Video Chat using mpegtsmux

[Gregory AE40]

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.

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,
>
> --
> Mathieu Duponchelle · https://www.centricular.com
>
> On 5/19/21 8:14 AM, Gregory AE40 via gstreamer-devel wrote:
>
> Hi,
>
> I am trying to implement a video chat using GStreamer on Windows 10. I
> intend to use gstreamer C API, but for now I am just using the command line
> for basic testing. In order to keep video and audio in sync, I am muxing
> them using mpegtsmux and then sending them via RTP. It works fine, but the
> latency is pretty high (I think it's about 1 second).
>
> My sending pipeline is as follows:
>
> gst-launch-1.0 mpegtsmux name=mux ! rtpmp2tpay ! udpsink host=127.0.0.1
> port=5555^
>  wasapisrc ! audioresample ! audioconvert ! avenc_aac ! queue ! mux.^
>  ksvideosrc ! videoconvert ! x264enc tune=zerolatency ! h264parse
> config-interval=-1 ! queue ! mux.^
>
> My receiving pipeline is as follows:
>
> gst-launch-1.0^
>  udpsrc address=127.0.0.1 port=5555 caps="application/x-rtp" !
> rtpmp2tdepay ! tsdemux name=demux^
>  demux. ! queue ! avdec_aac ! audioresample ! audioconvert ! wasapisink
> low-latency=true^
>  demux. ! queue ! h264parse ! avdec_h264 ! videoconvert ! autovideosink
>
> Questions:
> 1. Is this the correct approach for a video chat or should I use separate
> streams for audio and video?
> 2. If this is the correct approach, am I using the correct muxer or should
> I use something else such as asfmux?
> 3. If this is the correct approach and the correct muxer, is there
> something I can do in order to reduce the latency?
>
>
>
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>
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