Closed GoogleCodeExporter closed 8 years ago
webrtc2sip log say's:
*******************************************************************
Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>
LICENCE: GPLv3 or proprietary
VERSION: 2.2.0
'quit' to quit the application.
*******************************************************************
SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
*INFO: transport = udp://*:10060
*INFO: transport = ws://*:10060
*INFO: transport = wss://*:10062
*INFO: enable-rtp-symetric = yes
*INFO: enable-100rel = no
*INFO: enable-media-coder = no
*INFO: enable-videojb = yes
*INFO: video-size-pref = vga
*INFO: rtp-buffsize = 65535
*INFO: avpf-tail-length = [100-400]
*INFO: srtp-mode = optional
*INFO: srtp-type = sdes;dtls
*INFO: codecs = 70647920: pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: pipeR fd=6
*INFO: Socket added: fd=6, tail.count=1
*INFO: master fd=3
*INFO: Socket added: fd=3, tail.count=2
*INFO: pipeR fd=8
*INFO: Socket added: fd=8, tail.count=1
*INFO: master fd=4
*INFO: Socket added: fd=4, tail.count=2
*INFO: pipeR fd=10
*INFO: Socket added: fd=10, tail.count=1
*INFO: master fd=5
*INFO: Socket added: fd=5, tail.count=2
*INFO: SIP STACK -- START
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {130.10.20.220} on port {10062}
using fd {5} with type {128}...
*INFO: Starting [SIP transport] server with IP {130.10.20.220} on port {10060}
using fd {4} with type {64}...
*INFO: SIP STACK::run -- START
*INFO: Starting [SIP transport] server with IP {130.10.20.220} on port {10060}
using fd {3} with type {2}...
*INFO: ioctlt(4), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=12)
*INFO: Socket added: fd=12, tail.count=3
*INFO: WebSocket Peer accepted/connected with fd = 12
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 12
*INFO: WebSocket handshake message: GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: 130.10.20.220:10060
Origin: http://localhost
Sec-WebSocket-Protocol: sip
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: TCo0txnF/QZXGhTB/u+3tA==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: x-webkit-deflate-frame
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
*INFO: Sending DNS query to "130.10.130.1"
*INFO: DNS NAPTR (130.10.20.220) query returned zero result
*INFO:
SEND: REGISTER sip:130.10.20.220 SIP/2.0
Via: SIP/2.0/UDP
130.10.20.220:10060;branch=z9hG4bKQT1TMB6Ohai4d85FglpUC1Ix9Xznot2a;rport
From: <sip:web8000@130.10.20.220>;tag=8K0207Dgemzv2Fiox7ql
To: <sip:web8000@130.10.20.220>
Contact:
"web8000"<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-sr
c-ip=130.10.239.45;ws-src-port=47953;ws-src-proto=ws>;expires=200;click2call=no;
+g.oma.sip-im;+audio;language="en,fr"
Call-ID: d97b5cf3-44b0-ecd8-8d7a-b59bf2226a1c
CSeq: 9848 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.12
Organization: EDAG GmbH & Co. KGaA
Supported: path
Via: SIP/2.0/TCP
130.10.239.45:47953;rport;branch=z9hG4bKQT1TMB6Ohai4d85FglpUC1Ix9Xznot2a;ws-hack
ed=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
130.10.20.220:10060;branch=z9hG4bKQT1TMB6Ohai4d85FglpUC1Ix9Xznot2a;received=130.
10.20.220;rport=10060
Via: SIP/2.0/TCP
130.10.239.45:47953;rport;branch=z9hG4bKQT1TMB6Ohai4d85FglpUC1Ix9Xznot2a;ws-hack
ed=WS
From: <sip:web8000@130.10.20.220>;tag=8K0207Dgemzv2Fiox7ql
To: <sip:web8000@130.10.20.220>;tag=as5b72a417
Call-ID: d97b5cf3-44b0-ecd8-8d7a-b59bf2226a1c
CSeq: 9848 REGISTER
Server: asteriskdev2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="273a7a7a"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
*INFO: Sending DNS query to "130.10.130.1"
*INFO: DNS NAPTR (130.10.20.220) query returned zero result
*INFO:
SEND: REGISTER sip:130.10.20.220 SIP/2.0
Via: SIP/2.0/UDP
130.10.20.220:10060;branch=z9hG4bKnEioSBifBaXEmSwCC9acrR2OV3cLDoTr;rport
From: <sip:web8000@130.10.20.220>;tag=8K0207Dgemzv2Fiox7ql
To: <sip:web8000@130.10.20.220>
Contact:
"web8000"<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-sr
c-ip=130.10.239.45;ws-src-port=47953;ws-src-proto=ws>;expires=200;click2call=no;
+g.oma.sip-im;+audio;language="en,fr"
Call-ID: d97b5cf3-44b0-ecd8-8d7a-b59bf2226a1c
CSeq: 9849 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest
username="web8000",realm="asterisk",nonce="273a7a7a",uri="sip:130.10.20.220",res
ponse="ce7adbc0a75a4afe72bac67c7ca335fb",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.12
Organization: EDAG GmbH & Co. KGaA
Supported: path
Via: SIP/2.0/TCP
130.10.239.45:47953;rport;branch=z9hG4bKnEioSBifBaXEmSwCC9acrR2OV3cLDoTr;ws-hack
ed=WS
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP
130.10.20.220:10060;branch=z9hG4bKnEioSBifBaXEmSwCC9acrR2OV3cLDoTr;received=130.
