My guess would be that it's due to a discrepancy between WebRTC and RFC 5764. WebRTC uses a protocol string of "RTP/SAVPF", while the RFC says that "UDP/TLS/RTP/SAVPF" shall be used. You can try an SDP rewrite operation to substitute one for the other. Or teach your non-RTC client to use a different protocol string.
cheers On 04/03/14 10:22, jaflong jaflong wrote: > > > Hi List, > > Can anyone help me understand why this is getting rejected > > Please note the specific message further dow the log. > "Failed to parse SessionDescription. Failed to parse audio codecs correctly" > This is on Chrome. > > On Firefox There is a further message in the console > "Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: > Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: > Invalid port format (17296) specified for transport protocol (Unsupported), > parse failed." > > JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp > 2113937151 10.10.10.63 65223 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp > 2113937151 10.10.10.63 65223 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp > 2113937151 192.168.56.1 65224 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp > 2113937151 192.168.56.1 65224 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp > 1509957375 10.10.10.63 0 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp > 1509957375 10.10.10.63 0 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp > 1509957375 192.168.56.1 0 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp > 1509957375 192.168.56.1 0 typ host generation 0 > jssip-0.3.0.js:3369 > JsSIP | TRANSPORT | sending WebSocket message: > > INVITE sip:9822@10.1.1.101 SIP/2.0 > Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581 > Max-Forwards: 69 > To: <sip:9822@10.1.1.101> > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7435 INVITE > Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob> > Allow: ACK,CANCEL,BYE,OPTIONS,INVITE > Content-Type: application/sdp > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.0 > Content-Length: 1744 > > v=0 > o=- 3746191339358890844 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 > m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 10.10.10.63 > a=rtcp:65223 IN IP4 10.10.10.63 > a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 > a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 > a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host > generation 0 > a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host > generation 0 > a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 > a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 > a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 > a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 > a=ice-ufrag:Dgp8HIJdmr1lFPCQ > a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9 > a=ice-options:google-ice > a=fingerprint:sha-256 > C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=sendrecv > a=rtcp-mux > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY > a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 > 768e2e47-bc86-473d-bc2c-6e2340ace772 > a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 > a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772 > > jssip-0.3.0.js:519 > JsSIP | TRANSPORT | received WebSocket text message: > > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/WSS > flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63 > To: <sip:9822@10.1.1.101> > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7435 INVITE > Server: DXI WebRTC > Content-Length: 0 > Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455 > req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101 > out_uri=sip:9822@10.10.10.111:5443 via_cnt==1" > > jssip-0.3.0.js:670 > JsSIP | TRANSPORT | received WebSocket text message: > > SIP/2.0 200 OK > Via: SIP/2.0/WSS > flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581 > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > To: <sip:9822@10.1.1.101>;tag=as06b3db08 > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7435 INVITE > Server: Easycall > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Contact: <sip:9822@10.10.10.111:5443;transport=TLS> > Content-Type: application/sdp > Content-Length: 801 > > v=0 > o=root 431209641 431209641 IN IP4 10.10.10.111 > s=Asterisk PBX 12.2.0-rc1 > c=IN IP4 10.10.10.111 > t=0 0 > m=audio 30490 UDP/TLS/RTP/SAVPF 0 126 > a=rtpmap:0 PCMU/8000 > a=rtpmap:126 telephone-event/8000 > a=fmtp:126 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=maxptime:150 > a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138 > a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092 > a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host > a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx > a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host > a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx > a=connection:new > a=setup:active > a=fingerprint:SHA-256 > 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 > a=sendrecv > > jssip-0.3.0.js:670 > Failed to parse SessionDescription. Failed to parse audio codecs correctly. > jssip-0.3.0.js:4512 > JsSIP | DIALOG | new UAC dialog created with status CONFIRMED > jssip-0.3.0.js:2523 > JsSIP | TRANSPORT | sending WebSocket message: > > ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 > Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640 > Max-Forwards: 69 > To: <sip:9822@10.1.1.101>;tag=as06b3db08 > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7435 ACK > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.0 > Content-Length: 0 > > jssip-0.3.0.js:519 > JsSIP | TRANSPORT | sending WebSocket message: > > BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 > Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766 > Max-Forwards: 69 > To: <sip:9822@10.1.1.101>;tag=as06b3db08 > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7436 BYE > Reason: SIP ;cause=488; text="Not Acceptable Here" > Supported: path, outbound, gruu > User-Agent: JsSIP 0.3.0 > Content-Length: 0 > > jssip-0.3.0.js:519 > JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov > jssip-0.3.0.js:4193 > JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392 > JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted > jssip-0.3.0.js:2543 > JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187 > JsSIP | TRANSPORT | received WebSocket text message: > > SIP/2.0 200 OK > Via: SIP/2.0/WSS > flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766 > From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov > To: <sip:9822@10.1.1.101>;tag=as06b3db08 > Call-ID: 43oclsi0sva6n347bk5c > CSeq: 7436 BYE > Server: Easycall > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Length: 0 > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
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