Audio/Video conversation

[Kreuger]

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.

[senlegen]

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

[Montblanc]

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?

[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.

@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

[Montblanc]

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 .

[kakaroto]

ok, cool then!

[senlegen]

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

[kakaroto]

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!

[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 !!!!
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[MastaG]

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 !!!!
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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:)

[Kreuger]

I just tried MastaG's way of doing it. Got to the part of make for farsight and still getting the error

   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$

[farseeing]

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 ?

[kakaroto]

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!

[farseeing]

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 !  
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.

[kakaroto]

hehe, blame microsoft :p