Re: SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
Jonas Kellens <[email protected]> Thu, 9 Nov 2023 16:33:27 +0100
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <[email protected]> |
This is a multi-part message in MIME format. --===============4484041594758744660== Content-Type: multipart/alternative; boundary="------------OHqU890aJhOYWfUYYuMislRv" Content-Language: nl This is a multi-part message in MIME format. --------------OHqU890aJhOYWfUYYuMislRv Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit Hello List I would very much like to have some feedback on this. Where do I have to look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in my config ?! Kind regards. Op 28/06/2023 om 16:14 schreef Jonas Kellens: > > Hello list > > > when trying to set up webRTC communications with sipjs client package > (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file > the following : > > > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP c=IN IP4 99.88.77.66... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=candidate:4275385644 1 udp 2122260223 192.168.0.18 57987 typ > host generation 0 network-id 1... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=candidate:3686672562 1 udp 2122194687 172.21.64.1 57988 typ host > generation 0 network-id 2... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=candidate:4292167434 1 udp 1686052607 99.88.77.66 57987 typ > srflx raddr 192.168.0.18 rport 57987 generation 0 network-id 1... > UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=candidate:8380856 1 tcp 1518280447 192.168.0.18 9 typ host > tcptype active generation 0 network-id 1... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=candidate:622132262 1 tcp 1518214911 172.21.64.1 9 typ host > tcptype active generation 0 network-id 2... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=ice-ufrag:zBkv... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=ice-pwd:8LVdZW/AEq7hp898bLtsI/5W... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=ice-options:trickle... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=fingerprint:sha-256 > 92:6B:C7:79:41:B1:42:78:2B:3A:75:8B:0B:D0:C7:4C:7C:4E:4F:2D:03:A2:DA:D9:BB:CE:B2:39:5D:20:A0:EF... > OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=setup:actpass... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=mid:0... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level... > UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=extmap:2 > http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time... > UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=extmap:3 > http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01... > UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR > FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=sendrecv... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=msid:4f4db37d-65ff-4f57-8c1c-b404f976c3fb > cc4a3d72-3e9d-4926-b57c-056b6e7a6d6c... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtcp-mux... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:111 opus/48000/2... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtcp-fb:111 transport-cc... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=fmtp:111 minptime=10;useinbandfec=1... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:63 red/48000/2... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=fmtp:63 111/111... UNSUPPORTED OR FAILED. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:9 G722/8000... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:0 PCMU/8000... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:8 PCMA/8000... OK. > DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) > SDP a=rtpmap:13 CN/8000... OK > > > in sip.conf I have : > > icesupport = yes > > > in rtp.conf I have : > > icesupport=true > > stunaddr=stun.ekiga.net > > > sip peer has everything set for webrtc : > > > Realtime peer: Yes, cached > Prim.Transp. : WS > Allowed.Trsp : WSS > Codecs : (alaw|g729|gsm) > Useragent : SIP.js/0.10.0 > Reg. Contact : sip:[email protected];transport=ws > RTP Engine : asterisk > Encryption : Yes > RTCP Mux : Yes > > avpf = yes > > force_avp =yes > > icesupport = yes > > dtlsenable = yes > > dtlsverify = fingerprint > > dtlssetup = actpass > > dtlsfingerprint = sha-256 > > > Why is there "UNSUPPORTED OR FAILED" in the log when processing > "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag" and > "ice-pwd" in its "SIP/2.0 200 OK" response to the INVITE and thus > sipjs aborts the SIP call with a 488-"Not Acceptable Here". > > > I read about libuuid-devel and uuid-devel are necessary, but I have > these installed ! > > > What am I missing here ?! > > > > > Kind regards. > > --------------OHqU890aJhOYWfUYYuMislRv Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: 8bit <html> <head> <meta http-equiv="Content-Type" content="text/html; charset=UTF-8"> </head> <body> <p>Hello List</p> <p><br> </p> <p>I would very much like to have some feedback on this. Where do I have to look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in my config ?!<br> </p> <p><br> </p> <p>Kind regards.