<div dir="ltr">Thanks for the response. First, my application is already doing lots of other video/audio processing with GStreamer and I would really like to stick with it if possible. Second, I much prefer it over the massive Chromium based WebRTC implementation from Google...not that GStreamer is small. :-)<div><br></div><div>Can anyone say for sure that webrtcbin doesn't currently support data channels only?</div><div><br></div><div>Bill</div><div><div><br></div><div><br><div><br></div><div> </div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Jul 17, 2020 at 3:20 PM Anton Pryima <<a href="mailto:zingfrid@gmail.com">zingfrid@gmail.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"><div dir="ltr">Hello Bill.<div><br></div><div>As you need only a data channel, but neither video nor audio why do you need Gstreamer? Gstreamer is first of all audio/video framework. It is more useful to use 3rd party webrtc libraries like (<a href="https://webrtc.googlesource.com/src/+/master/api" target="_blank">https://webrtc.googlesource.com/src/+/master/api</a>).</div><div><br></div><div>Best regards,</div><div>Anton.</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Jul 17, 2020 at 5:10 PM William Gerecke <<a href="mailto:gerecke@gmail.com" target="_blank">gerecke@gmail.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"><div dir="ltr">I'm still having this problem first posted in March. I'm using GStreamer webrtcbin to send data over a WebRTC data channel and all is working well! The only problem is for the data channel to be established I also need to specify a dummy audio stream as follows:<div><br></div><div>pipe1 = gst_parse_launch("webrtcbin name=sendrecv "<br> "audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "<br> "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);<br></div><div><br></div><div>When I try webrtcbin by itself or with a fakesrc instead of an audio source I can still create the data channel as below without errors but I never get the "on-open" callback like I do when an audio source is present.</div><div><br></div><div>g_signal_emit_by_name(webrtc1, "create-data-channel", "channel", NULL, &send_channel);<br> if (send_channel) {<br> g_print("Created data channel\n");<br> connect_data_channel_signals(send_channel, session);</div><div>}<br></div><div><br></div><div>So in short - can webrtcbin be configured to work with only data channels, and if so, what am I missing?</div><div><br></div><div>Bill</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, Mar 31, 2020 at 11:30 AM Bill G <<a href="mailto:foatus@hotmail.com" target="_blank">foatus@hotmail.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"> <div dir="ltr"> <div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)"> </div> <span>Hello,<br> </span> <div><br> </div> <div>I'm trying to get a webrtcbin running which has data channels only (i.e. no audio, no video.) I started with the working sendrecv example and got to the point where I had a working data-channel with audio only. When I remove audio the data channels fail to connect - one data channel is created by GStreamer code, another intiated on the browser side. <br> </div> <div><br> </div> <div>I found and example (link below) for only receiving streams in which the gst_parse_launch() was removed and one-way transceivers manually added. I read that without a audio/video sink pad connected the transceivers need to be manually created. Maybe this is also related to data channels not functioning?<br> </div> <div><br> </div> <div><a href="https://github.com/centricular/gstwebrtc-demos/compare/master...a-morales:figure-out-transceivers?expand=1" target="_blank">https://github.com/centricular/gstwebrtc-demos/compare/master...a-morales:figure-out-transceivers?expand=1</a><br> </div> <div><br> </div> <span>So, I'm assuming there is something preventing these data-channels from getting established, and asking how to get past it? <span style="background-color:rgb(255,255,255);display:inline"> This is with everything running on one machine, Windows and GStreamer 1.16. </span> Thanks in advance!</span> <div style="font-family:Calibri,Helvetica,sans-serif;font-size:12pt;color:rgb(0,0,0)"> </div> </div> _______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br> </blockquote></div> _______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br> </blockquote></div> _______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a><br> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br> </blockquote></div>