<div dir="ltr">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.</div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Mar 27, 2020 at 3:15 AM Sebastian Dröge <<a href="mailto:sebastian@centricular.com">sebastian@centricular.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On Thu, 2020-03-26 at 19:47 -0400, Jerry Geis wrote:<br>
> How do I use webrtcbin on a local SIP call ?<br>
> <br>
> The same server running asterisk will be running webrtcbin.<br>
> <br>
> I need gstreamer to grab the H264 video and audio and run the rest of<br>
> the pipeline.<br>
<br>
webrtcbin is for WebRTC and can't do SIP. For SIP you'd need to either<br>
use Farstream, or build something similar to webrtcbin for SIP.<br>
<br>
-- <br>
Sebastian Dröge, Centricular Ltd · <a href="https://www.centricular.com" rel="noreferrer" target="_blank">https://www.centricular.com</a><br>
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</blockquote></div>