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

Tue Apr 1 13:51:05 MSD 2014

Thanks Anthony! It works now.

On Mon, Mar 31, 2014 at 7:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 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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>>>>>
>>>>> 
>>>>> 
>>>>>
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>>>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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