i re-check the farsight2 repository. Still have compilation problems.


mmh, and was libnice installation successful ? (
Code:
./autogen.sh --prefix=/usr
make
sudo make install
)

didn't you forget the
Code:
git checkout -b nice origin/nice
for farsight2 after having done
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
cd farsight2
? or did you forget to do
Code:
sudo apt-get install gtk-doc-tools
?


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 09:30:52 pm
I dont want to be rude Marc, but i did all that.... i have experience in packaging and compiling applications...
And yes, libnice compiled perfectly and i made the Intrepid package for it.

And btw, did someone compiled this successfully on Ubuntu Intrepid? lets start by there...


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 09:44:47 pm
Quote from: "GuS-Arg"
I dont want to be rude Marc, but i did all that.... i have experience in packaging and compiling applications...
And yes, libnice compiled perfectly and i made the Intrepid package for it.


I only wanted to be sure that you have made all of that  :P I'm not here to treat you as an "idiot", but I'm here to help you, there's a difference :wink: ( as I can of course )

I doubt that the problem comes from the architecture ( 64bits or 32bits )  :!: but did you try the method written on http://www.amsn-project.net/forums/viewtopic.php?p=33631#33631 so this :
Quote
git clone git://git.collabora.co.uk/git/user/kakaroto/farsight2.git farsight2
cd farsight2
git checkout -b nice origin/nice
cd ..
wget http://farsight.freedesktop.org/releases/farsight2/farsight2-0.0.3.tar.gz
tar zxfv farsight2-0.0.3.tar.gz
cd farsight2-0.0.3
cp -f ../farsight2/common/check.mak ./common/
cp -f ../farsight2/common/gst.supp ./common/
cp -f ../farsight2/common/gst-autogen.sh ./common/
cp -f ../farsight2/common/gtk-doc.mak ./common/
./autogen.sh --prefix=/usr --disable-python
nano -w farsight2.pc

Now change the version from 0.0.3 to 0.0.3.1
Press CTRL+O to save and CTRL+X to quit nano.

make
sudo make install
sudo /sbin/ldconfig


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 09:54:07 pm
humm.. I'll have a look at this in a little while...
marc, I don't know what that last method you posted is all about, but I'm not sure it's the right way to do things...


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 09:54:09 pm
The autogen still is failling....

Again i will ask: DID SOMEONE COMPILED THIS IN UBUNTU INTREPID? :P

Thanks.


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 10:02:11 pm
Quote from: "kakaroto"
humm.. I'll have a look at this in a little while...
marc, I don't know what that last method you posted is all about, but I'm not sure it's the right way to do things...


maybe it would have worked  :P even if it's written for fedora  :D ( because when I was on ubuntu intrepid 64bits, this method to install farsight worked )

@ GuS-Arg

mmh before I come back to fedora, I was on intrepid 8.10 64bits, and I remember the installation of farsight2 worked with the method I gave ( quoted from the link I gave ) but amsn failed to load farsight, it told me farsight preparing or farsight prepare 1

ps : it seems the problem is with "content_files" so, I'd say either the problem is with ubuntu, either it's with the sources themselves.


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 10:06:05 pm
When i've compiled Farsight on one month ago or so, was compiling too... so... Is not my problem.

And by the way, using the method you suggested Marc, farsight2 compiled (even with the autogen error) but amsn is not finding it.


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 10:09:46 pm
hey Gus,
thx for reporting. Olivier just fixed those two issues (autogen and compilation) and I've pushed those changes to git... so can you please retry by getting the latest git ?
it is indeed an issue with intrepid since it uses a very new version of gcc... I hope this fixes it.

edit yes, the method described by marc will make it compile but not work because it would be an old version of farsight2, but you need the latest changes for it to work with libnice...
Gus, if you still have a  problem, ping me on IRC!


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 10:14:50 pm
Good work, so now you can try and it must be successful

edit : effectively kakaroto, it didn't work in amsn ( audio test ), but when i was compiling amsn, it found farsight !


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 10:22:16 pm
yes it will find it because you spoof the version into 0.0.3.1 even though it's not :p


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 10:35:54 pm
Quote from: "kakaroto"
yes it will find it because you spoof the version into 0.0.3.1 even though it's not :p


Thanks for your explanation  :P


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 10:47:41 pm
well all the things installed well and amsn founds farsight ok  but when I call people (WLM8.5)  we didnt hear anything both sides ?  I`m behing a router/firewall  do I need to open some udp ports or so ?  

amsn console output :
Farsight : Preparing
Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000} {telephone-event 99 8000 0-16}
local candidates : {FThUGbEMXL345uDp3FiKB5Yqbk6/U3DH/TWQpUkkbBc= 1 8cDi7OoCSqBtnAeK5ju17g== UDP 0.8299999833106995 172.16.190.1 45577} {zKd90TK46N5sXeydB8mDHjesHsJ96leJm7PdvYjYM0o= 1 MyA+N5aB2+JGvH+xVTtvpg== UDP 0.8299999833106995 172.16.1.1 34191} {R5LtXsUz1RDL6Ayjyn30mJ41KgUvfgraxFKdkxJwoRI= 1 yh8wo8JrFJAxOakygYsK5w== UDP 0.8299999833106995 10.1.1.4 32866} {FThUGbEMXL345uDp3FiKB5Yqbk6/U3DH/TWQpUkkbBc= 2 8cDi7OoCSqBtnAeK5ju17g== UDP 0.8299999833106995 172.16.190.1 55805} {zKd90TK46N5sXeydB8mDHjesHsJ96leJm7PdvYjYM0o= 2 MyA+N5aB2+JGvH+xVTtvpg== UDP 0.8299999833106995 172.16.1.1 33337} {R5LtXsUz1RDL6Ayjyn30mJ41KgUvfgraxFKdkxJwoRI= 2 yh8wo8JrFJAxOakygYsK5w== UDP 0.8299999833106995 10.1.1.4 60688}


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 10:51:43 pm
Quote from: "lordamus"
well all the things installed well and amsn founds farsight ok  but when I call people (WLM8.5)  we didnt hear anything both sides ?  I`m behing a router/firewall  do I need to open some udp ports or so ?  

amsn console output :
Farsight : Preparing
Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000} {telephone-event 99 8000 0-16}
local candidates : {FThUGbEMXL345uDp3FiKB5Yqbk6/U3DH/TWQpUkkbBc= 1 8cDi7OoCSqBtnAeK5ju17g== UDP 0.8299999833106995 172.16.190.1 45577} {zKd90TK46N5sXeydB8mDHjesHsJ96leJm7PdvYjYM0o= 1 MyA+N5aB2+JGvH+xVTtvpg== UDP 0.8299999833106995 172.16.1.1 34191} {R5LtXsUz1RDL6Ayjyn30mJ41KgUvfgraxFKdkxJwoRI= 1 yh8wo8JrFJAxOakygYsK5w== UDP 0.8299999833106995 10.1.1.4 32866} {FThUGbEMXL345uDp3FiKB5Yqbk6/U3DH/TWQpUkkbBc= 2 8cDi7OoCSqBtnAeK5ju17g== UDP 0.8299999833106995 172.16.190.1 55805} {zKd90TK46N5sXeydB8mDHjesHsJ96leJm7PdvYjYM0o= 2 MyA+N5aB2+JGvH+xVTtvpg== UDP 0.8299999833106995 172.16.1.1 33337} {R5LtXsUz1RDL6Ayjyn30mJ41KgUvfgraxFKdkxJwoRI= 2 yh8wo8JrFJAxOakygYsK5w== UDP 0.8299999833106995 10.1.1.4 60688}


it's certainly idiot but did you do the audio test by doing CTRL + N ?


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 10:54:38 pm
Sure!  I did audio test...


Title: Audio/Video conversation
Post by: GuS-Arg on November 04, 2008, 10:55:29 pm
Hi again,

Anyone that have Ubuntu Intrepid could test these packages recenlty made:

http://download.tuxfamily.org/amsnskins/packages/ubuntu/amsn-svn-snapshots/

Just install all those packages with follow:

Code:
sudo dpkg -i *.deb


If dependencies errors shows, then:

Code:
sudo apt-get install -f



I will keep updating those, which includes my recent development of Sapphire skin 2.5 by default (which still is under development too).

Cheers.


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 10:57:18 pm
Quote from: "lordamus"
Sure!  I did audio test...


so it worked

mmh does your router support DMZ adress ?

HS : does it block ICMP pings ?


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 10:58:49 pm
yea dmz supported but I dont use it..  Only Nat and IP filter..  also my fedora firewall blocks some icmp requests


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 11:04:29 pm
Quote from: "lordamus"
yea dmz supported but I dont use it..  Only Nat and IP filter..  also my fedora firewall blocks some icmp requests


try to use just now dmz, which will redirect all the traffic to this address, and you will see if it works.

HS :
for icmp, it's just because I made a few minutes ago in my terminal
Code:
ping 172.16.1.1
to see if it works, and it didn't work, so good ( because if it would have worked, it would mean that you are exposed to hackers ! because you indicated your ip, but it seems there are a lot of ips in what you gave ! )
it's just for your information, because I don't like doing that to someone without telling him that I did that !


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 11:17:06 pm
I cant use dmz right now my router is old and dmz not easily configure..  I can ping 172.16.1.1  so what ?    my ext ip closed to icmp pinging...I want to know which ports do I need to forward in router/firewall to firesight audio getting to work ?


Title: Audio/Video conversation
Post by: kakaroto on November 04, 2008, 11:20:50 pm
wow, this thread is going way too fast for me to follow, lol!
ok, well, farsight is able to get prepared, which is good, but it looks like it can't find your external ip.. so I'm guessing turn/stun is not working..
You don't need to forward any ports to your computer and you don't need to put yourself in the DMZ... libnice should be able to do NAT traversal without you needing to do anything! but if stun/turn does not work, it must mean that UDP did not go well.. I think the only thing you need to do is to make sure that outgoing UDP is allowed... if it's allowed, then it should work... btw, when you did the call, did it tell you (in the terminal) that it succeeded or did it say that it couldn't connect the RTP component ? Could you give me the output from the terminal ? You might want to launch amsn with :
Code:
NICE_DEBUG=all amsn > log 2>&1
and  pastebin the content of the log file.


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 11:25:20 pm
Quote from: "kakaroto"
wow, this thread is going way too fast for me to follow, lol!
ok, well, farsight is able to get prepared, which is good, but it looks like it can't find your external ip.. so I'm guessing turn/stun is not working..
You don't need to forward any ports to your computer and you don't need to put yourself in the DMZ... libnice should be able to do NAT traversal without you needing to do anything! but if stun/turn does not work, it must mean that UDP did not go well.. I think the only thing you need to do is to make sure that outgoing UDP is allowed... if it's allowed, then it should work... btw, when you did the call, did it tell you (in the terminal) that it succeeded or did it say that it couldn't connect the RTP component ? Could you give me the output from the terminal ? You might want to launch amsn with :
Code:
NICE_DEBUG=all amsn > log 2>&1
and  pastebin the content of the log file.


but the solution with the dmz is also an idea  :P ( yes, okay, not the best but ...  :D )

++

ps : i also thought about the fact that farsight couldn't find the external ip but, as I don't know how to fix it, I didn't say it  :lol:  :lol:


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 11:34:49 pm
Quote from: "marc2009"
Quote from: "kakaroto"
wow, this thread is going way too fast for me to follow, lol!
ok, well, farsight is able to get prepared, which is good, but it looks like it can't find your external ip.. so I'm guessing turn/stun is not working..
You don't need to forward any ports to your computer and you don't need to put yourself in the DMZ... libnice should be able to do NAT traversal without you needing to do anything! but if stun/turn does not work, it must mean that UDP did not go well.. I think the only thing you need to do is to make sure that outgoing UDP is allowed... if it's allowed, then it should work... btw, when you did the call, did it tell you (in the terminal) that it succeeded or did it say that it couldn't connect the RTP component ? Could you give me the output from the terminal ? You might want to launch amsn with :
Code:
NICE_DEBUG=all amsn > log 2>&1
and  pastebin the content of the log file.


but the solution with the dmz is also an idea  :P ( yes, okay, not the best but ...  :D )

++

ps : i also thought about the fact that farsight couldn't find the external ip but, as I don't know how to fix it, I didn't say it  :lol:  :lol:


DMZ not a good idea to hackers also  :wink:


Title: Audio/Video conversation
Post by: marc2009 on November 04, 2008, 11:38:11 pm
Quote from: "lordamus"
Quote from: "marc2009"
Quote from: "kakaroto"
wow, this thread is going way too fast for me to follow, lol!
ok, well, farsight is able to get prepared, which is good, but it looks like it can't find your external ip.. so I'm guessing turn/stun is not working..
You don't need to forward any ports to your computer and you don't need to put yourself in the DMZ... libnice should be able to do NAT traversal without you needing to do anything! but if stun/turn does not work, it must mean that UDP did not go well.. I think the only thing you need to do is to make sure that outgoing UDP is allowed... if it's allowed, then it should work... btw, when you did the call, did it tell you (in the terminal) that it succeeded or did it say that it couldn't connect the RTP component ? Could you give me the output from the terminal ? You might want to launch amsn with :
Code:
NICE_DEBUG=all amsn > log 2>&1
and  pastebin the content of the log file.


but the solution with the dmz is also an idea  :P ( yes, okay, not the best but ...  :D )

++

ps : i also thought about the fact that farsight couldn't find the external ip but, as I don't know how to fix it, I didn't say it  :lol:  :lol:


DMZ not a good idea to hackers also  :wink:


I know but that's why I precised just now, because i wanted to be sure that the problem was coming from the external ip ( because i had the same problem, farsight detected my internal ip, so 192.168.... but not my external ip, but now it detects it )


Title: Audio/Video conversation
Post by: lordamus on November 04, 2008, 11:48:03 pm
Quote from: "kakaroto"
wow, this thread is going way too fast for me to follow, lol!
ok, well, farsight is able to get prepared, which is good, but it looks like it can't find your external ip.. so I'm guessing turn/stun is not working..
You don't need to forward any ports to your computer and you don't need to put yourself in the DMZ... libnice should be able to do NAT traversal without you needing to do anything! but if stun/turn does not work, it must mean that UDP did not go well.. I think the only thing you need to do is to make sure that outgoing UDP is allowed... if it's allowed, then it should work... btw, when you did the call, did it tell you (in the terminal) that it succeeded or did it say that it couldn't connect the RTP component ? Could you give me the output from the terminal ? You might want to launch amsn with :
Code:
NICE_DEBUG=all amsn > log 2>&1
and  pastebin the content of the log file.



here is some status logs during calling someone  http://pastebin.com/d2b562e

here is nice_debug   http://pastebin.com/m6a9cfa6b


Title: Audio/Video conversation
Post by: BW on November 04, 2008, 11:57:37 pm
Hi all,
first of all, thanks kakaroto for this great peace of work!

after seeing the binaries for windows and the "guess what.. it works! :)" on svn,
I've just tested it on a few windows machines, so here is what I got so far, fyi
in case you don't know it anyway (I hope it wasn't mentioned before anywhere,
and I'm not to early reporting things like that)

on Win2k Status log tells me:
Quote
Farsight : Preparing
Farsight Prepare error : couldn't load library "utils/farsight/tcl_farsight.dll": could not find specified procedure
(not the right version for win2k?)

on WinXP I get this starting from rev 10676, at login, and afterwords:
Quote
unable to register ".main.f.top.bigstate" as a droppable window
    while executing
"::dnd bindtarget $tw.$name Files <Drop> "fileDropHandler %D setdp self""
    (procedure "clickableDisplayPicture" line 16)
    invoked from within
"clickableDisplayPicture $pgBuddyTop mystatus bigstate {kill_balloon; tk_popup .my_menu %X %Y} [::skin::getKey bigstate_xpad] [::skin::getKey bigstate_..."
    (procedure "cmsn_draw_buildtop_wrapped" line 23)
    invoked from within
"cmsn_draw_buildtop_wrapped"
    ("eval" body line 1)
    invoked from within
"eval cmsn_draw_buildtop_wrapped"
    invoked from within
"catch "eval $command" errorMsg"
    (procedure "run_exclusive" line 20)
    invoked from within
"run_exclusive cmsn_draw_buildtop_wrapped draw_online "
    (procedure "cmsn_draw_online" line 17)
    invoked from within
"cmsn_draw_online 0 1"
    ("after" script)

on status log I couldn't see anything different apart from farsight is prepared and the error
so farsight works according to status log, but the error prevents me from testing it...

bye


Title: Audio/Video conversation
Post by: kakaroto on November 05, 2008, 01:25:35 am
@lordamus: cool, you helped me find a bug in libnice, can you update BOTH libnice and farsight2 from git and retry ? if it still doesn't work, then again, nice debug pastebin... thx
@BW: thx for the info... well, I'm still looking at this, it does work for windows, but it doesn't mean it's perfect yet... anyways, please update to the latest SVN, you shouldn't have this problem of not being able to load farsight anymore (the 'can't find specified procedure' error).
About the dnd problem, I don't know, there seems to be an incompatibility between tkdnd (drag and drop support) and farsight (more likely to be glib)... I'll have a look some other time....


Title: Audio/Video conversation
Post by: lordamus on November 05, 2008, 02:07:31 am
yea np...I updated both from git and installed but still no sound in call  sadly..   I also  open outgoing UDP traffic in router...


Quote from: "kakaroto"
@lordamus: cool, you helped me find a bug in libnice, can you update BOTH libnice and farsight2 from git and retry ? if it still doesn't work, then again, nice debug pastebin... thx
@BW: thx for the info... well, I'm still looking at this, it does work for windows, but it doesn't mean it's perfect yet... anyways, please update to the latest SVN, you shouldn't have this problem of not being able to load farsight anymore (the 'can't find specified procedure' error).
About the dnd problem, I don't know, there seems to be an incompatibility between tkdnd (drag and drop support) and farsight (more likely to be glib)... I'll have a look some other time....


Title: Audio/Video conversation
Post by: microcris on November 05, 2008, 02:18:04 am
Hi there :)

libnice complies well but in farsight I'm getting this error :S

Code:

...
gcc -g -O2 -o .libs/codec-discovery codec_discovery-codec-discovery.o codec_discovery-fs-rtp-discover-codecs.o codec_discovery-fs-rtp-codec-cache.o codec_discovery-fs-rtp-special-source.o codec_discovery-fs-rtp-dtmf-event-source.o codec_discovery-fs-rtp-dtmf-sound-source.o codec_discovery-fs-rtp-codec-negotiation.o codec_discovery-fs-rtp-specific-nego.o codec_discovery-fs-rtp-conference.o codec_discovery-fs-rtp-session.o codec_discovery-fs-rtp-stream.o codec_discovery-fs-rtp-substream.o codec_discovery-fs-rtp-participant.o codec_discovery-fs-rtp-marshal.o -pthread -pthread  ../../gst-libs/gst/farsight/.libs/libgstfarsight-0.10.so /usr/lib/libgstcheck-0.10.so -lcheck_pic -lm -lgstrtp-0.10 /usr/lib/libgstreamer-0.10.so /usr/lib/libgobject-2.0.so /usr/lib/libgmodule-2.0.so -ldl /usr/lib/libgthread-2.0.so -lrt /usr/lib/libxml2.so /usr/lib/libglib-2.0.so
codec_discovery-fs-rtp-codec-negotiation.o: In function `link_unlinked_pads':
/home/cris/farsight2/tests/rtp/../../gst/fsrtpconference/fs-rtp-codec-negotiation.c:62: undefined reference to `gst_bin_find_unlinked_pad'
collect2: ld returned 1 exit status
make[3]: *** [codec-discovery] Error 1
make[3]: Leaving directory `/home/cris/farsight2/tests/rtp'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/cris/farsight2/tests'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/cris/farsight2'
make: *** [all] Error 2


Title: Audio/Video conversation
Post by: kakaroto on November 05, 2008, 02:22:11 am
@lordamus... humm.. well, again, the nice_debug would be helpful...
@microcris: please follow the guide step by step and make sure you have the correct dependencies (your error is because you have a version of gstreamer that is too old). Check the wiki here : http://amsn-project.net/wiki/Farsight


Title: Audio/Video conversation
Post by: lordamus on November 05, 2008, 12:35:41 pm
here it is  http://pastebin.com/d60b0417e

Quote from: "kakaroto"
@lordamus... humm.. well, again, the nice_debug would be helpful...
@microcris: please follow the guide step by step and make sure you have the correct dependencies (your error is because you have a version of gstreamer that is too old). Check the wiki here : http://amsn-project.net/wiki/Farsight


Title: Audio/Video conversation
Post by: microcris on November 05, 2008, 04:20:55 pm
Quote from: "kakaroto"
@lordamus... humm.. well, again, the nice_debug would be helpful...
@microcris: please follow the guide step by step and make sure you have the correct dependencies (your error is because you have a version of gstreamer that is too old). Check the wiki here : http://amsn-project.net/wiki/Farsight


As you said, I had some outdated dependencies. I'm using Debian Lenny 64bit so I downloaded the updated packages from Debian Sid repository and installed it :)
Then I made

Code:

./autogen.sh --prefix=/usr
make


and I get a error that had to do whit  gtk-doc: Compiling scanner

After that I tried

Code:

./autogen.sh --prefix=/usr --disable-gtk-doc
make

Compiled and installed without problem.

After that I tried again

Code:

make clean
./autogen.sh --prefix=/usr
make


Strangely, it compiled a installed without problem lol

Now, I i think it is working :)
Code:
Farsight : Preparing
Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000} {telephone-event 101 8000 0-16}
local candidates : {PFAXVTdQQxVsx/ZO20AKp+4XY1Dj7pFkUPaa3U5pd4k= 1 EILdxbA5GDxISV7KzKPz6Q== UDP 0.8299999833106995 192.168.1.139 37344} {GhXBY5xRn7c7sacJyf0HdolWyqMVrmsBk8I4zxL6LQU= 1 43dgvNwx1gti011g6vuvjg== UDP 0.550000011920929 213.22.79.47 37344} {uE/eTVrif4ltjClWwYVn2ColtHRSzobUnHTyhePkeYo= 1 Gw5Fj2zujjHt9wy5YQGmGQ== UDP 0.44999998807907104 65.55.255.24 36865} {PFAXVTdQQxVsx/ZO20AKp+4XY1Dj7pFkUPaa3U5pd4k= 2 EILdxbA5GDxISV7KzKPz6Q== UDP 0.8299999833106995 192.168.1.139 51624} {GhXBY5xRn7c7sacJyf0HdolWyqMVrmsBk8I4zxL6LQU= 2 43dgvNwx1gti011g6vuvjg== UDP 0.550000011920929 213.22.79.47 51624} {uE/eTVrif4ltjClWwYVn2ColtHRSzobUnHTyhePkeYo= 2 Gw5Fj2zujjHt9wy5YQGmGQ== UDP 0.44999998807907104 65.55.255.24 34959}


Thank you :)


Title: Audio/Video conversation
Post by: kakaroto on November 05, 2008, 05:34:52 pm
@microcris : yep, it seems to be working just fine! :)
@lordamus : cool.. well, you still have the problem with the TURN server, I have honestly, no idea why... it seems that when you try to allocate the TURN port (the relay server in case you're all firewalled and you are both using a symetric NAT, which in not your case, so you should be good to go), it refuses your allocation and asks you to recreate a new temporary username/password.. but that's useless since you've just created those.. so I have no idea why it would do something like that... either way, because of that, you are not able to get your TURN candidate, and you don't get your server-reflexive candidate either (your external ip/port derived from a STUN or TURN server binding).
BUT, this is not really a problem.. why? simply because you get your peer's candidates, and although he doesn't know your external ip, so he can't connect to you, the ICE magic comes into play since you send him (the peer) your own connectivity check, when he receives it and authenticates it, he then accepts it as being your own message, but since it comes from your external IP, he will add to his own remote candidates list a new candidate based off your existing one and using your external IP address for it.. in his answer, he will tell you what's your external IP, as he received it, so you can yourself add your own peer-reflexive candidate (your external ip/port derived from a peer who acted just like a STUN server through a STUN binding)...
This leads to the following in your log :
Code:

** (<unknown>:12439): DEBUG: Agent 0x73f8300 : added a new peer-discovered pair with foundation of '4:DyT9byXz4qfMVepXMLKTgZDypBKKpth'.
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : conncheck 0x60af150 FAILED, 0x614fa10 DISCOVERED.
...
** (<unknown>:12439): DEBUG: Local candidate: C3vsy2b6Tc1NcFrozchksJSABXQ5uU47PlZKKWdQuroPJP1vJfPip8xV6lcwspOBkPKkEoqm2EmmKwj8wQxU9A== 1 nYPr2O2YyUAjBINp39vIPw== UDP 550 85.96.116.233 50808
 
** (<unknown>:12439): DEBUG: Remote candidate: DyT9byXz4qfMVepXMLKTgZDypBKKpthJpisI/MEMVPQ= 1 KI0W2WBfs/wkhfcyNLsXrA== UDP 0 94.122.91.24 29504
 


The last two lines show that aMSN was able to connect and considers the local/remote candidates as being 'connected/authenticated/ready to transfer data'...  which is subsequently proven by :
Code:

** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504

So you are sending data, so yeay, it's all working.. the problem is.. you're not receiving any data... and why is that? well, I'm guessing that either :
1 - WLM didn't accept your connectivity checks so it doesn't think you are both connected yet
2 - the SIP re-invite which tells the peer which candidates you decided to choose were not acceptable...

so.... first problem, I don't think it's true since we receive our peer-reflexive candidate by having the peer answer us.. since it answered us it means that it already accepted our connectivity check so it knows we're connected... second problem... well, I will need to have a look at that since I was never able to get that use case since I'm not in a symetric NAT with a friend on a full-cone NAT... but with your method (not using TURN or STUN), I can now reproduce this peer-reflexive use case...
Having the protocol log might help me here...
I'm also supposing that you're the one who sent the audio chat request... (which means that you're the controlling agent, so you're the one sending the SIP re-invite), and I'm guessing that if you *receive* the invite instead (you're the controlled agent, so you receive the SIP re-invite, not send it), then it should probably work!
If you could give me :
the nice_debug AND the protocol log pastebins for when you RECEIVE an invitation (both from the same session). I think I just might be able to help!
Make it quick please, I want to release libnice today!
thanks! :)

p.s.: if you update amsn (no need to update libnice/farsight), you shouldn't be having this problem anymore, BUT please do not update until you've tested and provided the logs requested above... thanks! (if you already updated, just do an 'svn update -r10679' to revert to the non-working version).


Title: Audio/Video conversation
Post by: BW on November 05, 2008, 07:40:12 pm
sorry kakaroto, but the 'can't find specified procedure' error is still there :?


Title: Audio/Video conversation
Post by: Fenix-TX on November 05, 2008, 08:15:47 pm
I've just installed libnice and farsight. I've compiled amsn and says FARSIGHT YES, but is not working:

Code:

Farsight : Preparing
Farsight Prepare error : Error while creating new stream (1): Could not create the nicesrc element


Code:

** (<unknown>:906): DEBUG: CODECS ARE READY

(<unknown>:906): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:906): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:906): DEBUG: stun ip : 64.14.48.28 : 3478
Farsight : Preparing
** (<unknown>:906): DEBUG: CODECS ARE READY

(<unknown>:906): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:906): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `buffer-time'
** (<unknown>:906): DEBUG: stun ip : 64.14.48.28 : 3478
** (<unknown>:906): DEBUG: FS: relay info = 0x9c4eb80 - 2
Farsight Prepare error : Error while creating new stream (1): Could not create the nicesrcelement


Title: Audio/Video conversation
Post by: kakaroto on November 05, 2008, 08:22:45 pm
@Fenix-TX : you didn't install libnice correctly, or you didn't give it --prefix=/usr, or something like that.. try doing in a terminal 'gst-inspect nice' it should give you something... Make sure the file  /usr/lib/gstreamer-0.10/libgstnice.so exists.

@BW: not possible (or at least, I hope that it's not )... :(


Title: Audio/Video conversation
Post by: Fenix-TX on November 05, 2008, 08:47:05 pm
Code:

gst-inspect-0.10 nice
Plugin Details:
  Name:                 nice
  Description:          Interactive UDP connectivity establishment
  Filename:             /usr/lib/gstreamer-0.10/libgstnice.so
  Version:              0.0.1
  License:              LGPL
  Source module:        nice
  Binary package:       nice
  Origin URL:           http://telepathy.freedesktop.org/wiki/

  nicesrc: ICE source
  nicesink: ICE sink

  2 features:
  +-- 2 elements


Any other suggestion?


Title: Audio/Video conversation
Post by: gothmog on November 05, 2008, 09:10:36 pm
Hi, I'm on windows XP SP3 and I have a problem similar to that of BW but in this case it says:
Quote
Farsight : Preparing
Farsight Prepare error : couldn't load library "utils/farsight/tcl_farsight.dll": this library or a dependent library could not be found in library path

BTW I'm using SVN -r 10682. Anyway, it doesn't show any message or error apart from that in status log and I'm able to mantain a conversation. Thx[


Title: Audio/Video conversation
Post by: kakaroto on November 05, 2008, 09:55:00 pm
@Fenix-TX :
Code:
Farsight Prepare error : Error while creating new stream (1): Could not create the nicesrc element

Code:
gst-inspect-0.10 nice
Plugin Details:
  Name:                 nice
  Description:          Interactive UDP connectivity establishment
  Filename:             /usr/lib/gstreamer-0.10/libgstnice.so
  Version:              0.0.1
  License:              LGPL
  Source module:        nice
  Binary package:       nice
  Origin URL:           http://telepathy.freedesktop.org/wiki/

  nicesrc: ICE source

  ^^^^^

explain that to me.. 'cause I can't :s

@gothmog/@BW : what i would need is for you to, humm... I think I know what the problem is! simple.. I'll fix it tonight.. I just need to recompile tcl_farsight for tcl/tk 8.5 (the new default with amsn 0.97.2) instead of 8.4 (what I use)...


Title: Audio/Video conversation
Post by: kakaroto on November 06, 2008, 01:22:54 am
@gothmog/@BW : Update the SVN, it should now work! tell me if it does indeed... hehe!


Title: Audio/Video conversation
Post by: harrydb on November 06, 2008, 03:21:55 am
Hmmm 'make install'  for farsight2 fails, anyone good ideas?

Code:
libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../gst-libs -I../../gst-libs -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -Werror -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -g -O2 -MT libfsrtpconference_la-fs-rtp-stream.lo -MD -MP -MF .deps/libfsrtpconference_la-fs-rtp-stream.Tpo -c fs-rtp-stream.c  -fPIC -DPIC -o .libs/libfsrtpconference_la-fs-rtp-stream.o
fs-rtp-stream.c: In function ‘_state_changed’:
fs-rtp-stream.c:717: error: ‘FS_TYPE_STREAM_STATE’ undeclared (first use in this function)
fs-rtp-stream.c:717: error: (Each undeclared identifier is reported only once
fs-rtp-stream.c:717: error: for each function it appears in.)
make[3]: *** [libfsrtpconference_la-fs-rtp-stream.lo] Error 1
make[3]: Leaving directory `/home/harry/bin/farsight2/gst/fsrtpconference'
make[2]: *** [install] Error 2
make[2]: Leaving directory `/home/harry/bin/farsight2/gst/fsrtpconference'
make[1]: *** [install-recursive] Error 1
make[1]: Leaving directory `/home/harry/bin/farsight2/gst'
make: *** [install-recursive] Error 1


Edit:
okay, I just typed 'make install' too quickly 'make' also gives the error above


Title: Audio/Video conversation
Post by: Fenix-TX on November 06, 2008, 08:03:19 am
Quote from: "kakaroto"
@Fenix-TX :
Code:
Farsight Prepare error : Error while creating new stream (1): Could not create the nicesrc element

Code:
gst-inspect-0.10 nice
Plugin Details:
  Name:                 nice
  Description:          Interactive UDP connectivity establishment
  Filename:             /usr/lib/gstreamer-0.10/libgstnice.so
  Version:              0.0.1
  License:              LGPL
  Source module:        nice
  Binary package:       nice
  Origin URL:           http://telepathy.freedesktop.org/wiki/

  nicesrc: ICE source

  ^^^^^

explain that to me.. 'cause I can't :s
.


Mmm, it's the output of command "gst-inspect-0.10 nice" as suggested me. And i have libgstnice.so on /usr/lib/gstreamer-0.10. Do i try again compile all (libnice, farsight and amsn) again? Do i remove previous installed files of libnice and farsigth?


Title: Audio/Video conversation
Post by: lordamus on November 06, 2008, 04:32:49 pm
Good explanation :D   I can send data but other side cant hear my voice...Yea I`m the one who sended audio chat requests... ok here is the logs during call invitation  http://pastebin.com/d73cf5a4f    I also PM`ed protocol logs to you check it out..  now I`ll try new svn...

Quote from: "kakaroto"
@microcris : yep, it seems to be working just fine! :)
@lordamus : cool.. well, you still have the problem with the TURN server, I have honestly, no idea why... it seems that when you try to allocate the TURN port (the relay server in case you're all firewalled and you are both using a symetric NAT, which in not your case, so you should be good to go), it refuses your allocation and asks you to recreate a new temporary username/password.. but that's useless since you've just created those.. so I have no idea why it would do something like that... either way, because of that, you are not able to get your TURN candidate, and you don't get your server-reflexive candidate either (your external ip/port derived from a STUN or TURN server binding).
BUT, this is not really a problem.. why? simply because you get your peer's candidates, and although he doesn't know your external ip, so he can't connect to you, the ICE magic comes into play since you send him (the peer) your own connectivity check, when he receives it and authenticates it, he then accepts it as being your own message, but since it comes from your external IP, he will add to his own remote candidates list a new candidate based off your existing one and using your external IP address for it.. in his answer, he will tell you what's your external IP, as he received it, so you can yourself add your own peer-reflexive candidate (your external ip/port derived from a peer who acted just like a STUN server through a STUN binding)...
This leads to the following in your log :
Code:

** (<unknown>:12439): DEBUG: Agent 0x73f8300 : added a new peer-discovered pair with foundation of '4:DyT9byXz4qfMVepXMLKTgZDypBKKpth'.
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : conncheck 0x60af150 FAILED, 0x614fa10 DISCOVERED.
...
** (<unknown>:12439): DEBUG: Local candidate: C3vsy2b6Tc1NcFrozchksJSABXQ5uU47PlZKKWdQuroPJP1vJfPip8xV6lcwspOBkPKkEoqm2EmmKwj8wQxU9A== 1 nYPr2O2YyUAjBINp39vIPw== UDP 550 85.96.116.233 50808
 
** (<unknown>:12439): DEBUG: Remote candidate: DyT9byXz4qfMVepXMLKTgZDypBKKpthJpisI/MEMVPQ= 1 KI0W2WBfs/wkhfcyNLsXrA== UDP 0 94.122.91.24 29504
 


The last two lines show that aMSN was able to connect and considers the local/remote candidates as being 'connected/authenticated/ready to transfer data'...  which is subsequently proven by :
Code:

** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504
** (<unknown>:12439): DEBUG: Agent 0x73f8300 : s1:1: sending 52 bytes to [94.122.91.24]:29504

So you are sending data, so yeay, it's all working.. the problem is.. you're not receiving any data... and why is that? well, I'm guessing that either :
1 - WLM didn't accept your connectivity checks so it doesn't think you are both connected yet
2 - the SIP re-invite which tells the peer which candidates you decided to choose were not acceptable...

so.... first problem, I don't think it's true since we receive our peer-reflexive candidate by having the peer answer us.. since it answered us it means that it already accepted our connectivity check so it knows we're connected... second problem... well, I will need to have a look at that since I was never able to get that use case since I'm not in a symetric NAT with a friend on a full-cone NAT... but with your method (not using TURN or STUN), I can now reproduce this peer-reflexive use case...
Having the protocol log might help me here...
I'm also supposing that you're the one who sent the audio chat request... (which means that you're the controlling agent, so you're the one sending the SIP re-invite), and I'm guessing that if you *receive* the invite instead (you're the controlled agent, so you receive the SIP re-invite, not send it), then it should probably work!
If you could give me :
the nice_debug AND the protocol log pastebins for when you RECEIVE an invitation (both from the same session). I think I just might be able to help!
Make it quick please, I want to release libnice today!
thanks! :)

p.s.: if you update amsn (no need to update libnice/farsight), you shouldn't be having this problem anymore, BUT please do not update until you've tested and provided the logs requested above... thanks! (if you already updated, just do an 'svn update -r10679' to revert to the non-working version).
:D


Title: Audio/Video conversation
Post by: Fenix-TX on November 06, 2008, 04:44:04 pm
Ouch! Now i have this other error:
Code:

Farsight : Preparing
Farsight Prepare error : Error while creating new session (1): Could not create the fsvalve element


Title: Audio/Video conversation
Post by: kakaroto on November 06, 2008, 05:06:45 pm
@harrydb: we'll look into that
@Fenix-TX: lol, yeah, I know what that was.. the 'explain this to me' was rethorical... it just meant that I have no idea why it shows that nicesrc exists but farsight can't create it... anyways, maybe the 'gst-inspect' was just needed to recreate the registry of plugins available...
Now the 'fsvalve' problem is different, try to see if you have it ('gst-inspect fsvalve' or ls /usr/lib/gstreamer-0.10/libfsvalve.so)... if you can't find it, it means that you didn't install the latest gst-plugins-farsight dependency correctly.
@lordamus : ok well it didn't work this time.. in the nice debug log, look at this :
Code:

** (<unknown>:17608): DEBUG: Error on BUS (11) Could not establish connection .. Could not establish connection on the RTP component

so it actually failed this time to connect...  did you try the call with a different person ? that could explain it...


