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
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
Hallo, my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end: !!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Regards Rainer
Am 03.04.2014 18:11, schrieb Richard Fuchs:
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
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
upps ... sorry ... *pass th**rough* and not path through :-[
2013-08-23 15:49 +0000 [r397524-397527] Matthew Jordan mjordan@digium.com
* CHANGES: Update CHANGES file to reflect pass through support for Opus/VP8
Source -> http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.2...
Regards Rainer
Am 04.04.2014 08:18, schrieb Rainer Piper:
Hallo, my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end: !!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!! Regards Rainer
Am 03.04.2014 18:11, schrieb Richard Fuchs:
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:
INVITEsip: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:
ACKsip: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:
BYEsip: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
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
-- *Rainer Piper* NOC - +49 (0)228 97167161 callto:004922897167161 - sip.soho-piper.de NOC - +49 (0)2247 9064188 callto:004922479064188 - sip.tele33.de - sip.tefonix.de - D293
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
El Fri, 04 Apr 2014 08:18:22 +0200 Rainer Piper rainer.piper@soho-piper.de escribió:
Hallo, my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end: !!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from the browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" manwe@aholab.ehu.es:
El Fri, 04 Apr 2014 08:18:22 +0200 Rainer Piper rainer.piper@soho-piper.de escribió:
Hallo, my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end: !!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
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
you can try to turn of the opus codec support in the browser.
At firefox ... open about:config and search for media.opus.enabled and set it to false.
At Chrome ... open about:flags and search for opus.
Regards Rainer
Am 04.04.2014 10:31, schrieb jaflong jaflong:
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from the browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" manwe@aholab.ehu.es:
El Fri, 04 Apr 2014 08:18:22 +0200 Rainer Piper rainer.piper@soho-piper.de escribió:
Hallo, my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end: !!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
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
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
-
jaflong jaflong -
Jon Bonilla (Manwe) -
Rainer Piper -
Richard Fuchs