[Freeswitch-users] RTCP FIR on floor change/new member join
Brian West
brian at freeswitch.org
Fri Mar 28 21:35:25 MSK 2014
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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 841 bytes
Desc: Message signed with OpenPGP using GPGMail
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140328/a07ec53b/attachment.bin
Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users
mailing list