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

[Hristo Trendev]

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 at freeswitch.org> wrote:

> Please make sure you're using the latest code, care to say what Rev you're
> on?
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9001
> ISN: 410*543
> Skype:briankwest
> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/0dd7995a/attachment.html