Hi, How does a server in a conference handle RTCP feedback packets and forward them to the clients?
The use case is several browsers connected to a server which relay RTP packets without mixing them. Each browser have one peer connection for sending audio and video and one peer connection for each every other client for receiving audio and video. It might be similar to section 3.4.3. Video conferencing system with central server in http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-14#section-3.4.3 Each client sends RTCP feedback for all the sources it receives. It sends SR for his publishing peer connection (if used) and RR for his other peer connections. If client 1 publishes audio and clients 2-10 send RR RTCP feedback packets, what does the server do with the RR packets? Can the server just forward the packets with the original sender SSRC? Does the server need to act as a peer and generate RTCP feedback of its own? How does the server handle FIR and NACK? Are there rfcs that deals with this use case and are supported by WebRTC clients? I'm interested in the easiest solution to implement which is supported by FF and also to learn about the most correct solution so I'll implement it in the future.. Thanks _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