10.20.220;rport=10060
Via: SIP/2.0/TCP
130.10.239.45:47953;rport;branch=z9hG4bKnEioSBifBaXEmSwCC9acrR2OV3cLDoTr;ws-hack
ed=WS
From: <sip:web8000@130.10.20.220>;tag=8K0207Dgemzv2Fiox7ql
To: <sip:web8000@130.10.20.220>;tag=as5b72a417
Call-ID: d97b5cf3-44b0-ecd8-8d7a-b59bf2226a1c
CSeq: 9849 REGISTER
Server: asteriskdev2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 200
Contact:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>;expires=200
Date: Thu, 14 Mar 2013 15:19:03 GMT
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
RECV:NOTIFY
sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.1
0.239.45;ws-src-port=47953;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK17b914bb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@130.10.20.220>;tag=as495fe0f6
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>
Contact: <sip:asterisk@130.10.20.220:5060>
Call-ID: 4d4725ff70f7ece56d67f80750d0184f@130.10.20.220:5060
CSeq: 102 NOTIFY
User-Agent: asteriskdev2
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@130.10.20.220
Voice-Message: 0/0 (0/0)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 130.10.20.220:5060;rport=5060;branch=z9hG4bK17b914bb
From: "asterisk"<sip:asterisk@130.10.20.220>;tag=as495fe0f6
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>
Call-ID: 4d4725ff70f7ece56d67f80750d0184f@130.10.20.220:5060
CSeq: 102 NOTIFY
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO:
RECV:INVITE
sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.1
0.239.45;ws-src-port=47953;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK667a029f;rport
Max-Forwards: 70
From: "Dennis Richter (8001)" <sip:8001@130.10.20.220>;tag=as324e425e
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>
Contact: <sip:8001@130.10.20.220:5060>
Call-ID: 2811193f29d633445d5f0c802a4b7266@130.10.20.220:5060
CSeq: 102 INVITE
User-Agent: asteriskdev2
Date: Thu, 14 Mar 2013 15:19:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Call-Info: <http://130.10.14.249:8080/xmlphone/picture?userid=8001>;purpose=icon
Content-Type: application/sdp
Content-Length: 440
v=0
o=root 2072700055 2072700055 IN IP4 130.10.20.220
s=Asterisk PBX 1.8.20.1~dfsg-1ubuntu1
c=IN IP4 130.10.20.220
b=CT:128
t=0 0
m=audio 12766 RTP/AVP 8 110 101
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12772 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO:
SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 130.10.20.220:5060;rport=5060;branch=z9hG4bK667a029f
From: "Dennis Richter (8001)"<sip:8001@130.10.20.220>;tag=as324e425e
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>
Call-ID: 2811193f29d633445d5f0c802a4b7266@130.10.20.220:5060
CSeq: 102 INVITE
Content-Length: 0
*INFO: is_ro_provisional_final_matching=0,
is_ro_media_lines_changed=0,
is_ro_network_info_changed=0,
is_ro_loopback_address=0,
is_media_type_changed=0
*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
**WARN: function: "tdav_session_av_prepare()"
file: "src/tdav_session_av.c"
line: "412"
MSG: DTLS-SRTP requested but not certificate provided, disabling this option :(
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
**WARN: function: "tdav_session_av_prepare()"
file: "src/tdav_session_av.c"
line: "412"
MSG: DTLS-SRTP requested but not certificate provided, disabling this option :(
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO:
SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 130.10.20.220:5060;rport=5060;branch=z9hG4bK667a029f
From: "Dennis Richter (8001)"<sip:8001@130.10.20.220>;tag=as324e425e
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>;tag=1362460085740
Contact: <sip:web8000@130.10.20.220:10060;transport=udp>
Call-ID: 2811193f29d633445d5f0c802a4b7266@130.10.20.220:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
***ERROR: function: "tsk_params_get_param_value()"
file: "src/tsk_params.c"
line: "219"
MSG: Invalid parameter
*INFO: Negotiated codecs with the left leg = 1441824
*INFO: State machine: x0500_Current_2_Current_X_oINVITE
*INFO: tsk_timer_manager_start
*INFO: tsk_timer_manager_start
*INFO: ICE CTX::run -- START
*INFO: Timer manager run()::enter
*INFO: ICE CTX::run -- START
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
[130.10.20.220:52944]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 130.10.20.220