Title: Audio/Video conversation
Post by: BW on November 06, 2008, 05:25:03 pm
Hi,
I also used tcl/tk 8.4 'til today (only updated through svn... :-) )
so for Win2k with 8.4 it's still the old message and with 8.5 I get:
Quote
Farsight : Preparing
Farsight Prepare error : couldn't load library "utils/farsight/tcl_farsight.dll": A function specified in the import table could not be resolved by the system.  Windows is not telling which one, I'm sorry.


Title: Audio/Video conversation
Post by: lordamus on November 06, 2008, 05:34:36 pm
I updated svn but I still cant get sound from both sides... also the other problem when other side invite to audio call it doesnt appear in the window only 30 seconds later  Accept / Reject came.. and when I did accept other side still see ringing call not accepted(other side using wlm8.5)... yea I tryed different persons same results..


Title: Audio/Video conversation
Post by: Fenix-TX on November 06, 2008, 06:28:34 pm
Quote from: "kakaroto"
@harrydb: we'll look into that
@Fenix-TX: lol, yeah, I know what that was.. the 'explain this to me' was rethorical... it just meant that I have no idea why it shows that nicesrc exists but farsight can't create it... anyways, maybe the 'gst-inspect' was just needed to recreate the registry of plugins available...
Now the 'fsvalve' problem is different, try to see if you have it ('gst-inspect fsvalve' or ls /usr/lib/gstreamer-0.10/libfsvalve.so)... if you can't find it, it means that you didn't install the latest gst-plugins-farsight dependency correctly.


Ok, reinstalled gstreamer-plugins-farsight, now other error (OMG)
Code:

Farsight : Preparing
Farsight Prepare error : Unable to set pipeline to PLAYING


Title: Audio/Video conversation
Post by: gothmog on November 06, 2008, 07:16:37 pm
Good news kakaroto, it works at least for me now  :P
Quote
Farsight : Preparing
Farsight : Preparing
logging in, destroying loginscreen : loggingIn
 Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000}
local candidates : {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 1 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2047} {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 2 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2048}
[19:13:51] Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000}
local candidates : {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 1 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2047} {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 2 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2048}

Just a question out of curiosity, what's the meaning of all the stuff appearing behind of Farsight is now prepared!?
Thank you   :)


Title: Audio/Video conversation
Post by: kakaroto on November 06, 2008, 07:16:54 pm
@BW : humm.. ok, well then, could you use 'dependency walker' to tell me which dll/function is missing? google for dependency walker, then open it, copy utils/farsight/tcl_farsight.dll to utils/windows/gstreamer/ then open utils/windows/gstreamer/tcl_farsight.dll with dependency walker, it should tell you if there's any warning/error... you could screenshot it for me... thx
@lordamus : the delay is normal, it's because it needs to first needs to authenticate with the TURN shared secret server, then once it has a username/password, then allocate a port on the TURN server, which will then need to timeout, then the candidates will be gathered.. once it's gathered, then we send the invite...  same thing happens when you receive an invitation, we first need to know our candidates before answering the call... so that once you accept, it can send the response right away... I will probably change that so it looks more fast even though it will still do all that stuff behind the scenes while waiting for you to click accept...

@Fenix-TX : I have no idea on what you did wrong.. I'll just tell you that you might want to retry installing everything from scratch, and make sure you follow the step by step guide...
(if you use oss, try to delete /usr/lib/gstreamer-0.10/libgstalsa*), this might be your problem...

edit :
@gothmog:
cool, glad it worked! :D
Code:
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000}
local candidates : {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 1 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2047} {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 2 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2048}
[19:13:51] Farsight : Farsight is now prepared!
local codecs : {SIREN 96 16000 bitrate=16000} {PCMU 0 8000} {PCMA 8 8000}
local candidates : {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 1 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2047} {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 2 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2048}

well, once farsight is 'prepared' (which means that it was able to do codec discovery and able to get all your local candidates), it will output on the status log what it found... this means :
local codecs : {SIREN 96 16000 birate=16000}, this means the codec named 'SIREN' with the payload type (an RTP specific thing) set to 96 with a clock rate of 16000 and an extra attribute telling us that the bitrate is 16000.
{PCMU 0 8000} : the PCMU codec with payload type 0 and a clock rate of 8000
{PCMA 8 8000} : the PCMA codec with payload type 8 and a clock rate of 8000

local candidates : {7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 1 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2047} , this means that it found a local candidate, for which it generated a random username 7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw=, it is for component '1' (which is RTP) and with a random password EgNRZ+ajHSPgiOXuDwS7Uw==, it uses UDP as a transport, has a qvalue (priority) of 0.829 and is bound to the ip *.*.*.* and using port 2047.
{7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw= 2 EgNRZ+ajHSPgiOXuDwS7Uw== UDP 0.829 *.*.*.* 2048}, this means that it found a local candidate, for which it generated a random username 7xp72qsn5jfVPoPi6Kvh11Vt4tNUChgBMD8eJhzyPFw=, it is for component '2' (which is RTCP) and with a random password EgNRZ+ajHSPgiOXuDwS7Uw==, it uses UDP as a transport, has a qvalue (priority) of 0.829 and is bound to the ip *.*.*.* and using port 2048.

with this info, I realize that TURN didn't work, so I'm assuming that you received that information from the audio/video assistant that you launched BEFORE connecting amsn. It also tells me that either STUN didn't work (since you only have one host candidate and no server reflexive candidate), OR that you have no router/NAT and that you're directly connected to the internet.
I also know that you have SIREN installed correctly and working, which means that you did indeed install gst-plugins-farsight correctly (well, I did it all for you:p), so if you had a problem, with just that info, I would know if you were using SIREN or not, and whether you were behind a router or not, or whether you were able to use TURN or not...
I hope it answers your question! :)


EDIT2 :
@harrydb: Here's what Olivier has to say about your problem :
Quote
<Tester> the fs-type...
<Tester> that means that the glib-mkenum failed
<Tester> in gst-libs/gst/farsight
<Tester> maybe does 'make clean all' in there will help

I would say that if it doesn't help, please pastebin the whole output of the make...


Title: Audio/Video conversation
Post by: gothmog on November 06, 2008, 08:28:20 pm
Wow, that was awesome, it's really unbelievable all this is done with only press a button  :o
And yes, you were right, I'm directly connected to the internet but you didn't on where I received this because it was what appeared in status log just login  :lol:


Title: Audio/Video conversation
Post by: kakaroto on November 06, 2008, 08:38:23 pm
yeah, right, it's all the same.. it tries to prepare farsight (to know if you can do an audio call, so it can enable your SIP capabilities flag) when you login, so since it prepares farsight before you were connected on msn, so it didn't use TURN either..
and yes, it's all magic, just with one press of a button :p


Title: Audio/Video conversation
Post by: Fenix-TX on November 06, 2008, 08:56:09 pm
Ok. Undoing all steps and made them again works. Hooray!!!!!


Title: Audio/Video conversation
Post by: kakaroto on November 06, 2008, 09:32:21 pm
great! :)


Title: Audio/Video conversation
Post by: BW on November 06, 2008, 10:07:54 pm
Hi,
here are the screenshots from dependency walker
http://www.savefile.com/files/1874325

it seems to be APPHELP.DLL missing on Win2k systems


Title: Audio/Video conversation
Post by: kakaroto on November 07, 2008, 12:22:59 am
@BW: Ahh! I see, it's not APPHELP.DLL, since that one is a delay-loaded, which means that if it can't find it, it won't load it.. and it's needed by a system DLL, so it's all good (otherwise no other app on your system would load...).
The real problem is ws2_32.dll (winsocket2 dll) because that's what we need... and it seems to be missing getaddrinfo/freeaddrinfo/getnameinfo...
Now that I'm reading the msdn* page correctly (thx to you), it seems that :
Quote

The getnameinfo function was added to the Ws2_32.dll on Windows XP and later. If you want to execute an application using this function on earlier versions of Windows (Windows 2000, Windows NT, and Windows Me/98/95), then you need to include the Ws2tcpip.h file and also include the Wspiapi.h file. When the Wspiapi.h include file is added, the getnameinfo function is defined to the WspiapiGetNameInfo inline function in the Wspiapi.h file. At runtime, the WspiapiGetNameInfo function is implemented in such a way that if the Ws2_32.dll or the Wship6.dll (the file containing getnameinfo in the IPv6 Technology Preview for Windows 2000) does not include getnameinfo, then a version of getnameinfo is implemented inline based on code in the Wspiapi.h header file. This inline code will be used on older Windows platforms that do not natively support the getnameinfo function.

Same goes for those three other functions... which basically means that I will have to add that include and recompile it for you and all those other obsolete windows users :p
thx for showing this to me! :)

* http://msdn.microsoft.com/en-us/library/ms738532(VS.85).aspx


EDIT : @BW: could also give me the screens dependency walker for libnice.dll in utils/windows/gstreamer/ (no need for the missing dlls with a 'clock' next to them) ? thanks!


Title: Audio/Video conversation
Post by: BW on November 07, 2008, 01:40:47 am
Hi,
here are the dependency walker screenshots for libnice.dll and the other files producing error messages, in case you need them too...
http://www.savefile.com/files/1874570


Title: Audio/Video conversation
Post by: kakaroto on November 07, 2008, 05:27:56 am
@BW: It should be fine now! update SVN
oh crap..I didn't see all those other dlls having the same problem.. ok, I'll look into it tomorrow hopefully!

EDIT ok, just update now, it should be fine!


Title: Audio/Video conversation
Post by: BW on November 07, 2008, 06:01:44 pm
Hi,
thanks kakaroto for the new dlls and the short introduction into dll dependencies.
Farsight is now prepared! yeay! :D

The only thing I don't understand is why we don't need to solve the dependency
problems on librawudp-transmitter.dll , but if it works without, I don't care...

The codeline you changed in rev 10679 adds another  .../windows/gstreamer
to env(PATH) every time an audio call is made, could that lead into problems?

Another thing i noticed is when I hangup on amsn side, WLM doesn't hangup as well

thanx again, bye


Title: Audio/Video conversation
Post by: harrydb on November 07, 2008, 06:40:49 pm
Yup, it compiles and installes now, but in amsn I now get:

Farsight : Preparing
Farsight Prepare error : couldn't load file "utils/farsight/tcl_farsight.so": utils/farsight/tcl_farsight.so: undefined symbol: fs_stream_set_remote_candidates


Title: Audio/Video conversation
Post by: kakaroto on November 07, 2008, 09:47:56 pm
@BW: cool!:)
well, I didn't have time to solve the librawudp-transmiter.dll but it's not needed anyways, I'll probably remove the file anyways.. it's the farsight transmitter.. we use the libnice-transmitter, and we don't use the librawudp-transmitter.. that's why it works without it.
about the env(PATH), nice catch! I'll have it fixed, but no, it shouldn't cause any problems anyways!
about the hanging up problem, it's a bug with our SIP implementation, i'll have to fix it, but it's unrelated to farsight//gstreamer, it's just bad tcl code...

@harrydb: make sure you are following the instructions correctly! you seem to be using an old version of farsight or something.... maybe you should start everything over!


Title: Audio/Video conversation
Post by: harrydb on November 08, 2008, 01:45:43 am
apparently there were some old libraries hanging around that messed things up, after cleaning up everything on my filesystem that had the name farsight in it I finally got everything to load. Thanks for the tips and patience. I will try to make a call tomorrow.

By the way, I'm on ubuntu intrepid and it already has gstreamer0.10-plugins-farsight 0.12.9 so I did not compile that one.


Title: Audio/Video conversation
Post by: lordamus on November 08, 2008, 02:38:20 am
kakaroto:  I`m on svn 10689  now my voice finally hear from other side wlm8.5  but user said the voice comes very deeply,parasite and noise..  Mic levels is high in alsa mixer also mic boost is at the middle..  I`m using this settings in skype without problems..  it seems still some connection prblem around...


Title: Audio/Video conversation
Post by: kakaroto on November 08, 2008, 06:01:08 am
@harrydb: cool, glad to hear that.
@lordamus: when did you update/recompile amsn ? I fixed a sound issue yesterday which should make the voice 'clear' instead of 'with interruptions and hard to hear'... you don't need to recompile libnice or farsight2, just svn update amsn and recompile amsn...


Title: Audio/Video conversation
Post by: lordamus on November 08, 2008, 11:54:57 pm
kakaroto:  I compiled at nov 8  1:00 am midnight  I dıd again svn update nothing updated I"m on svn 10700..  My voice still with interruptions and hard to hear and other sides voice didnt came to me nothing to hear...


Title: Audio/Video conversation
Post by: MastaG on November 09, 2008, 02:58:41 pm
Quote from: "lordamus"
kakaroto:  I compiled at nov 8  1:00 am midnight  I dıd again svn update nothing updated I"m on svn 10700..  My voice still with interruptions and hard to hear and other sides voice didnt came to me nothing to hear...


You could try running this command: gstreamer-properties
On fedora9 it brings up a panel to setup sound input and output for gstreamer.
I set it to use the mic inside my webcam and it works when I press the test button.
(http://www.akelo.ath.cx/~mastag/gstreamer.png)[/img]


Title: Audio/Video conversation
Post by: lordamus on November 09, 2008, 04:59:05 pm
I tryed that one for sure  input/output is default Alsa   device is all default...


Title: Audio/Video conversation
Post by: anubisg1 on November 09, 2008, 07:40:41 pm
hi.. i have a problem with libnice.. on my pc build fine, but fail on my chroot:

Code:
+ /usr/bin/make
/usr/bin/make  all-recursive
make[1]: Entering directory `/usr/src/packages/BUILD/libnice'
Making all in stun
make[2]: Entering directory `/usr/src/packages/BUILD/libnice/stun'
Making all in .
make[3]: Entering directory `/usr/src/packages/BUILD/libnice/stun'
/bin/sh ../libtool --tag=CC   --mode=compile gcc -DHAVE_CONFIG_H -I. -I..  -I..  -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter  -march=i586 -mtune=i686 -fmessage-length=0 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -O2 -MT stunagent.lo -MD -MP -MF .deps/stunagent.Tpo -c -o stunagent.lo stunagent.c
mkdir .libs
 gcc -DHAVE_CONFIG_H -I. -I.. -I.. -std=gnu99 -Wall -Werror -Wextra -Wundef -Wnested-externs -Wwrite-strings -Wpointer-arith -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wno-unused-parameter -march=i586 -mtune=i686 -fmessage-length=0 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -O2 -MT stunagent.lo -MD -MP -MF .deps/stunagent.Tpo -c stunagent.c  -fPIC -DPIC -o .libs/stunagent.o
cc1: warnings being treated as errors
stunagent.c: In function 'stun_agent_default_validater':
stunagent.c:66: error: unused parameter 'agent'
stunagent.c:67: error: unused parameter 'message'
make[3]: *** [stunagent.lo] Error 1
make[3]: Leaving directory `/usr/src/packages/BUILD/libnice/stun'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/src/packages/BUILD/libnice/stun'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/packages/BUILD/libnice'
make: *** [all] Error 2


are you able to tell me witch is the missing build-dependencie? thanks

Andrea


Title: Audio/Video conversation
Post by: kakaroto on November 09, 2008, 09:59:52 pm
@lordamus: I don't know, I'm still thinking about a volume issue from your side or your friend's side... try to do a call with yourself by opening two amsns...
@anubisg1: humm... I guess that's my bad... you can avoid that by editing the file common.mk in libnice's directly and remove the line with
Code:
  -Werror \

then run ./autogen.sh and redo the configure and make...


Title: Audio/Video conversation
Post by: anubisg1 on November 09, 2008, 11:00:09 pm
Quote from: "kakaroto"
@lordamus: I don't know, I'm still thinking about a volume issue from your side or your friend's side... try to do a call with yourself by opening two amsns...
@anubisg1: humm... I guess that's my bad... you can avoid that by editing the file common.mk in libnice's directly and remove the line with
Code:
  -Werror \

then run ./autogen.sh and redo the configure and make...


works in chroot too.

thanks

(i'm building suse rpms.. soon available on packman repository)


Title: Audio/Video conversation
Post by: kakaroto on November 09, 2008, 11:26:02 pm
cool, thx! :)
if you want an explanation, the problem was that in libnice, there is a function that doesn't use all the arguments it receives (it doesn't need them all), and in your chroot, you probably have a gcc option that tells it to warn when an argument is unused...
The problem is that I forgot to remove the option -Werror when releasing libnice, that option basically says "treat all warnings as errors" which is why it errored out.. without the option, now it just shows a warning and continues the compilation. I guess i'll have to add that magic option to my build environment to make sure such warnings do not happen... and remove the -Werror for future releases...


Title: Audio/Video conversation
Post by: anubisg1 on November 09, 2008, 11:35:51 pm
perfect, thanks.. i perfectly understood.

btw:

http://packman.links2linux.org/package/nice/

http://packman.links2linux.org/package/libfarsight2

http://packman.links2linux.org/package/gstreamer-0_10-plugins-farsight

amsn-svn is on queue :) and will be available here:

http://packman.links2linux.org/package/amsnsvn

btw that links are valid for manual download, you can use yast/zypper with packaman repo.
mirrors available here: http://en.opensuse.org/Additional_YaST_Package_Repositories#Packman

btw.. farsight gstreamer plugin fail on suse 11.1 because of that:

Code:
+ cd gst-plugins-farsight-0.12.9
+ autoreconf -f -i
m4/gst-feature.m4:49: error: m4_defn: undefined macro: _m4_divert_diversion
m4/gst-feature.m4:49: the top level
autom4te: /usr/bin/m4 failed with exit status: 1
aclocal: autom4te failed with exit status: 1
autoreconf: aclocal failed with exit status: 1
error: Bad exit status from /var/tmp/rpm-tmp.57268 (%build)


ideas?

thanks a lot


---edit---

btw part 2

farsight2 fail (on chroot only) if docs are built

i'll publish log next time.. a been deleted :(


Title: Audio/Video conversation
Post by: anubisg1 on November 10, 2008, 10:28:31 am
yeaah!!! farsight works :P (with libv4l too)

openSUSErs feel free to do that:

zypper in amsnSVN

 :lol:  :lol:

everything will works!!

you may want to wait mirrors syncs

btw.. that is libfarsight 2 with --enable-gtk-doc

now succeded ;) i don't know why.. spec file  is exactly the same of yesterday.. wait... not.. ok i understood...

paralled building with jobs made make fail! ;)

thanks and sorry for the inconvenient


Title: Audio/Video conversation
Post by: lordamus on November 11, 2008, 05:25:43 am
Nope all the volumes is high I'm using same volumes in skype its very clear.. friend side also he's calling other people without prblem but when he call me I cant hear something..Cant try that I get  Farsight Prepare error : Unable to set pipeline to PLAYING

Quote from: "kakaroto"
@lordamus: I don't know, I'm still thinking about a volume issue from your side or your friend's side... try to do a call with yourself by opening two amsns...
@anubisg1: humm... I guess that's my bad... you can avoid that by editing the file common.mk in libnice's directly and remove the line with
Code:
  -Werror \

then run ./autogen.sh and redo the configure and make...


Title: Audio/Video conversation
Post by: kakaroto on November 11, 2008, 11:33:25 am
@anubisg1: great, I'm glad you got it to build and thanks for the packages, I'm sure they'll come in handy to more than one!
@lordamus: I don't know then.. but the fact that it can't be 'PLAYING' when you try with two amsn means you don't have software mixing.. so either alsa doesn't have it.. or you're not using alsa...
If you are willing to give me ssh access to your machine so I can run some tests, then try to get online on IRC at some point and ping me in there so I can try to see where the problem comes from... otherwise, you'll have to wait and hope it gets fixed at some point... hehe


Title: Audio/Video conversation
Post by: lordamus on November 12, 2008, 01:09:17 am
now farsight totally broken on svn 10707

Farsight : Preparing
Farsight Prepare error : Unable to set pipeline to PLAYING

Yea I`m using only alsa  I removed pulse audio..lots of prblems before that with pulse..hah need to change some firewall/router sets for that...what that tests is ?  maybe I can try ?

Quote from: "kakaroto"
@anubisg1: great, I'm glad you got it to build and thanks for the packages, I'm sure they'll come in handy to more than one!
@lordamus: I don't know then.. but the fact that it can't be 'PLAYING' when you try with two amsn means you don't have software mixing.. so either alsa doesn't have it.. or you're not using alsa...
If you are willing to give me ssh access to your machine so I can run some tests, then try to get online on IRC at some point and ping me in there so I can try to see where the problem comes from... otherwise, you'll have to wait and hope it gets fixed at some point... hehe


Title: Audio/Video conversation
Post by: kakaroto on November 12, 2008, 01:26:49 am
nope it's not broken.. you just have the same problem of when you had too amsns open... you don't have dmix for your alsa or whatever is necessary for accessing the sound card from two processes... you probably had your mp3 player running or some other application using the sound card (even flash on a browser could do that).

edit oh and by the way, the tests are for testing gstreamer with your sound card.. so it's playing with gst-launch and gst-inspect...
if you can be on IRC, we could maybe try to fix it without ssh


Title: Creating a .deb package
Post by: fcastillo on November 15, 2008, 06:18:34 pm
I've notice that the farsight wiki has changed and now installing libnice and farsight is easier than before because there isn't git repositories involved. The new ubuntu intrepid has the farsight plugin for gstreamer, so I was wondering if somebody can create some .deb packages for libnice and farsight, so we can install them on ubuntu, and make it easier to distribute the svn version of amsn to friends. I'm new at creating .deb packages, I followed some guides around, but I couldn't make it work. Please can somebody help us making this packages. Thanks a lot!!!


Title: Audio/Video conversation
Post by: kakaroto on November 15, 2008, 09:04:14 pm
the packages are being created, just give it some time.


Title: Audio/Video conversation
Post by: fcastillo on November 15, 2008, 09:31:24 pm
Thank you so much!!!! I really appreciate it, thanks again


Title: Audio/Video conversation
Post by: marc2009 on November 18, 2008, 06:43:31 pm
Quote from: "kakaroto"
the packages are being created, just give it some time.


and for fedora  :D  :D ?


Title: Audio/Video conversation
Post by: kakaroto on November 18, 2008, 10:23:49 pm
also.. but hey, guess what.. we do the code, we don't do the packages... you should yell at your distro's maintainers instead! :)
but anyways, someone from Collabora was working on debian packages (but I first need to remove the openssl dependency ), and tjikkun should be working on the fedora packages...
and before you ask... mac binaries are in SVN now for intel, and ppc binaries will be there too *soon*.


Title: Audio/Video conversation
Post by: thomas13202 on November 24, 2008, 03:18:00 am
I had the source code for amsn 098b svn version 10529 and i installed that and it will log into msn but the latest svn version will not. i am not sure if anything between the two will log into msn or not. the problem i am having is that while i have the latest libnice and farsight2 from ur websight installed amsn does not recognize that farsight is installed. i know i compiled the programs right as i copyed and pasted the comands from ur instructions directly into the terminal and i did ldconfig. do i need to reinstall the older versions of libnice and farsight2 to get them to work with that above mentioned version of amsn? it used to work with the older versions before i updated them when trying to update to the latest svn version of amsn


Title: Audio/Video conversation
Post by: kakaroto on November 24, 2008, 07:26:47 pm
@thomas13202: you only need to post your question once, not twice, and make sure you post in the correct thread. As I said in the other thread, re-read the wiki and retry from scratch.


Title: Audio/Video conversation
Post by: thomas13202 on November 25, 2008, 06:37:20 pm
i followed ur advice and it says farsight preparing could not prepare new stream hear is the output from the terminal that i started amsn from
bash-3.2$ amsn
** (<unknown>:8707): DEBUG: bus message : farsight-codecs-changed
** (<unknown>:8707): DEBUG: bus message : farsight-codecs-changed
** (<unknown>:8707): DEBUG: CODECS ARE READY

(<unknown>:8707): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
** (<unknown>:8707): DEBUG: stun ip : 64.14.48.28 : 3478
** (<unknown>:8707): DEBUG: Receive bus message from the event proc : farsight-codecs-changed

(<unknown>:8707): GLib-GObject-CRITICAL **: g_object_get: assertion `G_IS_OBJECT (object)' failed
** (<unknown>:8707): DEBUG: CODECS ARE READY
** (<unknown>:8707): DEBUG: Receive bus message from the event proc : farsight-codecs-changed
Farsight : Preparing
** (<unknown>:8707): DEBUG: bus message : farsight-codecs-changed
** (<unknown>:8707): DEBUG: bus message : farsight-codecs-changed
** (<unknown>:8707): DEBUG: CODECS ARE READY

(<unknown>:8707): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
** (<unknown>:8707): DEBUG: stun ip : 64.14.48.28 : 3478
** (<unknown>:8707): DEBUG: FS: relay info = 0x9398c80 - 2

(<unknown>:8707): GLib-GObject-CRITICAL **: g_object_new_valist: assertion `G_TYPE_IS_OBJECT (object_type)' failed
Farsight Prepare error : Couldn't create new stream
** (<unknown>:8707): DEBUG: Receive bus message from the event proc : farsight-codecs-changed

(<unknown>:8707): GLib-GObject-CRITICAL **: g_object_get: assertion `G_IS_OBJECT (object)' failed
** (<unknown>:8707): DEBUG: CODECS ARE READY
** (<unknown>:8707): DEBUG: Receive bus message from the event proc : farsight-codecs-changed

@ kakaroto i posted this here and posted a observation i am having with the login issue in the thread for that is this the correct way to do it. i have no idea what alll the above output from the console means.


Title: Audio/Video conversation
Post by: McMajik on November 28, 2008, 06:20:33 pm
Hi. Uhh...when i configure the SVN version of amsn (before build), it says i don't have gstreamer and farsight installed. I know that i do, because i spent the past 2 days trying to get it to work. I've just compiled and installed all the packages it says it requires on the farsight page on the wiki, configured libnice and farsight with the extra things it said to put on the ./configure command, ran ldconfig after i installed it, and generally followed it word for word, but it still says i don't have it installed. The packages are all the version it says on the wiki.

Any ideas why?


Title: Audio/Video conversation
Post by: kakaroto on November 28, 2008, 07:01:42 pm
@thomas13202: Make sure your glib/gstreamer/farsight installation is correct
@McMajik: hi, if aMSN says it can't find it, then it can't find it, you may have the wrong versions... although you say you made sure everything was right, I would say that you missed something... retry everything from scratch just in case...
you could also give me the output of (the result I show is on my machine and it works fine with it) :
Code:
kakaroto% pkg-config --modversion gstreamer-0.10
0.10.21
kakaroto% pkg-config --modversion gstreamer-interfaces-0.10
0.10.21
kakaroto% pkg-config --modversion farsight2-0.10
0.0.4.1
kakaroto% pkg-config --modversion nice          
0.0.3.1


Title: Audio/Video conversation
Post by: McMajik on November 28, 2008, 09:34:36 pm
Quote

mcmajik@mcmajik-desktop:~$ pkg-config --modversion gstreamer-0.10
0.10.20
mcmajik@mcmajik-desktop:~$ pkg-config --modversion gstreamer-interfaces-0.10
0.10.20
mcmajik@mcmajik-desktop:~$ pkg-config --modversion farsight2-0.10
0.0.3
mcmajik@mcmajik-desktop:~$ pkg-config --modversion nice
0.0.2


Ahh, right, farsight is 0.0.3

I installed 0.0.4, but that must be from a previous instalation. I'll try reinstalling that.

EDIT: out of curiosity...how would one go about removing farsight?


Title: Audio/Video conversation
Post by: kakaroto on November 28, 2008, 11:29:04 pm
humm... install the new one over the old, that should do the trick...
apart from that, i'm guessing 'make uninstall' should work.


Title: Audio/Video conversation
Post by: McMajik on November 29, 2008, 12:12:51 am
Sorry about not being clearer, what i meant was i installed 0.0.4, so the 0.0.3 might be from a previous installation. Which would mean that installing it over the old one didn't work. And would i need the source for 0.0.3 to use make uninstall?

NVM, uninstalled 0.0.3, and suddenly it recognised 0.0.4. Installed 0.0.4 again anyway, just in case. It recognised and compiled with farsight, but now it can't find it in the audio/video assistant.

Quote
Farsight : Preparing
Farsight Prepare error : Couldn't create fsrtpconference


any help?


Title: Audio/Video conversation
Post by: kakaroto on November 29, 2008, 01:08:28 am
http://amsn-project.net/wiki/Farsight#Step-by-step_compilation_guide <-- search for 'fsrtpconference"


Title: Audio/Video conversation
Post by: McMajik on November 29, 2008, 01:28:16 am
i coppied and pasted the ./configure command with the extras from there, so not having the --prefix=/usr wouldn't be the problem...
I just recompiled and reinstalled farsight, and amsn, to make sure, and it still says it.

What else could cause that error?


Title: Audio/Video conversation
Post by: kakaroto on November 29, 2008, 05:18:26 am
try 'gst-inspect fsrtpconference'. if it says it can't find it, then try looking for 'libfsrtpconference.so' in /usr/lib/gstreamer-0.10/ It should be there, if it's not, then you did something wrong.. if it's there, I'm sure gst-inspect would tell you why it failed to load the .so...


Title: Audio/Video conversation
Post by: McMajik on November 29, 2008, 11:11:24 am
Ok, i found it. Its in the gstreamer folder. But when i run gst-inspect, it says "no such element or plugin 'fsrtpconference' "

*is VERRRRRRRRRRRRY confuzzled*  :?:


Title: Audio/Video conversation
Post by: marc2009 on November 29, 2008, 12:14:16 pm
I would like just to know something : I've just made yum list farsight* and here is the result :
Code:
farsight.i386                           0.1.28-2.fc10                     fedora
farsight-devel.i386                     0.1.28-2.fc10                     fedora


so, can we install this version, or wouldn't it work ?

++

ps : instructions to download ( git ... ) and install farsight2 and libpri didn't change, did they ?


Title: Audio/Video conversation
Post by: thomas13202 on November 29, 2008, 10:23:11 pm
i installed all the dependencies from the mandriva package manager with correct versions but the name of glib is libglib there is no glib gstreamer is libgstreamer there is no gstreamer and gstfarsightplugin is gstreamer farsight could these different names be causing the problem and lib nice and farsight r both got from the link on the wiki and installed on my comp from a package i made with check install,

also i installed everything on a different comp also running mandriva linux 2009.0 and while the audio assistant says everything is ok every time i try to make a voice call it crashes amsn. what would cause that?

mandriva linux uses the pulse audio server for its sound i wonder if pulse audio could be causing the problem. i know pulse audio gives me problems with sound events in gyachi the program i use for yahoo messenger


Title: Audio/Video conversation
Post by: kakaroto on November 30, 2008, 06:12:38 am
McMajik: yes it is confuzing.. what does 'gst-inspect' give you as an output ? how many 'plugins' does it find ? are all the .so files in /usr/lib/gstreamer-0.10 ? or are they in /usr/local/lib/gstreamer-0.10/ ? Maybe you installed gstreamer on /usr/local which is why it searches for the plugins in /usr/local/lib instead of /usr/lib ... can you look into that on your own a bit and try to figure it out ?
@marc2009: no, that's farsight, you need farsight2, read the wiki, it has everything you need to know.. and no, you don't need git anymore because we now released both libnice and farsight, re-read the wiki, it was updated with the appropriate dependency versions...
@thomas13202: the name of glib and gstreamer is perfectly normal, every distribution gives different names but it changes nothing in the end.
Please re-read the wiki as it has all the needed information, make sure you use the appropriate version of each package. if you use older versions it will crash! and your problem seems to be coming from a bad gstreamer installation, so make sure again, that you had the correct versions... if they are, then maybe the mandriva packages are bad, you might want to install gstreamer from the source code.
About pulse, it should still work, but I have no guarantees on how it will work.. try disabling pulse audio and see if it changes anything, but that's not the cause of the problem you posted before...


Title: Audio/Video conversation
Post by: marc2009 on November 30, 2008, 10:21:59 am
Quote from: "kakaroto"
McMajik: yes it is confuzing.. what does 'gst-inspect' give you as an output ? how many 'plugins' does it find ? are all the .so files in /usr/lib/gstreamer-0.10 ? or are they in /usr/local/lib/gstreamer-0.10/ ? Maybe you installed gstreamer on /usr/local which is why it searches for the plugins in /usr/local/lib instead of /usr/lib ... can you look into that on your own a bit and try to figure it out ?
@marc2009: no, that's farsight, you need farsight2, read the wiki, it has everything you need to know.. and no, you don't need git anymore because we now released both libnice and farsight, re-read the wiki, it was updated with the appropriate dependency versions...
@thomas13202: the name of glib and gstreamer is perfectly normal, every distribution gives different names but it changes nothing in the end.
Please re-read the wiki as it has all the needed information, make sure you use the appropriate version of each package. if you use older versions it will crash! and your problem seems to be coming from a bad gstreamer installation, so make sure again, that you had the correct versions... if they are, then maybe the mandriva packages are bad, you might want to install gstreamer from the source code.
About pulse, it should still work, but I have no guarantees on how it will work.. try disabling pulse audio and see if it changes anything, but that's not the cause of the problem you posted before...


Thanks a lot for your work, it works, and it looks as if it's quicker than before ( I mean the load of farsight in the audio assistant )

but, it's bizarre, why should we add --disable-python and --disable-gtk-doc ?? I didn't add them, and it worked !

and I have now a window called level_IN, what is it ? ( it stays at 0% )

++


Title: Audio/Video conversation
Post by: McMajik on November 30, 2008, 12:38:18 pm
ugh...
I copied libfsrtpconference.so, .a and .la over to the gstreamer folder in local. I also had to copy the libfsfunnel files over. They fixed two errors.
But now i get this error, and i have no idea what it means

Quote
Farsight : Preparing
Farsight debug : CODECS ARE READY
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x91e11b0 - 2
Farsight Prepare error : Unable to set pipeline to PLAYING


...help?


Title: Audio/Video conversation
Post by: lordamus on November 30, 2008, 04:43:27 pm
what is the level_in dialog  ?


Title: Audio/Video conversation
Post by: Xmister on November 30, 2008, 07:34:05 pm
The configure script doesn't find farsight2 and gstreamer, but they are installed, compiled from source.
The plugin details (gst-inspect fsrtpconference)
Quote
Plugin Details:
  Name:         fsrtpconference
  Description:      Farsight RTP Conference plugin
  Filename:      /usr/lib/gstreamer-0.10/libfsrtpconference.so
  Version:      0.0.2.1
  License:      LGPL
  Source module:   farsight2
  Binary package:   Farsight
  Origin URL:      http://farsight.freedesktop.org/

The end of configure script:
Quote
checking for GST... yes
checking for GST_INTERFACES... yes
checking for FARSIGHT2... no
checking for LIBV4L... no
configure: creating ./config.status
config.status: creating Makefile
config.status: creating utils/linux/capture/config.h
config.status: utils/linux/capture/config.h is unchanged

compile time options summary
============================

    X11          : yes
    Tcl       : 8.5
    TK        : 8.5
    DEBUG        : no
    STATIC       : no
    FARSIGHT     : no
    LIBV4L       : no

*** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
*** Read this for more information : http://amsn-project.net/wiki/Farsight


Title: Audio/Video conversation
Post by: marc2009 on November 30, 2008, 08:13:48 pm
Quote from: "Xmister"
The configure script doesn't find farsight2 and gstreamer, but they are installed, compiled from source.
The plugin details (gst-inspect fsrtpconference)
Quote
Plugin Details:
  Name:         fsrtpconference
  Description:      Farsight RTP Conference plugin
  Filename:      /usr/lib/gstreamer-0.10/libfsrtpconference.so
  Version:      0.0.2.1
  License:      LGPL
  Source module:   farsight2
  Binary package:   Farsight
  Origin URL:      http://farsight.freedesktop.org/

The end of configure script:
Quote
checking for GST... yes
checking for GST_INTERFACES... yes
checking for FARSIGHT2... no
checking for LIBV4L... no
configure: creating ./config.status
config.status: creating Makefile
config.status: creating utils/linux/capture/config.h
config.status: utils/linux/capture/config.h is unchanged

compile time options summary
============================

    X11          : yes
    Tcl       : 8.5
    TK        : 8.5
    DEBUG        : no
    STATIC       : no
    FARSIGHT     : no
    LIBV4L       : no

*** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
*** Read this for more information : http://amsn-project.net/wiki/Farsight


did you add the --prefix=/usr parameter to the configure of libnice and farsight2 ?

++


Title: Audio/Video conversation
Post by: kakaroto on November 30, 2008, 08:55:08 pm
@marc2009 :
cool, glad it works.. about the --disable-python and --disable-gtk-doc, yes it will work with them, it just adds python bindings for farsight and compiles farsight's documentation.. the reason I said to disable them is that if you don't have python-devel and gstreamer-python packages, and gtk-doc-tools installed, then it won't compile with those options.. so since we don't need python/docs, I just said to disable them instead of forcing everyone to install additional, unneeded dependencies...
About the level_IN window, read here :  http://www.amsn-project.net/forums/viewtopic.php?p=35276#35276
@McMajik: ok cool, so that helped.. now it still can't go into PLAYING.. don't know why.. it could be because your source (mic) isn't working... it could happen if for example you have installed libgstalsa but you use oss instead of alsa.. that stuff should be configurable and fixed soon though...
@lordamus : http://www.amsn-project.net/forums/viewtopic.php?p=35276#35276
@XMister: your farsight version is 0.0.2.1 which is OLD! Follow the guide from the wiki exactly as it says it.. you need version 0.0.4 of farsight2 otherwise amsn will not detect it because it knows previous versions won't work with amsn...
@marc2009: thanks for the help there! :)


Title: Audio/Video conversation
Post by: marc2009 on November 30, 2008, 10:40:11 pm
Quote from: "kakaroto"
@marc2009 :
cool, glad it works.. about the --disable-python and --disable-gtk-doc, yes it will work with them, it just adds python bindings for farsight and compiles farsight's documentation.. the reason I said to disable them is that if you don't have python-devel and gstreamer-python packages, and gtk-doc-tools installed, then it won't compile with those options.. so since we don't need python/docs, I just said to disable them instead of forcing everyone to install additional, unneeded dependencies...
About the level_IN window, read here :  http://www.amsn-project.net/forums/viewtopic.php?p=35276#35276
@McMajik: ok cool, so that helped.. now it still can't go into PLAYING.. don't know why.. it could be because your source (mic) isn't working... it could happen if for example you have installed libgstalsa but you use oss instead of alsa.. that stuff should be configurable and fixed soon though...
@lordamus : http://www.amsn-project.net/forums/viewtopic.php?p=35276#35276
@XMister: your farsight version is 0.0.2.1 which is OLD! Follow the guide from the wiki exactly as it says it.. you need version 0.0.4 of farsight2 otherwise amsn will not detect it because it knows previous versions won't work with amsn...
@marc2009: thanks for the help there! :)


Thanks for your answer, and good job !!!

