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

[Seven Du]

AFAIK FS has that so a jira would help
2014年3月29日 上午4:57于 "Hristo Trendev" <dist.lists at 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 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
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>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> 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
>>
>>
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>
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