</p> <p><br> </p> <p><br> </p> <div class="moz-cite-prefix">Op 28/06/2023 om 16:14 schreef Jonas Kellens:<br> </div> <blockquote type="cite" cite="mid:[email protected]"> <meta http-equiv="Content-Type" content="text/html; charset=UTF-8"> <p>Hello list</p> <p><br> </p> <p>when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following :</p> <p><br> </p> <p>DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:4275385644 1 udp 2122260223 192.168.0.18 57987 typ host generation 0 network-id 1... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:3686672562 1 udp 2122194687 172.21.64.1 57988 typ host generation 0 network-id 2... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:4292167434 1 udp 1686052607 99.88.77.66 57987 typ srflx raddr 192.168.0.18 rport 57987 generation 0 network-id 1... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:8380856 1 tcp 1518280447 192.168.0.18 9 typ host tcptype active generation 0 network-id 1... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:622132262 1 tcp 1518214911 172.21.64.1 9 typ host tcptype active generation 0 network-id 2... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ice-ufrag:zBkv... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ice-pwd:8LVdZW/AEq7hp898bLtsI/5W... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=ice-options:trickle... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fingerprint:sha-256 92:6B:C7:79:41:B1:42:78:2B:3A:75:8B:0B:D0:C7:4C:7C:4E:4F:2D:03:A2:DA:D9:BB:CE:B2:39:5D:20:A0:EF... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=setup:actpass... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=mid:0... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:2 <a class="moz-txt-link-freetext" href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" moz-do-not-send="true">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a>... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:3 <a class="moz-txt-link-freetext" href="http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01" moz-do-not-send="true">http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01</a>... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=msid:4f4db37d-65ff-4f57-8c1c-b404f976c3fb cc4a3d72-3e9d-4926-b57c-056b6e7a6d6c... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp-mux... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 opus/48000/2... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp-fb:111 transport-cc... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 minptime=10;useinbandfec=1... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:63 red/48000/2... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:63 111/111... UNSUPPORTED OR FAILED.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.<br> DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK</p> <p><br> </p> <p>in sip.conf I have :</p> <p>icesupport = yes</p> <p><br> </p> <p>in rtp.conf I have :</p> <p>icesupport=true</p> <p>stunaddr=stun.ekiga.net</p> <p><br> </p> <p>sip peer has everything set for webrtc :</p> <p><br> </p> <p> Realtime peer: Yes, cached<br> Prim.Transp. : WS<br> Allowed.Trsp : WSS<br> Codecs : (alaw|g729|gsm)<br> Useragent : SIP.js/0.10.0<br> Reg. Contact : <a class="moz-txt-link-abbreviated moz-txt-link-freetext" href="mailto:sip:[email protected];transport=ws" moz-do-not-send="true">sip:[email protected];transport=ws</a><br> RTP Engine : asterisk<br> Encryption : Yes<br> RTCP Mux : Yes<br> </p> <p>avpf = yes</p> <p>force_avp =yes</p> <p>icesupport = yes</p> <p>dtlsenable = yes</p> <p>dtlsverify = fingerprint</p> <p>dtlssetup = actpass</p> <p>dtlsfingerprint = sha-256</p> <p><br> </p> <p>Why is there "UNSUPPORTED OR FAILED" in the log when processing "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag" and "ice-pwd" in its "SIP/2.0 200 OK" response to the INVITE and thus sipjs aborts the SIP call with a 488-"Not Acceptable Here".<br> </p> <p><br> </p> <p>I read about libuuid-devel and uuid-devel are <span role="heading" aria-level="1" class="yKMVIe">necessary, but I have these installed !</span></p> <p><span role="heading" aria-level="1" class="yKMVIe"><br> </span></p> <p><span role="heading" aria-level="1" class="yKMVIe">What am I missing here ?!</span></p> <p><span role="heading" aria-level="1" class="yKMVIe"><br> </span></p> <p><span role="heading" aria-level="1" class="yKMVIe"><br> </span></p> <p><span role="heading" aria-level="1" class="yKMVIe"><br> </span></p> <p><span role="heading" aria-level="1" class="yKMVIe">Kind regards.</span></p> <p><span role="heading" aria-level="1" class="yKMVIe"><br> </span></p> </blockquote> </body> </html> --------------OHqU890aJhOYWfUYYuMislRv-- --===============4484041594758744660== Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: base64 Content-Disposition: inline LS0gCl9fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19f X19fX19fX19fX19fX19fXwotLSBCYW5kd2lkdGggYW5kIENvbG9jYXRpb24gUHJvdmlkZWQgYnkg aHR0cDovL3d3dy5hcGktZGlnaXRhbC5jb20gLS0KCkNoZWNrIG91dCB0aGUgbmV3IEFzdGVyaXNr IGNvbW11bml0eSBmb3J1bSBhdDogaHR0cHM6Ly9jb21tdW5pdHkuYXN0ZXJpc2sub3JnLwoKTmV3 IHRvIEFzdGVyaXNrPyBTdGFydCBoZXJlOgogICAgICBodHRwczovL3dpa2kuYXN0ZXJpc2sub3Jn L3dpa2kvZGlzcGxheS9BU1QvR2V0dGluZytTdGFydGVkCgphc3Rlcmlzay11c2VycyBtYWlsaW5n IGxpc3QKVG8gVU5TVUJTQ1JJQkUgb3IgdXBkYXRlIG9wdGlvbnMgdmlzaXQ6CiAgIGh0dHA6Ly9s aXN0cy5kaWdpdW0uY29tL21haWxtYW4vbGlzdGluZm8vYXN0ZXJpc2stdXNlcnM= --===============4484041594758744660==--