zhqiyi / webrtc2sip

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When RTCWeb Breaker is enabled - SIP INVITE authentication fails #67

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
What steps will reproduce the problem?
1.Enable RTCWeb breaker and register to Broadworks SIP server
2.Registration is challenged and then the correct credentials are passed.
3.Make a call - the SIP invite is challenged with a 401 which is acknowledged 
and then an invite with credentials are sent ( which are incorrect according to 
broadworks(sip server) logs ) and keeps getting challenged. The same call ( 
with same credentials ) works with the RTCWeb Breaker disabled.

What is the expected output? What do you see instead?
If correct credentials are provided the call should proceed but instead it 
keeps getting challenged with 401

What version of the product are you using? On what operating system?
webrtc2sip on CentOS release 6.3 (64 bit)..
*******************************************************************
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

Please provide any additional information below.

Original issue reported on code.google.com by nikesh...@gmail.com on 6 Mar 2013 at 8:58

GoogleCodeExporter commented 9 years ago
Additional observations. The call flow is below:

Chrome Browser --> webrtc2sip server --> Broadsoft BroadWorks application server

1. When web breaker is enabled, call authentication challenge dialog between 
the webrtc2sip server and the browser is not seen. Is this normal? Does the 
webrts2sip server handle this dialog without notifying the browser?

2. When calls are placed with web breaker enabled, the HA1 string is included 
in the contact header. Since we do not see dialog between the webrtc2sip server 
and the browser, what purpose does this serve?

3. Calls fail when the webrtc2sip server sends a re-invite with authentication 
and is not recognized by the BroadWorks server. The BroadWorks server then 
re-sends the challenge. At this point the webrtc2sip server issues the 
following error:

*INFO: State machine: x0000_Any_2_Any_X_i401_407_Challenge
***ERROR: function: "tsip_dialog_update_challenges()" 
file: "src/dialogs/tsip_dialog.c" 
line: "904" 
MSG: Failed to handle new challenge
*INFO: State machine: Exec function failed. Moving to terminal state.

Is the code designed to handle initial authentication failures?

4. While this dialog is taking place (web breaker enabled), the browser console 
shows a 100 trying message is received from the transaction layer, followed be 
a 180 ringing message and a 603 decline message. These messages seem to be 
generated by the browser SIP stack:

 tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport;branch=z9hG4bKeaaUbvuTblpscH69GWuXP2ss9R1XTZpc
From: <sip:YYYYYYYYYY@REALM>;tag=MnCvAkDXiIAbhcns1wmk
To: <sip:XXXXXXXXXX@REALM>
Call-ID: 5c52e263-3a84-54f4-7d6b-29391930ed32
CSeq: 55650 INVITE
Content-Length: 0

 tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport;branch=z9hG4bKeaaUbvuTblpscH69GWuXP2ss9R1XTZpc
From: <sip:YYYYYYYYYY@REALM>;tag=MnCvAkDXiIAbhcns1wmk
To: <sip:XXXXXXXXXX@REALM>;tag=1363503001173
Contact: 
<sips:XXXXXXXXXX@webrtc2sip_server:5060;transport=wss;ws-src-ip=zzz.zzz.zzz.zzz;
ws-src-port=33544;ws-src-proto=wss>
Call-ID: 5c52e263-3a84-54f4-7d6b-29391930ed32
CSeq: 55650 INVITE
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

 tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 603 Decline
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport;branch=z9hG4bKeaaUbvuTblpscH69GWuXP2ss9R1XTZpc
From: <sip:YYYYYYYYYY@REALM>;tag=MnCvAkDXiIAbhcns1wmk
To: <sip:XXXXXXXXXX@REALM>;tag=1363503001173
Call-ID: 5c52e263-3a84-54f4-7d6b-29391930ed32
CSeq: 55650 INVITE
Content-Length: 0
Reason: text="Decline";cause=603;text="Decline"

 tsk_utils.js:110
SEND: ACK sip:XXXXXXXXXX@REALM SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKeaaUbvuTblpscH69GWuXP2ss9R1XTZpc;rport
From: <sip:YYYYYYYYYY@REALM>;tag=MnCvAkDXiIAbhcns1wmk
To: <sip:XXXXXXXXXX@REALM>;tag=1363503001173
Call-ID: 5c52e263-3a84-54f4-7d6b-29391930ed32
CSeq: 55650 ACK

Content-Length: 0
Route: <sip:BroadWorks_server:5060;lr;transport=udp>
Max-Forwards: 70

 tsk_utils.js:110
State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE tsk_utils.js:110
=== INVITE Dialog terminated === tsk_utils.js:110
PeerConnection::stop() tsk_utils.js:110
==session event = i_ao_request call.htm:720
__on_state_change tsk_utils.js:110
==session event = terminated call.htm:720
The FSM is in the final state tsk_utils.js:116

The webrtc2sip server does not receive or generate any of these messages. Is 
this normal? When calls are made with web breaker disabled, normal SIP 
messaging is seen on the browser console.

