webrtcbin

[Jerry Geis-2]

How do I use webrtcbin on a local SIP call ?

The same server running asterisk will be running webrtcbin.

I need gstreamer to grab the H264 video and audio and run the rest of the pipeline.

How do you do that ?

Jerry

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[Sebastian Dröge-3]

On Thu, 2020-03-26 at 19:47 -0400, Jerry Geis wrote:
> How do I use webrtcbin on a local SIP call ?
>
> The same server running asterisk will be running webrtcbin.
>
> I need gstreamer to grab the H264 video and audio and run the rest of
> the pipeline.

webrtcbin is for WebRTC and can't do SIP. For SIP you'd need to either
use Farstream, or build something similar to webrtcbin for SIP.

--
Sebastian Dröge, Centricular Ltd · https://www.centricular.com

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[Shishir Pokharel]

Instead of Asterisk use Freeswitch which supports RTMP protocol, Use that RTMP stream with rtmpsrc in gstreamer to get the  H264 video and audio  on gstreamer pipeline.

On Fri, Mar 27, 2020 at 3:15 AM Sebastian Dröge <[hidden email]> wrote:

On Thu, 2020-03-26 at 19:47 -0400, Jerry Geis wrote:
> How do I use webrtcbin on a local SIP call ?
>
> The same server running asterisk will be running webrtcbin.
>
> I need gstreamer to grab the H264 video and audio and run the rest of
> the pipeline.

webrtcbin is for WebRTC and can't do SIP. For SIP you'd need to either
use Farstream, or build something similar to webrtcbin for SIP.

--
Sebastian Dröge, Centricular Ltd · https://www.centricular.com
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