About sip rtp and audio conversation

[denemeler]

Hi

I want to develop a toolkit but I cant understand something.
I start an audio conversation and at the same time I capture all packet.
but I can not see anything about sip authentication or rtp data.
there is only , I can see, TLSV, ssl packet. And some udp packet I think this is audio packet but this isnt caried on rtp protocol.

Please what is the backgroun on audio conversation?
please help me

thanks

[kakaroto]

Hi,
welcome to the forums!
the SIP is being done over TLS, so it won't show in wireshark (that's the TLS packets you see)...
about the RTP data, it's all over UDP, so the UDP packets, that's that.. if wireshark says it's unknown, it's simply because it didn't find the SIP packets to analyse them so it couldn't map the ip/ports to RTP.. if you do right-click, decode as, then choose RTP, it will show you the RTP... there's also RTCP in there... oh and of course, some STUN messages are also being sent, as part of ICE's connectivity checks for NAT traversal.
If you read this whole thread here : http://www.amsn-project.net/forums/viewtopic.php?t=4799
You will get every bit of information you might need.. (it starts with video conference, then at page 2, i changed my work to be on the SIP/RTP stuff instead).

I'm curious though, what are you developing ? and what do you mean by developing a "toolkit"  ?

[denemeler]

thanks kakaroto. I will listen you