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

Anthony Minessale anthony.minessale at gmail.com
Mon Mar 31 21:39:03 MSD 2014


It did not but I just added a patch so it will.  Try master.



On Mon, Mar 31, 2014 at 10:52 AM, Hristo Trendev <dist.lists at gmail.com>wrote:

> Hi,
>
> One of the browsers that I used for the initial testing was Chrome on
> Android 4.3. I guess this was causing some of the problems.
>
> I retested with the latest version from master (0876df6). When I make the
> test with two desktop browsers indeed a new keyframe seems to be requested
> on every floor change, as in this case the video stream switches almost
> instantly.
>
> However, the problem is still present for anyone who has just joined a
> conference. Unless the floor changes shortly after someone joins, he has to
> wait for the next keyframe. This is really simple to reproduce if there is
> only one speaker in the conference and all the others who are joining the
> conference are muted by default - in this case there is virtually no chance
> of a floor change to occur.
>
> Is a FIR packet also sent when someone joins a conference or is it
> currently only sent on floor change events?
>
> Best
> Hristo
>
>
> On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Fir is sent explicitly no settings req
>> On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang at gmail.com> wrote:
>>
>>> 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
>>>>> PO BOX 2531
>>>>> Brookfield, WI 53008-2531
>>>>> Twitter: @FreeSWITCH , @briankwest
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>>>>> http://www.freeswitchcookbook.com
>>>>>
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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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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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