Is there any good article or spec, which discusses when exactly to start and stop sending RTP? (beyond just being ready to receive RTP right after sending the INVITE/200 OK respectively.) For instance say a SIP Application Server wishes to connect Media server and end point for sake of playing an announcement to the end point. Is it critical who the AS will connect first? (e.g. I don't want the media server to start play the announcement too early...) Beyond RTCP Bye, is there any way that two RTP/RTCP end points can indicate start or stop of conversation?
_______________________________________________ Sip-implementors mailing list [email protected] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
