If you ever were in the situation to try to find out why the video quality of your WebRTC call was not good, you probably have also sworn at the encrypted RTP and RTCP. Instead of trying to put log statements into your locally compiled Firefox version, you can now simply request logging of the RTP and RTCP packets.
Tag archive: 'RTP'
RMCAT is an IETF Working Group which came out of proposal by myself and Harald Alvestrand, and an associated Congestion Control IAB/IRTF workshop at IETF 84 in Vancouver in 2012. The report from the workshop is RFC 7295.
The RMCAT WG is working to develop new congestion control protocols for realtime RTP traffic to improve on state-of-the-art, to ensure that media streams don’t harm other users of the networks (both non-RTP and other RTP media streams), and to maximize quality. There are several proposed algorithms, and this work will feed back into mainline WebRTC implementations to improve network usage and media quality. (more…)