Re: gstreamer webrtcbin
Jerry Geis via gstreamer-devel <[email protected]> Sat, 17 May 2025 05:25:46 -0400
| Newsgroups | gmane.comp.video.gstreamer.devel |
|---|---|
| Message-ID | <CABr8-B4Fhe8JNNEwD6+i5NAaoaWqEvcTNtF_+74t3ZfRg6sYAg@mail.gmail.com> |
--00000000000086195e0635517ca6 Content-Type: text/plain; charset="UTF-8" Content-Transfer-Encoding: quoted-printable 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=E2=80=AFAM 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 > --00000000000086195e0635517ca6 Content-Type: text/html; charset="UTF-8" Content-Transfer-Encoding: quoted-printable <div dir=3D"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 t= hat would use webrtc originate a call -=C2=A0 connect the call for audio an= d video (h264) and then output multicast audio / video.</div><div>I am tryi= ng to find a way to accomplish this.</div><div>Thanks</div><div><br></div><= div>Jerry</div></div><br><div class=3D"gmail_quote gmail_quote_container"><= div dir=3D"ltr" class=3D"gmail_attr">On Sat, May 17, 2025 at 4:00=E2=80=AFA= M Matthew Waters <<a href=3D"mailto:[email protected]">ystreet00@gmail= .com</a>> wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"mar= gin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1= ex">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=C2=A0to 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> --00000000000086195e0635517ca6--