5. The webrtc2sip debug shows incoming messages from the browser and BroadWorks 
server. Is there a way to also view the outgoing messages?

Thanks!

Original comment by GuyLeo...@gmail.com on 20 Mar 2013 at 10:32

GoogleCodeExporter commented 9 years ago
Hi,

is there any progress on this issue ? 
I have a similar problem - INVITE is challenged with a 407 and authentication 
fails. (realm in the digest challenge is different from domain)
doubango - branch 2.0 revision 982
webrtc2sip - trunk revision 112

Below is the debug trace. Any help would be appreciated.

*INFO: No all data in the WS buffer
*INFO: No all data in the WS buffer
*INFO: Receiving SIP o/ WebSocket message: INVITE sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKOmTRHXJiRTep3Ibz9yJebZzZ2fUz2oOa;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=Uq6TA2vqTgPPIcyvetHk
To: <sip:YYYYYYYYY@DOMAIN>
Contact: 
<sip:4822XXXXXXX@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport
=ws>;impi=4822XXXXXXXb%40DOMAIN;ha1=2357f1769210f06a47974826620ce5fe;+g.oma.sip-
im;+sip.ice;language="en,fr"
Call-ID: e9a121b7-7bff-4af1-d876-6db9cbf63858
CSeq: 48750 INVITE
Content-Type: application/sdp
Content-Length: 2365
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.06B
Organization: Doubango Telecom

v=0
o=- 3590444557885468700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS GfdQ2egdCHF7bT47VA4eLgoUaGbFqsql3c5c
m=audio 61127 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 89.77.189.69
a=rtcp:61127 IN IP4 89.77.189.69
a=candidate:21103810 1 udp 2113937151 192.168.1.79 61127 typ host generation 0
a=candidate:21103810 2 udp 2113937151 192.168.1.79 61127 typ host generation 0
a=candidate:2614420530 1 udp 2113937151 10.211.55.2 55840 typ host generation 0
a=candidate:2614420530 2 udp 2113937151 10.211.55.2 55840 typ host generation 0
a=candidate:868424551 1 udp 2113937151 10.37.129.2 53251 typ host generation 0
a=candidate:868424551 2 udp 2113937151 10.37.129.2 53251 typ host generation 0
a=candidate:4149378326 1 udp 1845501695 89.77.189.69 61127 typ srflx raddr 
192.168.1.79 rport 61127 generation 0
a=candidate:4149378326 2 udp 1845501695 89.77.189.69 61127 typ srflx raddr 
192.168.1.79 rport 61127 generation 0
a=candidate:1338112050 1 tcp 1509957375 192.168.1.79 0 typ host generation 0
a=candidate:1338112050 2 tcp 1509957375 192.168.1.79 0 typ host generation 0
a=candidate:3579254978 1 tcp 1509957375 10.211.55.2 0 typ host generation 0
a=candidate:3579254978 2 tcp 1509957375 10.211.55.2 0 typ host generation 0
a=candidate:2101405591 1 tcp 1509957375 10.37.129.2 0 typ host generation 0
a=candidate:2101405591 2 tcp 1509957375 10.37.129.2 0 typ host generation 0
a=ice-ufrag:VEmjbBbQGt4RCsdS
a=ice-pwd:cQO4vNis28mk9Vfk6FEjkHsa
a=ice-options:google-ice
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 
inline:3gTDabNT9I0+neHReApd4hBwNbS2aHckZ30xoqPd
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:3ku/dR4CEGDbYny/4cQ6XKBJhxUG+8+tSHJ4gpNx
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:2315227349 cname:a78fomb8NMqFmzZV
a=ssrc:2315227349 msid:GfdQ2egdCHF7bT47VA4eLgoUaGbFqsql3c5c 
GfdQ2egdCHF7bT47VA4eLgoUaGbFqsql3c5ca0
a=ssrc:2315227349 mslabel:GfdQ2egdCHF7bT47VA4eLgoUaGbFqsql3c5c
a=ssrc:2315227349 label:GfdQ2egdCHF7bT47VA4eLgoUaGbFqsql3c5ca0

