[SR-Users] What going on this SDP

Fri Apr 4 08:18:22 CEST 2014

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 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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>>
>
>
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-- 
*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
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