Webrtc plugin missed ssrc

[Hassan Muhammad]

Hi there,

I'm running into a similar issue with my webrtc pipeline when using H264
where ssrc is missed all the time however, it works just fine when I'm
using vp8. Can you tell me how exactly were you able to add a delay between
caps negotiation with the "notify::caps" signal, as I'm more of a beginner
and not entirely sure how to do this?

Thanks.

On Thu, Feb 4, 2021 at 9:53 PM Vladimir Tyutin <vladimir.tyutin at gmail.com>
wrote:

> 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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>>> gstreamer-devel at lists.freedesktop.org
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>>>
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