Video Chat using mpegtsmux
gregory at ae40.com
Thu May 20 04:26:17 UTC 2021
Ok, thank you all for the useful information and suggestions
*From:* Michael Gruner <michael.gruner at ridgerun.com>
*Sent:* Wednesday, May 19, 2021 10:06 PM
*To:* Discussion of the development of and with GStreamer <
gstreamer-devel at lists.freedesktop.org>
*Cc:* Nicolas Dufresne <nicolas at ndufresne.ca>; Gregory AE40 <
gregory at ae40.com>; Mathieu Duponchelle <mathieu at centricular.com>
*Subject:* Re: Video Chat using mpegtsmux
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
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> wrote:
Le mercredi 19 mai 2021 à 18:23 +0300, Gregory AE40 via gstreamer-devel a
> 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
> for each receiver.
Yet, you should still use RTP for mulicast. For your use case, just
RTSP server (using gst-rtp-server library) seem sufficient no ?
> On Wed, May 19, 2021, 15:02 Mathieu Duponchelle <mathieu at centricular.com>
> > 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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