gstreamer webrtcbin

[Matthew Waters]

I am not aware of anything out of the box for asterisk.

Signalling is explicitly out of scope for WebRTC and must be implemented 
independently for each service/application/framework. As such each 
service provides their own signalling API and that must be integrated 
with the pieces that GStreamer provides.

Someone would thus need to write that integration code with asterisk.

The webrtcsink/webrtcsrc elements from gst-plugins-rs does already have 
some integration with other services that provide WebRTC such as 
livekit, AWS KVS, etc.

Cheers
-Matt

On 17/5/25 19:25, Jerry Geis wrote:
> Hi Matt,
>
> I was "hoping" there was something out there that would just work?
> I need a command that would use webrtc originate a call - connect the 
> call for audio and video (h264) and then output multicast audio / video.
> I am trying to find a way to accomplish this.
> Thanks
>
> Jerry
>
> On Sat, May 17, 2025 at 4:00 AM Matthew Waters <ystreet00 at gmail.com> 
> wrote:
>
>     What have you tried?
>
>     You would need to at least write some kind of signalling conversion
>     between asterisk and the pieces that GStreamer provides.
>
>     Cheers
>     -Matt
>
>     On 16/5/25 05:49, Jerry Geis via gstreamer-devel wrote:
>     > Hi - I am trying to figure out how to use gst-launch-1.0 (ubuntu
>     > 24.04) with asterisk?
>     >
>     > I am trying to get the video/audio out of a call. WEBRTC is
>     working on
>     > asterisk well.
>     > I desire to connect with gstreamer and have audio video with
>     > autovideosink.
>     >
>     > How might I do that ?
>     > Thanks
>     >
>     > jerry
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20250517/83ccb493/attachment.htm>