*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO: Add call-id = 'e9a121b7-7bff-4af1-d876-6db9cbf63858' to peer with local 
fd = 17
*INFO: State machine: x0500_Current_2_Current_X_iINVITE
*INFO: tnet_ice_ctx_set_remote_candidates
*INFO: tsk_timer_manager_start
*INFO: ICE CTX::run -- START
*INFO: State machine: 
ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports 
[XX.196.175.177:48092]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = XX.196.175.177
*INFO: State machine: 
ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: stun.l.google.com:19302
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: 
ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexive
Candidates
*INFO: ICE reflexive candidates gathering ...0,500000
*INFO: ICE reflexive candidates gathering ...1,0
*INFO: srflx_addr_count_added=2, srflx_addr_count_skipped=0
*INFO: Candidate: i5EKy28NV 1 udp 2130706431 XX.196.175.177 48092 typ host
*INFO: Candidate: i5EKy28NV 2 udp 2130706430 XX.196.175.177 48093 typ host
*INFO: Candidate: srflxi5EK 2 udp 1694498814 XX.221.199.202 48093 typ srflx 
raddr XX.196.175.177 rport 48093
*INFO: Candidate: srflxi5EK 1 udp 1694498815 XX.221.199.202 48092 typ srflx 
raddr XX.196.175.177 rport 48092
*INFO: State machine: 
ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Succes
s
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: ICE: ignore processing SDP RO because version haven't changed
*INFO: is_ice_active=1,
is_ro_hold_resume_changed=0,
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,
is_ro_codecs_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: "422" 
MSG: DTLS-SRTP requested but no SSL certificates provided, disabling this 
option :(
*INFO: ICE enabled on RTP manager
*INFO: Remote SSRC = 2315227349
*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO: State machine: x0500_Current_2_Current_X_oINVITE
*INFO: tsk_timer_manager_start
*INFO: State machine: ICE_GatheringComplet_2_ConnChecking_X_ConnCheck
*INFO: ICE Pair: [i5EKy28NV 1 XX.196.175.177 48092] -> [21103810 1 192.168.1.79 
61127]
*INFO: ICE Pair: [i5EKy28NV 1 XX.196.175.177 48092] -> [2614420530 1 
10.211.55.2 55840]
*INFO: ICE Pair: [i5EKy28NV 1 XX.196.175.177 48092] -> [868424551 1 10.37.129.2 
53251]
*INFO: ICE Pair: [i5EKy28NV 1 XX.196.175.177 48092] -> [4149378326 1 
89.77.189.69 61127]
*INFO: ICE Pair: [i5EKy28NV 2 XX.196.175.177 48093] -> [21103810 2 192.168.1.79 
61127]
*INFO: ICE Pair: [i5EKy28NV 2 XX.196.175.177 48093] -> [2614420530 2 
10.211.55.2 55840]
*INFO: ICE Pair: [i5EKy28NV 2 XX.196.175.177 48093] -> [868424551 2 10.37.129.2 
53251]
*INFO: ICE Pair: [i5EKy28NV 2 XX.196.175.177 48093] -> [4149378326 2 
89.77.189.69 61127]
*INFO: ICE Pair: [srflxi5EK 2 XX.221.199.202 48093] -> [21103810 2 192.168.1.79 
61127]
*INFO: ICE Pair: [srflxi5EK 2 XX.221.199.202 48093] -> [2614420530 2 
10.211.55.2 55840]
*INFO: ICE Pair: [srflxi5EK 2 XX.221.199.202 48093] -> [868424551 2 10.37.129.2 
53251]
*INFO: ICE Pair: [srflxi5EK 2 XX.221.199.202 48093] -> [4149378326 2 
89.77.189.69 61127]
*INFO: ICE Pair: [srflxi5EK 1 XX.221.199.202 48092] -> [21103810 1 192.168.1.79 
61127]
*INFO: ICE Pair: [srflxi5EK 1 XX.221.199.202 48092] -> [2614420530 1 
10.211.55.2 55840]
*INFO: ICE Pair: [srflxi5EK 1 XX.221.199.202 48092] -> [868424551 1 10.37.129.2 
53251]
*INFO: ICE Pair: [srflxi5EK 1 XX.221.199.202 48092] -> [4149378326 1 
89.77.189.69 61127]
*INFO: ICE CTX::run -- START
*INFO: State machine: 
ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: Timer manager run()::enter
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports 
[XX.196.175.177:45832]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = XX.196.175.177
*INFO: State machine: 
ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: stun.l.google.com:19302
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: 
ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexive
Candidates
*INFO: TIMER MANAGER -- START
*INFO: ICE reflexive candidates gathering ...0,500000
*INFO: srflx_addr_count_added=2, srflx_addr_count_skipped=0
*INFO: Candidate: i0AONn7jl 1 udp 2130706431 XX.196.175.177 45832 typ host
*INFO: Candidate: i0AONn7jl 2 udp 2130706430 XX.196.175.177 45833 typ host
*INFO: Candidate: srflxi0AO 1 udp 1694498815 XX.221.199.202 45832 typ srflx 
raddr XX.196.175.177 rport 45832
*INFO: Candidate: srflxi0AO 2 udp 1694498814 XX.221.199.202 45833 typ srflx 
raddr XX.196.175.177 rport 45833
*INFO: State machine: 
ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Succes
s
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*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: "422" 
MSG: DTLS-SRTP requested but no SSL certificates provided, disabling this 
option :(
*INFO: ICE enabled on RTP manager
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: 