*INFO: ICE callback: Gathering host candidates succeed
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
[130.10.20.220:34226]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 130.10.20.220
*INFO: ICE callback: Gathering host candidates succeed
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: TIMER MANAGER -- START
*INFO: Candidate: kkd0s2Kbt 1 udp 2130706431 130.10.20.220 52944 typ host
*INFO: Candidate: kkd0s2Kbt 2 udp 2130706430 130.10.20.220 52945 typ host
*INFO: Candidate: srflxkkd0 1 udp 1694498815 62.156.178.2 33272 typ srflx raddr
130.10.20.220 rport 52944
*INFO: Candidate: srflxkkd0 2 udp 1694498814 62.156.178.2 33273 typ srflx raddr
130.10.20.220 rport 52945
*INFO: ICE callback: Gathering candidates completed
*INFO: Candidate: FRyNFGdGz 1 udp 2130706431 130.10.20.220 34226 typ host
*INFO: Candidate: FRyNFGdGz 2 udp 2130706430 130.10.20.220 34227 typ host
*INFO: Candidate: srflxFRyN 2 udp 1694498814 62.156.178.2 33274 typ srflx raddr
130.10.20.220 rport 34227
*INFO: Candidate: srflxFRyN 1 udp 1694498815 62.156.178.2 33275 typ srflx raddr
130.10.20.220 rport 34226
*INFO: ICE callback: Gathering candidates completed
*INFO: State machine: c0000_Started_2_Outgoing_X_oINVITE
*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
**WARN: function: "tdav_session_av_prepare()"
file: "src/tdav_session_av.c"
line: "412"
MSG: DTLS-SRTP requested but not certificate provided, disabling this option :(
*INFO: ICE enabled on RTP manager
**WARN: function: "tdav_session_av_prepare()"
file: "src/tdav_session_av.c"
line: "412"
MSG: DTLS-SRTP requested but not certificate provided, disabling this option :(
*INFO: ICE enabled on RTP manager
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
*INFO: === INVITE Dialog terminated ===
*INFO: === ICT terminated ===
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER -- STOP
*INFO: ICE CTX::run -- STOP
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER -- STOP
*INFO: ICE CTX::run -- STOP
*INFO: State machine: s0000_Ringing_2_Terminated_X_Reject
*INFO: State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699
*INFO:
SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/UDP 130.10.20.220:5060;rport=5060;branch=z9hG4bK667a029f
From: "Dennis Richter (8001)"<sip:8001@130.10.20.220>;tag=as324e425e
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>;tag=1362460085740
Call-ID: 2811193f29d633445d5f0c802a4b7266@130.10.20.220:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"
*INFO:
RECV:ACK sip:web8000@130.10.20.220:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 130.10.20.220:5060;branch=z9hG4bK667a029f;rport
Max-Forwards: 70
From: "Dennis Richter (8001)" <sip:8001@130.10.20.220>;tag=as324e425e
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=47953;ws-src-proto=ws>;tag=1362460085740
Contact: <sip:8001@130.10.20.220:5060>
Call-ID: 2811193f29d633445d5f0c802a4b7266@130.10.20.220:5060
CSeq: 102 ACK
User-Agent: asteriskdev2
Content-Length: 0
*INFO: === INVITE Dialog terminated ===
*INFO: State machine: tsip_transac_ist_Completed_2_Confirmed_ACK
*INFO: State machine: tsip_transac_ist_Any_2_Terminated_X_cancel
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: MPPeer object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: MPSipSession object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: State machine: No matching state found.
*INFO: *** Audio session destroyed ***
*INFO: MPProxyPluginConsumerVideo object destroyed
*INFO: MPProxyPluginProducerVideo object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Video session destroyed ***
*INFO: MPSipSession object destroyed
*INFO: *** SIP Session destroyed ***
*INFO: === ICT terminated ===
*INFO: *** SIP Session destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: MPProxyPluginConsumerVideo object destroyed
*INFO: MPProxyPluginProducerVideo object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Video session destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** ICT destroyed ***
Regards,
Dennis
Original comment by richter....@gmail.com
on 14 Mar 2013 at 3:27
This looks to be a bug in chrome on linux because setRemoteDescription is called
Original comment by boss...@yahoo.fr
on 18 Mar 2013 at 8:26
Thanks for help, but its the same on windows and chrome.
