Webrtc plugin missed ssrc

[Vladimir Tyutin]

Thanks Trey,
It seems you are right! Because if I set delay in 1 sec ssrc is added all
the time.
Thanks for your advice I will implement it


On Thu, Feb 4, 2021 at 6:46 PM Trey Hutcheson <trey.hutcheson at gmail.com>
wrote:

> 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.  "
>>
>>
>>
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>>
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