How to restart webrtcbin plugin without stopping whole pipeline

[Vladimir Tyutin]

Hi Zachary,
Thanks for your quick reply. Sounds like a very good idea. Let me try your
approach and will get back to you with the results.

Thanks,
Vladimir

On Mon, Dec 14, 2020 at 12:46 PM Zachary Hueras <ZHueras at amdtelemedicine.com>
wrote:

> ​I would recommend something akin to your latter approach. When a call
> comes in, you would modify the pipeline to include a new webrtcbin to
> handle it. Insert a tee element after capsfilter and link that to your
> webrtcbins
>
>
> When a call comes in, grab a new pad from the tee. Create a queue,
> rtph264pay, and a webrtcbin; add them to your pipeline, and link them
> together (queue -> rtph264pay -> webrtcbin). Link the pad from the tee to
> the queue's src pad. Do all your normal webrtc setup, and finally
> call gst_element_sync_state_with_parent on the queue, rtph264pay, and
> webrtcbin elements.​
>
>
> Zachary Hueras
> Principal Software Engineer
> AMD Global Telemedicine, Inc.
> Cell : 978-660-3812
> zhueras at amdtelemedicine.com
> www.amdtelemedicine.com | Follow us on LinkedIn
> ------------------------------
> *From:* gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org>
> on behalf of Vladimir Tyutin <vladimir.tyutin at gmail.com>
> *Sent:* Monday, December 14, 2020 3:58 AM
> *To:* gstreamer-devel at lists.freedesktop.org
> *Subject:* How to restart webrtcbin plugin without stopping whole pipeline
>
> Hello gstreamer experts,
> I need your advice on the issue below.
> I have a pipeline that records mpeg ts files (2 different video
> resolutions and audio) and webrtc (please see pipeline below.
> v536videosrc is my plugin that produces h264 encoded frames).
> Everything works fine for the first webrtc call. Now when the call is over
> I need somehow to reset webrtc to the initial state to be prepared for the
> new call.
> I tried two approaches:
> 1. Block capsfilter src before webrtc and move webrtcbin to state NULL and
> PLAYING again and unblock capsfilter.
> 2. Block capsfilter src before webrtc, remove webrtcbin from pipeline
> recreate it and add to pipeline again, link and unblock capsfilter.
>
> Both approaches do not work. The major issue I observe is that when webrtc
> moves to NULL state it tries to set NULL state to all plugins inside
> webrtcbin. And it hangs somewhere in rtpjittbuffer or so despite I do
> everything on a separate thread (not main loop thread).
>
> Please advise what is the correct way to reset webrtcbin to initial state
> and get it ready for a new incoming call without stopping the whole
> pipeline.
>
> Here is my pipeline example:
> #define 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 " " \
>                     "mpegtsmux name=fullhdts ! hlssink max-files=100
> target-duration=10 location=/mnt/ramdisk/fullhd_%09d.ts " \
>                     "mpegtsmux name=vgats ! hlssink max-files=100
> target-duration=10 location=/mnt/ramdisk/vga_%09d.ts " \
>                     "v536videosrc sys-init=false device=1 channel=0
> encoder=0 format=H264 width=1920 height=1080 ! video/x-h264,
> stream-format=byte-stream, alignment=au, profile=baseline ! queue !
> h264parse ! fullhdts. " \
>                     "v536videosrc sys-init=false device=1 channel=1
> encoder=1 format=H264 width=640 height=480 ! video/x-h264,
> stream-format=byte-stream, alignment=au, profile=baseline ! tee name=tv !
> queue ! h264parse ! vgats. " \
>                     "alsasrc ! tee name=t ! queue ! avenc_aac ! aacparse !
> fullhdts. " \
>                     "t. ! queue ! avenc_aac ! aacparse ! vgats. " \
>                     "tv. ! queue name=videoqueue leaky=downstream
> max-size-buffers=25 ! rtph264pay name=vrtp ! capsfilter name=vrtpcaps
> caps=" RTP_CAPS_H264 "96 ! webrtc. " \
>                     "t. ! queue name=audioqueue leaky=downstream !
> audioconvert name=aconvert ! opusenc ! rtpopuspay name=artp ! capsfilter
> name=artpcaps caps=" RTP_CAPS_OPUS "97 ! webrtc. "
>
> Thanks,
> Vladimir
>
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