webrtcbin

[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 <sebastian at centricular.com>
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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