Thank you for this post, we have all we need to implement a successful call. But I still have a problem with my ICE support. I have debian 32 bit installed, I didn't find the libuuid and I made apt-get install uuid uuid-dev (then http://forums.digium.com/configure for Asterisk and make && make install.), but I still have the error "Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd" in my Sipml5 debug. Can you help me?
herre is my debug and my file.conf:
sip.conf:
Code:
[general]
udpbindaddr=0.0.0.0:5060
realm=192.168.1.105
transport=udp,ws
[6001]
host=dynamic
context=from-internal
username=6001
callerid=6001 <6001>
secret=azerty
type=friend
disallow=all
allow=ulaw
allow=alaw
[6002]
type=friend
context=from-internal
host=dynamic
username=6002
callerid=6002 <6002>
secret=azerty
encryption=yes
avpf=yes
icesupport=yes
directmedia=no
disallow=all
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
http.conf
Code:
[general]
enabled=yes
bindport=8088
bindaddr=0.0.0.0
enablestatic=yes
rtp.conf:
Code:
[general]
icesupport=yes
stunaddr=
extensions.conf
Code:
[from-internal]
exten => 6002,1,Dial(SIP/6002,15)
exten => 6001,1,Dial(SIP/6001,15)
my rtp debug:
Code:
== Using SIP RTP CoS mark 5
-- Executing [6001@from-internal:1] Dial("SIP/6002-00000002", "SIP/6001,15") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6001
-- SIP/6001-00000003 is ringing
> 0x930e570 -- Probation passed - setting RTP source address to 192.168.1.105:8000
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032396, ts 2264604707, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032397, ts 2264604867, len 000160)
-- SIP/6001-00000003 answered SIP/6002-00000002
> 0x930e570 -- Probation passed - setting RTP source address to 192.168.1.105:8000
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032398, ts 2264605027, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028050, ts 2264605024, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032399, ts 2264605187, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028051, ts 2264605184, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032400, ts 2264605347, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028052, ts 2264605344, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032401, ts 2264605507, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028053, ts 2264605504, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032402, ts 2264605667, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028054, ts 2264605664, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032403, ts 2264605827, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028055, ts 2264605824, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032404, ts 2264605987, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028056, ts 2264605984, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032405, ts 2264606147, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028057, ts 2264606144, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032406, ts 2264606307, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028058, ts 2264606304, len 000160)
Got RTP packet from 192.168.1.105:8000 (type 00, seq 032407, ts 2264606467, len 000160)
Sent RTP packet to 192.168.1.104:52117 (type 00, seq 028059, ts 2264606464, len 000160)
my sip debug:
Code:
<--- SIP read from WS:192.168.1.104:59439 --->
INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
<------------->
--- (12 headers 38 lines) ---
Using INVITE request as basis request - c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
Found peer '6002' for '6002' from 192.168.1.104:59439
<--- Reliably Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport;received=192.168.1.104
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.105", nonce="6ad3c739"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' in 32000 ms (Method: INVITE)
<--- SIP read from WS:192.168.1.104:59439 --->
ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 ACK
Content-Length: 0
Max-Forwards: 70
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from WS:192.168.1.104:59439 --->
INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105",response="455fa3f20baf557b4a1bf192fdb1d935",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
<------------->
--- (13 headers 38 lines) ---
Using INVITE request as basis request - c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
Found peer '6002' for '6002' from 192.168.1.104:59439
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.104:42985
Looking for 6001 in from-internal (domain 192.168.1.105)
list_route: hop: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Content-Length: 0
<------------>
-- Executing [6001@from-internal:1] Dial("SIP/6002-00000006", "SIP/6001,15") in new stack
== Using SIP RTP CoS mark 5
Audio is at 25382
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.105:5061:
INVITE sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport
Max-Forwards: 70
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
Contact: <sip:6002@192.168.1.105:5060>
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.11.0
Date: Sat, 02 Aug 2014 21:06:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 2039524556 2039524556 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 25382 RTP/AVP 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=ptime:20