SEND: INVITE sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/UDP XX.196.175.177:10060;branch=z9hG4bK-887038794;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
To: <sip:YYYYYYYYY@DOMAIN>
Contact: 
<sip:4822XXXXXXX@XX.196.175.177:10060;ws-src-ip=89.77.189.69;ws-src-port=63516;w
s-src-proto=ws;transport=udp>
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
CSeq: 1085715611 INVITE
Content-Type: application/sdp
Content-Length: 1149
Max-Forwards: 70
User-Agent: webrtc2sip Media Server 2.5.1

v=0
o=doubango 1983 678901 IN IP4 XX.196.175.177
s=-
c=IN IP4 XX.196.175.177
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 45832 RTP/AVP 8 0 3
c=IN IP4 XX.196.175.177
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:UJuKvVxUJM7kjXV6kMNUjb0H3x8ouyGn2cDQzehj
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:WcOl5JLMFwYBRJhh7PwWOp5r397ZkJS9TQFoz3JY
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:294614736 cname:9e0dd8d7079419c13fbe18cd559e48c5
a=ssrc:294614736 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:294614736 label:doubango@audio
a=ice-ufrag:klhikTbGLGt9hJk
a=ice-pwd:tBTl12au6dxM3u0e5ekpd
a=candidate:i0AONn7jl 1 udp 2130706431 XX.196.175.177 45832 typ host
a=candidate:i0AONn7jl 2 udp 2130706430 XX.196.175.177 45833 typ host
a=candidate:srflxi0AO 1 udp 1694498815 XX.221.199.202 45832 typ srflx raddr 
XX.196.175.177 rport 45832
a=candidate:srflxi0AO 2 udp 1694498814 XX.221.199.202 45833 typ srflx raddr 
XX.196.175.177 rport 45833

*INFO: 

RECV:SIP/2.0 100 Trying
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
Via: SIP/2.0/UDP 
XX.196.175.177:10060;received=XX.221.199.202;branch=z9hG4bK-887038794;rport=1006
0
To: <sip:YYYYYYYYY@DOMAIN>
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
CSeq: 1085715611 INVITE
Date: Tue, 06 Aug 2013 17:13:10 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO: 

RECV:SIP/2.0 407 Proxy Authentication Required
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
Via: SIP/2.0/UDP 
XX.196.175.177:10060;received=XX.221.199.202;branch=z9hG4bK-887038794;rport=1006
0
To: <sip:YYYYYYYYY@DOMAIN>;tag=51e7df1b-1375809190751172-gm-po-lucentPCSF-098447
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
CSeq: 1085715611 INVITE
Date: Tue, 06 Aug 2013 17:13:10 GMT
Proxy-Authenticate: Digest realm="REALM",
  nonce="b7c9036dbf3054aea52012ea4940e9703dc8f84c1708",
  opaque="ALU:000332.23132", algorithm=MD5, qop="auth"
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

