[Live-devel] RTP Sink & SDP

[Ross Finlayson]

At 03:37 AM 7/27/04, you wrote:
>The playSIP is for receiving audio over RTP. I want to modify the playSIP 
>for sending audio over RTP using SIP. How can start RTP for sending audio 
>data to the other client using the SDP that i received in the SIP OK?

This should be possible to do, but will require a fair bit of new 
code.  There's not quite enough support for this in the current code base.

I suggest defining a new subclass of "ServerMediaSession" that - like 
"MediaSession" - can be initialized using a SDP description.  Your new 
subclass would need to parse the SDP description, and create/add new 
"ServerMediaSubsession" objects appropriately.  (You will probably also 
need to define a new subclass of "ServerMediaSubsession".)  You would then 
'initiate' these "ServerMediaSubsession" objects (analogous to 
"MediaSubsession::initiate()") to create appropriate "RTPSink" objects for 
each.  Then, call "startPlaying()" on each of these "RTPSink" objects (with 
appropriate sources).

Good luck :-)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>