Good evening

++


Title: Audio/Video conversation
Post by: Xmister on November 30, 2008, 11:22:05 pm
Quote from: "kakaroto"

@XMister: your farsight version is 0.0.2.1 which is OLD! Follow the guide from the wiki exactly as it says it.. you need version 0.0.4 of farsight2 otherwise amsn will not detect it because it knows previous versions won't work with amsn...


Yeah, I see.
But I tried the 0.0.4 version, but it wasn't compiled
Quote

fs-nice-stream-transmitter.c: In function ‘fs_nice_stream_transmitter_set_relay_info’:
fs-nice-stream-transmitter.c:883: error: ‘NiceRelayType’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:883: error: expected ‘;’ before ‘relay_type’
fs-nice-stream-transmitter.c:895: error: ‘relay_type’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:895: error: ‘NICE_RELAY_TYPE_TURN_TCP’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:897: error: ‘NICE_RELAY_TYPE_TURN_TLS’ undeclared (first use in this function)
fs-nice-stream-transmitter.c:902: error: too many arguments to function ‘nice_agent_set_relay_info’
make[3]: *** [libnice_transmitter_la-fs-nice-stream-transmitter.lo] Error 1
make[3]: Leaving directory `/home/xmister/downloads/farsight/farsight2-0.0.4/transmitters/nice'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/xmister/downloads/farsight/farsight2-0.0.4/transmitters'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/xmister/downloads/farsight/farsight2-0.0.4'
make: *** [all] Error 2

So after that I tried your git, but it seems like it's the version 0.0.2
So...what should I do now?


Title: Audio/Video conversation
Post by: kakaroto on December 01, 2008, 02:13:41 am
Xmister: farsight 0.0.4 didn't compile because you don't have libnice 0.0.3 installed.. you first need to use libnice 0.0.3 and install it, then try to recompile farsight2
and no, don't use git anymore, as I said, it's released now, so you don't need git anymore, just download the necessary versions from their respective websites.. or more precisely : follow the wiki!!!


Title: Audio/Video conversation
Post by: lordamus on December 01, 2008, 01:49:00 pm
hmm is it normal that level_in window appear every login to msn ?


Title: Audio/Video conversation
Post by: kakaroto on December 01, 2008, 01:59:02 pm
yes, because we test farsight every login to see if you can do an audio call and set the appropriate capabilities flag on your session so other will know about it... and no, it won't stay that way....


Title: Audio/Video conversation
Post by: Xmister on December 01, 2008, 04:26:57 pm
Quote from: "kakaroto"
Xmister: farsight 0.0.4 didn't compile because you don't have libnice 0.0.3 installed.. you first need to use libnice 0.0.3 and install it, then try to recompile farsight2
and no, don't use git anymore, as I said, it's released now, so you don't need git anymore, just download the necessary versions from their respective websites.. or more precisely : follow the wiki!!!

I have and had libnice 0.0.3 installed:
Quote
4$ gst-inspect nice
Plugin Details:
  Name:         nice
  Description:      Interactive UDP connectivity establishment
  Filename:      /usr/lib/gstreamer-0.10/libgstnice.so
  Version:      0.0.3
  License:      LGPL
  Source module:   nice
  Binary package:   nice
  Origin URL:      http://telepathy.freedesktop.org/wiki/

  nicesink: ICE sink
  nicesrc: ICE source

  2 features:
  +-- 2 elements

But farsight 2 don't compile.
I had done everything as the wiki says

EDIT:

Finally I managed to solve it.
I had to configure libnice and farsight2 with prefix=/usr/local too, and compile them this way.
Everything went fine, and amsn has found it too.


Title: Audio/Video conversation
Post by: lordamus on December 02, 2008, 07:41:13 pm
Quote from: "kakaroto"
yes, because we test farsight every login to see if you can do an audio call and set the appropriate capabilities flag on your session so other will know about it... and no, it won't stay that way....


I still cant get sound  and noone can hear me.. I think connection nat router prblem still exist


Title: Audio/Video conversation
Post by: kakaroto on December 02, 2008, 09:33:42 pm
lordamus, if status logs doesn't say anything about "unable to connect RTP component", then it's not the connection...
maybe it's because of that volume stuff I've been working on...  make sure to put the volume to 100% on both receiving and sending side once it's connected... (from the chatwindow).. for some reason it gets set to 0...


Title: Audio/Video conversation
Post by: lordamus on December 04, 2008, 02:30:12 pm
yea it sets to 0 and I cant slide to higher it doesnt move right...


Title: Audio/Video conversation
Post by: kakaroto on December 05, 2008, 12:04:58 am
yeah.. that should be fixed sometime soon... in the meantime, you can 'fix' it, by opening amsn's console and type :
Code:
::ChatWindow::UpdateVoipControls email_of_your_contact

it will then reactive them...


Title: Audio/Video conversation
Post by: Kalinda on December 06, 2008, 09:57:08 pm
Right, so I compiled libnice, farsight2 and installed the other packages (in Kubuntu Intrepid) just fine. The Configure sees Farsight and it compiles just fine. I was surprised how well it all went.

However, after aMSN opens fine, attempting to log in makes it hang, with the terminal giving me the following error:

Code:
(<unknown>:7278): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'


Dunno what the deal is, but I checked and I have all the required versions, some of them newer, of the gstreamer stuff as listed on the wiki.

Have the latest SVN.

All help appreciated :)


Title: Audio/Video conversation
Post by: kakaroto on December 07, 2008, 05:14:18 am
Hi Kalinda, long time no see...
well, it's good to hear that it all went well for you... that warning on the terminal is perfectly normal... amsn hanging is not normal though.. but I'm sure that it's all working correctly.. maybe you had some connection problems? either way, farsight would not block amsn for any reason... you could try however to see if farsight works from the audio/video assistant... if it does indeed freeze when checking for farsight, then try using gdb, once it freezes, press Ctrl-C, then type :
'thread apply all bt' (or a combination of the first 3 words.. can't remember their order...)
paste it here.


Title: Audio/Video conversation
Post by: Kalinda on December 07, 2008, 07:54:31 pm
Okay, I went through the audio/video assistant (I'm impressed how well it works with the mic now :)) and when I got to the step after it checks the mic, it hangs, with the terminal outputting the following:

Code:
Farsight : Preparing                                                          
Farsight debug : CODECS ARE READY                                              

(<unknown>:8716): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'                        
Farsight debug : stun ip : 64.14.48.28 : 3478


So, I pressed ctrl+C and it terminated the program in the terminal. After that typing your command doesn't do anything (I tried a couple different combos). I looked and found the key combos for opening the wish console in aMSN (thought maybe that's what you wanted), but it doesn't do it after its hung.

Anyhow, when I did it the second time, it spat out this error at me, without Farsight telling me anything:

Code:

(<unknown>:9275): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'


So it keeps changing the number, I wonder if that means anything...


Title: Audio/Video conversation
Post by: kakaroto on December 08, 2008, 11:13:55 pm
Problem of Kalinda was fixed. She was using farsight2 0.0.5 which had a little bug in there... so if anyone has the same issue, do one of these two solutions :
1 - patch your farsight2 0.0.5 with this patch : http://git.collabora.co.uk/?p=user/tester/farsight2.git;a=commitdiff_plain;h=7eabd4fe29191feda4fe5d4158dd4f1c256f8850
2 - use farsight2 0.0.4 instead...


Title: Audio/Video conversation
Post by: auris on December 11, 2008, 04:28:31 pm
I followed your solutions about farsight2, but I always get Kalinda's terminal output.
I used farsight 0.0.4 as the patched version 0.0.5, but I don't try to solve the problem (the mic bar is at zero level and so my friends don't hear my voice).
The Farsight extension is correctly loaded, the mic test is ok (I hear my voice at the mic).
The problem appeared  when it was introduced the level_IN window. Before that, the audio conversation went well.


Title: Audio/Video conversation
Post by: kakaroto on December 11, 2008, 05:41:52 pm
hi auris, welcome to the forums.
Make sure you update your SVN version.. the level_IN windows basically show the level of your microphone.. when adding that, I also added support for changing the volume of the call.. but I had a little bug that made the volume 0.0 on sending and receiving.. so you have to change the volume back to 100%... (for your own volume, resize the lower pane of the chat window to see the controls). Anyways, with the latest SVN version, I had fixed the bug, so it should work since the volume is left at 100%. Just update and see for yourself.
note that the test of the audio in the audio/video assistant is unrelated to farsight... it's libsnack used for voice clips, not farsight.
thanks!


Title: Audio/Video conversation
Post by: auris on December 11, 2008, 06:53:20 pm
Hi kakaroto, thanks for your greetings.
The panel on the right side of the chat window shows me the volume 0.0 and it's impossible to turn up the volume.
I updated amsn to SVN 10796, but with no luck.
With this version, I added a new problem: my webcam, using gspca module, is not recognized by amsn. I followed the solution about libv4l, but I got the same error.
Output:
Code:
aurelio@linux:~> amsn

(<unknown>:15188): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
libv4l2: error getting capabilities: Invalid argument
libv4l2: error getting capabilities: Invalid argument
libv4l2: error getting capabilities: Invalid argument
libv4l2: error getting capabilities: Invalid argument
Farsight : Preparing
Farsight debug : CODECS ARE READY

(<unknown>:15188): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x8cdddb0 - 2
Farsight debug : CANDIDATES ARE PREPARED
Farsight : Farsight is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 102 8000 0-16}
local candidates : {BM/NwKMjGfIs20TqZRRJJULqXlwJYqGLHb4gzYm29rg= 1 R7ahmPaa0dAL1o+Tj9EREQ== UDP 0.83 192.168.1.2 45769} {TyHmoo2xIW9dw1K5/jhNAZjqRPM0KAPcQL6ap1equcg= 1 ZtQNCzXlhFMylRui5PMLMA== UDP 0.55 82.50.49.131 45769} {Q4Vf0wppedxmstzJstel89N51KnY9NXCxHiQ+L6eRcQ= 1 Nb/t/rVrwAZ8yrw3xFm9dg== UDP 0.45 207.46.112.165 33436} {BM/NwKMjGfIs20TqZRRJJULqXlwJYqGLHb4gzYm29rg= 2 R7ahmPaa0dAL1o+Tj9EREQ== UDP 0.83 192.168.1.2 36747} {TyHmoo2xIW9dw1K5/jhNAZjqRPM0KAPcQL6ap1equcg= 2 ZtQNCzXlhFMylRui5PMLMA== UDP 0.55 82.50.49.131 36747} {Q4Vf0wppedxmstzJstel89N51KnY9NXCxHiQ+L6eRcQ= 2 Nb/t/rVrwAZ8yrw3xFm9dg== UDP 0.45 207.46.112.165 33834}

I get this error message when I try to turn up the volume:
Code:

(<unknown>:15188): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:15188): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

libv4l version is 0.5.0


Title: Audio/Video conversation
Post by: auris on December 11, 2008, 07:21:59 pm
I updated the libv4l sources to version 0.5.7: my webcam is not recognized by amsn.
 :(
Information about my webcam:
Code:

aurelio@linux:~> v4l-info /dev/video0

### video4linux device info [/dev/video0] ###
general info
    VIDIOCGCAP
name                    : "Creative PC-CAM 600"
type                    : 0x1 [CAPTURE]
channels                : 1
audios                  : 0
maxwidth                : 640
maxheight               : 480
minwidth                : 176
minheight               : 144

channels
    VIDIOCGCHAN(0)
channel                 : 0
name                    : "SPCA504C"
tuners                  : 0
flags                   : 0x0 []
type                    : CAMERA
norm                    : 0

tuner
ioctl VIDIOCGTUNER: Invalid argument

audio
ioctl VIDIOCGAUDIO: Invalid argument

picture
    VIDIOCGPICT
brightness              : 32768
hue                     : 0
colour                  : 4096
contrast                : 8192
whiteness               : 0
depth                   : 24
palette                 : RGB24

buffer
    VIDIOCGFBUF
base                    : (nil)
height                  : 0
width                   : 0
depth                   : 0
bytesperline            : 0

window
    VIDIOCGWIN
x                       : 0
y                       : 0
width                   : 320
height                  : 240
chromakey               : 0
flags                   : 0

amsn only recognizes my analog TV card:
Code:

### v4l2 device info [/dev/video1] ###
general info
    VIDIOC_QUERYCAP
driver                  : "saa7134"
card                    : "Terratec Cinergy 400 TV"
bus_info                : "PCI:0000:04:02.0"
version                 : 0.2.14
capabilities            : 0x5010015 [VIDEO_CAPTURE,VIDEO_OVERLAY,VBI_CAPTURE,TUNER,READWRITE,STREAMING]

standards
    VIDIOC_ENUMSTD(0)
index                   : 0
id                      : 0xff [PAL_B,PAL_B1,PAL_G,PAL_H,PAL_I,PAL_D,PAL_D1,PAL_K]
name                    : "PAL"
frameperiod.numerator   : 1
frameperiod.denominator : 25
framelines              : 625
    VIDIOC_ENUMSTD(1)
index                   : 1
id                      : 0x100 [PAL_M]
name                    : "PAL-M"
frameperiod.numerator   : 1001
frameperiod.denominator : 30000
framelines              : 525
    VIDIOC_ENUMSTD(2)
index                   : 2
id                      : 0x200 [PAL_N]
name                    : "PAL-N"
frameperiod.numerator   : 1
frameperiod.denominator : 25
framelines              : 625
    VIDIOC_ENUMSTD(3)
index                   : 3
id                      : 0x400 [PAL_Nc]
name                    : "PAL-Nc"
frameperiod.numerator   : 1
frameperiod.denominator : 25
framelines              : 625
    VIDIOC_ENUMSTD(4)
index                   : 4
id                      : 0x800 [PAL_60]
name                    : "PAL-60"
frameperiod.numerator   : 1001
frameperiod.denominator : 30000
framelines              : 525
    VIDIOC_ENUMSTD(5)
index                   : 5
id                      : 0xb000 [NTSC_M,NTSC_M_JP,?]
name                    : "NTSC"
frameperiod.numerator   : 1001
frameperiod.denominator : 30000
framelines              : 525
    VIDIOC_ENUMSTD(6)
index                   : 6
id                      : 0xff0000 [SECAM_B,SECAM_D,SECAM_G,SECAM_H,SECAM_K,SECAM_K1,SECAM_L,?ATSC_8_VSB]
name                    : "SECAM"
frameperiod.numerator   : 1
frameperiod.denominator : 25
framelines              : 625
......


Title: Audio/Video conversation
Post by: auris on December 11, 2008, 07:44:23 pm
Using amsn SVN 10784, my webcam is recognized (libv4l 0.5.7, as before with libv4l 0.5.0).
This is my amsn log:
Code:

aurelio@linux:~> amsn

(<unknown>:20816): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
Your IN volume is at 0.013888698071241379
Farsight : Preparing
Farsight debug : CODECS ARE READY

(<unknown>:20816): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x8ccbeb0 - 2
Your IN volume is at 0.013169287703931332
Farsight debug : CANDIDATES ARE PREPARED
Farsight : Farsight is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 102 8000 0-16}
local candidates : {wQVJGUDT/jxEaz682r+XxQWXVuCFkLxyyswByaLnp2A= 1 92GToJTaqbRMaVP1dCtPXg== UDP 0.83 192.168.1.2 35093} {5loHobS9sfpkZDoC8MWkAG3Ga2iuH+riSZ/IQeIIxnw= 1 6KJ7/uNEXuZKj02qin6g7g== UDP 0.55 82.50.49.131 35093} {uZljV2xPJe9NykN+9lwzVDJMDDtHIXepmeJNcHdYHuM= 1 hcb89+De63LKcvRmBr8VfQ== UDP 0.45 207.46.112.205 41902} {wQVJGUDT/jxEaz682r+XxQWXVuCFkLxyyswByaLnp2A= 2 92GToJTaqbRMaVP1dCtPXg== UDP 0.83 192.168.1.2 53622} {5loHobS9sfpkZDoC8MWkAG3Ga2iuH+riSZ/IQeIIxnw= 2 6KJ7/uNEXuZKj02qin6g7g== UDP 0.55 82.50.49.131 53622} {uZljV2xPJe9NykN+9lwzVDJMDDtHIXepmeJNcHdYHuM= 2 hcb89+De63LKcvRmBr8VfQ== UDP 0.45 207.46.112.205 41946}

The old version of amsn SVN seems compatible with v4l and v4l2 devices.


Title: Audio/Video conversation
Post by: kakaroto on December 11, 2008, 10:42:58 pm
@auris :
ok, first, the issue with your webcam is unrelated to this thread... because this is only about the audio conference, with no webcam. so please keep your messages on the appropriate threads. To answer you though, yes, we know, libv4l fails for some devices, we're working on it.
About your audio problem, well, the sound should be at 100%, make sure your microphone is working, that it's recording correctly from the correct device, and that your gnome configuration is set correctly to capture from the microphone.
Resize the lower pane of the chat window (where you type your text) to make sure you see the volume sliders (there are two, one for you and one for your friend), and make sure the volume is set correctly for YOUR side... yell in the microphone and see if the level_IN moves...


Title: Audio/Video conversation
Post by: auris on December 12, 2008, 02:40:32 pm
I set the microphone volume at max in GNOME sound mixer and I made a voice capture test from the microphone with audacity. No problem. Opening amsn the level_IN window shows me the volume at 8% (originally it was at 1%).
I'll test as soon as possible the audio conference. I've never seen the volume sliders in the lower panel of the chat window whether I have to be sincere.


Title: Audio/Video conversation
Post by: kakaroto on December 12, 2008, 09:07:40 pm
ok cool.. well, when you start an audio call, you'll see microphone volume/mute checkbox in the chatwindow (below your display picture and below, and below your contact's display picture).
the level_IN at startup is just when amsn checks if farsight is working...
if you saw it move from 1 to 8%, then it was recording some sound, which is cool... it should all work fine for you.


Title: Audio/Video conversation
Post by: auris on December 13, 2008, 12:57:17 pm
My friend hears me badly and the voice is really noisy. The microphone volume/mute checkbox in the chat window is not changeable (for example, it's impossible to turn up the microphone volume).
I'm boring, but before updating amsn SVN, my audio conversation went well. I could set the microphone volume at 70% in GNOME mixer and my friend heard me perfectly. This problem concerns both farsight2 0.0.4 and farsight 0.0.5 (the last one was patched as you suggested).
 :(


Title: Audio/Video conversation
Post by: kakaroto on December 14, 2008, 09:26:46 pm
humm... looks like we had some server issues yesterday and my answer to auris got deleted somehow.. here it is again :
Quote
humm.. usually bad voice is caused by a too high volume... the volume problems not being changeable are "going to be fixed"... but your own volume settings should always work (well, you do need the 'volume' element for gstreamer...)
Try lowering the volume from the gnome mixer until he hears you correctly...

oh and one thing.. please use the appropriate thread, or create a new thread for any question unrelated to the audio conversation/farsight2/gstreamer/SIP/libnice... thx

[edit by billiob: split ipv6 stuff to its own thread]


Title: Audio/Video conversation
Post by: auris on December 14, 2008, 10:01:43 pm
My volume settings always work in GNOME mixer. Thanks for your support.


Title: Audio/Video conversation
Post by: kakaroto on December 16, 2008, 02:11:32 am
auris, could you be more precise ?
1 - does the volume sliders in the chat window work for you (the output volume may not work, i know that, but what about your microphone volume sliders?).. and if not, then do you have the gstreamer volume elemnt (in a terminal, type 'gst-inspect volume')
2 - did lowering the volume from the gnome mixer fix the 'bad voice' issue you were having?


Title: Audio/Video conversation
Post by: auris on December 16, 2008, 07:23:05 pm
1)
Code:
aurelio@linux:~> gst-inspect volume
Factory Details:
  Long name: Volume
  Class: Filter/Effect/Audio
  Description: Set volume on audio/raw streams
  Author(s): Andy Wingo <wingo@pobox.com>
  Rank: none (0)

Plugin Details:
  Name: volume
  Description: plugin for controlling audio volume
  Filename: /usr/lib/gstreamer-0.10/libgstvolume.so
  Version: 0.10.21
  License: LGPL
  Source module: gst-plugins-base
  Binary package: GStreamer Base Plug-ins source release
  Origin URL: Unknown package origin

GObject
 +----GstObject
       +----GstElement
             +----GstBaseTransform
                   +----GstAudioFilter
                         +----GstVolume

Interfacce implementate:
  GstImplementsInterface
  GstMixer

Pad Templates:
  SRC template: 'src'
    Availability: Always
    Capabilities:
      audio/x-raw-float
                   rate: [ 1, 2147483647 ]
               channels: [ 1, 2147483647 ]
             endianness: 1234
                  width: { 32, 64 }
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 8
                  depth: 8
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 16
                  depth: 16
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 24
                  depth: 24
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 32
                  depth: 32
                 signed: true

  SINK template: 'sink'
    Availability: Always
    Capabilities:
      audio/x-raw-float
                   rate: [ 1, 2147483647 ]
               channels: [ 1, 2147483647 ]
             endianness: 1234
                  width: { 32, 64 }
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 8
                  depth: 8
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 16
                  depth: 16
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 24
                  depth: 24
                 signed: true
      audio/x-raw-int
               channels: [ 1, 2147483647 ]
                   rate: [ 1, 2147483647 ]
             endianness: 1234
                  width: 32
                  depth: 32
                 signed: true


Element Flags:
  no flags set

Element Implementation:
  Has change_state() function: gst_audio_filter_change_state
  Has custom save_thyself() function: gst_element_save_thyself
  Has custom restore_thyself() function: gst_element_restore_thyself

Element has no clocking capabilities.
Element has no indexing capabilities.
Element has no URI handling capabilities.

Pads:
  SRC: 'src'
    Implementation:
      Has getrangefunc(): gst_base_transform_getrange
      Has custom eventfunc(): gst_base_transform_src_event
    Pad Template: 'src'
  SINK: 'sink'
    Implementation:
      Has chainfunc(): gst_base_transform_chain
      Has custom eventfunc(): gst_base_transform_sink_event
      Has bufferallocfunc(): gst_base_transform_buffer_alloc
    Pad Template: 'sink'

Element Properties:
  name                : The name of the object
                        flags: leggibile, scrivibile
                        String. Default: null Current: "volume0"
  qos                 : Handle Quality-of-Service events
                        flags: leggibile, scrivibile
                        Boolean. Default: false Current: false
  mute                : mute channel
                        flags: leggibile, scrivibile, controllabile
                        Boolean. Default: false Current: false
  volume              : volume factor
                        flags: leggibile, scrivibile, controllabile
                        Double. Range:               0 -              10 Default:               1 Current:               1

My microphone volume sliders in aMSN stay at zero during the audio conversation, so they don't work for me.
2) My voice is clear after lowering the volume in GNOME mixer, but only using application as audacity or vlc. In aMSN my friends don't hear me, setting the volume at 70%. They only heard me when I set the volume in GNOME mixer at max, but my voice is really bad. (bad for my friend who hears me!)


Title: Audio/Video conversation
Post by: auris on December 17, 2008, 03:19:01 pm
Today I've updated gst-plugins-farsight to 0.12.10 version. I'll test it with audio conference.


Title: Audio/Video conversation
Post by: auris on December 19, 2008, 08:46:55 pm
After updating the gst-plugins-farsight sources to 0.12.10 version, my audio conversations go well. My voice is clear and I can setup the microphone volume in GNOME mixer about at 70%.
All's well that ends well!  :D


Title: Audio/Video conversation
Post by: kakaroto on December 20, 2008, 10:12:43 pm
oh cool! It was because of the fsvalve bug that renegociated caps on every packet... that caused latency for the audio... ok cool! so you heard him everyone! If anyone is having some problems with his audio quality.. update to gst-plugins-farsight 0.12.10
Thanks for thinking about it and doing it, auris! :)


Title: Audio/Video conversation
Post by: dpetka2001 on December 23, 2008, 01:54:20 am
hi there i get the following problem when trying to launch amsn with farsight2 support...
Code:
Farsight : Preparing
Farsight debug : CODECS ARE READY
Farsight debug : Creating source : (null)  --- (null) -- (null)
Farsight Prepare error : Couldn't create audio source
could anyone give some help regarding this issue? thanks in advance...


Title: Audio/Video conversation
Post by: kakaroto on December 23, 2008, 04:11:04 am
you have no audio source...
make sure you have either the oss or the alsa gstreamer elements installed... there might be a gstreamer0.10-alsa package in your distribution's repository.... try that!

p.s: you could also paste here the output of
Code:
::Farsight::Probe
in amsn's console (ctrl-shift-C), it will list all the audio source elements that you have installed on your system...


Title: Audio/Video conversation
Post by: dpetka2001 on December 23, 2008, 09:03:34 pm
ok i needed a package in my distribution gst-plugins-alsa...now it seems to work fine but i would like to ask something that seems weird to me...first of all the output from amsn console
Code:
(amsn) 1 % ::Farsight::Probe
{source alsasrc {Audio source (ALSA)} {Read from a sound card via ALSA} {hw:0,0 hw:0,2}} {source dtmfsrc {DTMF tone generator} {Generates DTMF tones}} {source audiotestsrc {Audio test source} {Creates audio test signals of given frequency and volume}} {sink alsasink {Audio sink (ALSA)} {Output to a sound card via ALSA} {hw:0,0 hw:0,1}} {sink autoaudiosink {Auto audio sink} {Wrapper audio sink for automatically detected audio sink}}
next when i launch amsn as my user
Code:
Farsight : Preparing
Farsight debug : CODECS ARE READY
Farsight debug : Creating source : (null)  --- (null) -- (null)
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x8a72040 - 2
Farsight debug : CANDIDATES ARE PREPARED
Farsight : Farsight is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 99 8000 0-16}
local candidates : {ggPTaV6ofnzmzWuEkq3WThOdYePl8BXT7hg5xmcxUSY= 1 atbsiziACp8wTEnlwIXWHQ== UDP 0.83 192.168.1.24 48193} {xqAT8/wIqVlTJq9tFKm65/8ki28/cFpyLxLz66QP0sk= 1 Z534vrou3axH4kNoMvIkbw== UDP 0.55 94.68.151.45 12678} {IqR2yva++TY0ZyZvedAD8vb6oytggPA7fAV0YDlWAac= 1 NnDFIB+v7WHqCwUzywGnuA== UDP 0.45 65.55.255.35 49136} {ggPTaV6ofnzmzWuEkq3WThOdYePl8BXT7hg5xmcxUSY= 2 atbsiziACp8wTEnlwIXWHQ== UDP 0.83 192.168.1.24 54699} {xqAT8/wIqVlTJq9tFKm65/8ki28/cFpyLxLz66QP0sk= 2 Z534vrou3axH4kNoMvIkbw== UDP 0.55 94.68.151.45 12679} {IqR2yva++TY0ZyZvedAD8vb6oytggPA7fAV0YDlWAac= 2 NnDFIB+v7WHqCwUzywGnuA== UDP 0.45 65.55.255.35 32681}
but it also pops up a window about level_IN...see picture (http://img395.imageshack.us/my.php?image=top12tm2.png) and it won't go away or show any progress...is this a problem? should i do anything else? thanks...


Title: Audio/Video conversation
Post by: dpetka2001 on December 24, 2008, 08:53:57 pm
well i tried to make a call with a friend of mine who uses windows live messenger and though it shows that the connection is established there is no audio from either sides...how could i tell what is wrong? any help would be appreciated...


Title: Audio/Video conversation
Post by: kakaroto on December 24, 2008, 10:45:38 pm
Hi dpetka2001, it looks fine to me. Probe shows that you have alsa installed and that it works, and the assistant shows farsight working correctly for you.. the level_IN window is just temporary (just like it says in the text box). If you wait long enough for farsight to finish getting tested, then you can close it...
During your call, make sure the volume is set correctly and that your microphone is working.. try talking in the microphone and you should see the bar in the level_IN window move.. when the other person is talking to you, the level_OUT bar should move too.
We'll soon add the possibility to configure which microphone to use (you seem to have two inputs in alsa), maybe that's your problem...
In the meantime, you can set it by typing this in amsn's console :
Code:
::Farsight::Config -source-device "hw:0,2"

(or "hw:0,0")
hope that helps...


Title: Audio/Video conversation
Post by: dpetka2001 on December 25, 2008, 12:37:24 am
excuse me but i forgot to paste the error amsn exited after the unsuccessful voice call i tried to make...here it is
Code:
invalid command name "::MSNSIP::SIPConnection11"
    while executing
"$sip Bye $callid"
    (procedure "::MSNSIP::HangUp" line 3)
    invoked from within
"::MSNSIP::HangUp $sip $callid"
    (procedure "::amsn::HangupSIPCall" line 6)
    invoked from within
"::amsn::HangupSIPCall kiki-xxx@sad-amsn-user.com ::MSNSIP::SIPConnection11 0346b276c46d27bc76898b5e0b84a549"
    invoked from within
".container_2.msg_2.f.bottom.left.buttons.inner.call invoke"
    ("uplevel" body line 1)
    invoked from within
"uplevel #0 [list $w invoke]"
    (procedure "tk::ButtonUp" line 22)
    invoked from within
"tk::ButtonUp .container_2.msg_2.f.bottom.left.buttons.inner.call"
    (command bound to event)
as far the level_IN window is concerned i left for about half an hour without doing anything and nothing happened...still it was stuck at 1% as provided in the picture in my previous post...so i don't really know where it stucks...will try tomorrow when i will be able to talk to my friend the command you suggested for the amsn console...thanks in advance...


Title: Audio/Video conversation
Post by: kakaroto on December 25, 2008, 04:38:16 am
Hi,
that error can happen if the call got canceled too fast or something like that.. it will be fixed before we release 0.98, just ignore it, and retry...
About the window being stuck at 1%, it's just because when you login, it will test if farsight is correctly configured, and when it is, you'll see that window, then once it finds farsight working, it will close everything, but it won't close that window.. of course it will be stuck since it gets no more input from your microphone because it closed it.. so just forget about it and close it... during the call, as long as the call is working, the level_IN should be there and should change with the input of your microphone..
anyways, as I said, don't pay attention to that window.


Title: Audio/Video conversation
Post by: dpetka2001 on December 26, 2008, 06:11:00 pm
well i tried again to talk to my friend without success...i start the call,the other end answers but after that there is no sound...also when i try to talk the level_IN flunctuates between 1% and 2%...but after that a windows pops up saying "Your audio call has ended"...take a look at the picture (http://img165.imageshack.us/my.php?image=top13fv2.png)...after that the level_IN stays stuck at 1%...at no point however is there a sound during the communication...i tried with both "hw:0,2" and "hw:0,0" without success...with "hw:0,0" it doesn't even start the call and shows the other end always busy...with "hw:0,2" happens what i've already told you...


Title: Audio/Video conversation
Post by: dpetka2001 on December 29, 2008, 11:15:05 pm
is this normal what i described in my previous post? to pop up an unknown conversation window saying my conversation has ended like in the picture i attached?


Title: Audio/Video conversation
Post by: vivia on December 29, 2008, 11:19:38 pm
Sorry, I cannot see your picture...

Have you tried raising the volume for your microphone? Make sure it's not too low


Title: Audio/Video conversation
Post by: kakaroto on December 29, 2008, 11:38:36 pm
dpetka2001: yes, that unknown conversation window is a bug, but it shouldn't affect your call.. the reason why it seems to end the call automatically is probably because it couldn't connect... could you paste here the output of the status log (Ctrl-S) ? That would be helpful!
also, what type of network do you have ? what router ? what configuration? do you have UDP blocked ?


Title: Audio/Video conversation
Post by: zyazhou on January 20, 2009, 10:07:54 am
Anyone tried the lastest MSN client v2009. I tried, but the audio call failed.


Title: Audio/Video conversation
Post by: kakaroto on January 21, 2009, 01:47:50 am
i haven't, but I will as soon as I get back home (currently in Australia http://linux.conf.au )


Title: Audio/Video conversation
Post by: zyazhou on January 21, 2009, 05:20:59 pm
MSN v2009 uses MSNP18, and it seems the protocol for audio has changed a lot. I'm not sure, just caught some packets with wireshark.


Title: Audio/Video conversation
Post by: kakaroto on January 21, 2009, 11:54:33 pm
yeah.. it did change.. but I don't know to what extent.. i'll look at it next week.. all I know is that when I tried to send an audio call to someone using WLM2009, he received the message :
Quote
KaKaRoTo is using a version of Messenger that does not have improvements to support a Computer Call.  Please ask your contact to upgrade to the latest version of Messenger and try your call again


so... I'll have to see if they changed the whole damn thing, or if it's just some minor changes that can be fixed...


Title: Audio/Video conversation
Post by: peter on January 26, 2009, 03:02:33 am
Hi there.

Even with the latest svn 10880 I'm having the same error regarding farsight config.

Well I do have /usr/lib/gstreamer-0.10 and /usr/lib/gstreamer-0.8.

I have openssl 0.9.8 ( libssl.so.0.9.7)


Quote


compile time options summary
============================

X11 : yes
Tcl : 8.4
TK : 8.4
DEBUG : no
STATIC : no
FARSIGHT : yes
LIBV4L : no

[pedro@localhost msnCopia]$ make
CC utils/farsight/src/tcl_farsight.o
LD utils/farsight/src/tcl_farsight.so
cp utils/farsight/src/tcl_farsight.so utils/farsight/tcl_farsight.so
[pedro@localhost msnCopia]$ ./amsn
Farsight : Preparing

(<unknown>:6323): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstice.so': libssl.so.0.9.7: cannot open shared object file: No such file or directory

(<unknown>:6323): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstvolume.so': /usr/lib/gstreamer-0.10/libgstvolume.so: undefined symbol: oil_function_class_ptr_scalarmultiply_f64_ns

(<unknown>:6316): GLib-GObject-CRITICAL **: g_type_class_unref: assertion `g_class != NULL' failed
Farsight Prepare error : Error while creating new session (1): Could not create GstRtpBin  



Quote

gst-inspect-0.10 nice

(gst-inspect-0.10:4653): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstice.so': libssl.so.0.9.7: cannot open shared object file: No such file or directory

(gst-inspect-0.10:4653): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstvolume.so': /usr/lib/gstreamer-0.10/libgstvolume.so: undefined symbol: oil_function_class_ptr_scalarmultiply_f64_ns
Plugin Details:
  Name:                 nice
  Description:          Interactive UDP connectivity establishment
  Filename:             /usr/lib/gstreamer-0.10/libgstnice.so
  Version:              0.0.4
  License:              LGPL
  Source module:        nice
  Binary package:       nice
  Origin URL:           http://telepathy.freedesktop.org/wiki/

  nicesink: ICE sink
  nicesrc: ICE source

  2 features:
  +-- 2 elements



Quote

gst-inspect-0.10 gstrtpbin

(gst-inspect-0.10:4661): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstice.so': libssl.so.0.9.7: cannot open shared object file: No such file or directory

(gst-inspect-0.10:4661): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstvolume.so': /usr/lib/gstreamer-0.10/libgstvolume.so: undefined symbol: oil_function_class_ptr_scalarmultiply_f64_ns
No such element or plugin 'gstrtpbin'



gstreamer-tools 0.10.21 and 0.8.12

gstreamer0.10-farsight 0.10.0.1


Sorry for posting in the other forum  :oops:


Title: Audio/Video conversation
Post by: peter on January 28, 2009, 12:29:23 am
Well I have to say that farsight is now working... (compile time... :P ), apart from creating source (null), but that's because right now I don't have permission to write to dsp :?. Nevertheless I can't use another account as it gets stuck after all that local codecs (...).

I still have to try amsn<->wlm call .