Hier the sipml5 log:
SIPML5 API version = 1.2.185 SIPml-api.js:1
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.22 (KHTML, like
Gecko) Chrome/25.0.1364.172 Safari/537.22 SIPml-api.js:1
WebSocket supported = yes SIPml-api.js:1
Navigator friendly name = chrome SIPml-api.js:1
OS friendly name = windows SIPml-api.js:1
Have WebRTC = yes SIPml-api.js:1
Have GUM = yes SIPml-api.js:1
Engine initialized SIPml-api.js:1
s_websocket_server_url=ws://130.10.20.220:10060 SIPml-api.js:1
s_sip_outboundproxy_url=(null) SIPml-api.js:1
b_rtcweb_breaker_enabled=yes SIPml-api.js:1
b_click2call_enabled=no SIPml-api.js:1
SIP stack start: proxy='sipml5.org:10062', realm='<sip:130.10.20.220>',
impi='web8000', impu='<sip:web8000@130.10.20.220>' SIPml-api.js:1
Connecting to 'ws://130.10.20.220:10060' SIPml-api.js:1
==stack event = starting SIPml-api.js:1
__tsip_transport_ws_onopen SIPml-api.js:1
==stack event = started SIPml-api.js:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js:1
SEND: REGISTER sip:130.10.20.220 SIP/2.0
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKAK3qFbwhS4pQ1NvAeT4RBZW3ZyhbxgkP;rport
From: <sip:web8000@130.10.20.220>;tag=hZBlFQxWv9zqv7Az59yg
To: <sip:web8000@130.10.20.220>
Contact:
"Dennis"<sip:web8000@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expir
es=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: f5a75b91-0ebe-4303-a987-c087a3b1f240
CSeq: 55004 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.14
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1
==session event = connecting SIPml-api.js:1
==session event = sent_request SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
130.10.20.220:10060;rport=10060;received=130.10.20.220;branch=z9hG4bKAK3qFbwhS4p
Q1NvAeT4RBZW3ZyhbxgkP
From: <sip:web8000@130.10.20.220>;tag=hZBlFQxWv9zqv7Az59yg
To: <sip:web8000@130.10.20.220>;tag=as1d502368
Call-ID: f5a75b91-0ebe-4303-a987-c087a3b1f240
CSeq: 55004 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP
130.10.239.45:64564;rport;branch=z9hG4bKAK3qFbwhS4pQ1NvAeT4RBZW3ZyhbxgkP;ws-hack
ed=WS
Server: asteriskdev2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest
realm="asterisk",nonce="13df96a6",stale=FALSE,algorithm=MD5
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js:1
SEND: REGISTER sip:130.10.20.220 SIP/2.0
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bK0FPEw7cFD0pXdA2XeUw2xivqvnPHuRTB;rport
From: <sip:web8000@130.10.20.220>;tag=hZBlFQxWv9zqv7Az59yg
To: <sip:web8000@130.10.20.220>
Contact:
"Dennis"<sip:web8000@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expir
es=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: f5a75b91-0ebe-4303-a987-c087a3b1f240
CSeq: 55005 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest
username="web8000",realm="asterisk",nonce="13df96a6",uri="sip:130.10.20.220",res
ponse="f49b6d5b9c6b5f6d94bc293950f495a6",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.03.14
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1
==session event = sent_request SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP
130.10.20.220:10060;rport=10060;received=130.10.20.220;branch=z9hG4bK0FPEw7cFD0p
XdA2XeUw2xivqvnPHuRTB
From: <sip:web8000@130.10.20.220>;tag=hZBlFQxWv9zqv7Az59yg
To: <sip:web8000@130.10.20.220>;tag=as1d502368
Contact:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=64564;ws-src-proto=ws>;expires=200
Call-ID: f5a75b91-0ebe-4303-a987-c087a3b1f240
CSeq: 55005 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP
130.10.239.45:64564;rport;branch=z9hG4bK0FPEw7cFD0pXdA2XeUw2xivqvnPHuRTB;ws-hack
ed=WS
Server: asteriskdev2
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 19 Mar 2013 7:35:0 GMT;19
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js:1
==session event = connected SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=NOTIFY
sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.1
0.239.45;ws-src-port=64564;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 130.10.20.220:5060;rport;branch=z9hG4bK6fd1d2fa
From: "asterisk"<sip:asterisk@130.10.20.220>;tag=as1e190a11
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=64564;ws-src-proto=ws>
Contact: <sip:asterisk@130.10.20.220:10060;transport=ws>
Call-ID: 323eff5f2605cddb205610f74b9087af@130.10.20.220:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 88
Max-Forwards: 70
User-Agent: asteriskdev2
Event: message-summary
Messages-Waiting: no
Message-Account: sip:*97@130.10.20.220
Voice-Message: 0/0 (0/0)
SIPml-api.js:1
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP 130.10.20.220:5060;rport=5060;branch=z9hG4bK6fd1d2fa
From: "asterisk"<sip:asterisk@130.10.20.220>;tag=as1e190a11
To:
<sip:web8000@130.10.20.220:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=130.
