[SR-Users] What going on this SDP
Rainer Piper
rainer.piper at soho-piper.de
Fri Apr 4 09:30:17 CEST 2014
upps ... sorry ... *pass th**rough* and not path through :-[
2013-08-23 15:49 +0000 [r397524-397527] Matthew Jordan <mjordan at digium.com>
* CHANGES: Update CHANGES file to reflect pass through support for
Opus/VP8
Source ->
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.2.0-rc1
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 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:
>>>
>>> ACKsip: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:
>>>
>>> BYEsip: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
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at 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 at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
*Rainer Piper*
NOC - +49 (0)228 97167161 - sip.soho-piper.de
NOC - +49 (0)2247 9064188 - sip.tele33.de - sip.tefonix.de - D293
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140404/679fbea4/attachment.html>
More information about the sr-users
mailing list