webrtcbin latency

[Mathieu Duponchelle]

Hey,

Under the hood, webrtcbin uses rtpbin, and its latency on the receiving side(s) is left
at the default of 200 ms, which is a reasonable enough value to let the jitter buffer
perform its work adequately.

That being said, a latency property could probably be exposed on webrtcbin, it should
be straightfoward if you feel like proposing a patch, bear in mind however that setting
the latency lower increases the chances of packets considered lost on the receiving end,
apart from the local network case.

Cheers,

-- 
Mathieu Duponchelle · https://www.centricular.com


On 06/26/2018 11:56 PM, antchi wrote:
> I've been testing out the new webrtcbin plugin and am encountering a
> noticeable addition in latency when streaming.  I used two machines with
> gstreamer to stream h264 video from one to the other via the webrtcbin
> plugin (pipelines look like this - sender: fdsrc ! h264parse ! rtph264pay !
> webrtcbin.  receiver: webrtcbin ! decodebin ! videoconvert ! autovideosink),
> which works like a charm.  If I replace the webrtcbin's with udpsink and
> udpsrc's, however, I get about a 150-200ms decrease in latency.  Has anybody
> else experienced the same?
>
> I'm under the impression that most of the work webrtcbin does is setting up
> the connection, and once that connection is set up, information should flow
> with very little overhead.  I'm wondering if this overhead is typical, or if
> there is anything that can be done to optimize this.
>
> 150-200ms may not seem that long, but depending on the real-time
> application, it could have a significant impact on experience.
>
> Thanks!
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel