<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"> <head> <meta http-equiv="Content-Type" content="text/html; charset=us-ascii"> <meta name="Generator" content="Microsoft Word 15 (filtered medium)"> <style></style> </head> <body lang="EN-US" link="#0563C1" vlink="#954F72"> <div class="WordSection1"> Hi all,<o:p></o:p> <o:p> </o:p> I've been trying to offer a WebRTC stream from a gstreamer pipeline to Chrome without much success. I'm able to reproduce the same issue with the Centricular webrtc-sendrecv example:<o:p></o:p> <o:p> </o:p> <a href="https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst/webrtc-sendrecv.c"> https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst/webrtc-sendrecv.c</a><o:p></o:p> <o:p> </o:p> The GStreamer app creates and sends offer, the Chrome app responds with an answer and ICE candidates are exchanged but things go downhill from there. On the Chrome side, webrtc-internals show that a candidate pair was successfully selected (googReadable/googWritable are true) but the PeerConnection's ICE connection state gets stuck in "checking". The Chrome debug logs show that there's a DTLS timeout.<o:p></o:p> <o:p> </o:p> On the GStreamer side, with *tls* logs set to level 5, I see this sequence repeatedly:<o:p></o:p> <o:p> </o:p> 0:00:11.835289000 15540 00000286D591D440 DEBUG dtlsdec gstdtlsdec.c:548:sink_chain:<dtlsdec0> received buffer from rtp_0_2873075822 with length 155<o:p></o:p> 0:00:11.847957000 15540 00000286D591D440 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000286D56A6890> process start: role=server buf=(00000286D5D3E130:0/155) 10110|0 before SSL initialization<o:p></o:p> 0:00:11.869501000 15540 00000286D591D440 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000286D56A6890> process after read: role=server buf=(00000286D5D3E130:0/155) 10110|0 before SSL initialization<o:p></o:p> 0:00:11.887298000 15540 00000286D591D440 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000286D56A6890> poll: before handshake: role=server buf=(00000286D5D3E130:0/155) 10110|0 before SSL initialization<o:p></o:p> 0:00:11.904438000 15540 00000286D591D440 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000286D56A6890> poll: after handshake: role=server buf=(00000286D5D3E130:0/155) 10110|0 before SSL initialization<o:p></o:p> 0:00:11.920380000 15540 00000286D591D440 DEBUG dtlsconnection gstdtlsconnection.c:758:openssl_poll:<GstDtlsConnection@00000286D56A6890> do_handshake encountered BIO error<o:p></o:p> 0:00:11.935830000 15540 00000286D591D440 ERROR dtlsconnection gstdtlsconnection.c:771:openssl_poll:<GstDtlsConnection@00000286D56A6890> SSL error<o:p></o:p> 0:00:11.946996000 15540 00000286D591D440 ERROR dtlsconnection gstdtlsconnection.c:727:ssl_err_cb:<GstDtlsConnection@00000286D56A6890> ssl error: 26156:error:140B4090:SSL routines:SSL_do_handshake:connection type not set:ssl/ssl_lib.c:3554:<o:p></o:p> <o:p> </o:p> 0:00:11.968253000 15540 00000286D591D440 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000286D56A6890> process after poll: role=server buf=(00000286D5D3E130:0/155) 10110|0 before SSL initialization<o:p></o:p> 0:00:11.987237000 15540 00000286D591D440 DEBUG dtlsconnection gstdtlsconnection.c:551:gst_dtls_connection_process:<GstDtlsConnection@00000286D56A6890> read result: -1<o:p></o:p> <o:p> </o:p> Based on the "SSL_do_handshake:connection type not set" error, it appears that SSL_set_connect_state()/SSL_set_accept_state() hasn't been called on GstDtlsConnection's OpenSSL handle which should happen in gst_dtls_connection_start() but I never see the corresponding "log_state:initial state set:" message. <o:p> </o:p> <o:p> </o:p> I *am* able to successfully run the gst-plugins-bad/tests/examples/webrtc/webrtc.c app. In that case, after the answer is processed, I see "initial state set" message preceded by creation of a new src pad on the webrtcbin instance which I *don't* see in my problem apps:<o:p></o:p> <o:p> </o:p> 0:00:01.867442000 