10.239.45;ws-src-port=64564;ws-src-proto=ws>
Call-ID: 323eff5f2605cddb205610f74b9087af@130.10.20.220:5060
CSeq: 102 NOTIFY
Content-Length: 0
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=INVITE sip:web8000@130.10.20.220:10060 SIP/2.0
Via: SIP/2.0/WS 130.10.20.220:10060;rport;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>
Contact: <sip:8001@130.10.20.220:10060;transport=ws>
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 INVITE
Content-Type: application/sdp
Content-Length: 2259
Max-Forwards: 70
Route: <sip:130.10.239.45:64564;transport=ws;lr>
User-Agent: webrtc2sip Media Server 2.0
v=0
o=doubango 1983 678901 IN IP4 130.10.20.220
s=-
c=IN IP4 130.10.20.220
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 60832 RTP/AVP 8 101
c=IN IP4 130.10.20.220
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80
inline:A9teaCjCzeZefQV/uRWdqBh/MSjIqby4KWRV3b57
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32
inline:UyzWx63WeZNBITbR4Oq2DP8CbXN4C7RfhEzM2lUu
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2522596638 cname:ldjWoB60jbyQlR6e
a=ssrc:2522596638 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2522596638 label:Doubango.Audio
a=ice-ufrag:O8G3C0JikKd0j4A
a=ice-pwd:7aehPcUnSloXhkN4woxoW
a=candidate:2lLa5ipB1 1 udp 2130706431 130.10.20.220 60832 typ host
a=candidate:2lLa5ipB1 2 udp 2130706430 130.10.20.220 60833 typ host
a=candidate:srflx2lLa 1 udp 1694498815 62.156.178.2 34554 typ srflx raddr
130.10.20.220 rport 60832
a=candidate:srflx2lLa 2 udp 1694498814 62.156.178.2 34553 typ srflx raddr
130.10.20.220 rport 60833
m=video 35990 RTP/AVP 104 103 34
c=IN IP4 130.10.20.220
a=rtpmap:104 H264/90000
a=imageattr:104 recv [x=[128:16:640],y=[96:16:480]] send
[x=[128:16:640],y=[96:16:480]]
a=fmtp:104 profile-level-id=42001e; packetization-mode=1
a=rtpmap:103 H263-1998/90000
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fBDQTho4KBqr70SyZLKOlQRfiVDNdqup1ASFI8Ma
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32
inline:lspfE5nPmYttdVwsawfLrTBHRVMNOmVXM6xsYqth
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2646761838 cname:ldjWoB60jbyQlR6e
a=ssrc:2646761838 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2646761838 label:Doubango.Video
a=ice-ufrag:tFvJJufXmn8lSMK
a=ice-pwd:cHluauTYPk5vMukLVUmZ6
a=candidate:FDa10Dxkp 1 udp 2130706431 130.10.20.220 35990 typ host
a=candidate:FDa10Dxkp 2 udp 2130706430 130.10.20.220 35991 typ host
a=candidate:srflxFDa1 1 udp 1694498815 62.156.178.2 34551 typ srflx raddr
130.10.20.220 rport 35990
a=candidate:srflxFDa1 2 udp 1694498814 62.156.178.2 34552 typ srflx raddr
130.10.20.220 rport 35991
SIPml-api.js:1
State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE SIPml-api.js:1
SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 130.10.20.220:10060;rport=10060;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 INVITE
Content-Length: 0
SIPml-api.js:1
PeerConnectionClass = function RTCPeerConnection() { [native code] }
SessionDescriptionClass = function RTCSessionDescription() { [native code] }
IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js:1
ICE
servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.ne
t:3478"},{"url":"stun:numb.viagenie.ca:3478"}] SIPml-api.js:1
setRemoteDescription(offer)
v=0
o=doubango 1983 678901 IN IP4 130.10.20.220
s=-
c=IN IP4 130.10.20.220
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 60832 RTP/SAVPF 8 101
c=IN IP4 130.10.20.220
a=ptime:20
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:A9teaCjCzeZefQV/uRWdqBh/MSjIqby4KWRV3b57
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:UyzWx63WeZNBITbR4Oq2DP8CbXN4C7RfhEzM2lUu
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2522596638 cname:ldjWoB60jbyQlR6e
a=ssrc:2522596638 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2522596638 label:Doubango.Audio
a=ice-ufrag:O8G3C0JikKd0j4A
a=ice-pwd:7aehPcUnSloXhkN4woxoW
a=candidate:2lLa5ipB1 1 udp 2130706431 130.10.20.220 60832 typ host
a=candidate:2lLa5ipB1 2 udp 2130706430 130.10.20.220 60833 typ host
a=candidate:srflx2lLa 1 udp 1694498815 62.156.178.2 34554 typ srflx raddr
130.10.20.220 rport 60832
a=candidate:srflx2lLa 2 udp 1694498814 62.156.178.2 34553 typ srflx raddr
130.10.20.220 rport 60833
m=video 35990 RTP/SAVPF 104 103 34
c=IN IP4 130.10.20.220
a=rtpmap:104 H264/90000
a=imageattr:104 recv [x=[128:16:640],y=[96:16:480]] send
[x=[128:16:640],y=[96:16:480]]
a=fmtp:104 profile-level-id=42001e; packetization-mode=1
a=rtpmap:103 H263-1998/90000
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fBDQTho4KBqr70SyZLKOlQRfiVDNdqup1ASFI8Ma
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:lspfE5nPmYttdVwsawfLrTBHRVMNOmVXM6xsYqth
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2646761838 cname:ldjWoB60jbyQlR6e
a=ssrc:2646761838 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2646761838 label:Doubango.Video
a=ice-ufrag:tFvJJufXmn8lSMK
a=ice-pwd:cHluauTYPk5vMukLVUmZ6
a=candidate:FDa10Dxkp 1 udp 2130706431 130.10.20.220 35990 typ host
a=candidate:FDa10Dxkp 2 udp 2130706430 130.10.20.220 35991 typ host
