Re: gstreamer webrtcbin
Matthew Waters via gstreamer-devel <[email protected]> Sat, 17 May 2025 21:13:12 +1000
| Newsgroups | gmane.comp.video.gstreamer.devel |
|---|---|
| Message-ID | <[email protected]> |
This is a multi-part message in MIME format. --------------gpVCIaQO6YH40zWLW0kAn6vB Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit 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 <[email protected]> > 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 > --------------gpVCIaQO6YH40zWLW0kAn6vB Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: 8bit <!DOCTYPE html> <html> <head> <meta http-equiv="Content-Type" content="text/html; charset=UTF-8"> </head> <body> <p>I am not aware of anything out of the box for asterisk.<br> <br> 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.<br> <br> Someone would thus need to write that integration code with asterisk.<br> <br> 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.<br> <br> Cheers<br> -Matt<br> </p> <div class="moz-cite-prefix">On 17/5/25 19:25, Jerry Geis wrote:<br> </div> <blockquote type="cite" cite="mid:CABr8-B4Fhe8JNNEwD6+i5NAaoaWqEvcTNtF_+74t3ZfRg6sYAg@mail.gmail.com"> <meta http-equiv="content-type" content="text/html; charset=UTF-8"> <div dir="ltr">Hi Matt, <div><br> </div> <div>I was "hoping" there was something out there that would just work?</div> <div>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.</div> <div>I am trying to find a way to accomplish this.</div> <div>Thanks</div> <div><br> </div> <div>Jerry</div> </div> <br> <div class="gmail_quote gmail_quote_container"> <div dir="ltr" class="gmail_attr">On Sat, May 17, 2025 at 4:00 AM Matthew Waters <<a href="mailto:[email protected]" moz-do-not-send="true" class="moz-txt-link-freetext">[email protected]</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">What have you tried?<br> <br> You would need to at least write some kind of signalling conversion <br> between asterisk and the pieces that GStreamer provides.<br> <br> Cheers<br> -Matt<br> <br> On 16/5/25 05:49, Jerry Geis via gstreamer-devel wrote:<br> > Hi - I am trying to figure out how to use gst-launch-1.0 (ubuntu <br> > 24.04) with asterisk?<br> ><br> > I am trying to get the video/audio out of a call. WEBRTC is working on <br> > asterisk well.<br> > I desire to connect with gstreamer and have audio video with <br> > autovideosink.<br> ><br> > How might I do that ?<br> > Thanks<br> ><br> > jerry<br> </blockquote> </div> </blockquote> </body> </html> --------------gpVCIaQO6YH40zWLW0kAn6vB--