***ERROR: function: "tsip_message_parser_execute()" 
file: "src/parsers/tsip_parser_message.c" 
line: "466" 
MSG: Failed to parse header - Proxy-Authenticate: Digest realm="REALM",
  nonce="b7c9036dbf3054aea52012ea4940e9703dc8f84c1708",
  opaque="ALU:000332.23132", algorithm=MD5, qop="auth"
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: 

SEND: ACK sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/UDP XX.196.175.177:10060;branch=z9hG4bK-887038794;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
To: <sip:YYYYYYYYY@DOMAIN>;tag=51e7df1b-1375809190751172-gm-po-lucentPCSF-098447
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
CSeq: 1085715611 ACK
Content-Length: 0
Max-Forwards: 70

*INFO: State machine: x0000_Any_2_Any_X_i401_407_Challenge
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: 

SEND: INVITE sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/UDP XX.196.175.177:10060;branch=z9hG4bK-1037984342;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
To: <sip:YYYYYYYYY@DOMAIN>
Contact: 
<sip:4822XXXXXXX@XX.196.175.177:10060;ws-src-ip=89.77.189.69;ws-src-port=63516;w
s-src-proto=ws;transport=udp>
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
CSeq: 1085715612 INVITE
Content-Type: application/sdp
Content-Length: 1149
Max-Forwards: 70
Proxy-Authorization: Digest 
username="4822XXXXXXXb@DOMAIN",realm="REALM",nonce="b7c9036dbf3054aea52012ea4940
e9703dc8f84c1708",uri="sip:YYYYYYYYY@DOMAIN",response="964d2104d0ce3ef6c7fa5e925
6faf8fe",algorithm=MD5,cnonce="6a4f2b70213e361e2535a1ca2b4643f3",opaque="ALU:000
332.23132",qop=auth,nc=00000001
User-Agent: webrtc2sip Media Server 2.5.1

v=0
o=doubango 1983 678901 IN IP4 XX.196.175.177
s=-
c=IN IP4 XX.196.175.177
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 45832 RTP/AVP 8 0 3
c=IN IP4 XX.196.175.177
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:UJuKvVxUJM7kjXV6kMNUjb0H3x8ouyGn2cDQzehj
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:WcOl5JLMFwYBRJhh7PwWOp5r397ZkJS9TQFoz3JY
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:294614736 cname:9e0dd8d7079419c13fbe18cd559e48c5
a=ssrc:294614736 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:294614736 label:doubango@audio
a=ice-ufrag:klhikTbGLGt9hJk
a=ice-pwd:tBTl12au6dxM3u0e5ekpd
a=candidate:i0AONn7jl 1 udp 2130706431 XX.196.175.177 45832 typ host
a=candidate:i0AONn7jl 2 udp 2130706430 XX.196.175.177 45833 typ host
a=candidate:srflxi0AO 1 udp 1694498815 XX.221.199.202 45832 typ srflx raddr 
XX.196.175.177 rport 45832
a=candidate:srflxi0AO 2 udp 1694498814 XX.221.199.202 45833 typ srflx raddr 
XX.196.175.177 rport 45833

*INFO: 

RECV:SIP/2.0 100 Trying
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
Via: SIP/2.0/UDP 
XX.196.175.177:10060;received=XX.221.199.202;branch=z9hG4bK-1037984342;rport=100
60
To: <sip:YYYYYYYYY@DOMAIN>
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
CSeq: 1085715612 INVITE
Date: Tue, 06 Aug 2013 17:13:10 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO: 

RECV:SIP/2.0 403 Forbidden
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
Via: SIP/2.0/UDP 
XX.196.175.177:10060;received=XX.221.199.202;branch=z9hG4bK-1037984342;rport=100
60
To: <sip:YYYYYYYYY@DOMAIN>;tag=51e7df1b-1375809190911165-gm-po-lucentPCSF-094305
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
CSeq: 1085715612 INVITE
Date: Tue, 06 Aug 2013 17:13:10 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(23)
*INFO: CloseSocket(24)
*INFO: 

SEND: ACK sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/UDP XX.196.175.177:10060;branch=z9hG4bK-1037984342;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=74534049
To: <sip:YYYYYYYYY@DOMAIN>;tag=51e7df1b-1375809190911165-gm-po-lucentPCSF-094305
Call-ID: 5644e3f9-645e-2270-f8dc-365a2317dd9e
CSeq: 1085715612 ACK
Content-Length: 0
Max-Forwards: 70

