[Freeswitch-users] RTCP FIR on floor change/new member join

[Hristo Trendev]

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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