Video Chat using mpegtsmux

[Nicolas Dufresne]

Le mercredi 19 mai 2021 à 21:44 +0300, Gregory AE40 a écrit :
> Yes, I understand that I should use RTP, my question is whether to mux audio
> and video to a single stream or to send two separate streams (with RTCP for
> every stream for synchronization). The first option seems more fail-safe to me
> in terms of audio/video synchronization.  However, I tested also the second
> option and its latency is lower. I guess that if you suggested to use WebRTC,
> I should stick with the second option, because AFAIK, WebRTC sends separate
> streams for video and audio. 

Correct, muxing and demuxing MPEG TS requires adding some latency due to how the
timing information is stored. We added a latency property to reduce it from the
default 700ms (maximum specified) for those that have the luxury to control the
muxing parameters, which should be your case here. Yet, you still need to
introduce a small amount of latency compare to streaming audio/video
individually.

> 
> On Wed, May 19, 2021, 20:54 Nicolas Dufresne <nicolas at ndufresne.ca> wrote:
> > Le mercredi 19 mai 2021 à 18:23 +0300, Gregory AE40 via gstreamer-devel a
> > écrit :
> > > Hi, thank you for the prompt response.
> > > I wasn't specific enough, the application will run on a LAN, so I can use
> > > multicast in order to send only one stream from each PC. 
> > > With WebRTC multicast is not possible, because it requires a separate
> > > stream
> > > for each receiver.
> > 
> > Yet, you should still use RTP for mulicast. For your use case, just creating
> > an
> > RTSP server (using gst-rtp-server library) seem sufficient no ?
> > 
> > > 
> > > On Wed, May 19, 2021, 15:02 Mathieu Duponchelle <mathieu at centricular.com>
> > > wrote:
> > > >  Hey,
> > > >  
> > > >  I would recommend using webRTC (webrtcbin in the GStreamer context) for
> > > >  such an application :)
> > > >  
> > > >  Best,
> > > >  
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> > > > gstreamer-devel at lists.freedesktop.org
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> > 
> >