<div dir="ltr"><div>Hello everyone, </div><div>I am trying to build a conference engine with WebRTC using GStremer 1.14.5 on Ubuntu 18.04. Been referring to the webrtc-sendrecv.c as an example.</div>Right now, I am trying to stream audio and video to and from the same Linux box. I am having trouble with set-remote-description on webrtcbin. I get this error after receiving a SDP Answer from the browser. ERROR:gstwebrtcbin.c:1152:_check_if_negotiation_is_needed: assertion failed: (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp)) >Aborted (core dumped) A GStreamer peer to GStreamer peer works when both have equal number of streams (medias, is it?) i.e. Just video or just audio or both audio and video. Here is the SDP Answer received from the browser. I think I am messing up the negotiation somewhere. I would really appreciate some help understanding what’s happening. v=0 o=mozilla...THIS_IS_SDPARTA-82.0 810706521954309706 0 IN IP4 0.0.0.0 s=- t=0 0 a=fingerprint:sha-256 C6:C3:86:8E:CB:D2:14:9C:C1:4A:B8:72:8D:60:00:08:34:51:33:79:FE:2D:39:4A:7C:E6:F5:97:92:5D:97:C4 a=ice-options:trickle a=msid-semantic:WMS * m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=sendrecv a=fmtp:96 max-fs=12288;max-fr=60 a=ice-pwd:4d7d2212bcd5f1f467855683f036f700 a=ice-ufrag:0594f02a a=mid:video0 a=msid:{1d029bf1-6123-4b39-be26-68202dad363f} {7c72666b-94ec-422b-8cc6-5010b8aa73ab} a=rtcp-fb:96 nack pli a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=setup:active a=ssrc:1864073429 cname:{4dad80e7-eb43-4706-8934-4a2d51202c21} m=audio 9 UDP/TLS/RTP/SAVPF 97 c=IN IP4 0.0.0.0 a=sendrecv a=fmtp:97 maxplaybackrate=48000;stereo=1;useinbandfec=1 a=ice-pwd:4d7d2212bcd5f1f467855683f036f700 a=ice-ufrag:0594f02a a=mid:audio1 a=msid:{1d029bf1-6123-4b39-be26-68202dad363f} {4607068a-ba2b-4d70-8561-12f13013b345} a=rtcp-mux a=rtpmap:97 opus/48000/2 a=setup:active a=ssrc:4273581052 cname:{4dad80e7-eb43-4706-8934-4a2d51202c21} Here is the C code I’m using for set-remote-description. static void c_peer_set_sdp_answer (CPeer *cpeer, Conference *conf, gchar *sdp_answer) { { g_message ("Setting SDP Answer: %s", sdp_answer); int ret; GstSDPMessage *sdp; GstWebRTCSessionDescription *answer; GstPromise *promise; ret = gst_sdp_message_new (&sdp); g_assert_cmphex (ret, ==, GST_SDP_OK); ret = gst_sdp_message_parse_buffer ((guint8 *) sdp_answer, strlen (sdp_answer), sdp); g_assert_cmphex (ret, ==, GST_SDP_OK); answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, sdp); g_assert_nonnull (answer); promise = gst_promise_new (); //This is where things go wrong g_signal_emit_by_name (conf->webrtcbin, "set-remote-description", answer, promise); gst_promise_interrupt (promise); gst_promise_unref (promise); } Is there something I could be missing in the process before set-remote-description? May be in set-local-description? From the GST_DEBUG=4 level logs, is this something I need to look into? 0:00:41.148395569 9264 0x55dc1f3b9590 INFO webrtcbin gstwebrtcbin.c:2795:_set_description_task:<webrtcbin> Attempting to set remote answer in the have-local-offer state <div>Here is the link to the what seems to be the relevant section of the logs <a href="https://pastebin.com/aiPMsZgG">https://pastebin.com/aiPMsZgG</a></div><div> </div><div>Regards</div><div>Mandar </div> </div>