[SR-Users] What going on this SDP

jaflong jaflong jaflong at yandex.com
Thu Apr 3 16:22:06 CEST 2014 


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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