[SR-Users] What going on this SDP
Richard Fuchs
rfuchs at sipwise.com
Thu Apr 3 18:11:33 CEST 2014
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 at 10.1.1.101 SIP/2.0
> Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581
> Max-Forwards: 69
> To: <sip:9822 at 10.1.1.101>
> From: <sip:webrtc at 10.10.10.48>;tag=6tmeble9ov
> Call-ID: 43oclsi0sva6n347bk5c
> CSeq: 7435 INVITE
> Contact: <sip:ce5egl03 at 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 at 10.1.1.101>
> From: <sip:webrtc at 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 at 10.1.1.101 out_uri=sip:9822 at 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 at 10.10.10.48>;tag=6tmeble9ov
> To: <sip:9822 at 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 at 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 at 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 at 10.1.1.101>;tag=as06b3db08
> From: <sip:webrtc at 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 at 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 at 10.1.1.101>;tag=as06b3db08
> From: <sip:webrtc at 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 at 10.10.10.48>;tag=6tmeble9ov
> To: <sip:9822 at 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
>
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