a=candidate:srflxFDa1 1 udp 1694498815 62.156.178.2 34551 typ srflx raddr
130.10.20.220 rport 35990
a=candidate:srflxFDa1 2 udp 1694498814 62.156.178.2 34552 typ srflx raddr
130.10.20.220 rport 35991
SIPml-api.js:1
State machine: s0000_Started_2_Ringing_X_iINVITE SIPml-api.js:1
__on_state_change SIPml-api.js:1
==stack event = m_permission_requested SIPml-api.js:1
onSetRemoteDescriptionError SIPml-api.js:1
SetRemoteDescription failed. SIPml-api.js:1
State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx SIPml-api.js:1
SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/WS 130.10.20.220:10060;rport=10060;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>;tag=ssa77baNGBZwmm3Iw1rl
Contact: <sip:web8000@df7jal23ls0d.invalid;transport=ws>
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
SIPml-api.js:1
==stack event = i_new_call SIPml-api.js:1
onGetUserMediaSuccess SIPml-api.js:1
createAnswer SIPml-api.js:1
onCreateSdpError SIPml-api.js:1
CreateAnswer can't be called before SetRemoteDescription. SIPml-api.js:1
State machine: s0000_Ringing_2_Terminated_X_Reject SIPml-api.js:1
=== INVITE Dialog terminated === SIPml-api.js:1
PeerConnection::stop() SIPml-api.js:1
==stack event = m_permission_accepted SIPml-api.js:1
State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699
SIPml-api.js:1
SEND: SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS 130.10.20.220:10060;rport=10060;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>;tag=ssa77baNGBZwmm3Iw1rl
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"
SIPml-api.js:1
__on_state_change SIPml-api.js:1
State machine: tsip_transac_ist_Any_2_Terminated_X_cancel SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=ACK sip:web8000@130.10.20.220:10060 SIP/2.0
Via: SIP/2.0/WS 130.10.20.220:10060;rport;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>;tag=ssa77baNGBZwmm3Iw1rl
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 ACK
Content-Length: 0
Max-Forwards: 70
Route: <sip:130.10.239.45:64564;transport=ws;lr>
SIPml-api.js:1
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 130.10.20.220:10060;rport=10060;branch=z9hG4bK1364804639856
From: <sip:8001@130.10.20.220>;tag=1365625024115
To: <sip:web8000@130.10.20.220:10060>;tag=ssa77baNGBZwmm3Iw1rl
Call-ID: ff6acce8-8ea9-761d-9dcf-79dcec48aedd
CSeq: 404884809 ACK
Content-Length: 0
Regards,
Dennis
Original comment by richter....@gmail.com
on 19 Mar 2013 at 7:41
VP8 is mandatory on Chrome. You must provide it when making video call all
disable video. Looks like you're using webrtc2sip which means you can enable
media coder if the remote party doesn't support vp8.
Original comment by boss...@yahoo.fr
on 19 Mar 2013 at 8:36
Thanks for help again,
in the webrtc2sip config.xml I set enable-media-coder to yes and now its
ringing in the Browser. After I click on Accept the Call start, but it doesn't
exist an audio transfer and the webrtc2sip crashes and I must reboot the
server. The SIPML5 Log have no ERRORS.
For safety here the webrtc2sip log:
*******************************************************************
Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>
LICENCE: GPLv3 or proprietary
VERSION: 2.2.0
'quit' to quit the application.
*******************************************************************
SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
warning: The VAD has been replaced by a hack pending a complete rewrite
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not
known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or
service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tsk_params_get_param_value()"
file: "src/tsk_params.c"
line: "219"
MSG: Invalid parameter
warning: The VAD has been replaced by a hack pending a complete rewrite
warning: The VAD has been replaced by a hack pending a complete rewrite
[h263 @ 0xb6d11080] Warning: not compiled with thread support, using thread
emulation
[h263 @ 0xb6d11080] The specified picture size of 640x480 is not valid for the
H.263 codec.
Valid sizes are 128x96, 176x144, 352x288, 704x576, and 1408x1152. Try H.263+.
***ERROR: function: "tdav_codec_h263_open_encoder()"
file: "src/codecs/h263/tdav_codec_h263.c"
line: "982"
MSG: Failed to open [H263-1996 codec] codec
***ERROR: function: "tmedia_codec_open()"
file: "src/tmedia_codec.c"
line: "148"
MSG: Failed to open [H263-1996 codec] codec
***ERROR: function: "tdav_session_video_start()"
file: "src/video/tdav_session_video.c"
line: "858"
MSG: Failed to open [H263-1996 codec] codec
***ERROR: function: "tmedia_session_mgr_start()"
file: "src/tmedia_session.c"
line: "803"
MSG: Failed to start video session
quit
quit
^C
Regards,
Dennis
Original comment by richter....@gmail.com
on 19 Mar 2013 at 2:07
Hi,
I try to use sipml5 with Asterisk(11.5) on chrome (30). I get the call
established ( for ws and wss) between chrome and a sip phone (connecetd to
asterisk), but no audio on both ends. My wireshark capture shows
audio-udp(srtp) packets flowing from the asterisk to the chrome (no srtp
packets from chrome). I use the PCMU or PCMA codec.
