[Live-devel] How to handle RTP/RTCP connection to send audio using SIPClient

[Antoine Lavier]

I'm trying to learn how to use SIP with Live555 to give a SIP call.
This is my first time with Live555, also I spent hours in the testProgs
section reading code and trying to understand how it works. Thanks for all
this section's content.

I focused my attention on the playSIP program. To be honest, if this one
could receive and send audio data (send from stdin or from a file like some
other sample programs), it will be the perfect documentation to me.

Reading this code, I don't understand how to send (audio) data. Should I
create an other MediaSession? An other MediaSubSession? Should I proceed
like in the test*Streamer.cpp and only create and use a RTP stream? How to
bind it to the current SIP session?

Reading SDP data from the SIP response, I can see some information like:

...

m=audio 4004 RTP/AVP 0

c=IN IP4 192.168.5.91

b=TIAS:650000

a=rtcp:4005 IN IP4 192.168.5.91

a=sendrecv

a=rtpmap:0 PCMU/8000

So I think I should create a RTP connection to 192.168.5.91:4004 and a RTCP
one to 192.168.5.91:4005? How?

Could you please help me by giving me informations based on playSIP.cpp?
Thanks!

Regards,

Antoine Lavier
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