Video Chat using mpegtsmux
michael.gruner at ridgerun.com
Wed May 19 19:05:32 UTC 2021
I echo the suggestions given by Nicolas and Mathieu. But also wanted to add that, if you chose to continue using TS, there is a “latency” property in the tsdemux element that you can tweak. By default it is set to 700ms (to comply with the TS standard), which sounds like the 1s lag you are experiencing.
> On 19 May 2021, at 12:44, Gregory AE40 via gstreamer-devel <gstreamer-devel at lists.freedesktop.org> wrote:
> 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.
> On Wed, May 19, 2021, 20:54 Nicolas Dufresne <nicolas at ndufresne.ca <mailto: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 <mailto: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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