For the call initiated from Chrome, I get the following error:
SetRemoteDescription failed: Failed to update session state: ERROR_CONTENT
SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
tmedia_session_jsep01.onSetRemoteDescriptionError SIPml-api.js:3
(anonymous function)
For the incoming call to chrome, no errors.
Below is the log for outgoing call from chrome:
SIPML5 API version = 1.3.203 SIPml-api.js:1
User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_8_2) AppleWebKit/537.36
(KHTML, like Gecko) Chrome/30.0.1599.101 Safari/537.36 SIPml-api.js:1
WebSocket supported = yes SIPml-api.js:1
Navigator friendly name = chrome SIPml-api.js:1
OS friendly name = mac SIPml-api.js:1
Have WebRTC = yes SIPml-api.js:1
Have GUM = yes SIPml-api.js:1
Engine initialized SIPml-api.js:1
s_websocket_server_url=wss://203.143.170.213:8089/ws SIPml-api.js:1
s_sip_outboundproxy_url=(null) SIPml-api.js:1
b_rtcweb_breaker_enabled=no SIPml-api.js:1
b_click2call_enabled=no SIPml-api.js:1
b_early_ims=yes SIPml-api.js:1
b_enable_media_stream_cache=no SIPml-api.js:1
o_bandwidth={} SIPml-api.js:1
o_video_size={} SIPml-api.js:1
SIP stack start: proxy='ns313841.ovh.net:11060', realm='<sip:asterisk>',
impi='nicsec01110', impu='<sip:nicsec01110@203.143.170.213>' SIPml-api.js:1
Connecting to 'wss://203.143.170.213:8089/ws' SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
__tsip_transport_ws_onopen SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js:1
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKmmSKI5PhXwDtetdCEaR59ZfKRZtebHDe;rport
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>
Contact:
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
s>;expires=200;click2call=no
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2347 REGISTER
Content-Length: 0
Max-Forwards: 70
Supported: path
SIPml-api.js:1
session event = type = connecting - description = Connecting...
wss_ast_call.html:42
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKmmSKI5Ph
XwDtetdCEaR59ZfKRZtebHDe
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>;tag=as77e33ffc
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2347 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest
realm="asterisk",nonce="6a604af9",stale=FALSE,algorithm=MD5
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js:1
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bK9RKzVx32PaGU0tfTe8h4Mf2NfJTeIWIO;rport
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>
Contact:
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
s>;expires=200;click2call=no
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2348 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest
username="nicsec01110",realm="asterisk",nonce="6a604af9",uri="sip:asterisk",resp
onse="31dd1166570b0aeac2c8363299e2fe10",algorithm=MD5
Supported: path
SIPml-api.js:1
session event = type = sent_request - description = REGISTER request
successfully sent wss_ast_call.html:42
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bK9RKzVx32
PaGU0tfTe8h4Mf2NfJTeIWIO
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>;tag=as77e33ffc
Contact:
<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=
200
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2348 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 24 Oct 2013 07:24:26 GMT;24
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js:1
session event = type = sent_request - description = REGISTER request
successfully sent wss_ast_call.html:42
session event = type = connected - description = Connected wss_ast_call.html:42
State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js:1
PeerConnectionClass = function RTCPeerConnection() { [native code] }
SessionDescriptionClass = function RTCSessionDescription() { [native code] }
IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js:1
ICE
servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.ne
t:3478"},{"url":"stun:numb.viagenie.ca:3478"}] SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
call event = connecting wss_ast_call.html:182
onGetUserMediaSuccess SIPml-api.js:1
createOffer SIPml-api.js:1
onCreateSdpSuccess SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
call event = m_stream_audio_local_added wss_ast_call.html:182
onSetLocalDescriptionSuccess SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
ICE GATHERING COMPLETED! SIPml-api.js:1
onIceGatheringCompleted SIPml-api.js:1
SEND: INVITE sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKUk5hRO2K7XVEvbf4lGq1tGlVFLMLxSNC;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact:
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 INVITE
Content-Type: application/sdp
Content-Length: 1525
Max-Forwards: 70
v=0
o=- 4518691561856059400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
m=audio 62449 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 203.143.170.138
a=rtcp:62449 IN IP4 203.143.170.138
a=candidate:1315546826 1 udp 2113937151 203.143.170.138 62449 typ host
generation 0
a=candidate:1315546826 2 udp 2113937151 203.143.170.138 62449 typ host
generation 0
a=candidate:15358522 1 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=candidate:15358522 2 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=ice-ufrag:/MiMNlcyslcybRWu
a=ice-pwd:2keuIHyKr/pI9WPhrhL+Tc+s