35404 0000017960746740 DEBUG webrtcbin gstwebrtcbin.c:3145:_update_transceiver_from_sdp_media:<recv> creating new receive pad for transceiver <webrtctransceiver1><o:p></o:p> 0:00:01.881440000 35404 00000179607466C0 DEBUG dtlsenc gstdtlsenc.c:378:src_activate_mode:<dtlsenc0> src pad activating in push mode<o:p></o:p> 0:00:01.896187000 35404 0000017960746740 DEBUG webrtcbin gstwebrtcbin.c:268:gst_webrtc_bin_pad_new:<'':src_0> new visible pad with direction src<o:p></o:p> 0:00:01.910867000 35404 00000179607466C0 DEBUG dtlsenc gstdtlsenc.c:330:gst_dtls_enc_change_state:<dtlsenc0> starting connection rtp_0_324247815<o:p></o:p> 0:00:01.921569000 35404 0000017960746740 INFO webrtcbin gstwebrtcbin.c:2910:_connect_output_stream:<recv> linking output stream 0<o:p></o:p> 0:00:01.936102000 35404 00000179607466C0 LOG dtlsconnection gstdtlsconnection.c:629:log_state:<GstDtlsConnection@00000179607C1EE0> initial state set: role=server buf=(0000000000000000:0/0) 10110|0 before SSL initialization<o:p></o:p> <o:p> </o:p> Here's the Chrome answer:<o:p></o:p> <o:p> </o:p> v=0<o:p></o:p> o=- 671422098252769522 2 IN IP4 127.0.0.1<o:p></o:p> s=-<o:p></o:p> t=0 0<o:p></o:p> a=group:BUNDLE video0 audio1<o:p></o:p> a=msid-semantic: WMS y6ekH83vxbrWGYgIHXshcbQUTSYjesz8cTQU<o:p></o:p> m=video 9 UDP/TLS/RTP/SAVPF 96<o:p></o:p> c=IN IP4 0.0.0.0<o:p></o:p> a=rtcp:9 IN IP4 0.0.0.0<o:p></o:p> a=ice-ufrag:UN/v<o:p></o:p> a=ice-pwd:5pyFZoXXTB6g/AAPq2ASVSdt<o:p></o:p> a=ice-options:trickle<o:p></o:p> a=fingerprint:sha-256 98:9C:D6:67:6D:C8:8D:80:61:A1:C0:08:89:6B:1C:99:56:AE:EF:00:61:22:1B:27:CB:20:2F:FB:13:C2:FE:32<o:p></o:p> a=setup:active<o:p></o:p> a=mid:video0<o:p></o:p> a=sendrecv<o:p></o:p> a=rtcp-mux<o:p></o:p> a=rtcp-rsize<o:p></o:p> a=rtpmap:96 VP8/90000<o:p></o:p> a=rtcp-fb:96 nack pli<o:p></o:p> a=ssrc:1873007401 cname:PbAeaJtZ3DsTAyc8<o:p></o:p> a=ssrc:1873007401 msid:y6ekH83vxbrWGYgIHXshcbQUTSYjesz8cTQU e5c18420-14d4-4ccd-b884-4f81b51081c5<o:p></o:p> a=ssrc:1873007401 mslabel:y6ekH83vxbrWGYgIHXshcbQUTSYjesz8cTQU<o:p></o:p> a=ssrc:1873007401 label:e5c18420-14d4-4ccd-b884-4f81b51081c5<o:p></o:p> m=audio 9 UDP/TLS/RTP/SAVPF 97<o:p></o:p> c=IN IP4 0.0.0.0<o:p></o:p> a=rtcp:9 IN IP4 0.0.0.0<o:p></o:p> a=ice-ufrag:UN/v<o:p></o:p> a=ice-pwd:5pyFZoXXTB6g/AAPq2ASVSdt<o:p></o:p> a=ice-options:trickle<o:p></o:p> a=fingerprint:sha-256 98:9C:D6:67:6D:C8:8D:80:61:A1:C0:08:89:6B:1C:99:56:AE:EF:00:61:22:1B:27:CB:20:2F:FB:13:C2:FE:32<o:p></o:p> a=setup:active<o:p></o:p> a=mid:audio1<o:p></o:p> a=sendrecv<o:p></o:p> a=rtcp-mux<o:p></o:p> a=rtpmap:97 OPUS/48000/2<o:p></o:p> a=fmtp:97 minptime=10;useinbandfec=1<o:p></o:p> a=ssrc:4026464680 cname:PbAeaJtZ3DsTAyc8<o:p></o:p> a=ssrc:4026464680 msid:y6ekH83vxbrWGYgIHXshcbQUTSYjesz8cTQU b7c37d92-042c-417e-a4ab-79fdbe577974<o:p></o:p> a=ssrc:4026464680 mslabel:y6ekH83vxbrWGYgIHXshcbQUTSYjesz8cTQU<o:p></o:p> a=ssrc:4026464680 label:b7c37d92-042c-417e-a4ab-79fdbe577974<o:p></o:p> <o:p> </o:p> And the webrtc.c example answer:<o:p></o:p> <o:p> </o:p> v=0<o:p></o:p> o=- 2260608518856376157 0 IN IP4 0.0.0.0<o:p></o:p> s=-<o:p></o:p> t=0 0<o:p></o:p> a=ice-options:trickle<o:p></o:p> m=video 9 UDP/TLS/RTP/SAVPF 96<o:p></o:p> c=IN IP4 0.0.0.0<o:p></o:p> a=ice-ufrag:tkt9ckCYhpGXYbx3vOHGSN68tTAOlomF<o:p></o:p> a=ice-pwd:Mvu5cUlmqpvF6xrssiXcYOr0ShtJCOwn<o:p></o:p> a=rtcp-mux<o:p></o:p> a=mid:video0<o:p></o:p> a=setup:active<o:p></o:p> a=rtpmap:96 VP8/90000<o:p></o:p> a=rtcp-fb:96 nack pli<o:p></o:p> a=recvonly<o:p></o:p> a=fingerprint:sha-256 DD:30:10:C2:F2:60:91:85:9F:6F:ED:BF:0A:23:AB:60:0B:FB:08:8E:04:83:D9:A1:DF:26:54:BC:A4:E8:59:BD<o:p></o:p> <o:p> </o:p> My system config:<o:p></o:p> Windows 10 <o:p></o:p> GStreamer x86_64 v1.15.2 (I've also tried v1.14.4 with the same result)<o:p></o:p> Chrome 73.0.3683.103<o:p></o:p> <o:p> </o:p> Any ideas as to what could be going wrong would be EXTREMELY appreciated!<o:p></o:p> <o:p> </o:p> Best,<o:p></o:p> Eric<o:p></o:p> <o:p> </o:p> </div> </body> </html>