Svn 10885

Quote

[23:20:56] Farsight : Preparing
[23:20:56] Farsight : Preparing
[23:20:56] Farsight debug : CODECS ARE READY
[23:20:56] Farsight debug : CODECS ARE READY
[23:20:56] Farsight debug : Creating source : (null)  --- (null) -- (null)
[23:20:56] Farsight debug : Creating source : (null)  --- (null) -- (null)
[23:20:56] Farsight debug : Using source gconfaudiosrc
[23:20:56] Farsight debug : Using source gconfaudiosrc
[23:20:56] Farsight debug : stun ip : 64.14.48.28 : 3478
[23:20:56] Farsight debug : stun ip : 64.14.48.28 : 3478
[23:20:56] logging in, destroying loginscreen : loggingIn
[23:20:57] Farsight debug : CANDIDATES ARE PREPARED
[23:20:57] Farsight debug : CANDIDATES ARE PREPARED
[23:20:57] Farsight : Farsight is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 98 8000 0-16}
local candidates : {DWLgGz+tLMCoUmYk/Hl49FVzduNeOlzHr612Pqv+2c4= 1 Sk/2w6vT025FBSwb58wUvw== UDP 0.83 192.168.1.64 32805} {tpXgAXaDvOIbNGZzVmdngXHDsP5L8F7sH9eN9U2pEz4= 1 6U0qqWWRSPQdph21177ndg== UDP 0.55 85.241.142.80 50472} {DWLgGz+tLMCoUmYk/Hl49FVzduNeOlzHr612Pqv+2c4= 2 Sk/2w6vT025FBSwb58wUvw== UDP 0.83 192.168.1.64 32806} {tpXgAXaDvOIbNGZzVmdngXHDsP5L8F7sH9eN9U2pEz4= 2 6U0qqWWRSPQdph21177ndg== UDP 0.55 85.241.142.80 50473}
[23:20:57] Farsight : Farsight is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 98 8000 0-16}
local candidates : {DWLgGz+tLMCoUmYk/Hl49FVzduNeOlzHr612Pqv+2c4= 1 Sk/2w6vT025FBSwb58wUvw== UDP 0.83 192.168.1.64 32805} {tpXgAXaDvOIbNGZzVmdngXHDsP5L8F7sH9eN9U2pEz4= 1 6U0qqWWRSPQdph21177ndg== UDP 0.55 85.241.142.80 50472} {DWLgGz+tLMCoUmYk/Hl49FVzduNeOlzHr612Pqv+2c4= 2 Sk/2w6vT025FBSwb58wUvw== UDP 0.83 192.168.1.64 32806} {tpXgAXaDvOIbNGZzVmdngXHDsP5L8F7sH9eN9U2pEz4= 2 6U0qqWWRSPQdph21177ndg== UDP 0.55 MyIP 50473}


Title: Audio/Video conversation
Post by: kakaroto on January 28, 2009, 01:40:10 am
ok cool, well it works.. that 'creating source (null)' is just a debug that shows if you manually configured farsight to use a specific device or pipeline (like read from a file instead of using a microphone, etc...).. there's no UI for this, so you can't specify it yet, so the creating source will always show (null).. and since it's NULL, it will automatically try to detect and use any audio device it finds... also note that the latest SVN version *should* (not really tested) work even if you have no microphone, it will just become a receive only session..


Title: Audio/Video conversation
Post by: kakaroto on January 28, 2009, 01:40:32 am
ok cool, well it works.. that 'creating source (null)' is just a debug that shows if you manually configured farsight to use a specific device or pipeline (like read from a file instead of using a microphone, etc...).. there's no UI for this, so you can't specify it yet, so the creating source will always show (null).. and since it's NULL, it will automatically try to detect and use any audio device it finds... also note that the latest SVN version *should* (not really tested) work even if you have no microphone, it will just become a receive only session..


Title: Audio/Video conversation
Post by: kakaroto on February 10, 2009, 03:03:19 am
Hi,
I wanted to post a bit of news to everyone who's interested in the Audio/Video conversation...
- the audio call may or may not work with WLM2009 users right now, sometimes it complains that the other contact should upgrade, sometimes it doesn't...
- I have just recently (today) made aMSN audio call work with WLM2009 using the newest tunneled SIP (SIP messages go through the NS using UUN commands, instead of using an external SIP server), and using the latest ICE-19 draft! I had to modify a bit libnice to make it work because they did some non-standard stuff in there (I'm guessing it's a bug they have), so if you want the latest WLM2009 audio call compatibility (and assuming you are using MSNP16), then you should update your SVN, recompile it, AND get the latest GIT of libnice, and recompile farsight2 too..
- I have also found that WLM2009 allows, when using MSNP16+ to do a video call using SIP.. the codec will be H263, and it would also be using libnice, farsight2 and gstreamer.. this means that yeay, we can finally have a real bidirectional audio+video call working AND it is not that crappy older protocol anymore.. it will be RTP, so it will be really nice for a change :)
More news soon, for now I'm still working on this, but hopefully, we'll soon be able to do audio and video calls correctly using either ICE6 or ICE19 specs...
See ya!


Title: Audio/Video conversation
Post by: Montblanc on February 11, 2009, 12:51:09 pm
Great, kakaroto! Simply, GREAT!
Just one thing: you forgot to say your git repo isn't nice.git anymore, but libnice.git:
Code:
git clone git://git.collabora.co.uk/git/user/kakaroto/libnice.git libnice

I'll look for someone using WLM 2009 and let you know!  :wink:


Title: Audio/Video conversation
Post by: kakaroto on February 11, 2009, 03:48:42 pm
humm.. yeah, right, the repo changed name, true. but you're not going to use git now, right... wait until I fix it all correctly, then make a release! :p


Title: Audio/Video conversation
Post by: Montblanc on February 11, 2009, 06:46:42 pm
Quote from: "kakaroto"
humm.. yeah, right, the repo changed name, true. but you're not going to use git now, right... wait until I fix it all correctly, then make a release! :p


Yeah, you're right. After compiling your libnice from git and farsight2 0.0.4 I noticed that wish was sucking ~60% off from my cpu... I don't know if it's due to libnice, I should remove it and see if it comes back to normal.


Title: Audio/Video conversation
Post by: kakaroto on February 11, 2009, 06:51:20 pm
probably not caused by it


Title: Audio/Video conversation
Post by: Montblanc on February 11, 2009, 07:27:20 pm
Quote from: "kakaroto"
probably not caused by it


Right guess, I reverted to nice 0.0.4 and it's still happening. aMSN is really sucking too much resources from 2 or 3 revisions, should I open a topic about this?


Title: Audio/Video conversation
Post by: vivia on February 11, 2009, 10:18:53 pm
Yes, open a new topic and report it. Make sure you find the exact revision number and give details about the files changed. :)


Title: Audio/Video conversation
Post by: kakaroto on February 12, 2009, 03:06:50 am
and just in case.. without changing anything.. unplug your webcam and see if it still happens...


Title: Audio/Video conversation
Post by: Montblanc on February 12, 2009, 12:44:20 pm
Unplugging my webcam makes aMSN use about 20% less cpu, but it still shouldn't suck that much. Anyway, i opened a topic (http://www.amsn-project.net/forums/viewtopic.php?t=6301) and really hope it helps.


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on February 15, 2009, 11:36:42 am
Quote from: "kakaroto"

- I have also found that WLM2009 allows, when using MSNP16+ to do a video call using SIP.. the codec will be H263, and it would also be using libnice, farsight2 and gstreamer.. this means that yeay, we can finally have a real bidirectional audio+video call working AND it is not that crappy older protocol anymore.. it will be RTP, so it will be really nice for a change :)
More news soon, for now I'm still working on this, but hopefully, we'll soon be able to do audio and video calls correctly using either ICE6 or ICE19 specs...
See ya!

AWESOME! :D

I've been away for ages, so I haven't tracked amsn forum. But today I decided to check what's new... and it's great news!
Amazing work kakaroto!


Title: Audio/Video conversation
Post by: messengerfreak on February 17, 2009, 01:38:31 pm
Simply amazing - thank you so much for this, kakaroto!


Title: Audio/Video conversation
Post by: supernaicol on February 17, 2009, 03:54:18 pm
I'm sorry, I didn't read all the 35 pages of this topic, so I don't know if someone else said this yet, anyway... using ubuntu jaunty (next 9.04 release), all the building dependencies to have amsn-svn working correctly with farsight (ver. 0.0.7) can be satisfied installing the libgstfarsight0.10-dev, with no need to manually compile nice and farsight frameworks.
It would be nice if you could make this present in the wiki page of farsight.
Thanks for all. Bye


Title: Audio/Video conversation
Post by: zyazhou on February 18, 2009, 03:49:19 pm
kakaroto,

I have one question about authorization header of sip register. I can see from the source code, when encoding $auth, $options(-password) is from $ticket. I can't find where $ticket come from, how can I get $ticket? Thx!


Title: Audio/Video conversation
Post by: kakaroto on February 18, 2009, 06:44:24 pm
it's the MessengerSecure ticket received from the SSO SOAP authentication that you use to authenticate with the notification server in MSNP15+


Title: Audio/Video conversation
Post by: zyazhou on February 19, 2009, 02:54:03 pm
I checked the ticket of SSO and password of sip authentication, and found them different. A example is list below:

SSO ticket is   t=EwBgAswbAQAUs1/.......
sip authentication password is  t=EwDgATIiAQAUno9......


Title: Audio/Video conversation
Post by: kakaroto on February 19, 2009, 06:34:39 pm
Code:
       proc createSIP {callbk {host "vp.sip.messenger.msn.com"}} {
                $::sso RequireSecurityToken MessengerSecure [list ::MSNSIP::createSIPSSOCB $callbk $host]
        }

Like I said, you need the MessengerSecure ticket, you should know that there are many tickets that you can get from SSO, one for each server (contacts.msn.com, or passport.net or messenger, or messengersecure, etc...) and you need the MessengerSecure one (the one with the endpoint reference address "messengersecure.live.com").
anyways, for tunneled sip, you don't need to authenticate to a SIP server anyways...


Title: Audio/Video conversation
Post by: zyazhou on February 20, 2009, 09:28:03 am
Thank you, Kakaroto. I got more clear.


Title: Audio/Video conversation
Post by: Montblanc on March 06, 2009, 10:50:00 am
I saw you recently updated your libnice from git, when you need some bugtracking just knock!  :D


Title: Audio/Video conversation
Post by: kakaroto on March 06, 2009, 11:08:11 am
Hi,
Nice to see some people who are following on the latest changes in libnice! :)
Yes indeed, I have just released today a new version of libnice... more info here : http://lists.freedesktop.org/archives/nice/2009-March/000278.html
Notice that it adds support for WLM2009 ICE19 compatibility too, which allows with the recent SVN version to have an audio and video call with WLM (right now sending an audio call invite sends actually a video+audio invite, but only if you're using MSNP18... )
There are still some things to figure out and hopefully we'll finish the UI soon for audio conferences... but also have the option to do a video call with MSNP18+ too...
note however that the ffmpeg h263 encoder doesn't seem to encode the video correctly, so WLM can't decode our video.. we'll hopefully fix this soon with Ole Andre's mscodecs which actually uses msn's dlls directly (you would need the dll on your linux machine though...)


Title: Audio/Video conversation
Post by: Montblanc on March 06, 2009, 01:54:15 pm
Ok, I'll give it a try as soon as possible!


Title: Audio/Video conversation
Post by: setzer on March 06, 2009, 02:27:57 pm
wow, that's absolutely amazing! I'm kinda short on time right now, but I'll try to test this tonight!! Thanks for your hard work kakaroto!!


Title: Audio/Video conversation
Post by: kakaroto on March 06, 2009, 08:49:08 pm
you're welcome! :)
and thanks for your appreciation! :)


Title: Audio/Video conversation
Post by: flomar34 on March 07, 2009, 04:48:20 pm
Hello,

Very nice to have news about this :)

Im sure that it will not help you but ... maybe

I use the last svn (1075) with libnice 0.0.5
I do not have (yet) a web cam

If i want to make an audio call with a wlm 9 contact using MSNP18 amsn crash
Code:
(<unknown>:26565): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'

(<unknown>:26565): GLib-GObject-WARNING **: value "3" of type `guint' is invalid or out of range for property `compatibility-mode' of type `guint'

(<unknown>:26565): GLib-GObject-WARNING **: value "3" of type `guint' is invalid or out of range for property `compatibility-mode' of type `guint'
Erreur de segmentation


With MSNP15 all is ok

Thanks a lot for all the work you are making for us and sorry not to be able to help you more that i would like


Title: Audio/Video conversation
Post by: kakaroto on March 07, 2009, 10:24:37 pm
Hi,
thanks :)
after installing libnice 0.0.5, you'll need to recompile farsight2....
If it still segfaults, please give me the backtrace from gdb


Title: Audio/Video conversation
Post by: flomar34 on March 08, 2009, 08:32:06 am
Hello

After compiling farsight2 0.0.7 i have the same issue
For the backtrace i m not sure that the result was what you exepected
On dbg mode amsn don't crash but freezed

Code:
(<unknown>:30812): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
[New Thread 0xb4adeb90 (LWP 30863)]
[New Thread 0xb65d6b90 (LWP 30864)]
[Thread 0xb65d6b90 (LWP 30864) exited]
[Thread 0xb5be0b90 (LWP 30858) exited]
[New Thread 0xb5be0b90 (LWP 30865)]
[New Thread 0xb65d6b90 (LWP 30866)]
[New Thread 0xb42ddb90 (LWP 30867)]
[New Thread 0xb32dbb90 (LWP 30868)]
[New Thread 0xb2adab90 (LWP 30869)]
[New Thread 0xb22d9b90 (LWP 30870)]
[New Thread 0xb1ad8b90 (LWP 30871)]
[New Thread 0xb12d7b90 (LWP 30872)]
[New Thread 0xb0ad6b90 (LWP 30873)]
[New Thread 0xb02d5b90 (LWP 30874)]

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0xb2adab90 (LWP 30869)]
0xafab6ba9 in ?? () from /usr/lib/libx264.so.59
(gdb) bt
#0  0xafab6ba9 in ?? () from /usr/lib/libx264.so.59
Cannot access memory at address 0x74
(gdb) bt full
#0  0xafab6ba9 in ?? () from /usr/lib/libx264.so.59
No symbol table info available.
Cannot access memory at address 0x74
(gdb)


Title: Audio/Video conversation
Post by: kakaroto on March 08, 2009, 11:11:38 am
Hi,
First, recompiling farsight fixed the warnings you had, which is nice...
secondly about the crash.. with gdb it also crashes, it doesn't freeze... look at what gdb says 'program received signal SIGSEGV, segmentation fault'.. it's just that gdb will freeze the program so you can see the backtrace instead of just exiting... hehe
anyways, you don't seem to have much debug symbols which isn't very helpful but it looks like the segmentation fault is in  /usr/lib/libx264.so
I'm guessing libx264.so is an h264 encoder/decoder library.. you should maybe uninstall it since that's doing the segfault...


Title: Audio/Video conversation
Post by: flomar34 on March 08, 2009, 11:40:46 am
Hello,
Thanks for your reply
Uninstall libx264-59 will uninstall



Code:
ffmpeg gstreamer0.10-ffmpeg gstreamer0.10-plugins-bad-multiverse
  libavcodec-unstripped-51 libavdevice52 libavformat52 libmjpegtools0c2a
  libquicktime1 libx264-59 libx264-dev mencoder mozilla-mplayer mplayer peertv
  vlc vlc-nox winff




But there is only this crash with MSNP18. With MSNP15 it works like a charm
SO it's not too important i will stay on th MSNP15 protocol (mostly if i am the only concerned :) )


Title: Audio/Video conversation
Post by: flomar34 on March 08, 2009, 05:08:41 pm
Hello,

So, i succeed with it
Uninstall libx264-59 and all its dependencies
Compile amsn
reinstall mplayer, vlc ... and it works :)

Concerning audio call and MSNP18

WLM9 --> amsn : works nicely

amsn -->WLM9 : nothing happens, and maybe its's because i have no video cam

Code:
[17:00:41] Farsight debug : AUDIO CODECS ARE READY
[17:00:41] Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
[17:00:41] Farsight debug : Testing source dshowaudiosrc
[17:00:41] Farsight debug : Testing source directsoundsrc
[17:00:41] Farsight debug : Testing source osxaudiosrc
[17:00:41] Farsight debug : Testing source gconfaudiosrc
[17:00:41] Farsight debug : Using audio_source gconfaudiosrc
[17:00:41] Farsight debug : stun ip : 64.xx.xx.xx : 3478
[17:00:41] Farsight debug : FS: relay info = 0xb115420 - 2
[17:00:41] Farsight debug : Creating video_source : videotestsrc pattern=4 is-live=TRUE ! ffmpegcolorspace ! video/x-raw-yuv,width=352,height=288  --- (null) -- (null)
[17:00:41] Farsight debug : stun ip : 64.xx.xx.xx : 3478
[17:00:41] Farsight debug : FS: relay info = 0xb115420 - 2
[17:00:42] Creating CW Voip controls
[17:00:42] Farsight debug : AUDIO CANDIDATES ARE PREPARED
[17:00:42] Farsight : Farsight audio is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.XXX.XXX.XXX 58510 192.XXX.XXX.XXX 58510 UDP 2013266431 host x3+5 lRlTrZ3WfnOCtiiL8e1Ynf} {4 1192.XXX.XXX.XXX 58510 192.168.3.2 58510 UDP 1677721855 srflx x3+5 lRlTrZ3WfnOCtiiL8e1Ynf} {5 1 192.XXX.XXX.XXX 52031 192.XXX.XXX.XXX 58510 UDP 1006633215 relay x3+5 lRlTrZ3WfnOCtiiL8e1Ynf} {1 2 192.XXX.XXX.XXX 54667 192.XXX.XXX.XXX 54667 UDP 2013266430 host x3+5 lRlTrZ3WfnOCtiiL8e1Ynf} {4 2 192.XXX.XXX.XXX 54667 192.XXX.XXX.XXX 54667 UDP 1677721854 srflx x3+5 lRlTrZ3WfnOCtiiL8e1Ynf} {5 2 192.XXX.XXX.XXX 45724 192.XXX.XXX.XXX 54667 UDP 1006633214 relay x3+5 lRlTrZ3WfnOCtiiL8e1Ynf}
[17:00:42] Fasight :Waiting for video preparation
[17:00:42] Farsight debug : VIDEO CANDIDATES ARE PREPARED
[17:00:42] TURN: Disconnecting


Then i can't close this chat windows. I have to quit amsn


Title: Audio/Video conversation
Post by: kakaroto on March 08, 2009, 08:31:23 pm
Hi, the crash will only happen with msnp18 because only msnp18  tries to do a video call and libx264 is a video codec...
About the 'nothing happens' i don't know, did you paste me ALL the status log or is something still missing in there ? Anyways, i think it might be an issue with farsight not detecting the codecs correctly or something like that... i'll have to look at that.
Btw, why can't you close the chat window ? is it because it says you must cancel the call but there's no button to cancel ? or does it bug or ... ?


Title: Audio/Video conversation
Post by: flomar34 on March 08, 2009, 08:42:19 pm
Hello,

If i want to close the chat windows i have this message (In the case amsn --> wml9)
Code:
[20:38:14]  we can't close, there's a sip call running ...


I don't see anything to hang up and my contact doesn't see anything for an audio call

For the crash with the H264 and MSNP18 i think it was a problem on my computer. It seems to be ok now


Title: Audio/Video conversation
Post by: kakaroto on March 08, 2009, 10:45:41 pm
ok, so that's what I thought.. farsight2 never finishes the codec discovery, so we never send the invite, we also wait for farsight to finish preparing before showing the invite in the chat window...
because of that, you can't close the chat... i told olivier about this and we'll look into it tomorrow...
if you can hang around in IRC channel #amsn on irc.freenode.net maybe we'll need you tomorrow for a bit more info...
Thanks for reporting!

p.s.: in the meantime, you could try to launch amsn with :
Code:
GST_DEBUG="*fs*:5,ffmpeg*:5,GST_SCHEDULING:5" amsn > log 2>&1

and upload the log file somewhere so we can have a look at it.. hopefully it will contain all the info we need...


Title: Audio/Video conversation
Post by: flomar34 on March 09, 2009, 06:26:13 pm
Hello

In the log file i have

Quote
bash:  GST_DEBUG=*fs*:5,ffmpeg*:5,GST_SCHEDULING:5 : commande introuvable


Do i need something to launch it or could you explain to me how i have to do (i'm not so good enough i think :) )


Title: Audio/Video conversation
Post by: flomar34 on March 09, 2009, 08:01:19 pm
Hello,

ok i found it
http://carnacp.chez-alice.fr/log.txt

Hope it helps


Title: Audio/Video conversation
Post by: kakaroto on March 09, 2009, 09:44:26 pm
humm... sur une console ca donne quoi la commande :
Code:
gst-inspect videotestsrc


Title: Audio/Video conversation
Post by: flomar34 on March 09, 2009, 09:50:30 pm
File wasn't usefull?

Code:
Factory Details:
  Long name: Video test source
  Class: Source/Video
  Description: Creates a test video stream
  Author(s): David A. Schleef <ds@schleef.org>
  Rank: none (0)

Plugin Details:
  Name: videotestsrc
  Description: Creates a test video stream
  Filename: /usr/lib/gstreamer-0.10/libgstvideotestsrc.so
  Version: 0.10.21
  License: LGPL
  Source module: gst-plugins-base
  Binary package: GStreamer Base Plugins (Ubuntu)
  Origin URL: https://launchpad.net/distros/ubuntu/+source/gst-plugins-base0.10

GObject
 +----GstObject
       +----GstElement
             +----GstBaseSrc
                   +----GstPushSrc
                         +----GstVideoTestSrc

Pad Templates:
  SRC template: 'src'
    Availability: Always
    Capabilities:
      video/x-raw-yuv
                 format: YUY2
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: UYVY
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: Y422
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: UYNV
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: YVYU
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: AYUV
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: IYU2
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: YVU9
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: YUV9
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: YV12
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: I420
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: NV12
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: NV21
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: Y41B
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: Y42B
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-yuv
                 format: Y800
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 24
               red_mask: 16711680
             green_mask: 65280
              blue_mask: 255
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 24
               red_mask: 255
             green_mask: 65280
              blue_mask: 16711680
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 24
               red_mask: -16777216
             green_mask: 16711680
              blue_mask: 65280
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 24
               red_mask: 65280
             green_mask: 16711680
              blue_mask: -16777216
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 32
               red_mask: 16711680
             green_mask: 65280
              blue_mask: 255
             alpha_mask: -16777216
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 32
               red_mask: 255
             green_mask: 65280
              blue_mask: 16711680
             alpha_mask: -16777216
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 32
               red_mask: -16777216
             green_mask: 16711680
              blue_mask: 65280
             alpha_mask: 255
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 32
             endianness: 4321
                  depth: 32
               red_mask: 65280
             green_mask: 16711680
              blue_mask: -16777216
             alpha_mask: 255
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 24
             endianness: 4321
                  depth: 24
               red_mask: 16711680
             green_mask: 65280
              blue_mask: 255
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 24
             endianness: 4321
                  depth: 24
               red_mask: 255
             green_mask: 65280
              blue_mask: 16711680
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 16
             endianness: 1234
                  depth: 16
               red_mask: 63488
             green_mask: 2016
              blue_mask: 31
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-rgb
                    bpp: 16
             endianness: 1234
                  depth: 15
               red_mask: 31744
             green_mask: 992
              blue_mask: 31
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]
      video/x-raw-bayer
                  width: [ 1, 2147483647 ]
                 height: [ 1, 2147483647 ]
              framerate: [ 0/1, 2147483647/1 ]


Element Flags:
  no flags set

Element Implementation:
  Has change_state() function: gst_base_src_change_state
  Has custom save_thyself() function: gst_element_save_thyself
  Has custom restore_thyself() function: gst_element_restore_thyself

Element has no clocking capabilities.
Element has no indexing capabilities.
Element has no URI handling capabilities.

Pads:
  SRC: 'src'
    Implementation:
      Has getrangefunc(): gst_base_src_pad_get_range
      Has custom eventfunc(): gst_base_src_event_handler
      Has custom queryfunc(): gst_base_src_query
        Provides query types:
      Has custom intconnfunc(): gst_pad_get_internal_links_default
    Pad Template: 'src'

Element Properties:
  name                : The name of the object
                        flags: accès en lecture, accès en écriture
                        String. Default: null Current: "videotestsrc0"
  blocksize           : Size in bytes to read per buffer (0 = default)
                        flags: accès en lecture, accès en écriture
                        Unsigned Long. Range: 0 - 4294967295 Default: 4096 Current: 4096
  num-buffers         : Number of buffers to output before sending EOS
                        flags: accès en lecture, accès en écriture
                        Integer. Range: -1 - 2147483647 Default: -1 Current: -1
  typefind            : Run typefind before negotiating
                        flags: accès en lecture, accès en écriture
                        Boolean. Default: false Current: false
  do-timestamp        : Apply current stream time to buffers
                        flags: accès en lecture, accès en écriture
                        Boolean. Default: false Current: false
  pattern             : Type of test pattern to generate
                        flags: accès en lecture, accès en écriture
                        Enum "GstVideoTestSrcPattern" Default: 0, "smpte" Current: 0, "smpte"
                           (0): smpte            - SMPTE 100% color bars
                           (1): snow             - Random (television snow)
                           (2): black            - 100% Black
                           (3): white            - 100% White
                           (4): red              - Red
                           (5): green            - Green
                           (6): blue             - Blue
                           (7): checkers-1       - Checkers 1px
                           (8): checkers-2       - Checkers 2px
                           (9): checkers-4       - Checkers 4px
                           (10): checkers-8       - Checkers 8px
                           (11): circular         - Circular
                           (12): blink            - Blink
  timestamp-offset    : An offset added to timestamps set on buffers (in ns)
                        flags: accès en lecture, accès en écriture
                        Integer64. Range: -9223372036854775808 - 9223372036854775807 Default: 0 Current: 0
  is-live             : Whether to act as a live source
                        flags: accès en lecture, accès en écriture
                        Boolean. Default: false Current: false
  peer-alloc          : Ask the peer to allocate an output buffer
                        flags: accès en lecture, accès en écriture
                        Boolean. Default: true Current: true


Title: Audio/Video conversation
Post by: kakaroto on March 09, 2009, 11:13:41 pm
humm... no... it looks fine.. the log made me think that you didn't have videotestsrc (a fake video source that just displays a grid.. I use that for testing which is why it doesn't matter if you don't have a webcam right now).
anyways, it looks like something could not be linked correctly or something, I'll have to look at it a bit more, maybe some more debug would be helpful, could you give me the log file for :
Code:
GST_DEBUG=*:5 amsn > log 2>&1

this should give me everything I need (but huge file...)


Title: Audio/Video conversation
Post by: flomar34 on March 10, 2009, 02:25:11 pm
Hello

Yesterday evening i had a problem with the command
Code:
gst-inspect videotestsrc


I had to uninstall and purge gstreamer tools and re install it and then, the command was ok
But i didn't try the audio call after it. (Peut il y avoir un lien de cause a effet sur le non fonctionnement de l'appel audio avec MSNP18 ?)

I will try it again this evening and if it's still doesn't work i will  try the GST_DEBUG=*:5 amsn > log 2>&1


Title: Audio/Video conversation
Post by: kakaroto on March 10, 2009, 05:50:48 pm
ok cool.. well, I asked if the command gave you anything because the first logs showed as if you didn't have it... now it should work fine if you just fixed it... well hopefully...
anyways, with MSNP18, it's not an audio call, it's a video call, so yes it is indeed linked, with no video source, it can't prepare do much right now...


Title: Audio/Video conversation
Post by: flomar34 on March 10, 2009, 06:43:28 pm
Hello,

Soory, im going to speak French to be sure to uderstand all.

Bonjour,
Je viens donc de recompiler amsn avec le gstreamer tools qui semble ok
Quand je lance l'appel appel audio vers wml9 j'ai maintenant cette erreur

Code:
<?xml version="1.0"?>
<bug version="0.3">
<error>
<date>1236706034</date>
<text>{{bad window path name &quot;.container_0.msg_0.f.bottom.f.voip&quot;}}</text>
<stack>bad window path name &quot;.container_0.msg_0.f.bottom.f.voip&quot;
    while executing
&quot;frame .container_0.msg_0.f.bottom.f.voip.level -class Progress&quot;
    (&quot;uplevel&quot; body line 1)
    invoked from within
&quot;uplevel 1 [list frame $w -class Progress] $args&quot;
    (procedure &quot;::dkfprogress::Progress&quot; line 2)
    invoked from within
&quot;::dkfprogress::Progress $frame_in.level&quot;
    (procedure &quot;::ChatWindow::AddVoipControls&quot; line 41)
    invoked from within
&quot;::ChatWindow::AddVoipControls $email $sip $callid&quot;
    (procedure &quot;::amsn::SIPPreparing&quot; line 4)
    invoked from within
&quot;::amsn::SIPPreparing $email $sip &quot;&quot;&quot;
    (procedure &quot;InviteUserCB&quot; line 15)
    invoked from within
&quot;InviteUserCB $email $sip&quot;
    (procedure &quot;::MSNSIP::InviteUser&quot; line 16)
    invoked from within
&quot;::MSNSIP::InviteUser $email&quot;
    (procedure &quot;::amsn::SIPCallInviteUser&quot; line 12)
    invoked from within
&quot;::amsn::SIPCallInviteUser darkflxxx@sad-amsn-user.com&quot;
    (&quot;eval&quot; body line 1)
    invoked from within
&quot;eval $command [lindex [lindex $itemlist 0] 1]&quot;
    (procedure &quot;::amsn::listChoose&quot; line 8)
    invoked from within
&quot;::amsn::listChoose $title $userlist $command 0 1&quot;
    (procedure &quot;::amsn::ShowChatList&quot; line 24)
    invoked from within
&quot;::amsn::ShowChatList &quot;Démarrer un appel audio&quot; [::ChatWindow::getCurrentTab .container_0] ::amsn::SIPCallInviteUser&quot;
    invoked from within
&quot;.container_0.#container_0#menu.#container_0#menu#actions invoke active&quot;
    (&quot;uplevel&quot; body line 1)
    invoked from within
&quot;uplevel #0 [list $w invoke active]&quot;
    (procedure &quot;tk::MenuInvoke&quot; line 50)
    invoked from within
&quot;tk::MenuInvoke .container_0.#container_0#menu.#container_0#menu#actions 1&quot;
    (command bound to event)</stack>
<code>NONE</code>
</error>
<system>
<amsn>0.98b</amsn>
<revision>11084</revision>
<date>03/10/2009 11:17:42</date>
<tcl>8.5.3</tcl>
<tk>8.5.3</tk>
<osversion>2.6.27-11-generic</osversion>
<pointersize>4</pointersize>
<byteorder>littleEndian</byteorder>
<threaded>1</threaded>
<machine>i686</machine>
<platform>unix</platform>
<os>Linux</os>
<user>cartes</user>
<wordsize>4</wordsize>
<msnprotocol>18</msnprotocol>
<loadedplugins>ColoredNicks {Desktop Integration} Nudge</loadedplugins>
<vendor>aMSN</vendor>
</system>
<extra>
<status_log>
0xad, 0x3f, 0x82, 0x98,
0x55, 0xc6, 0x36, 0x17}
[18:26:51] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x3f}
[18:26:51] TURN: Received response SHARED-SECRET-RESPONSE for id 73756b224b6825a053907c601a800bc0
[18:26:51] TURN: TURN server 213.199.163.63 : 0
[18:26:51] TURN: Received message of type SHARED-SECRET-RESPONSE
[18:26:51] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$
y[88][a9](J4[f3]m;k[ea][f3]C[d5][13][e4][16][0f]c[80]h[15]:[d6]=[b1]6Y66[e9]k[17][c8]5[00][07][00][10][82][a2][ee][e9](^s7,[d4][cc][b4][af]X[19][d1][00][0e][00][08][00][01][00][00][d5][c7][a3]?
[18:26:51] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[18:26:51] TURN: Received attribute USERNAME : {0x0a, 0x79, 0x88, 0xa9,
0x28, 0x4a, 0x34, 0xf3,
0x6d, 0x3b, 0x6b, 0xea,
0xf3, 0x43, 0xd5, 0x13,
0xe4, 0x16, 0x0f, 0x63,
0x80, 0x68, 0x15, 0x3a,
0xd6, 0x3d, 0xb1, 0x36,
0x59, 0x36, 0x36, 0xe9,
0x6b, 0x17, 0xc8, 0x35}
[18:26:51] TURN: Received attribute PASSWORD : {0x82, 0xa2, 0xee, 0xe9,
0x28, 0x5e, 0x73, 0x37,
0x2c, 0xd4, 0xcc, 0xb4,
0xaf, 0x58, 0x19, 0xd1}
[18:26:51] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x3f}
[18:26:51] TURN: Received response SHARED-SECRET-RESPONSE for id 31d9279d5df0462c0bc39575cbba6e9f
[18:26:51] TURN: TURN server 213.199.163.63 : 0
[18:26:51] TURN: Disconnecting
[18:26:51] Turn prepared {213.199.163.63 0 YTQM9Es9og6SOmvq80PVE+QWD2MQ+S18YpaEeayBCpg1++Eq oAQXIkl0Xz6tP4KYVcY2Fw== 1 udp} {213.199.163.63 0 CnmIqShKNPNtO2vq80PVE+QWD2OAaBU61j2xNlk2NulrF8g1 gqLu6Sheczcs1My0r1gZ0Q== 2 udp}
[18:26:51] Farsight debug : AUDIO CODECS ARE READY
[18:26:51] Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
[18:26:51] Farsight debug : Testing source dshowaudiosrc
[18:26:51] Farsight debug : Testing source directsoundsrc
[18:26:51] Farsight debug : Testing source osxaudiosrc
[18:26:51] Farsight debug : Testing source gconfaudiosrc
[18:26:51] Farsight debug : Using audio_source gconfaudiosrc
[18:26:51] Farsight debug : stun ip : 64.14.48.28 : 3478
[18:26:51] Farsight debug : FS: relay info = 0xa42e4c0 - 2
[18:26:51] Farsight debug : Creating video_source : videotestsrc pattern=4 is-live=TRUE ! ffmpegcolorspace ! video/x-raw-yuv,width=352,height=288  --- (null) -- (null)
[18:26:52] Farsight debug : stun ip : 64.14.48.28 : 3478
[18:26:52] Farsight debug : FS: relay info = 0xa42e4c0 - 2
[18:26:52] Creating CW Voip controls
[18:26:52] Farsight debug : AUDIO CANDIDATES ARE PREPARED
[18:26:52] Farsight : Farsight audio is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.168.3.2 60641 192.168.3.2 60641 UDP 2013266431 host PWtz DrJ5jfr0KSypBz+OKKn/xU} {4 1 86.66.161.236 60641 192.168.3.2 60641 UDP 1677721855 srflx PWtz DrJ5jfr0KSypBz+OKKn/xU} {5 1 213.199.163.63 38692 192.168.3.2 60641 UDP 1006633215 relay PWtz DrJ5jfr0KSypBz+OKKn/xU} {1 2 192.168.3.2 46824 192.168.3.2 46824 UDP 2013266430 host PWtz DrJ5jfr0KSypBz+OKKn/xU} {4 2 86.66.161.236 46824 192.168.3.2 46824 UDP 1677721854 srflx PWtz DrJ5jfr0KSypBz+OKKn/xU} {5 2 213.199.163.63 41050 192.168.3.2 46824 UDP 1006633214 relay PWtz DrJ5jfr0KSypBz+OKKn/xU}
[18:26:52] Fasight :Waiting for video preparation
[18:26:52] Farsight debug : VIDEO CANDIDATES ARE PREPARED
[18:26:52] TURN: Disconnecting


</status_log>
<protocol_log>
[18:26:33] &lt;-ns-sock31 PRP 13 PHW %20%20
[18:26:33] &lt;-ns-sock31 PRP 14 PHM %20%20
[18:26:34] -&gt;ns-sock31 PNG

[18:26:34] &lt;-ns-sock31 QNG 42
[18:26:37] -&gt;ns-sock31 XFR 15 SB

[18:26:39] &lt;-ns-sock31 XFR 15 SB 64.4.36.46:1863 CKI 1128828274.56218199.56106167 U messenger.msn.com 1
[18:26:39] &lt; Connected to: 64.4.36.46 1863 &gt;
[18:26:39] -&gt;ns-sock31 UUX 16 71
&lt;EndpointData&gt;&lt;Capabilities&gt;2687762468:16&lt;/Capabilities&gt;&lt;/EndpointData&gt;
[18:26:39] -&gt;ns-sock31 UUX 17 126
&lt;PrivateEndpointData&gt;&lt;EpName&gt;aMSN&lt;/EpName&gt;&lt;Idle&gt;false&lt;/Idle&gt;&lt;State&gt;NLN&lt;/State&gt;&lt;ClientType&gt;1&lt;/ClientType&gt;&lt;/PrivateEndpointData&gt;
[18:26:39] -&gt;ns-sock31 UUX 18 151
&lt;Data&gt;&lt;PSM&gt;&lt;/PSM&gt;&lt;CurrentMedia&gt;&lt;/CurrentMedia&gt;&lt;MachineGuid&gt;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&lt;/MachineGuid&gt;&lt;SignatureSound&gt;&lt;/SignatureSound&gt;&lt;/Data&gt;
[18:26:42] -&gt;::MSN::SB1-sock19 USR 19 cartexxx@sad-amsn-user.com;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2} 1128828274.56218199.56106167

[18:26:42] &lt;-ns-sock31 UUX 16 0
[18:26:43] &lt;-ns-sock31 UBX 1:cartexxx@sad-amsn-user.com 436
[18:26:43] Message Contents:
&lt;Data&gt;&lt;PSM&gt;&lt;/PSM&gt;&lt;CurrentMedia&gt;&lt;/CurrentMedia&gt;&lt;MachineGuid&gt;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&lt;/MachineGuid&gt;&lt;SignatureSound&gt;&lt;/SignatureSound&gt;&lt;EndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;Capabilities&gt;2687762468:16&lt;/Capabilities&gt;&lt;/EndpointData&gt;&lt;PrivateEndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;EpName&gt;aMSN&lt;/EpName&gt;&lt;Idle&gt;false&lt;/Idle&gt;&lt;State&gt;NLN&lt;/State&gt;&lt;ClientType&gt;1&lt;/ClientType&gt;&lt;/PrivateEndpointData&gt;&lt;/Data&gt;
[18:26:43] &lt;-ns-sock31 UBX 1:darkflxxx@sad-amsn-user.com 558
[18:26:43] Message Contents:
&lt;Data&gt;&lt;CurrentMedia&gt;&lt;/CurrentMedia&gt;&lt;PSM&gt;i am someone else. &amp;#x5E;o)&lt;/PSM&gt;&lt;SignatureSound&gt;&lt;/SignatureSound&gt;&lt;Scene&gt;&amp;#x3C;msnobj Creator&amp;#x3D;&quot;darkflxxx@sad-amsn-user.com&quot; Type&amp;#x3D;&quot;16&quot; SHA1D&amp;#x3D;&quot;DXpmpt9Jq38X274YNCDWReR1uaI&amp;#x3D;&quot; Size&amp;#x3D;&quot;57226&quot; Location&amp;#x3D;&quot;0&quot; Friendly&amp;#x3D;&quot;RwByAGEAZgBmAGkAdABpAAAA&quot;/&amp;#x3E;&lt;/Scene&gt;&lt;ColorScheme&gt;-1643538&lt;/ColorScheme&gt;&lt;MachineGuid&gt;&amp;#x7B;703F5C66-3559-4136-911D-58EC4A074D1E&amp;#x7D;&lt;/MachineGuid&gt;&lt;DDP&gt;&lt;/DDP&gt;&lt;EndpointData id=&quot;{703f5c66-3559-4136-911d-58ec4a074d1e}&quot;&gt;&lt;Capabilities&gt;2788999228:48&lt;/Capabilities&gt;&lt;/EndpointData&gt;&lt;/Data&gt;
[18:26:43] &lt;-ns-sock31 UUX 17 0
[18:26:43] &lt;-ns-sock31 UBX 1:cartexxx@sad-amsn-user.com 436
[18:26:43] Message Contents:
&lt;Data&gt;&lt;PSM&gt;&lt;/PSM&gt;&lt;CurrentMedia&gt;&lt;/CurrentMedia&gt;&lt;MachineGuid&gt;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&lt;/MachineGuid&gt;&lt;SignatureSound&gt;&lt;/SignatureSound&gt;&lt;EndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;Capabilities&gt;2687762468:16&lt;/Capabilities&gt;&lt;/EndpointData&gt;&lt;PrivateEndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;EpName&gt;aMSN&lt;/EpName&gt;&lt;Idle&gt;false&lt;/Idle&gt;&lt;State&gt;NLN&lt;/State&gt;&lt;ClientType&gt;1&lt;/ClientType&gt;&lt;/PrivateEndpointData&gt;&lt;/Data&gt;
[18:26:44] &lt;-ns-sock31 UUX 18 0
[18:26:44] &lt;-ns-sock31 UBX 1:cartexxx@sad-amsn-user.com 436
[18:26:44] Message Contents:
&lt;Data&gt;&lt;PSM&gt;&lt;/PSM&gt;&lt;CurrentMedia&gt;&lt;/CurrentMedia&gt;&lt;MachineGuid&gt;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&lt;/MachineGuid&gt;&lt;SignatureSound&gt;&lt;/SignatureSound&gt;&lt;EndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;Capabilities&gt;2687762468:16&lt;/Capabilities&gt;&lt;/EndpointData&gt;&lt;PrivateEndpointData id=&quot;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}&quot;&gt;&lt;EpName&gt;aMSN&lt;/EpName&gt;&lt;Idle&gt;false&lt;/Idle&gt;&lt;State&gt;NLN&lt;/State&gt;&lt;ClientType&gt;1&lt;/ClientType&gt;&lt;/PrivateEndpointData&gt;&lt;/Data&gt;
[18:26:45] &lt;-::MSN::SB1-sock19 USR 19 OK {cartexxx@sad-amsn-user.com;{42aeb402-b3b3-7b2b-113e-e6bb3f7291b2}} cartexxx%40sad-amsn-user.com%2Efr
[18:26:45] -&gt;::MSN::SB1-sock19 CAL 20 cartexxx@sad-amsn-user.com

[18:26:45] -&gt;::MSN::SB1-sock19 CAL 21 darkflxxx@sad-amsn-user.com

[18:26:46] &lt;-::MSN::SB1-sock19 CAL 20 RINGING 1128828274
[18:26:47] &lt;-::MSN::SB1-sock19 JOI cartexxx@sad-amsn-user.com cartexxx%40sad-amsn-user.com%2Efr 2687762468:16
[18:26:47] &lt;-::MSN::SB1-sock19 CAL 21 RINGING 1128828274
[18:26:47] &lt;-::MSN::SB1-sock19 JOI {darkflxxx@sad-amsn-user.com;{703f5c66-3559-4136-911d-58ec4a074d1e}} flo,the%20rock&apos;n%20roll%20is%20the%20best%20style%20of%20song 2788999228:48
[18:26:47] -&gt;::MSN::SB1-sock19 MSG 22 U 96
MIME-Version: 1.0
Content-Type: text/x-clientcaps

Client-Name: aMSN 0.98b
Chat-Logging: Y

[18:26:48] &lt;-::MSN::SB1-sock19 JOI darkflxxx@sad-amsn-user.com flo,the%20rock&apos;n%20roll%20is%20the%20best%20style%20of%20song 2788999228:48


</protocol_log>
</extra>
<user>
<comment>

</comment>
</user>
</bug>




J'ignore l'erreur et il ne se passe rien et je ne peux fermer la fenetre de discussion

Quand vous dites it can't prepare do much right now... est ce que cela veut dire qu'en l'état, cela ne peux fonctionner sans webcam?

