webrtcbin
Sebastian Dröge
sebastian at centricular.com
Fri Mar 27 09:08:26 UTC 2020
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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 963 bytes
Desc: This is a digitally signed message part
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20200327/67efceda/attachment-0001.sig>
More information about the gstreamer-devel
mailing list