*INFO: State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
*INFO: === INVITE Dialog terminated ===
*INFO: === ICT terminated ===
*INFO: *** ICT destroyed ***
*INFO: === ICT terminated ===
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER -- STOP
*INFO: ICE CTX::run -- STOP
*INFO: CloseSocket(22)
*INFO: CloseSocket(21)
*INFO: State machine: s0000_Ringing_2_Terminated_X_Reject
*INFO: State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699
*INFO: === INVITE Dialog terminated ===
*INFO: State machine: tsip_transac_ist_Any_2_Terminated_X_cancel
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER -- STOP
*INFO: === ICT terminated ===
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** ICT destroyed ***
*INFO: ICE CTX::run -- STOP
*INFO: CloseSocket(18)
*INFO: CloseSocket(19)
*INFO: [Stream] Removed call-id = 'e9a121b7-7bff-4af1-d876-6db9cbf63858' from 
peer with local fd = 17
*INFO: [Transport] Removed call-id = 'e9a121b7-7bff-4af1-d876-6db9cbf63858' 
from transport with type = 64
*INFO: [Transport Layer] Removed call-id = 
'e9a121b7-7bff-4af1-d876-6db9cbf63858' from transport layer
*INFO: *** INVITE Dialog destroyed ***
*INFO: MPPeer object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: *** SIP Session destroyed ***
*INFO: MPSipSession object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: MPSipSession object destroyed
*INFO: *** SIP Session destroyed ***
*INFO: Receiving SIP o/ WebSocket message: ACK sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKOmTRHXJiRTep3Ibz9yJebZzZ2fUz2oOa;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=Uq6TA2vqTgPPIcyvetHk
To: <sip:YYYYYYYYY@DOMAIN>;tag=1958521788
Call-ID: e9a121b7-7bff-4af1-d876-6db9cbf63858
CSeq: 48750 ACK
Content-Length: 0
Max-Forwards: 70

*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(18)
*INFO: CloseSocket(19)
*INFO: Sending DNS query to "XX.16.0.23"
*INFO: CloseSocket(18)
*INFO: CloseSocket(19)
*INFO: 

SEND: ACK sip:YYYYYYYYY@DOMAIN SIP/2.0
Via: SIP/2.0/UDP 
XX.196.175.177:10060;branch=z9hG4bKOmTRHXJiRTep3Ibz9yJebZzZ2fUz2oOa;rport
From: <sip:4822XXXXXXX@DOMAIN>;tag=Uq6TA2vqTgPPIcyvetHk
To: <sip:YYYYYYYYY@DOMAIN>;tag=1958521788
Call-ID: e9a121b7-7bff-4af1-d876-6db9cbf63858
CSeq: 48750 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 
89.77.189.69:63516;rport;branch=z9hG4bKOmTRHXJiRTep3Ibz9yJebZzZ2fUz2oOa;ws-hacke
d=WS

Original comment by dawid.mi...@gmail.com on 7 Aug 2013 at 11:45

GoogleCodeExporter commented 9 years ago
We had the same issue with webrtc2sip and asterisk. Our problem was that the 
webrtc breaker run the response algorithm with the realm parameter from the web 
browser and not the realm parameter that asterisk supplied in the SIP/2.0 401 
Unauthorized WWW-Authenticate: Digest message.

Realm parameter from website was 192.168.1.125

This is the WEBRTC-BREAKER -><- ASTERISK sip signaling
<--- Received SIP request (1473 bytes) from UDP:192.168.1.125:10060 --->
INVITE sip:06@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.125:10060;branch=z9hG4bK-1258469664;rport
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>
Contact: 
<sip:rtc_phone_erik@192.168.1.125:10060;ws-src-ip=192.168.1.159;ws-src-port=5435
4;ws-src-proto=ws;transport=udp>
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
CSeq: 333404276 INVIT
Content-Type: application/sdp
Content-Length: 973
Max-Forwards: 70
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 192.168.1.125
s=-
c=IN IP4 192.168.1.125
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 51398 RTP/AVP 8 0 101
c=IN IP4 192.168.1.125
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/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:YP95A+pZfF0p2rBvTW8iZznP6xvEbP6i8Wq0JMrL
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:8k51OjSCbq4zuDpAPHPNK5Fyldx3KeqM7e+7DN19
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:3380065476 cname:e57cb64764a77a7f24727b617b27628f
a=ssrc:3380065476 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3380065476 label:doubango@audio
a=ice-ufrag:sdmlmitEG7rB3sw
a=ice-pwd:nbXi8RV1zn4X0ck7EmmDBl
a=candidate:HgZfhbx9n 1 udp 2130706431 192.168.1.125 51398 typ host
a=candidate:HgZfhbx9n 2 udp 2130706430 192.168.1.125 51399 typ host

