questions on rtp jitterbuffer

[Yu You]

In general it would be very useful to have more information about the
GStreamer you used like the versions and platforms when asking questions.
Have you tried to increase the value of the "latency" property in the
"webrtcbin" element e.g. to 500? The "latency" value should go to the
rtpjitterbuffer. The default value is 200 ms.

Regards,

Yu

On Thu, 29 Dec 2022 at 11:30, Pradeep Acharya via gstreamer-devel <
gstreamer-devel at lists.freedesktop.org> wrote:

> Hi,
>    i'm using webtcbin plugin to develop video SFU server. . we see video
> freezing very often on the browser side as the frames get dropped and when
> PLI requests are sent by the browser, it's forwarded to the appropriate
> client to generate the keyframe. video freeze recovers when the browser
> sends a key frame..
>   SDP negotiations for RTX payload are proper and RTX packets are also
> sent and received properly by client and our SFU server.
>
> As per the jitterbuffer logs, However, in some situations , when the RTT
> time increases to ~200ms , if packets are not arrived with-in the
> configured jitter buffer latency (for example 500ms), the packets are
> dropped by rtpjitterbuffer. Even the RTX packets sent by the client are
> dropped. What are changes required to be done in rtpjitterbuffer element so
> that packets are not dropped ?
>
> Regards
> Pradeep
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20221230/57a10b89/attachment.htm>