WebRTC Inconsistent Playout Across Browsers

[Nathan Stratton]

I am seeing the same thing.

><>
Nathan Stratton
Founder, CTO
Vocinity, Inc.


On Mon, Apr 26, 2021 at 11:56 AM Cameron Blomquist via gstreamer-devel <
gstreamer-devel at lists.freedesktop.org> wrote:

> I have been testing webrtc and came across a strange behavior with
> browsers on the receiving side. To verify this behavior, I used
> gst-examples/webrtc/sendrecv/gst-rust. Changed audiotestsrc to use the
> ticks wave and videotestsrc to sweep and use running-time so that the tick
> should line up with a single sweep of the ball.
>
> Using GStreamer 1.18 on Ubuntu 18.04.
>
>    - On Firefox for Android, the received video and audio streams were as
>    expected
>
>
>    - On Safari for iOS, the received audio stream saw significant delay
>    from the video stream in the range of 250-750ms by ear (no exact
>    measurements, just comparing the motion.) Such a delay makes it unusable
>    for real-time streams.
>
>
>    - On Chrome for Android, the video would not start. I believe this to
>    be unrelated though to the subject (chrome not allowing any autoplay and
>    hiding video controls without a source?)
>
> This delay was initially seen with an H264 video stream on an Android
> device and was reproduced in an audio-only stream before testing sendrecv.
> I've had others able to reliably reproduce the issue on other iOS devices
> but it seems harder to replicate on any given Android.
>
> With the problem on the receiving end, I would first think that the issue
> is with browser implementations of webrtc but with control being extremely
> limited on that side, I'm not really sure where else to look except at what
> can be done on the GStreamer and webrtcbin side.
>
> Has anyone else seen this behavior and found a means to counteract it
> without delaying the whole stream?
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>
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