<--- Transmitting SIP response (466 bytes) to UDP:192.168.1.125:10060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.1.125:10060;rport;received=192.168.1.125;branch=z9hG4bK-1258469664
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>;tag=z9hG4bK-1258469664
CSeq: 333404276 INVITE
WWW-Authenticate: Digest  
realm="asterisk",nonce="1394638916/e54eb22826f2cbcff90061cb62be6aff",opaque="415
187fb4a71d790",algorithm=md5,qop="auth"
Content-Length:  0

<--- Received SIP request (318 bytes) from UDP:192.168.1.125:10060 --->
ACK sip:06@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.125:10060;branch=z9hG4bK-1258469664;rport
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>;tag=z9hG4bK-1258469664
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
CSeq: 333404276 ACK
Content-Length: 0
Max-Forwards: 70

<--- Received SIP request (1763 bytes) from UDP:192.168.1.125:10060 --->
INVITE sip:06@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.125:10060;branch=z9hG4bK-22495700;rport
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>
Contact: 
<sip:rtc_phone_erik@192.168.1.125:10060;ws-src-ip=192.168.1.159;ws-src-port=5435
4;ws-src-proto=ws;transport=udp>
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
CSeq: 333404277 INVITE
Content-Type: application/sdp
Content-Length: 973
Max-Forwards: 70
Authorization: Digest 
username="rtc_phone_erik",realm="asterisk",nonce="1394638916/e54eb22826f2cbcff90
061cb62be6aff",uri="sip:06@192.168.1.125",response="2c9aa18a205a65b5f4247731a5d8
caef",algorithm=md5,cnonce="2b5e0bb91955628cc9bea97db326db95",opaque="415187fb4a
71d790",qop=auth,nc=00000001
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 192.168.1.125
s=-
c=IN IP4 192.168.1.125
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 51398 RTP/AVP 8 0 101
c=IN IP4 192.168.1.125
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/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:YP95A+pZfF0p2rBvTW8iZznP6xvEbP6i8Wq0JMrL
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:8k51OjSCbq4zuDpAPHPNK5Fyldx3KeqM7e+7DN19
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:3380065476 cname:e57cb64764a77a7f24727b617b27628f
a=ssrc:3380065476 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3380065476 label:doubango@audio
a=ice-ufrag:sdmlmitEG7rB3sw
a=ice-pwd:nbXi8RV1zn4X0ck7EmmDBl
a=candidate:HgZfhbx9n 1 udp 2130706431 192.168.1.125 51398 typ host
a=candidate:HgZfhbx9n 2 udp 2130706430 192.168.1.125 51399 typ host

<--- Transmitting SIP response (462 bytes) to UDP:192.168.1.125:10060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.1.125:10060;rport;received=192.168.1.125;branch=z9hG4bK-22495700
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>;tag=z9hG4bK-22495700
CSeq: 333404277 INVITE
WWW-Authenticate: Digest  
realm="asterisk",nonce="1394638916/e54eb22826f2cbcff90061cb62be6aff",opaque="4d8
e3aa71aec5e51",algorithm=md5,qop="auth"
Content-Length:  0

<--- Received SIP request (314 bytes) from UDP:192.168.1.125:10060 --->
ACK sip:06@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.125:10060;branch=z9hG4bK-22495700;rport
From: <sip:rtc_phone_erik@192.168.1.125>;tag=584297862
To: <sip:06@192.168.1.125>;tag=z9hG4bK-22495700
Call-ID: cdae4454-4041-6ba6-027a-4110f0bb6793
CSeq: 333404277 ACK
Content-Length: 0
Max-Forwards: 70

Original comment by wx3...@gmail.com on 13 Mar 2014 at 3:19

GoogleCodeExporter commented 9 years ago
Good morning,
Is that have you successfully deployed a webphone  locally(without connected to 
a ICE)?
when I try to call I get this error message: 
"State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE",
Please,if anyone can help me?!
Thank you

Original comment by l_zaoua...@esi.dz on 19 Jun 2014 at 2:06