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:lf04HXQ0nEA2Q1031haVgU50ZIZPIEamcbchSdZV
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:yUj0De1AXYe9kRXjCTd5ZsC7pcmVk2g/z8tQvoXr
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:107 CN/48000
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:3539336787 cname:Kdv90oeX0650TQ6k
a=ssrc:3539336787 msid:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
a=ssrc:3539336787 mslabel:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
a=ssrc:3539336787 label:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKUk5hRO2K
7XVEvbf4lGq1tGlVFLMLxSNC
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as6423d2ee
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest
realm="asterisk",nonce="051134d3",stale=FALSE,algorithm=MD5
SIPml-api.js:1
SEND: ACK sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKUk5hRO2K7XVEvbf4lGq1tGlVFLMLxSNC;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as6423d2ee
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 ACK
Content-Length: 0
Max-Forwards: 70
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js:1
SEND: INVITE sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKATMMli90AddZ5IIkBJihxvJBEgBB0iH2;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact:
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Type: application/sdp
Content-Length: 1525
Max-Forwards: 70
Authorization: Digest
username="nicsec01110",realm="asterisk",nonce="051134d3",uri="sip:20@203.143.170
.213",response="b16ffbfdf7321a04273e6bbe2ba5ba28",algorithm=MD5
v=0
o=- 4518691561856059400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
m=audio 62449 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 203.143.170.138
a=rtcp:62449 IN IP4 203.143.170.138
a=candidate:1315546826 1 udp 2113937151 203.143.170.138 62449 typ host
generation 0
a=candidate:1315546826 2 udp 2113937151 203.143.170.138 62449 typ host
generation 0
a=candidate:15358522 1 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=candidate:15358522 2 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=ice-ufrag:/MiMNlcyslcybRWu
a=ice-pwd:2keuIHyKr/pI9WPhrhL+Tc+s
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:lf04HXQ0nEA2Q1031haVgU50ZIZPIEamcbchSdZV
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:yUj0De1AXYe9kRXjCTd5ZsC7pcmVk2g/z8tQvoXr
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:107 CN/48000
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:3539336787 cname:Kdv90oeX0650TQ6k
a=ssrc:3539336787 msid:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
a=ssrc:3539336787 mslabel:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
a=ssrc:3539336787 label:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
call event = i_ao_request wss_ast_call.html:182
call event = i_ao_request wss_ast_call.html:182
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
call event = i_ao_request wss_ast_call.html:182
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Type: application/sdp
Content-Length: 742
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
v=0
o=root 1449102145 1449102145 IN IP4 203.143.170.213
s=Asterisk PBX 11.5.0
c=IN IP4 203.143.170.213
t=0 0
m=audio 10016 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:662cde007895691646e75b180d979e91
a=ice-pwd:1446e3d55206319258de8afc0daae4df
a=candidate:Hcb8faad5 1 UDP 2130706431 203.143.170.213 10016 typ host
a=candidate:Ha0a046e 1 UDP 2130706431 10.10.4.110 10016 typ host
a=candidate:Hcb8faad5 2 UDP 2130706430 203.143.170.213 10017 typ host
a=candidate:Ha0a046e 2 UDP 2130706430 10.10.4.110 10017 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_80
inline:vsKbW0mHuIZEMHKavHaeltWcT/75oSSkYJEVrNaT
SIPml-api.js:1
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js:1
setRemoteDescription(answer)
v=0
o=root 1449102145 1449102145 IN IP4 203.143.170.213
s=Asterisk PBX 11.5.0
c=IN IP4 203.143.170.213
t=0 0
m=audio 10016 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:662cde007895691646e75b180d979e91
a=ice-pwd:1446e3d55206319258de8afc0daae4df
a=candidate:Hcb8faad5 1 UDP 2130706431 203.143.170.213 10016 typ host
a=candidate:Ha0a046e 1 UDP 2130706431 10.10.4.110 10016 typ host
a=candidate:Hcb8faad5 2 UDP 2130706430 203.143.170.213 10017 typ host
a=candidate:Ha0a046e 2 UDP 2130706430 10.10.4.110 10017 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_80
inline:vsKbW0mHuIZEMHKavHaeltWcT/75oSSkYJEVrNaT
SIPml-api.js:1
SEND: ACK sip:20@203.143.170.213:5060;transport=WS SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK7ain7moi7JrXk6Tqkmvq;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact:
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest
username="nicsec01110",realm="asterisk",nonce="051134d3",uri="sip:20@203.143.170
.213:5060;transport=WS",response="c7e81073d068c223d16b77eeae08eed4",algorithm=MD
5
SIPml-api.js:1
onSetRemoteDescriptionError SIPml-api.js:1
SetRemoteDescription failed: Failed to update session state: ERROR_CONTENT
SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
tmedia_session_jsep01.onSetRemoteDescriptionError SIPml-api.js:3
(anonymous function) SIPml-api.js:3
call event = m_early_media wss_ast_call.html:182
call event = connected
Original comment by ath...@gmail.com
on 24 Oct 2013 at 7:35
Original issue reported on code.google.com by
richter....@gmail.com
on 14 Mar 2013 at 3:25