Quoi qu'il en soit, si je suis seul concerné par ce probleme cela veut tres surement dire que le probleme est lié a mon ordi.
Donc ne perdez surtout pas du temps la dessus si cela ne sert pas a faire avancer votre projet que je trouve plein d'espoir pour le petit monde de Linux
Cela n'est pas bloquant pour moi car fonctionnel avec MSNP15.

Cordialement[/quote]


Title: Audio/Video conversation
Post by: kakaroto on March 11, 2009, 04:38:07 pm
ok, bon, bein alors j'attendrais ton log avec GST_DEBUG=*:5 amsn :p
bein c'est le meme problem qu'avant, pourquoi ca fait le bug, je sais pas, mais c'est lie avec le GUI, histoire de mettre des controles du volume dans la fenetre de chat, donc ca a pas rapport...
et puis, si c'est un probleme lie a ton ordi :
1 - aMSN doit quand meme s'en sortir
2 - t'es surement pas le seul au monde avec la meme config...
3 - ton probleme montre peut-etre un bug dans notre code qu'on ne voyait pas avant avec d'autres configs...

Sinon, bein je disais que son source video ca aurait pas pu marcher (oui, ca c un bug que tu viens de m'aider a trouver), mais t'en fais pas, quand je dis "source video" bein videotestsrc c'est une source video et c'est ce qu'on utilise pour l'instant...


Title: Audio/Video conversation
Post by: flomar34 on March 11, 2009, 06:38:10 pm
Hello,

Here it is and not a little one :)

http://carnacp.chez-alice.fr/log.txt

Hope it could help

Edit : correction of the adress


Title: Audio/Video conversation
Post by: flomar34 on March 13, 2009, 03:20:09 pm
Hello,

Concerning the crash with the H264 when you want to make an audio call with MSNP18, it could maybe come from gstreamer0.10-ffmpeg on my computer

If gstreamer0.10-ffmpeg is installed, i compile amsn, make the audio call and it crash (with an error about H264)

If it's not installed, amsn doesn't crash, it's just waiting for something (i've seen that now there is a message if you want to close the windows that you can't do it and that you have to hang up first)


Title: Audio/Video conversation
Post by: kakaroto on March 14, 2009, 08:52:38 am
ok thanks, i'll have a look to see why nothing happens.. about the gst-ffmpeg bug, yes h264 seems to cause the crash, i'd like to have farsight not do those checks (since we know we don't need h264), i'll have to discuss this with Olivier (farsight2 maintainer).


Title: Audio/Video conversation
Post by: lamaresh on March 18, 2009, 07:10:23 pm
hi all.
I've installed last amsn on ubuntu hardy. I've correct version of libnice, farsight and gstreamer-plugins-farsigth as explained in the wiki.
Compiling amsn farisht has been found, so i can use audio call: bun when i try it i obtain this error:
Code:
GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'


What could i do?

EDIT: i tryed on debian squeeze too, with newer packages version, but i have the same error


Title: Audio/Video conversation
Post by: zyazhou on March 19, 2009, 03:25:06 pm
kakaroto, one question about MSNP15 SIP. I found in sip log, why INVITE is sent again after MSN accepts our first INVITE? Thanks!


Title: Audio/Video conversation
Post by: kakaroto on March 19, 2009, 05:06:33 pm
@lamaresh: that's not an error, it's a warning, it's normal, everyone gets it.
@zyazhou: read the SIP and ICE specs, it's a reinvite, it's needed to tell the other user what's the selected-candidate-pair in ICE.


Title: Audio/Video conversation
Post by: lamaresh on March 20, 2009, 11:10:11 am
maybe a reported an incorrect string, but when i try to start an audio call i have a window reportin a tk bug. But i can't click the button for details because it's someway blocked, so i reported that warning.

i tryed both tk 8.5 and 8.6, but i have the same problem (in ubuntu and debian)

Do you know about a similar behavior?


Title: Audio/Video conversation
Post by: kakaroto on March 20, 2009, 01:16:41 pm
no I don't sorry. If the details button does't work, then ignore the error, go to the contact list and press Ctrl-S  to open the status log window and paste here the bug details, it will appear as a lot of text with a black background in the status window.


Title: Audio/Video conversation
Post by: lamaresh on March 20, 2009, 03:44:30 pm
Is not only detail button that doesn't work, but all graphic become unusable.
However after several attempts i managed in getting the details:

Code:
invalid command name ".container_0.msg_0.f.bottom.f.voip.amplifier"
    while executing
"$frame_in.amplifier configure -state normal"
    (procedure "::ChatWindow::UpdateVoipControls" line 25)
    invoked from within
"::ChatWindow::UpdateVoipControls $chatid $sip $callid"
    (procedure "::amsn::SIPInviteSent" line 16)
    invoked from within
"::amsn::SIPInviteSent $email $sip $callid"
    (procedure "::MSNSIP::invitePrepared" line 11)
    invoked from within
"::MSNSIP::invitePrepared ::MSNSIP::SIPConnection1 halflxxx@sad-amsn-user.com"
    ("eval" body line 1)
    invoked from within
"eval $options(-prepared)"


while in terminal i get this:

Code:
Farsight : Preparing
Farsight debug : AUDIO CODECS ARE READY
Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
Farsight debug : Testing source dshowaudiosrc
Farsight debug : Testing source directsoundsrc
Farsight debug : Testing source osxaudiosrc
Farsight debug : Testing source gconfaudiosrc
Farsight debug : Using audio_source gconfaudiosrc

(<unknown>:5298): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstGConfAudioSrc' has no property named `blocksize'
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x9edaea0 - 2
Farsight debug : AUDIO CANDIDATES ARE PREPARED
Farsight : Farsight Audio is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.168.0.107 39938 192.168.0.107 39938 UDP 0.83 host F2cGeSdypBarW7tvmJwdKcy1aHKJC75YNW41tlpq+SQ= a7oRybNJsOZGJtihB1s1sA==} {2 1 79.19.41.222 10940 192.168.0.107 39938 UDP 0.55 srflx YvylkbGa5lx5owP6yfYBZBpqKJaWsiFXWopEzvnmwas= ZKsctzZI0vWsJ3qfnxn3mA==} {3 1 213.199.163.66 40775 192.168.0.107 39938 UDP 0.45 relay dFAXhlf9f2rHiNJX9oNKhxDDWtp8Lf8dP/LNk+H1PSM= WClKU63N2YmTzHYJFSl6fg==} {1 2 192.168.0.107 50173 192.168.0.107 50173 UDP 0.83 host F2cGeSdypBarW7tvmJwdKcy1aHKJC75YNW41tlpq+SQ= a7oRybNJsOZGJtihB1s1sA==} {2 2 79.19.41.222 10941 192.168.0.107 50173 UDP 0.55 srflx YvylkbGa5lx5owP6yfYBZBpqKJaWsiFXWopEzvnmwas= ZKsctzZI0vWsJ3qfnxn3mA==} {3 2 213.199.163.66 44249 192.168.0.107 50173 UDP 0.45 relay dFAXhlf9f2rHiNJX9oNKhxDDWtp8Lf8dP/LNk+H1PSM= WClKU63N2YmTzHYJFSl6fg==}


The last thing: i noticed that if i start amsn with "wish /usr/bin/amsn" farsight plugin is always recognized, while if i only use amsn sometimes i can't have farsight recognized :o


Title: Audio/Video conversation
Post by: kakaroto on March 23, 2009, 05:49:21 pm
@lamaresh:
1 - you don't have the latest SVN version, make sure you update to the latest!
2 - that 'bug' is because you didn't install amsn correctly and is just a UI bug, nothing to do with farsight
3 - the messages in the terminal are perfectly normal
4 - you might have 2 amsn installed, one in /usr/ and one in /usr/local/ that might be why it sometimes detects amsn and sometimes it doesn't.


Title: Audio/Video conversation
Post by: lamaresh on March 23, 2009, 09:20:56 pm
i used make deb of amsn source. What should I do?
However it's not just a gui problem, beacuse even if i can use gui afrter the error i can't hear any sound


Title: Audio/Video conversation
Post by: kakaroto on March 23, 2009, 09:37:52 pm
no, the gui problem makes everything go bad! and it's not about 'make deb' or not, it's about having the latest SVN version! so update it!


Title: Audio/Video conversation
Post by: lamaresh on March 24, 2009, 02:14:34 am
tnk, now it's perfect :)


Title: Audio/Video conversation
Post by: Kreuger on March 24, 2009, 04:48:38 pm
Hey guys, I recently tried this again. I got through everything until I ran make for farsight, it came back with this.

Quote
mkdir .libs
 gcc -DHAVE_CONFIG_H -I. -I../.. -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -Wall -g -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -pthread -I/usr/include/gstreamer-0.10 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2 -g -O2 -MT libfsrtcpfilter_la-fs-rtcp-filter.lo -MD -MP -MF .deps/libfsrtcpfilter_la-fs-rtcp-filter.Tpo -c fs-rtcp-filter.c  -fPIC -DPIC -o .libs/libfsrtcpfilter_la-fs-rtcp-filter.o
fs-rtcp-filter.c: In function ‘fs_rtcp_filter_transform_ip’:
fs-rtcp-filter.c:206: error: invalid use of void expression

make[3]: *** [libfsrtcpfilter_la-fs-rtcp-filter.lo] Error 1
make[3]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2-0.0.8/gst/rtcpfilter'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2-0.0.8/gst'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/kreuger/Downloads/Updates/farsight2-0.0.8'
make: *** [all] Error 2
kreuger@kreuger-desktop:~/Downloads/Updates/farsight2-0.0.8$


Title: Audio/Video conversation
Post by: kakaroto on March 24, 2009, 05:55:57 pm
@Kreuger: You need to upgrade to gstreamer 0.10.22!
In theory, the ./configure should have told you this with an error...

EDIT and by the way, you shouldn't use the 0.0.8 version of farsight unless you read this : http://www.amsn-project.net/forums/viewtopic.php?t=6425


Title: Audio/Video conversation
Post by: Kreuger on March 25, 2009, 02:20:49 am
Yeah it told me to grab gst-plugins-base 0.10.22 but the rest it didnt say. Plus, the guide says 0.10.20 and newer should work and thats what I have.


Title: Audio/Video conversation
Post by: kakaroto on March 25, 2009, 05:56:09 pm
yeah, the guide wasn't updated... the guide basically says that the minimal configuration should work, if you use a newer version of farsight and that version depends on a newer version of gstreamer, then you will also need to upgrade gstreamer... anyways!


Title: Audio/Video conversation
Post by: jack_ut on March 26, 2009, 12:20:03 am
Hi all,
I've updated amsn with libnice 0.0.5 and farsight2.0.0.8 and it seems working fine (i've tried audio conversation on two pc on the same lan, first with linux i686 and second with Linux x86_64, i've tried the audio conversation with WLM 2009 too), but reading the previous posts i understand that i could have some problems (i didn't apply the patch) could you tell me if there's something wrong in my installation procedure?

This is my installation procedure

First i've installed
gstreamer-0.10.22 and gst-plugins-base-0.10.22 in /opt/gstreamer
gst-plugins-farsight-0.12.10 in /usr

After ì've installed libnice and compiled farsight2.0.0.8 with command

Code:
export PKG_CONFIG_PATH=/opt/gstreamer/lib/pkgconfig


This is the farsight log in amsn

Code:
Farsight : Preparing
Farsight debug : AUDIO CODECS ARE READY
Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
Farsight debug : Testing source dshowaudiosrc
Farsight debug : Testing source directsoundsrc
Farsight debug : Testing source osxaudiosrc
Farsight debug : Testing source gconfaudiosrc
Farsight debug : Using audio_source gconfaudiosrc
Farsight debug : stun ip : 64.14.48.28 : 3478
Farsight debug : FS: relay info = 0x7fac040b2660 - 2
Farsight debug : AUDIO CANDIDATES ARE PREPARED
Farsight : Farsight Audio is now prepared!
local codecs : {SIREN 111 16000 bitrate=16000} {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.168.1.4 36610 {} 0 UDP 0.8299999833106995 host MON2HmUQGngXKO+58AINFaq+yjMWiRi1UHtDCTIGN3c= /SiKpR8fWEuQK0Y5QIRu1w==} {2 1 192.168.1.2 33613 {} 0 UDP 0.8299999833106995 host /ZSXreZgSIcC+y1cgxSjYyGbh8eiqiT4f4fU5fEkwOA= i2TAbvNmDkpHfFCktEJKpQ==} {3 1 93.148.42.151 36610 192.168.1.4 36610 UDP 0.550000011920929 srflx rYffi+QJn1hxYcBM5PbRha17R8NtlkZRM7F5aR89Xhk= CUopVgZg7ShBWdD9qucmOg==} {4 1 213.199.163.40 30916 192.168.1.4 36610 UDP 0.44999998807907104 relay 6DRL7ZAdQtn+8CohRKhlk3x6o7fyjayfay6trpZR/mM= UgrfH8f9KFIXXM7eKnUsag==} {5 1 93.148.42.151 33613 192.168.1.2 33613 UDP 0.550000011920929 srflx +7XC0gIx9pzkXpb6/9L3ZJ2io5/ishu5aJ+fRzDbIBw= gFnEHNGGjcxMjAv01uUpHQ==} {1 2 192.168.1.4 56801 {} 0 UDP 0.8299999833106995 host MON2HmUQGngXKO+58AINFaq+yjMWiRi1UHtDCTIGN3c= /SiKpR8fWEuQK0Y5QIRu1w==} {2 2 192.168.1.2 39347 {} 0 UDP 0.8299999833106995 host /ZSXreZgSIcC+y1cgxSjYyGbh8eiqiT4f4fU5fEkwOA= i2TAbvNmDkpHfFCktEJKpQ==} {3 2 93.148.42.151 56801 192.168.1.4 56801 UDP 0.550000011920929 srflx rYffi+QJn1hxYcBM5PbRha17R8NtlkZRM7F5aR89Xhk= CUopVgZg7ShBWdD9qucmOg==} {4 2 213.199.163.40 53259 192.168.1.4 56801 UDP 0.44999998807907104 relay 6DRL7ZAdQtn+8CohRKhlk3x6o7fyjayfay6trpZR/mM= UgrfH8f9KFIXXM7eKnUsag==} {5 2 93.148.42.151 39347 192.168.1.2 39347 UDP 0.550000011920929 srflx +7XC0gIx9pzkXpb6/9L3ZJ2io5/ishu5aJ+fRzDbIBw= gFnEHNGGjcxMjAv01uUpHQ==}


and this is the answer to the gst-inspect fsrtpconference command
Code:
(gst-inspect-0.10:8644): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libfsrtpconference.so': /opt/gstreamer/lib/libgstbase-0.10.so.0: undefined symbol: gst_util_seqnum_next
element plugin couldn't be loaded
Plugin Details:
  Name: fsrtpconference
  Description: Farsight RTP Conference plugin
  Filename: /usr/lib/gstreamer-0.10/libfsrtpconference.so
  Version: 0.0.8
  License: LGPL
  Source module: farsight2
  Binary package: Farsight
  Origin URL: http://farsight.freedesktop.org/

  fsrtpconference: Farsight RTP Conference

  1 features:
  +-- 1 elements


Sorry for my english, hoping in your answer, thank you


Title: Audio/Video conversation
Post by: kakaroto on March 26, 2009, 06:18:52 pm
Hi jack_ut, welcome to the forums!
Your english is quite good, so no need to be sorry about it!
Well.. it works for you, so no need to complain! :)
Without the patch, you would actually have it fail directly, not working at all. But it would only happen if you have gst-plugins-farsight-0.12.11, but you have version 0.12.10, so you're fine... For now, just don't upgrade gst-plugins-farsight, and gst-plugins-bad. If you upgrade them both, then you'll need the patch! Otherwise, it should just work! :)


Title: Audio/Video conversation
Post by: jack_ut on March 27, 2009, 03:25:54 pm
Thank you for your answer, now it's all clear, you're great :D


Title: Audio/Video conversation
Post by: kakaroto on March 27, 2009, 06:41:05 pm
of course I am! lol, j/k, you're welcome! :)


Title: Audio/Video conversation
Post by: jones on March 30, 2009, 04:01:44 pm
I get the aMSN svn version, and I find the SDP messages contain the following line.
"a=x-caps:34 65537:352:288:15.0:256000:1;131074:176:144:15.0:180000:1\r\n"
I guess that 352:288 is resolution, 15.0 is fps, and 256000 is bitrate.
But I don't understand the other parameters.
What does the first parameter(65537) and the last parameter(1) mean?
Is the first parameter a codec profile ID? If yes, how does WLM specify the profile ID in a video streaming? Is it can be changed adaptively?
Thank you.

p.s. My English is not good, sorry about that.


Title: Audio/Video conversation
Post by: kakaroto on March 31, 2009, 06:33:35 am
Hi jones,
welcome to the forums! some quite good questions there, however, why not take the time to say hi, and tell me a bit more about you and what you're doing? (yes, i'm a very curious person!). I'd like to know what you need this information for? What are you trying to achieve exaclty? Are you writing your own client too ?
anyways, the x-caps attribute was copied the same as what WLM sends, I also don't know about the first and last arguments, it looks more like a bit field (first one is shifted left by 1 compared to the 176x144 one). Amyways, there's no open source implementation of a H263 or msvc1 codec.. the ffmpeg implementation of h263 seems to be incompatible with WLM...
If you find a solution, please let me know! Thanks!


Title: Audio/Video conversation
Post by: jones on April 01, 2009, 02:39:54 pm
Hi kakaroto,
Thank you for your answer.
Yes, I trying to do the video call with WLM2009, and I find solution to solve the ffmpeg issue.
I observe the packets that generate from farsight2 will be fragment in IP layer.
And I trying to set the argument "rtp-payload-size" of ffenc_h263, then it works.... :twisted:
Just set the argument(rtp-payload-size) to the size that smaller than MTU - headers. ex: smaller than 1300.
But the farsight2 does not have the API to set the argument of encoder, or do I miss something?
I set the argument by modify the farsight2 now.
Although it can work well, but I don't know the root cause of this issue, does every packet need to contain the rtp and h263 headers?

Now we have another issue, new p2p protocol of WLM2009.......


Title: Audio/Video conversation
Post by: kakaroto on April 01, 2009, 04:23:13 pm
humm... you stil didn't tell me what client you're working on :p
anyways, I tried that, I tried quite a lot of stuff, I modified gst-ffmpeg directly, it didn't work... I also modified ffmpeg so it can send an event on each GOB so the payloader can correctly packetize it without adding the GOB header, like WLM does, but it still didn't work...
I even tried making the frame  all fit into one single packet but WLM still refused it. I also tried modifying WLM a little to make it accept a bigger MTU, but it still didn't work. So no, something different is wrong. Also, the max payload size should be 1024, not 1300, that's what WLM uses (look at their assembly code).
by the way, ffmpeg has a bug with the rtp-payload-size, even if you set it, it doesn't respect it.. if you want less than 1024, you should set that property to 500 or 600.
Anyways, you said "then it works", I guess you meant "Then it should work", right? were you ever able to make it work correctly some way or another using ffenc_h263? I'd really like to see that!


Title: Audio/Video conversation
Post by: jones on April 02, 2009, 02:17:30 pm
I am trying to let pidgin can do the video call and other features with WLM2009....
It's my mistake. Because I don't have webcam, so I use the videotestsrc instead of v4l2src, and it works with WLM when I set the rtp-payload-size only.
When I find the webcam and use v4l2src to do the video call, and it failed......
But finally it works again. :twisted:
I enable the property "do-timestamp" of v4l2src, and the WLM can see the video of my webcam, but it performance not so good, the video will stop frequently.
And then I look for the difference of v4l2src and videotestsrc. I find out the frame rate of v4l2src and videotestsrc is different, so I add a videorate to source as following (v4l2src -- videorate -- fsrtpconference), and set the frame rate to 30, and the performance looks better. I don't know why....
But the parameters still need to be refine.


Title: Audio/Video conversation
Post by: kakaroto on April 02, 2009, 05:01:19 pm
what :| that's not normal, I spent weeks on that, and I modified ffmpeg, gstreamer and all sorts of stuff and i couldn't find the problem! So you're saying that the rtp-payload-size was the only problem? damn! What WLM did you test it with? was it on Win XP? because I had a windows Vista to test with and maybe the behavior is different in Vista... And yes, I was also using videotestsrc (and I tested with a few different paterns in videotestsrc and I tested with a real cam too)
anyways, thanks for the info, i'll try again now!
By the way, do NOT use videorate, it will just duplicate all your buffers which is very bad considering you are in a live pipeline! The reception was very good for me, but I don't know about sending... I suggest you come talk to us in #farsight, maybe we'll be able to help!
By the way, maybe what made it work for you was different, maybe something in the SIP... could you compare what you do with what aMSN does and see if there's a difference? or allow me to test your code, where can I find it ?
Thanks! :)


Title: Audio/Video conversation
Post by: flomar34 on April 06, 2009, 06:42:41 pm
Hello,

Just to give you some news about my problems with audio call and MSNP18

With the last release.

Amsn didn't freeze anymore when i want't to try an audio call

with amsn => wlm9
and
wlm9 => amsn i can start an audio call but it hang up in the 4 or 5 seconds

the only difference is, as you said, with amsn => wlm9 it starts the audio with the video (before hanging up)

Thanks for all    :)


Title: Audio/Video conversation
Post by: kakaroto on April 07, 2009, 12:30:18 am
@flomar34: sorry if i didn't answer you, I found the bug last week that caused it to not work, and its fixed in SVN... it should now work correctly with sending/receiving the audio or video call with MSNP18. Just make sure you use the latest releases of farsight (0.0.8) and libnice (0.0.6).

@jones: I'm still waiting for more info from you!!!! I'm really stuck and I would gladly accept any help you could give me! Please contact me!


Title: Audio/Video conversation
Post by: jones on April 07, 2009, 04:14:33 pm
Sorry for the later reply.
The WLM that I test the video call is in XP, and I will try this in Vista later. Does that depend on the os?
I will try to use the aMSN to do the video call as soon as I can, and find the different. And then I will told you the result later.
Now in my implementation, I use the following send bin.
(v4l2src -(video/x-raw-yuv)- videorate -(video/x-raw-yuv,framerate=15/1)- videoscale -(video/x-raw-yuv,width=352,height=288)- ffmpegcolorspace) ---- fsrtpconference
and set the "do-timestamp" = true of v4l2src,  "rtp-payload-size" = 256 of ffenc_h263.
I know the videorate is just duplicate the frame but if we don't set the frame rate, the videotestsrc will generate a high bitrate data about 1MB/s...


Title: Audio/Video conversation
Post by: kakaroto on April 07, 2009, 05:19:08 pm
Ok jones, thanks, I'll have a look! By the way, I also tested with win XP and it made no difference, so I don't think it depends on the os!
I will try to use the same pipeline as you and see if it helps, although I doubt it.. we'll see!
If you could provide me with the source code of what you did, so i can do my own sniffing and see what difference there is, I should be able to figure it out hopefully.. It will allow me to work much much faster too...
By the way, you said you patched farsight, can we get the patch? maybe what you changed in farsight is what helped it work... Are you also using the normal gstreamer versions or did you patch the h263 payloader or something like that?
Could you also give me a list of what exact version of gstreamer, gst-plugins-base/good/bad/ugly/farsight and which version of farsight/libnice you are using ? Maybe the versions are different and I should try with the same versions as you...
Thanks!


Title: Audio/Video conversation
Post by: flomar34 on April 07, 2009, 06:17:48 pm
Hi,

Just to finish with this .
With the same version of amsn svn i've update to ffarsight2 0.0.9 and gst-plugins-farsight 0.12.11

When vista wml9 ask for a video call, amsn accept it and it works nicely
(video and audio)

Thanks a lot.
Maybe i shoult by a webcam now  :D


Title: Audio/Video conversation
Post by: kakaroto on April 07, 2009, 06:21:12 pm
flomar34! wait, what?! ok, I know that audio should work, and I know that video reception should work, but you're saying video and audio work nicely, are you saying in both directions? Or is WLM not receiving anything? Because it shouldn't be receiving anything (while we send but WLM refuses to show the cam)...
and if you have no webcam, it's ok, because right now amsn sends a virtual webcam (basically just a pattern image) for testing purposes...
please be clear on what exactly worked! Thanks!


Title: Audio/Video conversation
Post by: flomar34 on April 07, 2009, 06:43:12 pm
I have no webcam so of course wlm receive no video (désolé pour la fausse joie)

It's just that i recieve both video and audio (tres fluide).

And amsn didn't hang up anymore as i said yesterday


Title: Audio/Video conversation
Post by: kakaroto on April 07, 2009, 07:54:47 pm
ah ok... well, it should have actually sent video (une fausse video, juste une image statique en fait) even if you don't have one.. so it didn't show anything, means it still doens't work...
now all my hope is with jones! :)


Title: Audio/Video conversation
Post by: MastaG on April 08, 2009, 03:30:02 pm
I'm on Fedora 10 x64 and I have it updated
These are the latest packages in the repo's which I have installed:
gstreamer-0.10.21-2.fc10.x86_64
gstreamer-plugins-ugly-0.10.10-2.fc10.x86_64
gstreamer-plugins-flumpegdemux-0.10.15-4.fc10.x86_64
gstreamer-plugins-good-0.10.13-1.fc10.x86_64
gstreamer-plugins-good-devel-0.10.13-1.fc10.x86_64
gstreamer-plugins-base-devel-0.10.21-2.fc10.x86_64
gstreamer-tools-0.10.21-2.fc10.x86_64
gstreamer-python-0.10.12-1.fc10.x86_64
gstreamer-plugins-bad-extras-0.10.9-1.fc10.x86_64
gstreamer-plugins-farsight-0.12.9-3.fc10.x86_64
PackageKit-gstreamer-plugin-0.3.14-1.fc10.x86_64
gstreamer-plugins-base-0.10.21-2.fc10.x86_64
gstreamer-devel-0.10.21-2.fc10.x86_64
gstreamer-plugins-bad-devel-0.10.9-1.fc10.x86_64
gstreamer-ffmpeg-0.10.5-1.fc10.x86_64
gstreamer-plugins-bad-0.10.9-1.fc10.x86_64

libnice-0.0.6 compiles and installs fine with:
./configure --prefix=/usr --libdir=/usr/lib64 --sysconfdir=/etc --localstatedir=/var
make && sudo make install

However farsight2-0.0.8 and 0.0.9 dont compile because they complain about my gstreamer being too old.
So I´m using 0.0.7 compiled and installed with:
./configure --prefix=/usr --libdir=/usr/lib64 --sysconfdir=/etc --localstatedir=/var --disable-python
make && sudo make install

The latest svn of amsn builds fine even though I'm not using the latest version of farsight2.
Is it a big problem?
Since I don't feel like building the complete gstreamer suite just to satisfy farsight2.

The Audio/Video wizard in aMSN passes the audio conversation check.


Title: Audio/Video conversation
Post by: kakaroto on April 08, 2009, 06:32:00 pm
@MastaG: if you can make a call then why complain? :p
the newest farsight needs gstreamer 0.10.22, it's always better to use the latest versions available, it makes things more stable, better, etc.. but I think you can still with farsight2 0.0.7.
I've just updated today the wiki to state the new dependencies' versions.


Title: Audio/Video conversation
Post by: kakaroto on April 11, 2009, 03:48:07 am
@jones: hellooooo, come on! I'm still waiting to see what you did, I'm still unable to get it to work!


Title: Audio/Video conversation
Post by: jones on April 11, 2009, 05:04:40 am
Hi kakaroto,
I find some diffenenecs. I create two fsrtpconference, and aMSN create two session in the same fsrtpconference, if we use the same fsrtpconference, and the foundations of ICE candidates are not similar to WLM2009 exactly. For example, foundations will be the following: audio 1 3, video 2 4, not audio 1 2, video 1 2. Is it a problem, I don't know.....
By the way,  I get the svn version aMSN, and when I do the video call with WLM2009, WLM2009 can get the successful connect message, but aMSN doesn't show any video(both self preview and remote), is it normal? or I should make some config?


Title: Audio/Video conversation
Post by: kakaroto on April 11, 2009, 09:36:56 pm
ok jones, thanks, i'll try separating this into two fsrtpconference *for testing* .. but it should not be done this way, it has to be one conference with two sessions! It's necessary for audio/video synchronisation, RTCP, and it has to be the same SSRC, the concept of sessions is there for a reason... ICE processing also needs this for candidate nomination and speeding up the connection, etc...
Also the ICE foundations are just strings, they can be absolutely anything, so if it's "1" or "2" or "foobar" it doesn't make any difference.
Anyways, yes, if you try the SVN version, it will be only audio call, not video call that's why it won't show anything... You'd have to connect using MSNP18 if you want to do the video call.. open the amsn console with ctrl-shift-C from the main window and type
Code:
::config::setKey protocol 18
before connecting... then when you make an audio call it will actually be a video call too..
the preview window won't show though, only the remote video will be shown for you.

Anyways, jones, this is like the 4th time I'm asking for your work, everytime you say a little thing and it's not very helfpul.. I'm sure that even if I use two fsrtconference instead of one, it still won't work, then i'll ask you again about the code and you still will only tell me what difference you see, etc... and we'll never finish!
Jones, can you please send me the source code of what you've done, you said it's for pidgin, which is open source software anyways, so the license makes it that it has to stay open source and you must provide the source code... so it's not like you're doing it for some company who wants the code to stay closed source...
if you don't want to make your work public yet (you want it to be finished/stable/etc... before releasing it), then I can understand, no problem then, I wouldn't give the code to anyone, so if you could just send it to me directly by email, I would only use it to find what I'm doing wrong. This will make things 100 times easier/faster....
You can mail me at : kakaroto  AT users.sourceforge.net

Thanks!


Title: Audio/Video conversation
Post by: zyazhou on April 13, 2009, 10:35:23 am
Hi, kakaroto
I'm updating my own client to support MSN2009 audio. But something wrong happpens after ice connectivity check. MSN2009 side can hear my audio, but I cannot hear audio from MSN2009. Furthermore, after about 30 seconds MSN2009 hang up this call. I only send Binding Request to MSN2009 TURN candidate(not local or stun candiate, I want to make it simple first), and can receive Bindng Response from this candidate. I don't know what's going wrong. Could you give me some help? Thanks in advance.

By the way, MSN2009 is behind a symmetric NAT, and my client is on a public IP. As my understanding, if I only use TURN candidate, there should be no any deprived candidate for symmetric NAT reason. That's why I want to make it simple first.


Title: Audio/Video conversation
Post by: jones on April 13, 2009, 05:12:09 pm
Hi kakaroto,
Sorry about the source code, there are some reasons....so.....
But I find a mistake of the UUN command that aMSN sent.
The UUN command need to carry the machine guid except the first sip invite message if the role is inviter. Because the MPOP  feature, we should assign the machine guid.
Example: "UUN TrID xxx@mail;{XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXX} 12 length"
But aMSN always not carry the machine guid, but after I add the machine guid, it still won't work.

And I find out that you mean it won't work is the WLï¼­ shows the no webcam picture not the gray frame...
so I think it may have some problems in sip messages not in pipeline,
I assume that WLM get the gray video at all times.....because when I don't set the do-timestamp and rtp-payload-size, the WLM will show the gray frame not the no webcam..

Anyways, I will keep to find the issue of aMSN...


Title: Audio/Video conversation
Post by: kakaroto on April 13, 2009, 06:07:30 pm
@zyazhou: Make sure that the ICE connectivity checks worked correctly and that both amsn and WLM say that it was connected. Also make sure that the volume is correctly set so you can hear stuff and try testing the microphone on WLM...
About the 30 seconds timeout, it was a bug in libnice at some point, so make sure you are using the 0.0.6 release.

@jones: Ok, thanks for the help! I understand your issue about the source code, but if you could be nice enough to send me the source code in private by email, I'll make sure to keep it for myself and I won't release it to anyone, I'll only use it to test and fix aMSN. But thanks anyway for trying to find the issue with aMSN and helping in fixing it!
Yes, I've noticed that it shows the no webcam icon.. I thought it was because it couldn't decode the h263 and showed it, but I also tried without sending any video at all and it still showed that icon so it's not be the stream itself.. I'm pretty sure it's the SIP signaling that is wrong somewhere but I couldn't find any differences... I'll try to fix the UUN too and see if it changes anything. I also tried sending the UUN 11 stuff too (where it sends "5 1 1" with "11" instead of "12" in the UUN line) once the INVITE/200 is sent, but it didn't help.. do you send that stuff too ?
I noticed that if I try with a WLM that doesn't have a webcam, it shows the same icon, so I'm thinking maybe my signaling is telling WLM somehow that I don't have a webcam... but I wasn't able to see any difference between the signaling when WLM has or when it doesn't have a webcam plugged in, but there's obviously something there to tell it "I have no webcam".
I'll continue looking at this tomorrow, hopefully, I'll find something.. otherwise, I'll be waiting for your code...
If you really *really* can't send me the source code, even though I'll keep it private, then maybe try emailing me both the protocol log and a wireshark dump, so at least I can compare with that!
Thank you again! :)


Title: Audio/Video conversation
Post by: kakaroto on April 15, 2009, 01:02:29 am
Hey Jones! Thanks for all the help, in the end, I found out what the problem was.. I just wasted weeks on this... the solution was SO simple...
The clientid needed the 'HasWebcam' capability enabled... So if you actually had aMSN configured to show your webcam (so others can see a little 'webcam' icon next to your name), then the video conference with H263 would have worked.. if you didn't have it, then the 'no webcam' icon would be shown...


