Check out fs_video2 branch or wait for 1.6 merge later this week. All of that code is drasticly different now. On Saturday, May 23, 2015, Jon Lederman <jonlederman@gmail.com> wrote:
Hi,
I am trying to use WebRTC with Freeswitch. WebRTC relies upon RTCP for bandwidth estimation. I have tried to get forwarding of RTCP packets working with Freeswitch to make this work.
However, a more fundamental issue seems to be occurring in that Freeswitch is not able to process the RTCP messages arriving from the WebRTC client. I have traced the problem to the function process_rtcp_frame in switch_rtp.c. In that function, the version of RTP appearing in the RTCP packets is 0, not 2, which is required and the packets are never processed (i.e., no fresh frames). I am not sure what the cause of this is? Could it be related to muxing of RTP/RTCP, which webrtc utilizes? Any thoughts on resolving this would be greatly appreciated.
Thanks.
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