AFAIK FS has that so a jira would help 2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists@gmail.com>写道：
Hi Brian,
It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I can test with the master branch on Monday if needed. However, I am not really sure if the functionality to request a keyframe via RTCP FIR is even present in conference/freeswitch. In case it is indeed missing, then it will at least explain the behavior I see, otherwise I've probably missed something while testing.
Best, Hristo
On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian@freeswitch.org> wrote:
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On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists@gmail.com> wrote:
Hi,
I've been testing a bit with WebRTC video using Chrome on Mac and a Freeswitch conference. The problem that I have is that the WebRTC client (Chrome) sends video keyframes every 3000 frames, which means that when a new participant joins the conference it can be some time before he/she gets a keyframe. The same problem occurs when the floor changes and the video stream source of the conference is switched to another participant, whose WebRTC client won't necessarily send a keyframe on floor change.
As far as I understood after reading through a couple of Chrome tickets, a keyframe can be requested by the receiving party - in this case a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture Loss Indication) packet.
Ticket FS-5596 mentions FIR in its description, so it seems as if FS already has support for FIR (or maybe I got it all wrong and the case described in this ticket refers to RTCP packets, which are simply being proxied by FS). Additionally FS indicates in the SDP that it supports FIR.
BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR received. webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in RTCP RR extended highest sequence number.
The real question is if the conference app can be configured/patched to send an RTCP FIR packet to the member who has the floor whenever a new participant joins the conference or in case the floor itself just moved to someone else.
Has anyone tested something similar?
BR Hristo
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