Title: Audio/Video conversation
Post by: MastaG on April 16, 2009, 07:00:38 am
Hi kakarot, nice work man:)
I've updated my fedora to rawhide and now ships with libnice-0.0.6 and farsight2-0.0.9.
I'll give the video-chat a try and report here as soon as possible.
BTW: Why isnt msnp18 the default protocol for aMSN?


Title: Audio/Video conversation
Post by: kakaroto on April 16, 2009, 09:53:21 am
You're welcome!
MSNP18 isn't enabled by default because MPOP support is not stable and more importantly, it's because the P2P stack is completely different and not yet reverse engineered... so all MSNP2P stuff won't work (display picture, custom emoticons, file transfer, etc..)


Title: Audio/Video conversation
Post by: flomar34 on April 16, 2009, 10:05:55 am
Hello,

I still have no webcam :)

With revision 11138
amsn call wlm 9
amsn recieve webcam and audio
wlm9 don't recieve audio

With revision 11143
amsn call wlm 9
amsn don't recieve audio, recieve an video screen with something like a "mire" : some colors like if it can't decode it. But The video screen appear before wlm9 accept the video call
amsn don't recieve sound
wlm9 don't recieve audio


I've made the tests with libnice 6 and farsight 9.


Title: Audio/Video conversation
Post by: MastaG on April 16, 2009, 10:16:58 am
Quote from: "kakaroto"
You're welcome!
MSNP18 isn't enabled by default because MPOP support is not stable and more importantly, it's because the P2P stack is completely different and not yet reverse engineered... so all MSNP2P stuff won't work (display picture, custom emoticons, file transfer, etc..)


I always thought MS were forced to give the specs of their protocol.
Doesn't the European law state something like that?
So they can't have a monopoly position on their messenger.


Title: Audio/Video conversation
Post by: jones on April 16, 2009, 01:23:43 pm
Oh...I just want to check the clientid and than you found it, both we forget the clientid....
But congratulation, it works eventually.
By the way, the new p2p protocol is not so friendly......

Quote from: "kakaroto"
Hey Jones! Thanks for all the help, in the end, I found out what the problem was.. I just wasted weeks on this... the solution was SO simple...
The clientid needed the 'HasWebcam' capability enabled... So if you actually had aMSN configured to show your webcam (so others can see a little 'webcam' icon next to your name), then the video conference with H263 would have worked.. if you didn't have it, then the 'no webcam' icon would be shown...


Title: Audio/Video conversation
Post by: kakaroto on April 16, 2009, 08:56:32 pm
@flomar34: You need 11143 indeed, the 'mire' is actually what aMSN will be sending, that window that pops up before WLM accepts is your "preview", this is what I have set for now as a virtual webcam...
so once you do the call, video works? just audio doesn't? then maybe the microphones aren't correct.. you may need to configure which mic to use, use oss or alsa, etc... try in a terminal 'gstreamer-properties' to set up your audio stuff before doing the call...
@MastaG: only for monopols.. so Office.. they give all the specs for ICE, TURN, SIP, etc.. because it's part of Office Communicator, but they don't give specs about the MSN protocol... (also they have small differences between Office Communicator and MSN, even if it's the same dll they use I guess). Also their specs aren't really that good.. and finally, MSN isn't a monopoly, there's yahoo, ICQ, jabber, blablablabla, there are tons of messengers and they don't have a monopoly (in china for example, noone uses msn!)
@jones: hehe, yeah, it finally worked! and I was SO mad that the issue was just with the clientid.. but anyways, now it works and that's all that matters!
About P2PV2, I know, it's not easy, but you can find some reverse engineering info here : http://forums.fanatic.net.nz/index.php?showtopic=19372&st=0&gopid=108224
If you can figure out some missing parts and you can write some documentation on how it all works, that would be very helpful! I'm also soon going to start reverse engineering it!
If you want, we can work together, if you don't mind, we could add each other to MSN.. I'll PM you with my MSN address, add me if you want!


Title: Audio/Video conversation
Post by: flomar34 on April 17, 2009, 07:54:55 am
Hi,

Thanks kakaroto for your help.
For the audio it needed gstreamer-pulseaudio even if pulseaudio is not installed on my xfce. With gstreamer-alsa it didn't work.

Pour la video, avec la version 11143 je ne recois de wlm que la mire(ni video, ni audio)(wlm lui, recoit bien l'audio). En reception, Ceci marchait tres bien avec la revision 11138 (je viens d'en refaire le test)

Quoi qu'il en soit, si cela marche pour les possesseurs de webcam, je suis super heureux, c'est  vraiment un tres grand jour pour le libre et amsn!!!

Sinceres félicitations

Sorry billiob for the french here  :D

Code:
[09:06:16] CallInviteUser xxxxxxxx@hotmail.fr
[09:06:16] User xxxxxxxx@hotmail.fr supports SIP
[09:06:16] MSNSIP : Inviting user xxxxxxxx@hotmail.fr to a SIP call
[09:06:16] Sending Tunneled SIP invite to xxxxxxxxxxxxxxxx@hotmail.fr
[09:06:16] Farsight : Preparing
[09:06:17] TURN: Connecting
[09:06:17] TURN: Sending [00][02][02]|H[17][08]c[90][fa][95][16][ba][9f](p}z[ad][1a][00][06][02]xRPS_t=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&p=[00][00][00]
[09:06:18] TURN: Sending [00][02][02]|[a4][af][b1][ee][0b][dd][88][df][90][9d]O[9c][04][fc][16]$[00][06][02]xRPS_t=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&p=[00][00][00]
[09:06:18] TURN: Received message of type SHARED-SECRET-ERROR
[09:06:18] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00])23176562;BE2A7DEC69C2888B609AAE0C58756A1F[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[09:06:18] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:06:18] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[09:06:18] TURN: Received attribute REALM : {0x32, 0x33, 0x31, 0x37,
0x36, 0x35, 0x36, 0x32,
0x3b, 0x42, 0x45, 0x32,
0x41, 0x37, 0x44, 0x45,
0x43, 0x36, 0x39, 0x43,
0x32, 0x38, 0x38, 0x38,
0x42, 0x36, 0x30, 0x39,
0x41, 0x41, 0x45, 0x30,
0x43, 0x35, 0x38, 0x37,
0x35, 0x36, 0x41, 0x31,
0x46}
[09:06:18] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[09:06:18] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[09:06:18] TURN: Received response SHARED-SECRET-ERROR for id 4817086390fa9516ba9f28707d7aad1a
[09:06:18] TURN: Parsing MAGIC-COOKIE
[09:06:18] TURN: Parsing NONCE
[09:06:18] TURN: Parsing REALM
[09:06:18] TURN: Parsing ERROR-CODE
[09:06:18] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[09:06:18] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][df][91][03][c3][a1][c0][16]d[d8][82][c0]
>[99][f1][09][b2][00][06][02]xRPS_t=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&p=[00][00][00][00][14][00])23176562;BE2A7DEC69C2888B609AAE0C58756A1F[00][15][00][1a]"voice.messenger.live.com"
[09:06:18] TURN: Sending [00][02][02][df][91][03][c3][a1][c0][16]d[d8][82][c0]
>[99][f1][09][b2][00][06][02]xRPS_t=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&p=[00][00][00][00][14][00])23176562;BE2A7DEC69C2888B609AAE0C58756A1F[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][b2][d9][fe][cb][c7]2o[f7]P[17]cE[a4][13][88][9a][b5][05]@[a9]
[09:06:18] TURN: Received message of type SHARED-SECRET-ERROR
[09:06:18] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00])23176687;21139E5CF69D5A6730ABFAF9D501477A[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[09:06:18] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:06:18] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[09:06:18] TURN: Received attribute REALM : {0x32, 0x33, 0x31, 0x37,
0x36, 0x36, 0x38, 0x37,
0x3b, 0x32, 0x31, 0x31,
0x33, 0x39, 0x45, 0x35,
0x43, 0x46, 0x36, 0x39,
0x44, 0x35, 0x41, 0x36,
0x37, 0x33, 0x30, 0x41,
0x42, 0x46, 0x41, 0x46,
0x39, 0x44, 0x35, 0x30,
0x31, 0x34, 0x37, 0x37,
0x41}
[09:06:18] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[09:06:18] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[09:06:18] TURN: Received response SHARED-SECRET-ERROR for id a4afb1ee0bdd88df909d4f9c04fc1624
[09:06:18] TURN: Parsing MAGIC-COOKIE
[09:06:18] TURN: Parsing NONCE
[09:06:18] TURN: Parsing REALM
[09:06:18] TURN: Parsing ERROR-CODE
[09:06:18] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[09:06:18] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][df][d5][a0]
l[bf][de][fe]<[c3]qX[14]x[bc][db][b4][00][06][02]xRPS_t=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&p=[00][00][00][00][14][00])23176687;21139E5CF69D5A6730ABFAF9D501477A[00][15][00][1a]"voice.messenger.live.com"
[09:06:19] TURN: Sending [00][02][02][df][d5][a0]
l[bf][de][fe]<[c3]qX[14]x[bc][db][b4][00][06][02]xRPS_t=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&p=[00][00][00][00][14][00])23176687;21139E5CF69D5A6730ABFAF9D501477A[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][ca][90][c7][f1][e9][c2][ac][8c]V[ba][1e]%[e0]y[18][d9][b7]7[05]=
[09:06:19] TURN: Received message of type SHARED-SECRET-RESPONSE
[09:06:19] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$I[aa][c5][ac][88]Zf/[b4][a7]a[01][f3]C[d5][13][e4][16][0f]c[c8][a0]q-[ba]V7#[eb]&[d8][87][c8][92]E[e6][00][07][00][10]IM[c3][e6][dc]x[81][94][8f]-[90][c2][f0][99][e6][95][00][0e][00][08][00][01][00][00][d5][c7][a3]&
[09:06:19] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:06:19] TURN: Received attribute USERNAME : {0x49, 0xaa, 0xc5, 0xac,
0x88, 0x5a, 0x66, 0x2f,
0xb4, 0xa7, 0x61, 0x01,
0xf3, 0x43, 0xd5, 0x13,
0xe4, 0x16, 0x0f, 0x63,
0xc8, 0xa0, 0x71, 0x2d,
0xba, 0x56, 0x37, 0x23,
0xeb, 0x26, 0xd8, 0x87,
0xc8, 0x92, 0x45, 0xe6}
[09:06:19] TURN: Received attribute PASSWORD : {0x49, 0x4d, 0xc3, 0xe6,
0xdc, 0x78, 0x81, 0x94,
0x8f, 0x2d, 0x90, 0xc2,
0xf0, 0x99, 0xe6, 0x95}
[09:06:19] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x26}
[09:06:19] TURN: Received response SHARED-SECRET-RESPONSE for id 9103c3a1c01664d882c00d3e99f109b2
[09:06:19] TURN: TURN server xx.xx.xx.xx.38 : 0
[09:06:20] TURN: Received message of type SHARED-SECRET-RESPONSE
[09:06:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$aL[a8][86][14][f9]-%[8f][a8]a[01][f3]C[d5][13][e4][16][0f]cO[d9][c0]Fh%3+[c5][fd]U[bb][a6][b7][a4]
[00][07][00][10][03]$[b9][d3][0c]=~I[dd]sw[aa]Fh][93][00][0e][00][08][00][01][00][00][d5][c7][a3]&
[09:06:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:06:20] TURN: Received attribute USERNAME : {0x61, 0x4c, 0xa8, 0x86,
0x14, 0xf9, 0x2d, 0x25,
0x8f, 0xa8, 0x61, 0x01,
0xf3, 0x43, 0xd5, 0x13,
0xe4, 0x16, 0x0f, 0x63,
0x4f, 0xd9, 0xc0, 0x46,
0x68, 0x25, 0x33, 0x2b,
0xc5, 0xfd, 0x55, 0xbb,
0xa6, 0xb7, 0xa4, 0x0d}
[09:06:20] TURN: Received attribute PASSWORD : {0x03, 0x24, 0xb9, 0xd3,
0x0c, 0x3d, 0x7e, 0x49,
0xdd, 0x73, 0x77, 0xaa,
0x46, 0x68, 0x5d, 0x93}
[09:06:20] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x26}
[09:06:20] TURN: Received response SHARED-SECRET-RESPONSE for id d5a00a6cbfdefe3cc371581478bcdbb4
[09:06:20] TURN: TURN server xx.xx.xx.xx.38 : 0
[09:06:20] TURN: Disconnecting
[09:06:20] Turn prepared {xx.xx.xx.xx.38 0 SarFrIhaZi+0p2EB80PVE+QWD2PIoHEtulY3I+sm2IfIkkXm SU3D5tx4gZSPLZDC8JnmlQ== 1 udp} {xx.xx.xx.xx.38 0 YUyohhT5LSWPqGEB80PVE+QWD2NP2cBGaCUzK8X9Vbumt6QN AyS50ww9fkndc3eqRmhdkw== 2 udp}
[09:06:20] Farsight debug : AUDIO CODECS ARE READY
[09:06:20] Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
[09:06:20] Farsight debug : Testing source dshowaudiosrc
[09:06:20] Farsight debug : Testing source directsoundsrc
[09:06:20] Farsight debug : Testing source osxaudiosrc
[09:06:20] Farsight debug : Testing source gconfaudiosrc
[09:06:20] Farsight debug : Using audio_source gconfaudiosrc
[09:06:20] Farsight debug : stun ip : 64.14.48.28 : 3478
[09:06:20] Farsight debug : FS: relay info = 0x9c095a0 - 2
[09:06:20] Farsight debug : Creating video_source : videotestsrc is-live=true ! video/x-raw-yuv,width=352,height=288 ! ffmpegcolorspace ! tee name="t" t. ! autovideosink t. ! identity  --- (null) -- (null)
[09:06:20] Farsight debug : stun ip : 64.14.48.28 : 3478
[09:06:20] Farsight debug : FS: relay info = 0x9c095a0 - 2
[09:06:21] Creating CW Voip controls
[09:06:21] TURN: Disconnecting
[09:06:25] Farsight debug : AUDIO CANDIDATES ARE PREPARED
[09:06:25] Farsight : Farsight audio is now prepared!
local codecs : {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.168.3.2 52186 {} 0 UDP 2013266431 host e169 6agFBhLq40sz2KlXlAnyh+} {3 1 xx.xx.xx.xx 52186 192.168.3.2 52186 UDP 1677721855 srflx e169 6agFBhLq40sz2KlXlAnyh+} {4 1 xx.xx.xx.xx.38 37858 192.168.3.2 52186 UDP 1006633215 relay e169 6agFBhLq40sz2KlXlAnyh+} {1 2 192.168.3.2 33325 {} 0 UDP 2013266430 host e169 6agFBhLq40sz2KlXlAnyh+} {3 2 xx.xx.xx.xx 33325 192.168.3.2 33325 UDP 1677721854 srflx e169 6agFBhLq40sz2KlXlAnyh+} {4 2 xx.xx.xx.xx.38 37859 192.168.3.2 33325 UDP 1006633214 relay e169 6agFBhLq40sz2KlXlAnyh+}
[09:06:25] Fasight :Waiting for video preparation
[09:06:25] Farsight debug : VIDEO CANDIDATES ARE PREPARED
[09:06:25] Farsight debug : Setting window id 0 on sink
[09:06:26] Farsight debug : VIDEO CODECS ARE READY
[09:06:26] Farsight : Farsight Video is now prepared!
local codecs : {H263 34 90000}
local candidates : {2 1 192.168.3.2 37143 {} 0 UDP 2013266431 host U7Ye DFEfWyrw3OTqiSQc3gLcZK} {5 1 xx.xx.xx.xx 37143 192.168.3.2 37143 UDP 1677721855 srflx U7Ye DFEfWyrw3OTqiSQc3gLcZK} {2 2 192.168.3.2 47636 {} 0 UDP 2013266430 host U7Ye DFEfWyrw3OTqiSQc3gLcZK} {5 2 xx.xx.xx.xx 47636 192.168.3.2 47636 UDP 1677721854 srflx U7Ye DFEfWyrw3OTqiSQc3gLcZK}
[09:06:26] SIP : Registering : REGISTERED
[09:06:26] SIP : Sending Invite
[09:06:26] SIP invite sent
[09:06:26] Got unknown NS input!! --> UUN
UUN 21 OK
[09:06:27] Got Tunneled SIP invite from xxxxxxxx@hotmail.fr;{703f5c66-3559-4136-911d-58ec4a074d1e}
[09:06:27] Received INVITE response
[09:06:27] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection1 xxxxxxxx@hotmail.fr 7028cfec4fbe752177c383799af873c2 TRYING
[09:06:28] Got Tunneled SIP invite from xxxxxxxx@hotmail.fr;{703f5c66-3559-4136-911d-58ec4a074d1e}
[09:06:28] Received INVITE response
[09:06:28] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection1 xxxxxxxx@hotmail.fr 7028cfec4fbe752177c383799af873c2 RINGING
[09:06:32] Got Tunneled SIP invite from xxxxxxxx@hotmail.fr;{703f5c66-3559-4136-911d-58ec4a074d1e}
[09:06:32] Received INVITE response
[09:06:32] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection1 xxxxxxxx@hotmail.fr 7028cfec4fbe752177c383799af873c2 OK
[09:06:32] SIP callee accepted our call
[09:06:33] Farsight starting : {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000} - {1 1 192.168.3.3 20235 {} 0 UDP 2130706431 host CewI w7C0nrDuzYmYq9UOt3+BMVNa} {1 2 192.168.3.3 27707 {} 0 UDP 2130705918 host CewI w7C0nrDuzYmYq9UOt3+BMVNa} {2 1 xx.xx.xx.xx.72 52063 xx.xx.xx.xx.72 52063 UDP 16647679 relay CewI w7C0nrDuzYmYq9UOt3+BMVNa} {2 2 xx.xx.xx.xx.72 52064 xx.xx.xx.xx.72 52064 UDP 16647678 relay CewI w7C0nrDuzYmYq9UOt3+BMVNa} {4 1 xx.xx.xx.xx 19210 192.168.3.3 19210 UDP 1694235647 srflx CewI w7C0nrDuzYmYq9UOt3+BMVNa} {4 2 xx.xx.xx.xx 29140 192.168.3.3 29140 UDP 1694235134 srflx CewI w7C0nrDuzYmYq9UOt3+BMVNa} -- {H263 34 90000} - {1 1 192.168.3.3 9655 {} 0 UDP 2130706431 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P} {1 2 192.168.3.3 9890 {} 0 UDP 2130705918 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P} {2 1 xx.xx.xx.xx.46 43324 xx.xx.xx.xx.46 43324 UDP 16647679 relay bcYd IG9WCMSF8s3maGjv9Pi/FD1P} {2 2 xx.xx.xx.xx.46 43325 xx.xx.xx.xx.46 43325 UDP 16647678 relay bcYd IG9WCMSF8s3maGjv9Pi/FD1P} {4 1 xx.xx.xx.xx 28253 192.168.3.3 28253 UDP 1694235647 srflx bcYd IG9WCMSF8s3maGjv9Pi/FD1P} {4 2 xx.xx.xx.xx 30701 192.168.3.3 30701 UDP 1694235134 srflx bcYd IG9WCMSF8s3maGjv9Pi/FD1P}
[09:06:33] Farsight debug : New remote audio codec : 8 PCMA 8000
[09:06:33] Farsight debug : New remote audio codec : 0 PCMU 8000
[09:06:33] Farsight debug : New remote audio codec : 101 telephone-event 8000
[09:06:33] Farsight debug : New remote video codec : 34 H263 90000
[09:06:33] Farsight debug : New Remote candidate: 1 1 192.168.3.3 20235 - 0 UDP 2130706431 host CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 1 2 192.168.3.3 27707 - 0 UDP 2130705918 host CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 2 1 xx.xx.xx.xx.72 52063 xx.xx.xx.xx.72 52063 UDP 16647679 relay CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 2 2 xx.xx.xx.xx.72 52064 xx.xx.xx.xx.72 52064 UDP 16647678 relay CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 4 1 xx.xx.xx.xx 19210 192.168.3.3 19210 UDP 1694235647 srflx CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 4 2 xx.xx.xx.xx 29140 192.168.3.3 29140 UDP 1694235134 srflx CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New Remote candidate: 1 1 192.168.3.3 9655 - 0 UDP 2130706431 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New Remote candidate: 1 2 192.168.3.3 9890 - 0 UDP 2130705918 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New Remote candidate: 2 1 xx.xx.xx.xx.46 43324 xx.xx.xx.xx.46 43324 UDP 16647679 relay bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New Remote candidate: 2 2 xx.xx.xx.xx.46 43325 xx.xx.xx.xx.46 43325 UDP 16647678 relay bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New Remote candidate: 4 1 xx.xx.xx.xx 28253 192.168.3.3 28253 UDP 1694235647 srflx bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New Remote candidate: 4 2 xx.xx.xx.xx 30701 192.168.3.3 30701 UDP 1694235134 srflx bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Got unknown NS input!! --> UUN
UUN 22 OK
[09:06:33] Farsight debug : New active candidate pair (audio) :
[09:06:33] Farsight debug : Local candidate: 1 1 192.168.3.2 52186 - 0 UDP 2013266431 host e169 6agFBhLq40sz2KlXlAnyh+
[09:06:33] Farsight debug : Remote candidate: 1 1 192.168.3.3 20235 - 0 UDP 2130706431 host CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight debug : New active candidate pair (audio) :
[09:06:33] Farsight debug : Local candidate: 1 2 192.168.3.2 33325 - 0 UDP 2013266430 host e169 6agFBhLq40sz2KlXlAnyh+
[09:06:33] Farsight debug : Remote candidate: 1 2 192.168.3.3 27707 - 0 UDP 2130705918 host CewI w7C0nrDuzYmYq9UOt3+BMVNa
[09:06:33] Farsight : New AUDIO_ACTIVE active candidate pair
[09:06:33] Farsight debug : New active candidate pair (video) :
[09:06:33] Farsight debug : Local candidate: 2 1 192.168.3.2 37143 - 0 UDP 2013266431 host U7Ye DFEfWyrw3OTqiSQc3gLcZK
[09:06:33] Farsight debug : Remote candidate: 1 1 192.168.3.3 9655 - 0 UDP 2130706431 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight debug : New active candidate pair (video) :
[09:06:33] Farsight debug : Local candidate: 2 2 192.168.3.2 47636 - 0 UDP 2013266430 host U7Ye DFEfWyrw3OTqiSQc3gLcZK
[09:06:33] Farsight debug : Remote candidate: 1 2 192.168.3.3 9890 - 0 UDP 2130705918 host bcYd IG9WCMSF8s3maGjv9Pi/FD1P
[09:06:33] Farsight : New VIDEO_ACTIVE active candidate pair
[09:06:34] Got unknown NS input!! --> UUN
UUN 23 OK
[09:06:34] Got Tunneled SIP invite from xxxxxxxx@hotmail.fr;{703f5c66-3559-4136-911d-58ec4a074d1e}
[09:06:34] Received INVITE response
[09:06:35] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection1 xxxxxxxx@hotmail.fr 7028cfec4fbe752177c383799af873c2 TRYING
[09:06:35] Got Tunneled SIP invite from xxxxxxxx@hotmail.fr;{703f5c66-3559-4136-911d-58ec4a074d1e}
[09:06:35] Received INVITE response
[09:06:36] Got unknown NS input!! --> UUN
UUN 24 OK


[/code]


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2009, 04:40:18 pm
@flomar:
Quote
Pour la video, avec la version 11143 je ne recois de wlm que la mire(ni video, ni audio)(wlm lui, recoit bien l'audio).

hummm... la mire, c'est aMSN qui te l'affiche (c'est genre ton 'preview'...) si tu dis que ni la video ni l'audio marche alors il y'a un probleme.. ton status log montre que vous vous connectez bien en fait tant pour l'audio que pour la video...
il faut peut-etre configurer pour que l'audio fonctionne correctement, test bien tant le micro que les hauts parleurs avec la commande gstreamer-properties. Pour la video, je sais pas... ca devrait juste fonctionner la...


Title: Audio/Video conversation
Post by: flomar34 on April 17, 2009, 04:49:13 pm
Merci,

Une derniere question et j'arrette de "polluer" ce topic. (promis)
Si je redescend a la revision 11138 amsn recois tout (audio +video) lors de l'appel vision. si je remonte  la 11143 (amsn ne recoit plus rien pour la visio)
Est ce que cela peut me donner un indice sur ce qui me manque?
Probleme d'ouverture de port different, codec different, une version de gstreamer <> ... ?


Title: Audio/Video conversation
Post by: jones on April 17, 2009, 05:43:08 pm
Hi kakaroto,
I'd like to work with you, we can share the information and do a good job.
By the way, the informations of this website are very useful, thanks.


Title: Audio/Video conversation
Post by: kakaroto on April 17, 2009, 09:08:43 pm
@flomar34: humm ok, je sais peut-etre pourquoi.. mais j'ai pas le temps de m'en occuper la, je verrais ca apres... le probleme c'est peut-etre juste un "3" qui devrait etre un "2" dans le SIP... mais ca fera bugger autre chose.. je corrigerais ca de la bonne maniere apres...
@jones: I sent you a PM, working together would be great! :)


Title: Audio/Video conversation
Post by: flomar34 on May 16, 2009, 05:57:17 pm
Hi,

Now i get back to msnp15

If wlm make the call it works nicely
If amsn initiate the call it hangs up immediately

Code:

[18:51:29] Received INVITE response
[18:51:29] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 darkxxxx@hotmail.fr 46970a0123f70fed55f403e6a08f5178 TRYING
[18:51:30] Received INVITE response
[18:51:30] SIP : Registering : REGISTERED
[18:51:30] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 darkxxxx@hotmail.fr 46970a0123f70fed55f403e6a08f5178 ERROR 500 Server Internal Error
[18:51:30] SIP call ended
[18:51:30] Removing CW Voip controls
[18:51:31] MSNSIP : Destroying ::MSNSIP::SIPConnection3
[18:51:31] SIP : Unregistering
[18:51:31] Received INVITE response
[18:51:31] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 darkdarkxxxx@hotmail.fr 46970a0123f70fed55f403e6a08f5178 CLOSED LOCAL_BYE
[18:51:31] SIP call ended


Im using the last svn
Is there something that i can try?

Thanks


Title: Apple users audio with msn
Post by: smiles on May 18, 2009, 06:09:24 am
I have windows live messenger and my friend has Apple Mac. With your download, which he did last year, we are now able to webcam but not mic. Is a download now available so we can start actually talking instead of typing? If so please tell me what he needs to do to get this going? Would make my day !!! :D Thank you again for all you do to help all of us with a computer to communicate. Unlike Microsoft you guys are very helpful. Anything you comment to them about regarding msn or windows is just not actually realllllllllll :?


Title: Audio/Video conversation
Post by: kakaroto on May 19, 2009, 06:38:45 pm
@flomar: I think it's the same issue i answered you about before, just be patient...
@smiles: i already answered you in the other thread, you don't need to ask the same thing more than once...


Title: Audio/Video conversation
Post by: smiles on May 19, 2009, 11:32:44 pm
Sorry yes you did. I typed it in here first by accident then realised I did it in wrong area then typed where you responded. Was a genuine mistake here.. Sorry :-)


Title: Audio/Video conversation
Post by: kakaroto on May 20, 2009, 02:29:15 am
no problem then...


Title: Audio/Video conversation
Post by: ranmori on May 28, 2009, 08:14:06 am
Thanks for the feedback. That is very useful.
Regards,:D
simulation credit auto (http://simulationcreditauto.net/)


Title: Audio/Video conversation
Post by: flomar34 on May 28, 2009, 05:39:19 pm
Hi,
i'm not sure that my feedback are interesting or not for you, if it's not tell me please.

With the last svn it's very better for me

With no webcam

Wlm ask for audio call it's workink
wlm ask for video call ....... it's working  :)

That's great !

(in the other way it's still not working but not so important)

Thanks, very great job


Title: Audio/Video conversation
Post by: kakaroto on May 28, 2009, 09:45:10 pm
flomar: thanks, I found out why sending a call doesn't work, i'll fix it, don't worry! just.. not now, hehe


Title: Audio/Video conversation
Post by: flomar34 on June 09, 2009, 06:20:20 pm
Hi,

With the last update everything is working for me even if  amsn is making the call

Everything is perfect now

congratulation   :)


Title: Audio/Video conversation
Post by: kakaroto on June 10, 2009, 01:37:11 am
oh yes, sure, it's supposed to work now! I forgot to tell you i've fixed it!
Thanks for giving me the feedback! :)


Title: Audio/Video conversation
Post by: marc2009 on June 12, 2009, 03:03:12 pm
Hi all ! :D

I've a problem : I installed yesterday fedora 11, and this morning, amsn ( from the svn ) ... but when I launch the audio/video assistant ( CTRL + N ), amsn doesn't detect my sound card ! :? I mean there is nothing in both "choices".. it's the third and fourth screen. I hope you understood me ^^

Thanks

ps : I'm using pulseaudio


Title: Audio/Video conversation
Post by: kakaroto on June 12, 2009, 08:55:12 pm
the screens are dependent on what your system has, so it doesn't mean anything.. anyways, the assistant doesn't allow you to choose your audio setup (yet) for farsight, those choices are for libsnack (for voice clips) only, so it's irrelevant.. for audio calls, I suggest you use 'gstreamer-properties' or your gnome sound panel for now...


Title: Audio/Video conversation
Post by: marc2009 on June 13, 2009, 09:35:47 am
Quote from: "kakaroto"
the screens are dependent on what your system has, so it doesn't mean anything.. anyways, the assistant doesn't allow you to choose your audio setup (yet) for farsight, those choices are for libsnack (for voice clips) only, so it's irrelevant.. for audio calls, I suggest you use 'gstreamer-properties' or your gnome sound panel for now...


No, you don't understand ! ( it would be easier in French )

You open the audio/video assistant by typing ctrl + N, right ?

On the screen of the third and fourth next, you can configure your audio, and the video ... right ? And about the audio, there's nothing in the devices list ! and when I click on play, it tells me "could not gain access to /dev/sound/dsp for writing" !

And when I launch amsn from the terminal ( so when I type amsn ) : Unable to open mixer /dev/mixer

Thanks :D


Title: Audio/Video conversation
Post by: kjir on June 13, 2009, 12:07:39 pm
There is a french forum, if you want.
Basically what is happening is that you have the OSS devices already in use by pulseaudio I guess, but it is not related to the new system that uses gstreamer/farsight, so you're in the wrong thread ;)


Title: Audio/Video conversation
Post by: kakaroto on June 14, 2009, 06:55:42 pm
marc2009, I perfectly understood you, but it looks like you didn't understand my answer....


Title: Audio/Video conversation
Post by: marc2009 on June 15, 2009, 12:36:49 pm
Quote from: "kakaroto"
marc2009, I perfectly understood you, but it looks like you didn't understand my answer....


I would have asked you to repeat, but as kjir said, it's not the right forum  :D  :D I posted in the "linux" thread ...


Title: Audio/Video conversation
Post by: kakaroto on June 19, 2009, 02:34:12 am
Hello again! Just a quick note!
The latest SVN revision should be able to send and receive Audio+Video calls Using MSNP15.. so you don't need to go to MSNP18 anymore!
Enjoy! :)

p.s: please test and start reporting bugs!


Title: Not working on Fedora 11 x86_64
Post by: MastaG on June 22, 2009, 09:26:04 am
I'm running Fedora 11 x64 and having some issues.
F11 comes with the following packages:
Code:

gstreamer-plugins-good-0.10.15-1.fc11.x86_64
gstreamer-ffmpeg-0.10.7-2.fc11.x86_64
gstreamer-python-0.10.14-2.fc11.x86_64
gstreamer-tools-0.10.23-2.fc11.x86_64
gstreamer-plugins-flumpegdemux-0.10.15-6.fc11.x86_64
gstreamer-0.10.23-2.fc11.x86_64
PackageKit-gstreamer-plugin-0.4.8-1.fc11.x86_64
gstreamer-plugins-farsight-0.12.10-2.fc11.x86_64
gstreamer-plugins-bad-extras-0.10.12-2.fc11.x86_64
gstreamer-plugins-good-devel-0.10.15-1.fc11.x86_64
gstreamer-plugins-bad-devel-0.10.12-2.fc11.x86_64
gstreamer-devel-0.10.23-2.fc11.x86_64
gstreamer-plugins-base-0.10.23-1.fc11.x86_64
gstreamer-plugins-bad-0.10.12-2.fc11.x86_64
gstreamer-plugins-base-devel-0.10.23-1.fc11.x86_64
gstreamer-plugins-ugly-0.10.11-2.fc11.x86_64


So I started up with libnice-0.0.8
Configured with: ./autogen.sh --prefix=/usr --libdir=/usr/lib64 --sysconfdir=/etc --localstatedir=/var
make && sudo make install

Then farsight-0.0.12
Configured with: ./autogen.sh --prefix=/usr --libdir=/usr/lib64 --sysconfdir=/etc --localstatedir=/var --disable-python --disable-gtk-doc
make && sudo make install

Then run ldconfig: sudo ldconfig

Clean amsn: make clean distclean
Then update amsn:
Code:

U    chatwindow.tcl
U    lang/langen
U    lang/langes
U    lang/langit
U    lang/langfr
U    lang/langnl
U    protocol.tcl
U    gui.tcl
U    sip.tcl
U    assistant.tcl
U    utils/gupnp/gupnp.c
U    utils/farsight/src/tcl_farsight.c
Updated to revision 11263.

