Webrtc plugin missed ssrc

Trey Hutcheson trey.hutcheson at gmail.com
Thu Feb 4 15:46:34 UTC 2021


You need to wait until webrtcbin's sink pads get caps from upstream. You
can connect to the notify::caps signal on both sink pads and verify caps
are there before generating the offer.

On Thu, Feb 4, 2021 at 9:38 AM Vladimir Tyutin <vladimir.tyutin at gmail.com>
wrote:

> Hi all,
> I use webrtcbin for video and audio streaming (see my pipeline below).
> The issue that when webrtcbin generates SDP offers in 60% cases a:ssrc
> parameter is missed for video stream. In 40% cases it's generated.
> It's important because Android client does not create remote video track
> if ssrc is missed.
> So how to force webrtcbin to generate ssrc all the time?
>
> Here is my pipeline:
> #define WEBRTC_PIPELINE    "webrtcbin name=webrtc " STUN_SERVER_PROP "="
> STUN_1 " " STUN_SERVER_PROP "=" STUN_2 " " STUN_SERVER_PROP "=" STUN_3 " " \
>                     STUN_SERVER_PROP "=" STUN_4 " " STUN_SERVER_PROP "="
> STUN_5 " " TURN_SERVER_PROP "=" TURN_1 " " \
>                     "v536videosrc sys-init=false push_mode=true device=1
> channel=2 encoder=2 format=H264 width=640 height=480 ! video/x-h264,
> stream-format=byte-stream, alignment=au, profile=baseline ! rtph264pay
> rtp-h264aggregate-mode=2 ! capsfilter caps=" RTP_CAPS_H264 "96 ! queue
> leaky=downstream ! webrtc. " \
>                     "alsasrc ! queue leaky=downstream ! audioconvert !
> opusenc ! rtpopuspay ! capsfilter caps=" RTP_CAPS_OPUS "97 ! webrtc.  "
>
>
>
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20210204/f7748872/attachment.htm>


More information about the gstreamer-devel mailing list