Configure with: ./configure --prefix=/usr --enable-debug
Code:

compile time options summary
============================

    X11          : yes
    Tcl : 8.5
    TK : 8.5
    DEBUG        : yes
    STATIC       : no
    FARSIGHT     : yes
    LIBV4L       : yes
    GUPNP-IGD    : yes

make && sudo make install


Now I run amsn from bash and when it opens the following appears:
Code:

mastag@evilside src]$ amsn
Unable to open mixer /dev/mixer
vid-probe: trying: v4l2...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0

(<unknown>:8381): GLib-GObject-CRITICAL **: g_object_unref: assertion `G_IS_OBJECT (object)' failed

Now I run the av assistant untill the message appears saying farsight is not loaded:
Code:

vid-probe: trying: v4l2...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0
vid-probe: trying: v4l2...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0
vid-probe: trying: v4l...
open(/dev/video1): No such file or directory
open(/dev/video2): No such file or directory
open(/dev/video3): No such file or directory
Found AOC Webcam at /dev/video0
vid-open: trying: v4l2...
v4l2: open
v4l2: init
v4l2: device info:
  uvcvideo 0.1.0 / AOC Webcam @ usb-0000:00:04.1-4.4
v4l2: close
vid-open: ok: v4l2
vid-open: flags: 2
v4l2: fini
vid-open: trying: v4l2...
v4l2: open
v4l2: init
v4l2: device info:
  uvcvideo 0.1.0 / AOC Webcam @ usb-0000:00:04.1-4.4
v4l2: close
vid-open: ok: v4l2
vid-open: flags: 2
v4l2: open
ng_dev_open: opened AOC Webcam [refcount 1]
v4l2: new capture params (320x240, RGB3, 230400 byte)
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 0
v4l2: start ts=1245658777632275000
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 230400
v4l2: close
ng_dev_close: closed AOC Webcam [refcount 0]
v4l2: fini
vid-open: trying: v4l2...
v4l2: open
v4l2: init
v4l2: device info:
  uvcvideo 0.1.0 / AOC Webcam @ usb-0000:00:04.1-4.4
v4l2: close
vid-open: ok: v4l2
vid-open: flags: 2
v4l2: open
ng_dev_open: opened AOC Webcam [refcount 1]
v4l2: new capture params (320x240, RGB3, 230400 byte)
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 0
v4l2: start ts=1245658839640627000
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 230400
v4l2: close
ng_dev_close: closed AOC Webcam [refcount 0]
v4l2: fini

(<unknown>:8381): GLib-GObject-CRITICAL **: g_object_unref: assertion `G_IS_OBJECT (object)' failed


And now the status log when preparing fails:
Code:

[10:20:42] Farsight : Preparing
[10:20:42] TURN: Connecting
[10:20:42] TURN: Sending [00][02][02][88][12][bd][96][a0]F[bc][13]4[d5][a4][99][cf][1e]@[b0][bc][00][06][02][84]RPS_t=EwDYATIiAQAUno9QpucDnNi8T5syqJTstzjjm3WAABCXnIJqoAOknpaJHXhXhGtybCoaz4seLasVVDMbJYN/VfKZhlAOkLhNfbHsegKcKWAhVL2K8LFcajfqRc68s3/7tOYhq8KG7L/qsFD5WBT8GimlvQID216atZm6knY8JP9ummmKMkVXrxXvRybMqZb+qLla3vB+zKIu+83GyVMZA2YAAAjvWEl4tUWqRSgB58J8ipp2h6TAK8KXLSgB20E0MhoscC1iEih1rDFoTeilEb86YqC3yx2P/9Sghb6FVcxByHnJbf+cwpoOtOwSYqxDZPfsSm0EP29ByL6mCUY1gbINH+LPH3LxC9NoX8GT7MD14ggMCBFbFJdbBCLfPyJICR8WD5UCKn31jTvUYihBLMLUAm0xYTXxJR4sGJ/mjcFdk+z0RSI2h2y/t2eOv7/AlS6evyhCtlguD85y8y8RDhZvC12CRZUEpPAPaVKwYLjEXFBX2pj03ve8X0d3VWbv3dNxRTC9qVb25YFL9AxhpP+3gZCo7DhQ1Iv+7EX4mkxkqVnEz9xVhD+IrxigyaKnRd5aC2ChjcLfkdRXeDqitY932o/p2uFlFOM2b6RhfPPb0s8b7/RGAQ==&p=[00][00][00]
[10:20:43] TURN: Sending [00][02][02][88]v7[e9]|v[8f][86]][81][cb][f9][f9][a8]?f[00][00][06][02][84]RPS_t=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&p=[00][00][00]
[10:20:43] TURN: Received message of type SHARED-SECRET-ERROR
[10:20:43] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]*205263578;E97E5F7411C3BC9ACD6CF0E2549B970B[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[10:20:43] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[10:20:43] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[10:20:43] TURN: Received attribute REALM : {0x32, 0x30, 0x35, 0x32,
0x36, 0x33, 0x35, 0x37,
0x38, 0x3b, 0x45, 0x39,
0x37, 0x45, 0x35, 0x46,
0x37, 0x34, 0x31, 0x31,
0x43, 0x33, 0x42, 0x43,
0x39, 0x41, 0x43, 0x44,
0x36, 0x43, 0x46, 0x30,
0x45, 0x32, 0x35, 0x34,
0x39, 0x42, 0x39, 0x37,
0x30, 0x42}
[10:20:43] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[10:20:43] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[10:20:43] TURN: Received response SHARED-SECRET-ERROR for id 12bd96a046bc1334d5a499cf1e40b0bc
[10:20:43] TURN: Parsing MAGIC-COOKIE
[10:20:43] TURN: Parsing NONCE
[10:20:43] TURN: Parsing REALM
[10:20:43] TURN: Parsing ERROR-CODE
[10:20:43] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[10:20:43] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ec][e7]
LA$[fa]9[1a][af][c2]eA[a2][13][c9][9a][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*205263578;E97E5F7411C3BC9ACD6CF0E2549B970B[00][15][00][1a]"voice.messenger.live.com"
[10:20:43] TURN: Sending [00][02][02][ec][e7]
LA$[fa]9[1a][af][c2]eA[a2][13][c9][9a][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*205263578;E97E5F7411C3BC9ACD6CF0E2549B970B[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][bf][8c][99].[98]-c[ec]5[8b]NP[a8]aZ[97]k[c2]h[1b]
[10:20:43] TURN: Received message of type SHARED-SECRET-ERROR
[10:20:43] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]*205263609;B2981FCA504CFFE41C36B25C69FFF85A[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[10:20:43] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[10:20:43] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[10:20:43] TURN: Received attribute REALM : {0x32, 0x30, 0x35, 0x32,
0x36, 0x33, 0x36, 0x30,
0x39, 0x3b, 0x42, 0x32,
0x39, 0x38, 0x31, 0x46,
0x43, 0x41, 0x35, 0x30,
0x34, 0x43, 0x46, 0x46,
0x45, 0x34, 0x31, 0x43,
0x33, 0x36, 0x42, 0x32,
0x35, 0x43, 0x36, 0x39,
0x46, 0x46, 0x46, 0x38,
0x35, 0x41}
[10:20:43] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[10:20:43] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[10:20:43] TURN: Received response SHARED-SECRET-ERROR for id 7637e97c768f865d81cbf9f9a83f6600
[10:20:43] TURN: Parsing MAGIC-COOKIE
[10:20:43] TURN: Parsing NONCE
[10:20:43] TURN: Parsing REALM
[10:20:43] TURN: Parsing ERROR-CODE
[10:20:43] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[10:20:43] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ec]=[86][81][d5]2l[d0][ff]1'[a4]n[cd][ee],[ee][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*205263609;B2981FCA504CFFE41C36B25C69FFF85A[00][15][00][1a]"voice.messenger.live.com"
[10:20:43] TURN: Sending [00][02][02][ec]=[86][81][d5]2l[d0][ff]1'[a4]n[cd][ee],[ee][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*205263609;B2981FCA504CFFE41C36B25C69FFF85A[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14]%[11][99][be]~i[d7][0e]`[f9][ff][f2][c6]ftO[e6]8[cb][9f]
[10:20:43] TURN: Received message of type SHARED-SECRET-RESPONSE
[10:20:43] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$R[d9][06][ef][87]x[9d]'[18][13]<[0c][99][dc]70[e5][87][b4][1c][0f]*[d3][e9]z[f4][a7][05][90][a4]Id[c3] [e6][12][00][07][00][10][16][93]2NK[e4][06][c3][1e]e[f2]tW&:[08][00][0e][00][08][00][01][00][00][d5][c7][a3](
[10:20:43] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[10:20:43] TURN: Received attribute USERNAME : {0x52, 0xd9, 0x06, 0xef,
0x87, 0x78, 0x9d, 0x27,
0x18, 0x13, 0x3c, 0x0c,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0x0f, 0x2a, 0xd3, 0xe9,
0x7a, 0xf4, 0xa7, 0x05,
0x90, 0xa4, 0x49, 0x64,
0xc3, 0x20, 0xe6, 0x12}
[10:20:43] TURN: Received attribute PASSWORD : {0x16, 0x93, 0x32, 0x4e,
0x4b, 0xe4, 0x06, 0xc3,
0x1e, 0x65, 0xf2, 0x74,
0x57, 0x26, 0x3a, 0x08}
[10:20:43] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x28}
[10:20:43] TURN: Received response SHARED-SECRET-RESPONSE for id e70a4c4124fa391aafc26541a213c99a
[10:20:43] TURN: TURN server 213.199.163.40 : 0
[10:20:43] TURN: Received message of type SHARED-SECRET-RESPONSE
[10:20:43] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$T[a8][a0][c4]0[18][8f][eb]G[13]<[0c][99][dc]70[e5][87][b4][1c]0[e8][96][a7]c[af]*[e1][e6][c1]~'[a3]0c[a8][00][07][00][10][ac][8b][a7][f8][9c]Z[e1][c8][9d][90][06][b0]4[cd]Z[b4][00][0e][00][08][00][01][00][00][d5][c7][a3](
[10:20:43] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[10:20:43] TURN: Received attribute USERNAME : {0x54, 0xa8, 0xa0, 0xc4,
0x30, 0x18, 0x8f, 0xeb,
0x47, 0x13, 0x3c, 0x0c,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0x30, 0xe8, 0x96, 0xa7,
0x63, 0xaf, 0x2a, 0xe1,
0xe6, 0xc1, 0x7e, 0x27,
0xa3, 0x30, 0x63, 0xa8}
[10:20:43] TURN: Received attribute PASSWORD : {0xac, 0x8b, 0xa7, 0xf8,
0x9c, 0x5a, 0xe1, 0xc8,
0x9d, 0x90, 0x06, 0xb0,
0x34, 0xcd, 0x5a, 0xb4}
[10:20:43] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x28}
[10:20:43] TURN: Received response SHARED-SECRET-RESPONSE for id 3d8681d5326cd0ff3127a46ecdee2cee
[10:20:43] TURN: TURN server 213.199.163.40 : 0
[10:20:43] TURN: Disconnecting
[10:20:43] Turn prepared {213.199.163.40 0 UtkG74d4nScYEzwMmdw3MOWHtBwPKtPpevSnBZCkSWTDIOYS FpMyTkvkBsMeZfJ0VyY6CA== 1 udp} {213.199.163.40 0 VKigxDAYj+tHEzwMmdw3MOWHtBww6JanY68q4ebBfiejMGOo rIun+Jxa4cidkAawNM1atA== 2 udp}
[10:20:43] Farsight debug : Error: Error while creating new audio_session (1): Could not create the valve element
[10:20:43] Farsight Prepare error :
[10:20:43] TURN: Disconnecting


EDIT: UPDATE with aMSN SVN Rev 11271 (which adds TURN support to video-conversation in msnp15)
Code:
09:56:15] Webcam-Assistant: Stopped grabbing
[09:56:18] Webcam-Assistant: Stopped grabbing
[09:56:20] Farsight : Preparing
[09:56:20] TURN: Connecting
[09:56:20] TURN: Sending [00][02][02][88][bb]%[b7][c3][da][f2][be]m;[e7]{[e4]D[dd]eg[00][06][02][84]RPS_t=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&p=[00][00][00]
[09:56:20] TURN: Sending [00][02][02][88][XD[1a][ad]kOc[f4][9c][d6]iA[d4][7f]~[00][06][02][84]RPS_t=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&p=[00][00][00]
[09:56:20] TURN: Received message of type SHARED-SECRET-ERROR
[09:56:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]*367433000;04C0C3BA9934FB7A391F1404315FB0F5[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[09:56:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:56:20] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[09:56:20] TURN: Received attribute REALM : {0x33, 0x36, 0x37, 0x34,
0x33, 0x33, 0x30, 0x30,
0x30, 0x3b, 0x30, 0x34,
0x43, 0x30, 0x43, 0x33,
0x42, 0x41, 0x39, 0x39,
0x33, 0x34, 0x46, 0x42,
0x37, 0x41, 0x33, 0x39,
0x31, 0x46, 0x31, 0x34,
0x30, 0x34, 0x33, 0x31,
0x35, 0x46, 0x42, 0x30,
0x46, 0x35}
[09:56:20] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[09:56:20] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[09:56:20] TURN: Received response SHARED-SECRET-ERROR for id bb25b7c3daf2be6d3be77be444dd6567
[09:56:20] TURN: Parsing MAGIC-COOKIE
[09:56:20] TURN: Parsing NONCE
[09:56:20] TURN: Parsing REALM
[09:56:20] TURN: Parsing ERROR-CODE
[09:56:20] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[09:56:20] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ec][87][df][a1][e1][c3][cf][b1][8d]j[80][a2][b7]$[8f][e0]|[00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*367433000;04C0C3BA9934FB7A391F1404315FB0F5[00][15][00][1a]"voice.messenger.live.com"
[09:56:20] TURN: Sending [00][02][02][ec][87][df][a1][e1][c3][cf][b1][8d]j[80][a2][b7]$[8f][e0]|[00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*367433000;04C0C3BA9934FB7A391F1404315FB0F5[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][03][87][97][b7][cf][c9]>[fe][c0][8d]Mjy[eb]h
[b7][aa]c|
[09:56:20] TURN: Received message of type SHARED-SECRET-ERROR
[09:56:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]*367433031;20B2083411E2245660F2A525CDACF976[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[09:56:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:56:20] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[09:56:20] TURN: Received attribute REALM : {0x33, 0x36, 0x37, 0x34,
0x33, 0x33, 0x30, 0x33,
0x31, 0x3b, 0x32, 0x30,
0x42, 0x32, 0x30, 0x38,
0x33, 0x34, 0x31, 0x31,
0x45, 0x32, 0x32, 0x34,
0x35, 0x36, 0x36, 0x30,
0x46, 0x32, 0x41, 0x35,
0x32, 0x35, 0x43, 0x44,
0x41, 0x43, 0x46, 0x39,
0x37, 0x36}
[09:56:20] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[09:56:20] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[09:56:20] TURN: Received response SHARED-SECRET-ERROR for id 5b58441aad6b4f63f49cd66941d47f7e
[09:56:20] TURN: Parsing MAGIC-COOKIE
[09:56:20] TURN: Parsing NONCE
[09:56:20] TURN: Parsing REALM
[09:56:20] TURN: Parsing ERROR-CODE
[09:56:20] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[09:56:20] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ec]\[f2][ea]![bb]?K[7f]j[b3][f1][eb][b1][f4][df][93][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*367433031;20B2083411E2245660F2A525CDACF976[00][15][00][1a]"voice.messenger.live.com"
[09:56:20] TURN: Sending [00][02][02][ec]\[f2][ea]![bb]?K[7f]j[b3][f1][eb][b1][f4][df][93][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]*367433031;20B2083411E2245660F2A525CDACF976[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][be]c[06][1f][b5]d*[b7][1f]om[c5][a6][98]%[b2][ff][fa][bc][91]
[09:56:20] TURN: Received message of type SHARED-SECRET-RESPONSE
[09:56:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$+W[8e][d8][dd][04]U[d2]v[95][e6][15][99][dc]70[e5][87][b4][1c]j
[13][18]>[1a]O#[ef][c1]Yl[06][80]I[c5][00][07][00][10][d1]$[a1][cb][e4][cd]-[91]fU[df][d2][14][10][1f][a4][00][0e][00][08][00][01][00][00][d5][c7][a3][1f]
[09:56:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:56:20] TURN: Received attribute USERNAME : {0x2b, 0x57, 0x8e, 0xd8,
0xdd, 0x04, 0x55, 0xd2,
0x76, 0x95, 0xe6, 0x15,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0x6a, 0x0a, 0x13, 0x18,
0x3e, 0x1a, 0x4f, 0x23,
0xef, 0xc1, 0x59, 0x6c,
0x06, 0x80, 0x49, 0xc5}
[09:56:20] TURN: Received attribute PASSWORD : {0xd1, 0x24, 0xa1, 0xcb,
0xe4, 0xcd, 0x2d, 0x91,
0x66, 0x55, 0xdf, 0xd2,
0x14, 0x10, 0x1f, 0xa4}
[09:56:20] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x1f}
[09:56:20] TURN: Received response SHARED-SECRET-RESPONSE for id 87dfa1e1c3cfb18d6a80a2b7248fe07c
[09:56:20] TURN: TURN server 213.199.163.31 : 0
[09:56:20] TURN: Received message of type SHARED-SECRET-RESPONSE
[09:56:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$O[bd]t[dc][f9]{[fe]j[a5][95][e6][15][99][dc]70[e5][87][b4][1c]6,pC[d3]|[d5]y[9d]E[fb][ee][04]m#[fc][00][07][00][10][cd]e\[8b][ca][aa][90]:[18][98][cd]1xF[80][c2][00][0e][00][08][00][01][00][00][d5][c7][a3][1f]
[09:56:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[09:56:20] TURN: Received attribute USERNAME : {0x4f, 0xbd, 0x74, 0xdc,
0xf9, 0x7b, 0xfe, 0x6a,
0xa5, 0x95, 0xe6, 0x15,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0x36, 0x2c, 0x70, 0x43,
0xd3, 0x7c, 0xd5, 0x79,
0x9d, 0x45, 0xfb, 0xee,
0x04, 0x6d, 0x23, 0xfc}
[09:56:20] TURN: Received attribute PASSWORD : {0xcd, 0x65, 0x5c, 0x8b,
0xca, 0xaa, 0x90, 0x3a,
0x18, 0x98, 0xcd, 0x31,
0x78, 0x46, 0x80, 0xc2}
[09:56:20] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0xd5, 0xc7, 0xa3, 0x1f}
[09:56:20] TURN: Received response SHARED-SECRET-RESPONSE for id 5cf2ea21bb3f4b7f6ab3f1ebb1f4df93
[09:56:20] TURN: TURN server 213.199.163.31 : 0
[09:56:20] TURN: Disconnecting
[09:56:20] Turn prepared {213.199.163.31 0 K1eO2N0EVdJ2leYVmdw3MOWHtBxqChMYPhpPI+/BWWwGgEnF 0SShy+TNLZFmVd/SFBAfpA== 1 1 udp} {213.199.163.31 0 T7103Pl7/mqlleYVmdw3MOWHtBw2LHBD03zVeZ1F++4EbSP8 zWVci8qqkDoYmM0xeEaAwg== 1 2 udp}
[09:56:20] Farsight debug : Error: Error while creating new audio_session (1): Could not create the valve element
[09:56:20] Farsight Prepare error :
[09:56:20] TURN: Disconnecting
[09:56:22] Getting local IP
[09:56:22] Finished


What's wrong?


Title: Audio/Video conversation
Post by: kakaroto on June 26, 2009, 06:50:30 pm
Hi, the error here is clear : "Could not create the valve element" which means you don't have the "valve" element installed.. it should be in gst-plugins-bad, but you're saying you have it installed... unless fedora was stupid enough not to include it (in that case, go complain to them).. could you check for the file /usr/lib/gstreamer-0.10//libgstvalve.so ? If it exists, try 'gst-inspect valve' on a terminal...
Also, I'm worried about this  critical error you're getting
Quote
(<unknown>:8381): GLib-GObject-CRITICAL **: g_object_unref: assertion `G_IS_OBJECT (object)' failed

Could you run amsn with
Code:
G_DEBUG=fatal_warnings gdb --args wish amsn

If glib was compiled with debug enabled, it should crash amsn when that warning is displayed... and since the command launches it in gdb, you can type 'bt' to get the backtrace so I can know where that warning comes from... (you need to type 'r' first to run the program)
thanks


Title: Audio/Video conversation
Post by: gilaparis75 on June 27, 2009, 04:02:03 pm
$ rpm -q gstreamer-plugins-bad --filesbypkg

(...)

Quote
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstsubenc.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgsttrm.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgsttta.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstvcdsrc.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstvideosignal.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstvmnc.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstx264.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstxdgmime.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libgstxvid.so
gstreamer-plugins-bad     /usr/lib/gstreamer-0.10/libresindvd.so


no valve. pas de valve. so if we have to compile just for this library, I think we've to do without...


Title: Audio/Video conversation
Post by: kakaroto on June 27, 2009, 09:23:36 pm
well then, you go yell at the fedora package maintainers.. it looks like they updated gst-plugins-bad to 0.10.12 but forgot to add the new gst elements that comes with this new release... tell them they missed some files and they'll add it.. or compile it yourself!


Title: Audio/Video conversation
Post by: MastaG on June 29, 2009, 07:04:01 pm
I've filled a bugzilla at rpmfusion.org: https://bugzilla.rpmfusion.org/show_bug.cgi?id=695
And he's right, libgstvalve has indeed moved into gstreamer-plugins-good, it's in updates-testing for Fedora 11.
Too bad I still have to dist-upgrade from F10 to F11 on this machine. But I'll test it out as soon as possible and report back. (I have other machine's running F11).

Beslemma


Title: Audio/Video conversation
Post by: kakaroto on June 30, 2009, 03:06:40 am
ok cool, thanks... apparently, gst-plugins-bad contains some potentially patented stuff, so fedora doesn't want to ship it, so they moved some of the non-patented plugins from gst-plugins-bad to gst-plugins-good, but in reality those plugins are not in gst-plugins-good officially,


Title: Audio/Video conversation
Post by: MastaG on June 30, 2009, 01:12:06 pm
EDIT: oops sorry for double posting, something went wrong


Title: Audio/Video conversation
Post by: MastaG on June 30, 2009, 01:14:59 pm
It works:)
Tested it with an WLM 8.5 user and an aMSN user.

For the ones who want to get it to work on Fedora, you'll at least need F11 with the rpmfusion repo' s installed.
Remove the gstreamer-plugins-bad packages if you have them installed, or else you'll get a conflict:
Code:
yum remove gstreamer-plugins-bad*

Then update gstreamer-plugins* from updates-testing, rpmfusion-free-updates-testing and rpmfusion-nonfree-updates-testing (they'll soon hit stable):
Code:
yum install --enablerepo=updates-testing --enablerepo=rpmfusion-nonfree-updates-testing --enablerepo=rpmfusion-free-updates-testing gstreamer-plugins*

You can use the versions of farsight2 and libnice which are also in the testing repo's (currently farsight2-0.0.12 and libnice-0.0.6, stable currently being farsight2-0.0.9 and libnice-0.0.6 which will also work):
Code:
yum install --enablerepo=updates-testing farsight2 farsight2-devel libnice libnice-devel

You should now have the following packages installed:
Code:
gstreamer-plugins-ugly-0.10.12-1.fc11
gstreamer-plugins-flumpegdemux-0.10.15-6.fc11
gstreamer-plugins-bad-extras-0.10.13-3.fc11
PackageKit-gstreamer-plugin-0.4.8-1.fc11
gstreamer-plugins-schroedinger-1.0.7-1.fc11
gstreamer-plugins-base-0.10.23-3.fc11
gstreamer-plugins-good-devel-0.10.15-3.fc11
gstreamer-plugins-bad-devel-docs-0.10.13-3.fc11
gstreamer-plugins-good-0.10.15-3.fc11
gstreamer-plugins-bad-devel-0.10.13-3.fc11
gstreamer-plugins-base-devel-0.10.23-3.fc11
gstreamer-plugins-bad-0.10.13-3.fc11
gstreamer-devel-0.10.23-2.fc11
farsight2-python-0.0.12-1.fc11
farsight2-devel-0.0.12-1.fc11
farsight2-0.0.12-1.fc11
libnice-0.0.6-1.fc11
libnice-devel-0.0.6-1.fc11


Then compile the latest svn of amsn with --prefix=/usr and everything will work.

Btw I also got rid of the:
Code:
(<unknown>:8381): GLib-GObject-CRITICAL **: g_object_unref: assertion `G_IS_OBJECT (object)' failed

After updating to F11.

Now the only thing is left is fixing the testing buttons in the new AV-wizard.


Title: Audio/Video conversation
Post by: kakaroto on June 30, 2009, 07:40:09 pm
Great!
Thanks MastaG for the feedback! :)
Next week, Billiob and me will be attending Grand Canaria Desktop Summit and we'll have a week to work on all this and finish up 0.98 for the release!


Title: Audio/Video conversation
Post by: marc2009 on July 16, 2009, 09:05:54 am
bizarre ! I don't do that :D

What I do :

I use the svn version of amsn of course ! I've gstreamer-* installed ( so all ), and farsight2 and libnice, all by yum ! Moreover, I've libv4l-devel and gupnp-igd-devel installed ... by yum too

Quote
yum install farsight2* libnice* tcl-snack* tcltls* libv4l* gupnp-igd* tk-devel tcl-devel


Then, I go to amsn directory, I do :

Code:
svn update
./configure
make clean
make
su -c 'make install'


 :P  And everything is working ! By the way, a Great Thanks to the Devs for the support of pulseaudio :D ( or maybe it's thanks to the farsight2 package ?! but anyway, Thanks ! )


Title: Audio/Video conversation
Post by: kakaroto on July 17, 2009, 03:07:07 pm
Hi marc, thanks for the feedback, I'm glad it all works fine for you! It is supposed to work, so that's cool! :)
The support of pulseaudio is because gstreamer has a pulsesrc and pulsesink elements that makes it use pulse, so yes, thanks to gstreamer pulse is supported! :)


Title: Audio/Video conversation
Post by: marc2009 on July 17, 2009, 06:23:20 pm
Quote from: "kakaroto"
Hi marc, thanks for the feedback, I'm glad it all works fine for you! It is supposed to work, so that's cool! :)
The support of pulseaudio is because gstreamer has a pulsesrc and pulsesink elements that makes it use pulse, so yes, thanks to gstreamer pulse is supported! :)


Thanks Michter ! :D


Title: Audio/Video conversation
Post by: Kalinda on July 28, 2009, 07:46:53 pm
Sooo, I'm pretty happy that aMSN's audio call is now working for me and I used it on two occasions to chat with my WLM 2009 using friend for a few hours. It works pretty damn flawlessly, aside from some crackling (or so he told me), though that could just be my mic or MSN's audio protocol being sucky.

A weird thing, though, was how the audio call kept going the first night after I closed the window. Seems it only ends if one of the two people actually ends it by hitting the button. This may have been brought up already, but that's ok. I'm just glad it works so very well :)

Thanks for all the hard work, guys!


Title: Audio/Video conversation
Post by: kakaroto on July 28, 2009, 10:00:09 pm
thanks Kalinda for the feedback! I don't really understand the issue with the audio staying when you closed the chatwindow.. in theory the chatwindow will not close if you're in a call, it will ask you to first hangup before being able to close it...
ahh.. I just checked a possible usecase, and yeah, that's a bug! Actually, if it's inside a tab, it will get closed.. the 'you have to hangup before closing' message seems to only happen if the chatwindow is all alone (not as a tab).. thanks, we'll fix that!
Glad it worked so well for you! :)
Did you try the video call too or just audio ?


Title: Audio/Video conversation
Post by: Kalinda on August 03, 2009, 04:35:14 am
I haven't tried video. A while back I tried to do a video+audio call but it didn't work for me. However, I may have some issues of my own since audio didn't work for me for the longest time, either. I can try it with regular webcam+audio next time I MSN voice chat with my friend.

Anyhow I didn't get any message about hanging up before closing and I don't use the tabs. It's not really a huge deal anyhow, though, I just hit the button again before closing the convo.


Title: Audio/Video conversation
Post by: kakaroto on August 03, 2009, 06:04:26 pm
On linux, it should work fine now, could you try it out again ?


Title: Audio/Video conversation
Post by: why.arent.guests.allowed on August 03, 2009, 06:11:42 pm
wooooooooooooow!!!!!!! :D
Amazing news!
I've been away from the forums for quite some time and it's awesome to know that amsn now supports bi-directional audio-video conversation!
Amazing work!

I've also read that you're finishing up 0.98, so maybe I'll wait for it. Any expected date for it?

Thanks for your hard work with the audio-video! :D


Title: Audio/Video conversation
Post by: kevin57 on August 03, 2009, 07:46:04 pm
I tried but it doesn't work on my computer. In the configuration tool for audio and video there is an error message saying that farsight isn't installed (I'm sorry I cannot post the exact message because it's a french version of aMSN so I'm not sure it will help you...). I don't understand why there is this problem because I already installed all the dependencies I had to install. I have installed :
glib 2.20.4-1.fc11
gstreamer 0.10.23.2-1.fc11
gstreamer-plugins-base 0.10.23-3.fc11
gstreamer-plugins-good 0.10.15-4.fc11
gstreamer-plugins-bad 0.10.13-3.fc11
libnice 0.0.6-1.fc11
farsight2 0.0.12-1.fc11

As you maybe already saw, I have fedora 11, 64 bits (x86_64)

Do you have an idea where the problem comes from?

Thank you very much

Kevin57

PS : I'm sorry for the mistakes I surely made but I'm french. I hope you can understand what I wrote!

edit : I forgot that I also have an error message in English : Farsight prepare error : can't find package Farsight. But yum list farsight* shows :
Code:
Paquets installés
farsight.x86_64                    0.1.28-3.fc11            @fedora
farsight2.x86_64                   0.0.12-1.fc11            @updates
Paquets disponibles
farsight.i586                      0.1.28-3.fc11            fedora
farsight-devel.i586                0.1.28-3.fc11            fedora
farsight-devel.x86_64              0.1.28-3.fc11            fedora
farsight2.i586                     0.0.12-1.fc11            updates
farsight2-devel.i586               0.0.12-1.fc11            updates
farsight2-devel.x86_64             0.0.12-1.fc11            updates
farsight2-python.x86_64            0.0.12-1.fc11            updates


Title: Audio/Video conversation
Post by: kakaroto on August 03, 2009, 09:41:16 pm
Hi kevin57, welcome to the forums,
your english is very good, no need to be sorry...
you probably have all the necessary requirements installed, but I think that you forgot to do this simple task :
recompile amsn.... Make sure you do ./configure and it says Farsight : YES, then recompile it, it should then work.
And you can post your french output in here, it's always more helpful than nothing...


Title: Audio/Video conversation
Post by: kevin57 on August 03, 2009, 09:49:21 pm
OK thank you very much! So I'll try to recompile it tomorrow! Do I have to uninstall anything before recompiling?

Edit : I just did ./configure and it's written : *** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
But when I do yum install gstreamer or yum install farsight2 it's written that I already installed them... :?
In the case that it can help you, here the result for ./configure :

Code:
[root@Kevin amsn]# ./configure                                      
checking for prefix by checking for wish... /usr/bin/wish          
checking for gcc... gcc                                            
checking for C compiler default output file name... a.out          
checking whether the C compiler works... yes                        
checking whether we are cross compiling... no                      
checking for suffix of executables...                              
checking for suffix of object files... o                            
checking whether we are using the GNU C compiler... yes            
checking whether gcc accepts -g... yes                              
checking for gcc option to accept ISO C89... none needed            
checking for g++... g++                                            
checking whether we are using the GNU C++ compiler... yes          
checking whether g++ accepts -g... yes                              
checking tcl build dir... using tcl library in /usr/lib64          
checking tk build dir... using tk library in /usr/lib64            
checking for main in -lstdc++... yes                                
checking how to run the C preprocessor... gcc -E                    
checking for X... libraries , headers                              
checking for gethostbyname... yes                                  
checking for connect... yes                                        
checking for remove... yes                                          
checking for shmat... yes                                          
checking for IceConnectionNumber in -lICE... no                    
checking for png_read_info in -lpng... yes                          
checking png.h usability... yes                                    
checking png.h presence... yes                                      
checking for png.h... yes                                          
checking for jpeg_CreateDecompress in -ljpeg... yes                
checking jpeglib.h usability... yes                                
checking jpeglib.h presence... yes                                  
checking for jpeglib.h... yes                                      
checking jerror.h usability... yes                                  
checking jerror.h presence... yes                                  
checking for jerror.h... yes                                        
checking for ftello... yes                                          
checking for fseeko... yes                                          
checking for getpt... yes                                          
checking for strcasestr... yes                                      
checking for memmem... yes                                          
checking for dlopen... no                                          
checking for pthread_create in -lpthread... yes                    
checking if mmx should be used... no                                
checking for pkg-config... yes                                      
checking for pkg-config... /usr/bin/pkg-config                      
checking pkg-config is at least version 0.9.0... yes                
checking for GLIB... yes                                            
checking for GST... no                                              
checking for GST_INTERFACES... no                                  
checking for LIBV4L... no                                          
checking for GUPNP... no                                            
configure: creating ./config.status                                
config.status: creating Makefile                                    
config.status: creating utils/linux/capture/config.h                
config.status: utils/linux/capture/config.h is unchanged            

compile time options summary
============================

    X11          : yes
    Tcl          : 8.5
    TK           : 8.5
    DEBUG        : no
    STATIC       : no
    FARSIGHT     : no
    LIBV4L       : no
    GUPNP-IGD    : no

*** You do not seem to have gstreamer and farsight2 installed.
*** You will not be able to build the required component for audio conversations.
*** Read this for more information : http://amsn-project.net/wiki/Farsight
[root@Kevin amsn]#


Edit 2 : The message in the configuration tool is :
Verifie si l'extension Farsight est chargée. Vous ne pourrez pas avoir d'appels Audio. Vous trouverez des informations sur Farsight sur notre Wiki :
http://www.amsn-project.net/wiki/Farsight
And then :
Farsight : Preparing
Farsight Prepare error : can't find package Farsight

I hope this can help you, thank you very much.


Title: Audio/Video conversation
Post by: kakaroto on August 03, 2009, 10:45:21 pm
kevin, you have gstreamer, yes, but you don't have the associated -devel packages.. you need those in order to compile the farsight extension for amsn (header files from gstreamer and farsight...)


Title: Audio/Video conversation
Post by: kevin57 on August 04, 2009, 07:29:40 pm
Yes you're right now it seems to work. I'm donna try as soon as possible! Thank you very much!


Title: Audio/Video conversation
Post by: MastaG on August 05, 2009, 11:32:09 pm
I've just tested with two aMSN clients (over the internet)
Both running Fedora 11 x86_64 and the following packages installed (both updated equally):
Code:
farsight2-0.0.12-1.fc11.x86_64
farsight2-devel-0.0.12-1.fc11.x86_64
telepathy-farsight-0.0.6-1.fc11.x86_64
farsight2-python-0.0.12-1.fc11.x86_64

libnice-devel-0.0.6-1.fc11.x86_64
libnice-0.0.6-1.fc11.x86_64

gstreamer-plugins-ugly-0.10.12-1.fc11.x86_64
gstreamer-ffmpeg-0.10.7-2.fc11.1.x86_64
gstreamer-plugins-base-devel-0.10.23-3.fc11.x86_64
gstreamer-plugins-bad-0.10.13-3.fc11.x86_64
PackageKit-gstreamer-plugin-0.4.8-2.fc11.x86_64
gstreamer-plugins-bad-devel-0.10.13-3.fc11.x86_64
gstreamer-plugins-good-0.10.15-4.fc11.x86_64
gstreamer-0.10.23.2-1.fc11.x86_64
gstreamer-plugins-good-devel-0.10.15-4.fc11.x86_64
gstreamer-plugins-bad-devel-docs-0.10.13-3.fc11.x86_64
gstreamer-python-0.10.15-1.fc11.x86_64
gstreamer-devel-0.10.23.2-1.fc11.x86_64
gstreamer-plugins-bad-extras-0.10.13-3.fc11.x86_64
gstreamer-plugins-flumpegdemux-0.10.15-6.fc11.x86_64
gstreamer-tools-0.10.23.2-1.fc11.x86_64
gstreamer-plugins-base-0.10.23-3.fc11.x86_64

tcltls-1.6-4.fc11.x86_64
tcl-snack-2.2.10-11.fc11.x86_64
tcl-8.5.6-6.fc11.x86_64
tcl-devel-8.5.6-6.fc11.x86_64
tcllib-1.11.1-2.fc11.noarch

tklib-0.5-3.fc11.noarch
tktable-2.9-12.fc11.x86_64
tk-devel-8.5.6-4.fc11.x86_64
tk-8.5.6-4.fc11.x86_64


Both computers have updated and recompiled amsn from svn rev 11436

When I attemp to make a sip video call this happens:
Code:
[00:23:18] CallInviteUser xxx@hotmail.com
[00:23:18] User xxx@hotmail.com supports SIP
[00:23:18] MSNSIP : Inviting user xxx@hotmail.com to a SIP call
[00:23:18] MSNSIP : Creating SIP connection to vp.sip.messenger.msn.com
[00:23:18] MSNSIP : SIP connection created : ::MSNSIP::SIPConnection3
[00:23:18] Creating CW Voip controls
[00:23:18] Farsight : Preparing
[00:23:18] TURN: Connecting
[00:23:18] TURN: Sending [00][02][02][88][b0][fc]r[19][16][8b][09][ee][d2],[9a]H[ed]2[f3][8e][00][06][02][84]RPS_t=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&p=[00][00][00]
[00:23:18] TURN: Sending [00][02][02][88]D[ab][bf][b2]3[98][cd][12][96]c[a5]0
[9a][1d][05][00][06][02][84]RPS_t=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&p=[00][00][00]
[00:23:19] TURN: Not enough header : 0
[00:23:19] TURN: Not enough header : 0
[00:23:19] TURN: Not enough header : 0
[00:23:19] TURN: Not enough header : 0
[00:23:19] TURN: Received message of type SHARED-SECRET-ERROR
[00:23:19] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]+1526251593;C8D9F76659FA557D1393BD4674A60A7C[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[00:23:19] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[00:23:19] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[00:23:19] TURN: Received attribute REALM : {0x31, 0x35, 0x32, 0x36,
0x32, 0x35, 0x31, 0x35,
0x39, 0x33, 0x3b, 0x43,
0x38, 0x44, 0x39, 0x46,
0x37, 0x36, 0x36, 0x35,
0x39, 0x46, 0x41, 0x35,
0x35, 0x37, 0x44, 0x31,
0x33, 0x39, 0x33, 0x42,
0x44, 0x34, 0x36, 0x37,
0x34, 0x41, 0x36, 0x30,
0x41, 0x37, 0x43}
[00:23:19] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[00:23:19] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[00:23:19] TURN: Received response SHARED-SECRET-ERROR for id b0fc7219168b09eed22c9a48ed32f38e
[00:23:19] TURN: Parsing MAGIC-COOKIE
[00:23:19] TURN: Parsing NONCE
[00:23:19] TURN: Parsing REALM
[00:23:19] TURN: Parsing ERROR-CODE
[00:23:19] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[00:23:19] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ed]$=[0e] [a6]3yA|0[d2][14]N[81]
9[00][06][02][84]RPS_t=EwDYATIiAQAUno9QpucDnNi8T5syqJTstzjjm3WAANV/P2V/5MZnuQ4qHI7h38fX57JjYy+VYcuZeyB/Y1oltl1PNEZJl4eNsuvkqUffQ8MYItZqAUO2Uw7nRf2fKz4xz84m5RlBjlAsP1DvMhV9Qhq8PeAI4smAdHSwx7anusVKYjr5QLtmyjzq4LCQiyKcZ6QZbq/MMX5XQyY/+K9xA2YAAAhKpky193IHCygB/NQrMOf+qpWWprSWr7u4IWRBRBMo90CDrvvBkrNYgeW6Ux7VuwCWoKQmHXrmk6ti4dOSTNDcItm720ju90junCi+yhAQg+hFY8MvEiUOS3WDmXXmTmOZ+X3HjfLqSB5ZFGQCT4VUnvIjF+4riTcH9YMumddhB3ETJEuqUG8A1mR0pagdLx7knoZoUTN3kVeCO12eMK5KlivztAMWHJF4M9c2loPaG8tugCO4futPYEIjxDRX4TVpFsq4fgiMtL6cnJkMHSYB3Gqd0M/JSZMqWseExMYOVyhjaiJdSAUlBbZq1hyEX0Zn4moKw6tg0m1b/xoRZynL1K4+A/xlsazFb2aoQWq/b3BVctvlfEpFfW7azIYQXOZ0Zaorwhl6sMiTUZpwsTjxc81GAQ==&p=[00][00][00][00][14][00]+1526251593;C8D9F76659FA557D1393BD4674A60A7C[00][15][00][1a]"voice.messenger.live.com"
[00:23:19] TURN: Sending [00][02][02][ed]$=[0e] [a6]3yA|0[d2][14]N[81]
9[00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]+1526251593;C8D9F76659FA557D1393BD4674A60A7C[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][bd][8b][d5]Q"[f8][aa][e1]F[a1]y-[97][f5][eb][b1][f8]!S[ff]
[00:23:20] TURN: Received message of type SHARED-SECRET-ERROR
[00:23:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][15][00][1a]"voice.messenger.live.com"[00][14][00]+1526251781;81FE825B1775BCC2E5A1E1BA6BAFC5DF[00][09][00][10][00][00][04][01]Unauthorized[80][04][00][0b]RPS_MBI_SSL
[00:23:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[00:23:20] TURN: Received attribute NONCE : {0x22, 0x76, 0x6f, 0x69,
0x63, 0x65, 0x2e, 0x6d,
0x65, 0x73, 0x73, 0x65,
0x6e, 0x67, 0x65, 0x72,
0x2e, 0x6c, 0x69, 0x76,
0x65, 0x2e, 0x63, 0x6f,
0x6d, 0x22}
[00:23:20] TURN: Received attribute REALM : {0x31, 0x35, 0x32, 0x36,
0x32, 0x35, 0x31, 0x37,
0x38, 0x31, 0x3b, 0x38,
0x31, 0x46, 0x45, 0x38,
0x32, 0x35, 0x42, 0x31,
0x37, 0x37, 0x35, 0x42,
0x43, 0x43, 0x32, 0x45,
0x35, 0x41, 0x31, 0x45,
0x31, 0x42, 0x41, 0x36,
0x42, 0x41, 0x46, 0x43,
0x35, 0x44, 0x46}
[00:23:20] TURN: Received attribute ERROR-CODE : {0x00, 0x00, 0x04, 0x01,
0x55, 0x6e, 0x61, 0x75,
0x74, 0x68, 0x6f, 0x72,
0x69, 0x7a, 0x65, 0x64}
[00:23:20] TURN: Received attribute UNKNOWN_ATTRIBUTE_32772 : {0x52, 0x50, 0x53, 0x5f,
0x4d, 0x42, 0x49, 0x5f,
0x53, 0x53, 0x4c}
[00:23:20] TURN: Received response SHARED-SECRET-ERROR for id 44abbfb23398cd129663a5300d9a1d05
[00:23:20] TURN: Parsing MAGIC-COOKIE
[00:23:20] TURN: Parsing NONCE
[00:23:20] TURN: Parsing REALM
[00:23:20] TURN: Parsing ERROR-CODE
[00:23:20] TURN: Parsing UNKNOWN_ATTRIBUTE_32772
[00:23:20] TURN: Doing integrity check ("voice.messenger.live.com") on [00][02][02][ed][98][0c][94][fd][f8]G[c5]&[8b][eb]h[97][f9]c[a7][f6][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]+1526251781;81FE825B1775BCC2E5A1E1BA6BAFC5DF[00][15][00][1a]"voice.messenger.live.com"
[00:23:20] TURN: Sending [00][02][02][ed][98][0c][94][fd][f8]G[c5]&[8b][eb]h[97][f9]c[a7][f6][00][06][02][84]RPS_t=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&p=[00][00][00][00][14][00]+1526251781;81FE825B1775BCC2E5A1E1BA6BAFC5DF[00][15][00][1a]"voice.messenger.live.com"[00][08][00][14][1b][e2][08][0b][d2] [9d][09]'DdE[f1]v1[f7]W[a8][ca][b5]
[00:23:20] TURN: Received message of type SHARED-SECRET-RESPONSE
[00:23:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$[8a][b5]%u[0c][82][f1]w[d0][c1][f8]Z[99][dc]70[e5][87][b4][1c][82]p9[d5][c7][99][a6][bd][df][a5][f7][8f]"R%[d9][00][07][00][10]~[0c]}[17][9e]dM[b1][85][8a][f3][ba][e5][f1][b0]A[00][0e][00][08][00][01][00][00]A6[ac][cd]
[00:23:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[00:23:20] TURN: Received attribute USERNAME : {0x8a, 0xb5, 0x25, 0x75,
0x0c, 0x82, 0xf1, 0x77,
0xd0, 0xc1, 0xf8, 0x5a,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0x82, 0x70, 0x39, 0xd5,
0xc7, 0x99, 0xa6, 0xbd,
0xdf, 0xa5, 0xf7, 0x8f,
0x22, 0x52, 0x25, 0xd9}
[00:23:20] TURN: Received attribute PASSWORD : {0x7e, 0x0c, 0x7d, 0x17,
0x9e, 0x64, 0x4d, 0xb1,
0x85, 0x8a, 0xf3, 0xba,
0xe5, 0xf1, 0xb0, 0x41}
[00:23:20] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0x41, 0x36, 0xac, 0xcd}
[00:23:20] TURN: Received response SHARED-SECRET-RESPONSE for id 243d0e20a63379417c30d2144e810a39
[00:23:20] TURN: TURN server 65.54.172.205 : 0
[00:23:20] TURN: Received message of type SHARED-SECRET-RESPONSE
[00:23:20] TURN: Received [00][0f][00][04]r[c6]K[c6][00][06][00]$e[f5][a0]*[bf][8f]rs[8b][c2][f8]Z[99][dc]70[e5][87][b4][1c][c9]c[9d]
}U5[ef][bd][cb][99]o[f2]G[bd]k[00][07][00][10]e[00]6`2[05][cb][9f][ce][a5]Og[eb][f5][8c][c2][00][0e][00][08][00][01][00][00]A6[ac][cd]
[00:23:20] TURN: Received attribute MAGIC-COOKIE : {0x72, 0xc6, 0x4b, 0xc6}
[00:23:20] TURN: Received attribute USERNAME : {0x65, 0xf5, 0xa0, 0x2a,
0xbf, 0x8f, 0x72, 0x73,
0x8b, 0xc2, 0xf8, 0x5a,
0x99, 0xdc, 0x37, 0x30,
0xe5, 0x87, 0xb4, 0x1c,
0xc9, 0x63, 0x9d, 0x0d,
0x7d, 0x55, 0x35, 0xef,
0xbd, 0xcb, 0x99, 0x6f,
0xf2, 0x47, 0xbd, 0x6b}
[00:23:20] TURN: Received attribute PASSWORD : {0x65, 0x00, 0x36, 0x60,
0x32, 0x05, 0xcb, 0x9f,
0xce, 0xa5, 0x4f, 0x67,
0xeb, 0xf5, 0x8c, 0xc2}
[00:23:20] TURN: Received attribute ALTERNATE-SERVER : {0x00, 0x01, 0x00, 0x00,
0x41, 0x36, 0xac, 0xcd}
[00:23:20] TURN: Received response SHARED-SECRET-RESPONSE for id 980c94fdf847c5268beb6897f963a7f6
[00:23:20] TURN: TURN server 65.54.172.205 : 0
[00:23:20] TURN: Disconnecting
[00:23:20] Turn prepared {65.54.172.205 0 irUldQyC8XfQwfhamdw3MOWHtByCcDnVx5mmvd+l948iUiXZ fgx9F55kTbGFivO65fGwQQ== 1 1 udp} {65.54.172.205 0 ZfWgKr+PcnOLwvhamdw3MOWHtBzJY50NfVU1773LmW/yR71r ZQA2YDIFy5/OpU9n6/WMwg== 1 2 udp}
[00:23:20] Farsight debug : AUDIO CODECS ARE READY
[00:23:20] Farsight debug : Creating audio_source : (null)  --- (null) -- (null)
[00:23:20] Farsight debug : Testing source dshowaudiosrc
[00:23:20] Farsight debug : Testing source directsoundsrc
[00:23:20] Farsight debug : Testing source osxaudiosrc
[00:23:20] Farsight debug : Testing source gconfaudiosrc
[00:23:20] Farsight debug : Using audio_source gconfaudiosrc
[00:23:20] Farsight debug : stun ip : 64.14.48.28 : 3478
[00:23:20] Farsight debug : FS: relay info = 0x2f92f60 - 2
[00:23:20] Farsight debug : VIDEO CODECS ARE READY
[00:23:20] TURN: Disconnecting
[00:23:20] Farsight debug : AUDIO CANDIDATES ARE PREPARED
[00:23:20] Farsight : Farsight audio is now prepared!
local codecs : {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000 0-16}
local candidates : {1 1 192.168.0.2 56671 {} 0 UDP 0.8299999833106995 host Rb8/jb/7ABt5tetlfP2dJHPEeram/w8OxsyuEbLYu4c= N159xXK//tHbDGDw3LocLg==} {2 1 195.240.48.10 56671 192.168.0.2 56671 UDP 0.550000011920929 srflx 8bamAIL2D0Ukc1FPiOOLEIeyNwSVMaWUJ1ZIiZT57jk= 7C1I69anbFRkbo19fPBQ+g==} {3 1 65.54.172.205 50535 192.168.0.2 56671 UDP 0.44999998807907104 relay ePojqXQrjHvnzDlqe7f1M8FMIJVkNT6Bk545+bMTlcE= LoFglMCj79joSAeFe6nlHg==} {1 2 192.168.0.2 43199 {} 0 UDP 0.8299999833106995 host Rb8/jb/7ABt5tetlfP2dJHPEeram/w8OxsyuEbLYu4c= N159xXK//tHbDGDw3LocLg==} {2 2 195.240.48.10 43199 192.168.0.2 43199 UDP 0.550000011920929 srflx 8bamAIL2D0Ukc1FPiOOLEIeyNwSVMaWUJ1ZIiZT57jk= 7C1I69anbFRkbo19fPBQ+g==} {3 2 65.54.172.205 56725 192.168.0.2 43199 UDP 0.44999998807907104 relay ePojqXQrjHvnzDlqe7f1M8FMIJVkNT6Bk545+bMTlcE= LoFglMCj79joSAeFe6nlHg==}
[00:23:20] SIP : Registering :
[00:23:20] SIP invite sent
[00:23:21] SIP : Registered
[00:23:21] SIP : Sending Invite
[00:23:21] Received INVITE response
[00:23:21] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 xxx@hotmail.com 7f6102ba95713bdaede69fa0fa474f9b TRYING
[00:23:35] ::MSNP2P::GetUser: filename is 5445572517947774c48723059684464525a54726f6d42744333303d3
[00:23:36] Send text/x-clientcaps
[00:23:36] Received INVITE response
[00:23:36] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 xxx@hotmail.com 7f6102ba95713bdaede69fa0fa474f9b RINGING
[00:23:40] SIP Keepalive
[00:23:57] Going to Read : 711
[00:23:57] Received INVITE response
[00:23:57] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 xxx@hotmail.com 7f6102ba95713bdaede69fa0fa474f9b OK
[00:23:57] SIP callee accepted our call
[00:23:57] Farsight starting : {PCMA 8 8000} {PCMU 0 8000} {telephone-event 101 8000} - {DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob 1 192.168.1.71 40108 {} 0 UDP 0.830 {} DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob/NJrWknu/Nw= 4Ofav2IUQMB59Tk0+OaPmA==} {DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob 2 192.168.1.71 57104 {} 0 UDP 0.830 {} DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob/NJrWknu/Nw= 4Ofav2IUQMB59Tk0+OaPmA==} {pj9HmqZsPOIMARIujPF674sDwOlt4Y45 1 77.167.203.163 62223 {} 0 UDP 0.550 {} pj9HmqZsPOIMARIujPF674sDwOlt4Y45jyzcozifQZ0= /FFNVFbw+D+KYDAJ8g7yrg==} {pj9HmqZsPOIMARIujPF674sDwOlt4Y45 2 77.167.203.163 62224 {} 0 UDP 0.550 {} pj9HmqZsPOIMARIujPF674sDwOlt4Y45jyzcozifQZ0= /FFNVFbw+D+KYDAJ8g7yrg==} --  -
[00:23:57] Farsight debug : New remote audio codec : 8 PCMA 8000
[00:23:57] Farsight debug : New remote audio codec : 0 PCMU 8000
[00:23:57] Farsight debug : New remote audio codec : 101 telephone-event 8000
[00:23:57] Farsight debug : New Remote candidate: DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob 1 192.168.1.71 40108 - 0 UDP 830 host DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob/NJrWknu/Nw= 4Ofav2IUQMB59Tk0+OaPmA==
[00:23:57] Farsight debug : New Remote candidate: DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob 2 192.168.1.71 57104 - 0 UDP 830 host DrhXDBxv2NVwyLKI0/GPOGcrKrJY7Dob/NJrWknu/Nw= 4Ofav2IUQMB59Tk0+OaPmA==
[00:23:57] Farsight debug : New Remote candidate: pj9HmqZsPOIMARIujPF674sDwOlt4Y45 1 77.167.203.163 62223 - 0 UDP 550 host pj9HmqZsPOIMARIujPF674sDwOlt4Y45jyzcozifQZ0= /FFNVFbw+D+KYDAJ8g7yrg==
[00:23:57] Farsight debug : New Remote candidate: pj9HmqZsPOIMARIujPF674sDwOlt4Y45 2 77.167.203.163 62224 - 0 UDP 550 host pj9HmqZsPOIMARIujPF674sDwOlt4Y45jyzcozifQZ0= /FFNVFbw+D+KYDAJ8g7yrg==
[00:24:00] SIP Keepalive
[00:24:07] Farsight debug : Error on BUS (108) Could not establish connection .. Could not establish connection on the RTP component
[00:24:07] Farsight debug : An error occured : Farsight error
[00:24:07] Farsight : got error Farsight error
[00:24:07] Farsight : Closed
[00:24:07] MSNSIP : InviteClosed ::MSNSIP::SIPConnection3 xxx@hotmail.com 7f6102ba95713bdaede69fa0fa474f9b 1
[00:24:07] SIP : Registering : REGISTERED
[00:24:07] SIP call ended
[00:24:07] Removing CW Voip controls
[00:24:07] MSNSIP : Destroying ::MSNSIP::SIPConnection3
[00:24:07] SIP : Unregistering
[00:24:07] Received INVITE response
[00:24:07] MSNSIP : inviteSIPCB : ::MSNSIP::SIPConnection3 xxx@hotmail.com 7f6102ba95713bdaede69fa0fa474f9b CLOSED REMOTE_BYE
[00:24:07] SIP callee closed the call
[00:24:07] Removing CW Voip controls
[00:24:07] MSNSIP : Destroying ::MSNSIP::SIPConnection3
[00:24:07] Got Disconnected from SIP
[00:24:07] MSNSIP: Got an error
[00:24:07] MSNSIP : Destroying ::MSNSIP::SIPConnection3
[00:24:17] Got Disconnected from SIP


Seems like an farsight error?


Title: Audio/Video conversation
Post by: kakaroto on August 06, 2009, 12:32:57 am
Humm.. seems like a libnice issue.. First problem is that you have libnice 0.0.6, but the latest version is 0.0.9, there were lots of fixes since then, so please make sure you upgrade!
There's also a farsight2 0.0.13 release that you should try...

From what I could see... for your friend (the one not providing the status log), TURN relay candidates weren't available (TURN allocation failed maybe)... although it should still work since you have TURN candidates... Also, unless you both are behind a symetric NAT, you both have STUN candidates, so connectivity should be established...

IF and only IF you upgrade to farsight2 0.0.13 and libnice 0.0.9, you *still* have the *same* error, then could you retry by providing these informations :
1 - a wireshark dump of the connection attempt
2 - launch amsn with :
Code:
NICE_DEBUG=all amsn > log.txt 2>&1

and provide the 'log.txt' file


p.s: I also noticed that you don't have siren, which is included in gst-plugins-bad 0.0.13, that's not good, I think the fedora package is broken because they forgot to include the libgstsiren.so in the rpm... You don't really need it, but you're gonna be using 16 times more bandwidth for your calls...


Title: Audio/Video conversation
Post by: MastaG on August 06, 2009, 10:24:59 am
Ah thanks for sorting that out.
As for the siren codec it got removed from the gstreamer-plugins-bad package on purpose.
The maintainer assumed it relied on libmimic which is patented, but that was not the case.
Look at: https://bugzilla.rpmfusion.org/show_bug.cgi?id=749 lol I hope I explained it the right way lol :P

The next update for gstreamer-plugins-bad in rpmfusion will include the siren codec.
As for libnice and farsight2, they will probably get updated along the way in the near future.
So within a few weeks all Fedora 11 users should have all the dependencies in the repo's :)

I've rebuild gstreamer-plugins-bad-0.10.13 (siren included by default), farsight2-0.0.14 and libnice-0.0.9 all from source with: --prefix=/usr --libdir=/usr/lib64 --sysconfdir=/etc --localstatedir=/var and installed them (overwriting the ones from the packages).
I should be set now and waiting for my cousin to respond so I can do the same on his computer.
Then I'll retest and report here.


Title: Audio/Video conversation
Post by: MastaG on August 17, 2009, 10:45:50 pm
For all Fedora users:
A new version of gstreamer-plugins-bad with the siren codec has entered the rpmfusion testing repo, it will hit stable soon.
As for farsight2 it's still at 0.0.12 and libnice at 0.0.6 so we'll have to compile the latest versions ourself.
I'll ask the maintainers for an update (but I don't think they'll put much pressure on it since farsight and nice get updated very regularly).

I've tested it with farsight2-0.0.14 and libnice-0.0.9 and it works kakaroto :D
Only one thing, my cousin uses a bluetooth headset for source and sink.
It works great but the wizard doesn't seem to save the settings for it.
(http://akelo.ath.cx/~mastag/Schermafdruk%201.png)

He pairs the device before starting aMSN, but each time he runs the wizard it will default to his webcam mic and soundcard.
Also does the wizard save the device settings globally or does he have to do it for each profile he logs in with?
I think its best to save the device settings so it will be the same for each profile we login with.

Other then that it works awesome, sound is crystal clear using his headset:)


Title: Audio/Video conversation
Post by: kakaroto on August 17, 2009, 11:20:23 pm
Hi MastaG,
The assistant is still the last small part that we're working on for the release... If you run the assistant, it will configure farsight to use whatever you specified, but that setting won't be saved for now... I will try to finish up the assistant (the video part) and I'm guessing tonight or tomorrow, it will save the settings...
However, those settings are not global, they are per profile... you've got a good point, but we're used to always have everything setup per profile.. i'll have to think about it and see if i'll make it a general config option or not..
Thanks for the feedback, glad it works so smoothly! :D :)


Title: Audio/Video conversation
Post by: kakaroto on August 18, 2009, 09:56:01 pm
Hi,
settings will now be saved and used for all your calls.. video settings are now working too, and I made them global configuration like requested.. now go test it out and report any problems!
However, the problem is that if you setup your webcam on one account, for it to work, we must advertise that we have a webcam... this means that you can't have one account that doesn't want to "Share my webcam" ...


Title: Audio/Video conversation
Post by: BW on August 19, 2009, 07:26:07 am
Hi,
Quote from: "kakaroto"
I made them global configuration like requested ...
... this means that you can't have one account that doesn't want to "Share my webcam" ...

so how about this:
we store the settings locally and put an option "Save configuration only for this account" into the video settings
if it's not selected we copy the local vars into global;
Afterwards, when selecting a profile we copy them back to local if the "Save configuration only for this account" option
is not active and the globals have been set before
(then it might be possible to have more than one audio call, forwarding one to the telephone next door or whatever ...)

anyway, thanks for the great work kakaroto


Title: Audio/Video conversation
Post by: kakaroto on August 19, 2009, 05:58:58 pm
humm.. i thought about it, but adds too much complication for me and for the user...
i'll leave it as it is right now.. if there's a need to fix this, we'll fix it later..
about having more than one audio call, if your source is pulsesrc or alsasrc, then there's no reason not to have more than one audio call...


Title: Audio/Video conversation
Post by: BW on August 19, 2009, 06:07:37 pm
hi kakaroto,
when the audio/video assistant was done before login
the follow error occurs on windows at login
it also occurs if you run the assistant a second time
Quote
bad command "devices": must be configure, cget, xview, or yview
    while executing
".webcam_preview devices"
    (procedure "::CAMGUI::camPresent" line 16)
    invoked from within
"::CAMGUI::camPresent"
    (procedure "::MSN::connect" line 34)
    invoked from within
"::MSN::connect $password"
    (procedure "::loginscreen::Snit_methodlogin" line 45)
    invoked from within

it wasn't in r11333 (with the ones in between 'til r11486 farsight fails for me)
so I would say it's caused by the Farsight video selection

About Step 4 in the assistant: maybe we shouldn't automatically redirect to
the audio setting, when farsight was successfully detected and then giving the sound
from micro directly to the loudspeaker - the Audio feedback can be really bad there...

Quote from: "kakaroto"
about having more than one audio call, if your source is pulsesrc or alsasrc, then there's no reason not to have more than one audio call...

I thought about having different audio calls at the same time with different devices...
so you can route the sound from 2 different applications connected to the same virtual device
to different real devices?


Title: Audio/Video conversation
Post by: kakaroto on August 19, 2009, 06:57:08 pm
ah, no, I misunderstood, i thought you were on linux and was wondering if you could do multiple calls because the audio device would be "in use"... no indeed, you're right, you would have to reconfigure it... well, of course, if you open two amsn instances and configure one of them, the other one won't be affected until it's restarted, so it would work... but I realized that having it as a global setting is useless
because everyone should have it set to 'automatic' which is just the best setting you could use... (I worked hard on that!) so it's useless, if someone wants a custom setting it should be custom to his session.. i'll remove the global prefs and make it per-profile...
about your other bug, i spent hours on this yesterday and i'm in discussion with the author of the tkvideo extension and it's actually the only thing preventing us from doing a release right now.. so don't worry, we'll fix it up before releasing!


Title: Audio/Video conversation
Post by: MastaG on August 24, 2009, 07:23:31 pm
Quote from: "kakaroto"
Hi,
settings will now be saved and used for all your calls.. video settings are now working too, and I made them global configuration like requested.. now go test it out and report any problems!
However, the problem is that if you setup your webcam on one account, for it to work, we must advertise that we have a webcam... this means that you can't have one account that doesn't want to "Share my webcam" ...


Thumbs up bro!
Sorry for the late response, I've been busy.
It works great, everything does.
Well I don't care if other users know I have a cam.
All settings are saved and my cousin uses it now to chat with relatives over in morocco (with his sexy bluetooth headset:P)

Btw: Fedora 11 updated its packages: farsight2-0.0.14 libnice-0.0.9 and gstreamer-plugins-bad-0.10.13-6 (including the siren codec)
So everything works out of the box.

To sum things up for Fedora users:
1. Goto http://rpmfusion.org/Configuration and setup the repo's
2. Be sure that your system is fully updated:
Code:
yum -y update

3. Install the following packages:
Code:
yum install tcl-devel tk-devel tktable cabextract gnash gupnp-igd-devel tcllib tklib tcl-snack farsight2-devel libnice-devel gstreamer-plugins-base-devel gstreamer-plugins-bad-devel gstreamer-plugins-bad-extras gstreamer-plugins-good-devel gstreamer-devel gstreamer-plugins-ugly subversion

4. Get the latest amsn source:
Code:
svn co https://amsn.svn.sourceforge.net/svnroot/amsn/trunk/amsn amsn

5. Now compile and install it:
Code:
cd amsn
./configure --prefix=/usr
make
su
make install


Voila! You'll have a working aMSN with upnp, soundclips, audio/video-conversation and winks support!

Can't wait for the release :)
You did a great job kakaroto!


Title: Audio/Video conversation
Post by: Kalinda on October 18, 2009, 06:12:10 pm
Hallo!
So aMSN's voice chat works great. However, recently I got myself one of those Logitech Wireless USB headsets. It works fine with Skype and Linux sees it and everything. So I run through the wizard for aMSN's audio call, selecting plughw:2 and hw:2 and everything along the way. It plays back the audio just fine in the headset. However, when I get to the microphone part, it hangs. Or at least it did with the gdp output I have below. My first attempt with gdp didn't produce any useful info in the output (those farsight errors weren't there, either; I think I might've loaded it wrong the first time). The first time I managed to get it to record with the wireless mic in the part of the wizard where it tests recording. It then hanged when I clicked next and it went to load the Farsight extension.

And the whole time wish was using tons of CPU, but that was probably because of gdp. I hope this is helpful, however. The output starts from around the point where I open the wizard (I think).
Code:
[New Thread 0xb5167b70 (LWP 3046)]                                          
[Thread 0xb5167b70 (LWP 3046) exited]                                        
Error while mapping shared library sections:                                
utils/webcamsn/webcamsn.so: No such file or directory.                      
Error while mapping shared library sections:                                
utils/linux/capture/capture.so: No such file or directory.                  
Error while mapping shared library sections:                                
./utils/linux/capture/libng/plugins/conv-mjpeg.so: No such file or directory.
Error while mapping shared library sections:                                
./utils/linux/capture/libng/plugins/drv0-v4l2.so: No such file or directory.
Error while mapping shared library sections:                                
./utils/linux/capture/libng/plugins/drv1-v4l.so: No such file or directory.  
Error while mapping shared library sections:                                
./utils/linux/capture/libng/plugins/sn9c10x.so: No such file or directory.  
vid-probe: trying: v4l2...                                                  
open(/dev/video1): No such file or directory                                
open(/dev/video2): No such file or directory                                
open(/dev/video3): No such file or directory                                
Found USB Camera (041e:401e) at /dev/video0                                  
vid-probe: trying: v4l...                                                    
open(/dev/video1): No such file or directory                                
open(/dev/video2): No such file or directory                                
open(/dev/video3): No such file or directory                                
Found USB Camera (041e:401e) at /dev/video0                                  
vid-probe: trying: v4l2...                                                  
open(/dev/video1): No such file or directory                                
open(/dev/video2): No such file or directory                                
open(/dev/video3): No such file or directory                                
Found USB Camera (041e:401e) at /dev/video0                                  
vid-probe: trying: v4l...                                                    
open(/dev/video1): No such file or directory                                
open(/dev/video2): No such file or directory                                
open(/dev/video3): No such file or directory                                
Found USB Camera (041e:401e) at /dev/video0                                  
vid-open: trying: v4l2...                                                    
v4l2: open                                                                  
v4l2: init                                                                  
v4l2: device info:                                                          
  zc3xx 2.6.0 / USB Camera (041e:401e) @ usb-0000:00:0b.0-3                  
v4l2: close                                                                  
vid-open: ok: v4l2                                                          
vid-open: flags: 2                                                          
v4l2: fini                                                                  
vid-open: trying: v4l2...                                                    
v4l2: open                                                                  
v4l2: init                                                                  
v4l2: device info:                                                          
  zc3xx 2.6.0 / USB Camera (041e:401e) @ usb-0000:00:0b.0-3                  
v4l2: close                                                                  
vid-open: ok: v4l2                                                          
vid-open: flags: 2                                                          
v4l2: open                                                                  
ng_dev_open: opened USB Camera (041e:401e) [refcount 1]                      
v4l2: new capture params (320x240, RGB3, 230400 byte)                        
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 13028                      
v4l2: start ts=244440538104000                                              
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 230400                      
v4l2: close                                                                  
ng_dev_close: closed USB Camera (041e:401e) [refcount 0]                    
v4l2: fini                                                                  
vid-open: trying: v4l2...                                                    
v4l2: open                                                                  
v4l2: init                                                                  
v4l2: device info:                                                          
  zc3xx 2.6.0 / USB Camera (041e:401e) @ usb-0000:00:0b.0-3                  
v4l2: close
vid-open: ok: v4l2
vid-open: flags: 2
v4l2: open
ng_dev_open: opened USB Camera (041e:401e) [refcount 1]
v4l2: new capture params (320x240, RGB3, 230400 byte)
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 13286
v4l2: start ts=244443351254000
v4l2: buf 0: video-cap 0xabcdef00+16777216, used 230400
v4l2: close
ng_dev_close: closed USB Camera (041e:401e) [refcount 0]
v4l2: fini

Program received signal SIGTERM, Terminated.
0xb7fe1424 in __kernel_vsyscall ()
(gdb) quit


Title: Audio/Video conversation
Post by: kakaroto on October 20, 2009, 07:16:08 am
when you get the SIGTERM signal in gdb, you should type "bt" not "quit".. read the FAQ for how to generate proper reports. thanks!

p.s.: that's weird, didn't know there was a bug somewhere... you're using the SVN version right ?


Title: Audio/Video conversation
Post by: Kalinda on October 24, 2009, 03:36:42 am
Yeah, I've got the latest SVN. I've had this problem for quite a while, actually, with regard to the new headset.


Title: Audio/Video conversation
Post by: Kalinda on October 27, 2009, 08:12:10 pm
Ok, well.. now it's working, which is weird. It almost seems like trial and error to get it to go through the wizard, but it did it and I tested it and it works :)


Title: Audio/Video conversation
Post by: arantes on October 28, 2009, 05:37:03 pm
We are getting spammed ...


Title: Audio/Video conversation
Post by: deschansons on November 23, 2009, 06:23:43 pm
Hey everyone, I just recently updated my aMSN, but video/audio calls won't work. It will ring/person will 'pick up', but then it tells me the call has been cancelled. Apologies if I'm missing something obvious.


Title: Audio/Video conversation
Post by: kakaroto on November 23, 2009, 10:28:47 pm
ctrl-s from the main window to see debug messages


Title: Audio/Video conversation
Post by: Torocatala on November 26, 2009, 05:41:40 pm
Hi i make the ctrl+s for the debug window. After that i was invited for a audio&video conversation from a machine whit Live Messenger but when i accept the conversation automatically end. The debug show:

[17:33:42] Farsight::Start Error : Could not set the video remote codecs
[17:33:42] Farsight : Closed
[17:33:42] MSNSIP: answerClosed : ::MSNSIP::SIPConnection15 d387eba1e1074b8eaee0e68011a1df06 1
[17:33:42] SIP call ended
[17:33:42] Removing CW Voip controls
[17:33:42] SIP : Registering : REGISTERED
[17:33:42] MSNSIP : Destroying ::MSNSIP::SIPConnection15
[17:33:42] SIP : Unregistering
[17:33:42] Got Disconnected from SIP
[17:33:42] MSNSIP: Got an error
[17:33:42] MSNSIP : Destroying ::MSNSIP::SIPConnection15
[17:33:49] DEBUG: Closing old Log fileid in set (this shouldn't happen)
[17:33:49] DEBUG: Closing log file for 0
[17:33:49] DEBUG: Calling unset on an unexisting variable
[17:33:49] DEBUG: Calling unset on an unexisting variable
[17:33:52] Got Disconnected from SIP


Title: Audio/Video conversation
Post by: kakaroto on November 26, 2009, 10:42:08 pm
You don't have the H263 encoder/decoder, make sure you install the gstreamer-plugins-ffmpeg package (or similar) and that it was compiled with the encoders enabled. (if on debian, check the debian-multimedia project).


Title: error in "make" of farsight2..
Post by: dsevastakis on December 03, 2009, 05:46:39 pm
hi! i am new to this forum! And after i searched a bit, i still have the problem.. i get this output after the "make" :

Code:

Making all in rtcpfilter
  CC    fs-rtcp-filter.o
fs-rtcp-filter.c:39:35: error: gst/rtp/gstrtcpbuffer.h: No such file or directory
fs-rtcp-filter.c: In function ‘fs_rtcp_filter_transform_ip’:
fs-rtcp-filter.c:189: warning: implicit declaration of function ‘gst_rtcp_buffer_validate’
fs-rtcp-filter.c:199: error: ‘GstRTCPPacket’ undeclared (first use in this function)
fs-rtcp-filter.c:199: error: (Each undeclared identifier is reported only once
fs-rtcp-filter.c:199: error: for each function it appears in.)
fs-rtcp-filter.c:199: error: expected ‘;’ before ‘packet’
fs-rtcp-filter.c:200: warning: ISO C90 forbids mixed declarations and code
fs-rtcp-filter.c:202: warning: implicit declaration of function ‘gst_rtcp_buffer_get_first_packet’
fs-rtcp-filter.c:202: error: ‘packet’ undeclared (first use in this function)
fs-rtcp-filter.c:206: warning: implicit declaration of function ‘gst_rtcp_packet_get_type’
fs-rtcp-filter.c:206: error: ‘GST_RTCP_TYPE_SR’ undeclared (first use in this function)
fs-rtcp-filter.c:208: error: expected ‘;’ before ‘nextpacket’
fs-rtcp-filter.c:211: warning: implicit declaration of function ‘gst_rtcp_packet_move_to_next’
fs-rtcp-filter.c:211: error: ‘nextpacket’ undeclared (first use in this function)
fs-rtcp-filter.c:212: error: ‘GST_RTCP_TYPE_RR’ undeclared (first use in this function)
fs-rtcp-filter.c:214: warning: implicit declaration of function ‘gst_rtcp_packet_remove’
fs-rtcp-filter.c:222: error: ‘GST_RTCP_VERSION’ undeclared (first use in this function)
fs-rtcp-filter.c:243: warning: implicit declaration of function ‘gst_rtcp_buffer_end’
make[3]: *** [libfsrtcpfilter_la-fs-rtcp-filter.lo] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2


what do i have to do? and is there an easy way to install all the dependencies of farsight2? coz every time i try it drives me crazy.. and i always fail:S  1 time i succeeded.. but still aMSN said i have to install farsight2 blah blah! Any guesses?:)
 thanks in advance!!!


Title: Audio/Video conversation
Post by: kakaroto on December 03, 2009, 08:03:06 pm
Quote

fs-rtcp-filter.c:39:35: error: gst/rtp/gstrtcpbuffer.h: No such file or directory

you're missing a gstreamer dev package..
an easy way to install it? depends on which system you use! If you're on debian sid or latest ubuntu, I guess installing the libgstfarsight and libgstfarsight-dev packages is enough.


Title: Re: Audio/Video conversation
Post by: zyazhou on February 23, 2010, 12:02:54 pm
It seems audio call doesn't work from this morning, and vp.sip.messenger.msn.com can't be connected. Could anyone verify this?


Title: Re: Audio/Video conversation
Post by: alexandernst on February 23, 2010, 03:07:17 pm
vp.sip.messenger.msn.com doesn't pong, so, yes. Anyways, this isn't directly an amsn prob.


Title: Re: Audio/Video conversation
Post by: zyazhou on February 23, 2010, 03:55:28 pm
Is there any possibility that msn will remove this sip server forever?


Title: Re: Audio/Video conversation
Post by: kakaroto on February 23, 2010, 07:46:38 pm
if they do, they would break their own client, so I would say no, never.. just like the feature for old audio calls that was there since about 10 years ago and no client ever uses it, is still available if you wanted to use it...


Title: Re: Audio/Video conversation
Post by: zyazhou on February 24, 2010, 09:41:45 am
Hi, Kakaroto. MSN has already blocked all login which version is below v2009, and force you to upgrade to the latest one, so that's why I think they'll remove audio call of MSNP15 permanently. As we know, the default protocol for MSN v2009 is MSNP18, so no MSN client will use this kind of audio call. As for the old audio call you mentioned, MSN removed that early last year.

if they do, they would break their own client, so I would say no, never.. just like the feature for old audio calls that was there since about 10 years ago and no client ever uses it, is still available if you wanted to use it...


Title: Re: Audio/Video conversation
Post by: kakaroto on February 25, 2010, 08:55:34 pm
zyazhou: they removed it? nope, I don't think so.. not if you try to call 'windows messenger'.. I've seen it myself, when I made amsn act like it supported the old rtp protocol, I still got that old invitation which we haven't seen in years...
and no, they don't break support.. they force you to upgrade, but they always stay backward compatible..
don't forget that WLM 8.5 is the latest version available for windows NT/98 for example.. so the people on those systems don't upgrade.. and although they don't support win 98 anymore, it's still their policy to always stay backward compatible...
yes, they use MSNP18, but you can still connect with MSNP12 if you wanted to, they force you to upgrade, but they don't block older protocols...
Also, this whole SIP/RTP stack is being used in MSNP18 too (actually, SIP/RTP video calls is only possible with MSNP18)


Title: Re: Audio/Video conversation
Post by: zyazhou on February 26, 2010, 10:30:31 am
kakaroto, thank you so much for your explaination. Probably you are right. But the thing is, vp.sip.messenger.msn.com (which is used in MSNP15 for audio call) doesn't work from Monday. If msn supports this, why it has stopped working for so long?
 
zyazhou: they removed it? nope, I don't think so.. not if you try to call 'windows messenger'.. I've seen it myself, when I made amsn act like it supported the old rtp protocol, I still got that old invitation which we haven't seen in years...


Title: Re: Audio/Video conversation
Post by: alexandernst on February 27, 2010, 12:00:30 am
Probably it's a temporal down


Title: Re: Audio/Video conversation
Post by: kakaroto on February 28, 2010, 06:33:43 pm
yep, temporary service down time is common..
however, I just did some tests, and it looks like MSNP18<->MSNP18 still works because it's using a 'tunneled sip' which means it doesn't dpeend anymore on that external vp.sip.messenger.msn.com server...
If I try from WLM to aMSN, it says "service is unavailable right now, please retry later"...
I might have been wrong, maybe you're right, they could drop the support, just like they seem to have stopped using that third party server for webcam when behind a NAT/firewall... They keep the support in the client, but they just bring down some servers that aren't needed anymore, which ends up breaking the support itself...
We'll have to wait a little to see if the server comes back online or not.. if it doesn't, then we're screwed :(
We'll have to either remove the feature or force the move to MSNP18... sucks!


Title: Re: Audio/Video conversation
Post by: alexandernst on February 28, 2010, 07:29:26 pm
Nope, it seems that M$ just closed their servers for all clients that aren't msnp18.

Edit: Not confirmed


Title: Re: Audio/Video conversation
Post by: cuteboy on April 18, 2010, 05:42:34 pm
Haaaa ben là chu triste en tabarnac!  :'(


Title: Re: Audio/Video conversation
Post by: falde on June 28, 2011, 02:55:32 pm
So what is needed in order to bring us forward to MSNP18? I may be willing to do some work on this.

Would it be possible to build a protocol on top of the MSN protocol for usage in amsn to amsn conversions, or amsn to pidgin conversions? With that we would not be as pressed to keep up with Microsoft's changes in the protocol. Instead we could just implement something that are stable, but incompatible with their clients.


Title: Re: Audio/Video conversation
Post by: kakaroto on July 02, 2011, 01:13:06 am
So what is needed in order to bring us forward to MSNP18? I may be willing to do some work on this.

Would it be possible to build a protocol on top of the MSN protocol for usage in amsn to amsn conversions, or amsn to pidgin conversions? With that we would not be as pressed to keep up with Microsoft's changes in the protocol. Instead we could just implement something that are stable, but incompatible with their clients.

For MSNP18, we need P2PV2 support, we already refactored the P2P stack in order to more easily allow P2Pv2 implementation but it hasn't been done yet.
For a non-MSN protocol, etc.. just use XMPP, it's an open standard IM/VoIP protocol, so just use that if you want (that's what Jabber, google talk and facebook chat uses).