a=sendrecv
---
-- Called SIP/6001
<--- SIP read from UDP:192.168.1.105:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport=5060
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.105:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport=5060
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34
From: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 102 INVITE
User-Agent: Zoiper r21155
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
-- SIP/6001-00000007 is ringing
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.105:5061 --->
<------------->
Really destroying SIP dialog '9f5b3baf-2d7e-6900-494f-c4773252203c' Method: REGISTER
> 0x92f18b0 -- Probation passed - setting RTP source address to 192.168.1.105:8000
<--- SIP read from UDP:192.168.1.105:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK7e0e84b5;rport=5060
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34
From: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r21155
Allow-Events: presence, kpml
Content-Length: 308
v=0
o=Z 0 3 IN IP4 197.31.123.195
s=Z
c=IN IP4 197.31.123.195
t=0 0
m=audio 8000 RTP/AVP 0 3 110 98 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 197.31.123.195:8000
list_route: hop: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
set_destination: Parsing <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP> for address/port to send to
set_destination: set destination to 197.31.123.195:5061
Transmitting (NAT) to 192.168.1.105:5061:
ACK sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK6403d8d9;rport
Max-Forwards: 70
From: "6002" <sip:6002@192.168.1.105>;tag=as698a9a90
To: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34
Contact: <sip:6002@192.168.1.105:5060>
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.11.0
Content-Length: 0
---
-- SIP/6001-00000007 answered SIP/6002-00000006
Audio is at 21190
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport;received=192.168.1.104
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 2120884642 2120884642 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 21190 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
<------------>
> 0x92f18b0 -- Probation passed - setting RTP source address to 192.168.1.105:8000
<--- SIP read from WS:192.168.1.104:59439 --->
ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQCwpPqNNDIZ481uSBwI;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105:5060;transport=WS",response="9954906e8debce0780742e3b9237ea1a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.105:5061 --->
BYE sip:6002@192.168.1.105:5060 SIP/2.0
Via: SIP/2.0/UDP 197.31.123.195:5061;branch=z9hG4bK-d8754z-43d5b88923bed3f0-1---d8754z-
Max-Forwards: 70
Contact: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>
To: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90
From: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 2 BYE
User-Agent: Zoiper r21155
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.105:5061 (NAT)
Scheduling destruction of SIP dialog '4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 192.168.1.105:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 197.31.123.195:5061;branch=z9hG4bK-d8754z-43d5b88923bed3f0-1---d8754z-;received=192.168.1.105;rport=5061
From: <sip:6001@197.31.123.195:5061;rinstance=ebc175da0ec03d8d;transport=UDP>;tag=b31dcf34
To: "6002"<sip:6002@192.168.1.105>;tag=as698a9a90
Call-ID: 4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060
CSeq: 2 BYE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (from-internal, 6001, 1) exited non-zero on 'SIP/6002-00000006'
Scheduling destruction of SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Reliably Transmitting (no NAT) to 192.168.1.104:5060:
BYE sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc
Max-Forwards: 70
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="6002", realm="192.168.1.105", algorithm=MD5, uri="sip:192.168.1.105", nonce="6ad3c739", response="4188a9e02c874579f65ebf7b91182d76"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from WS:192.168.1.104:59439 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0' Method: INVITE
<--- SIP read from UDP:192.168.1.105:5061 --->
<------------->
Really destroying SIP dialog '4c35c0026845ff8313b0c6e14ebe9679@192.168.1.105:5060' Method: BYE
debian*CLI>
my Sipml5 debug:
Code:
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js?svn=224:116
ICE servers:[{"url":"stun:null"}] tsk_utils.js?svn=224:116
==stack event = m_permission_requested tsk_utils.js?svn=224:116
==session event = connecting tsk_utils.js?svn=224:116
onGetUserMediaSuccess tsk_utils.js?svn=224:116
createOffer tsk_utils.js?svn=224:116
==stack event = m_permission_accepted tsk_utils.js?svn=224:116
onCreateSdpSuccess tsk_utils.js?svn=224:116
==session event = m_stream_audio_local_added tsk_utils.js?svn=224:116
onSetLocalDescriptionSuccess tsk_utils.js?svn=224:116
5
onIceCandidate = undefined tsk_utils.js?svn=224:116
ICE GATHERING COMPLETED! tsk_utils.js?svn=224:116
onIceGatheringCompleted tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2732090651842997000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
m=audio 44683 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:44683 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:xokyb72LGstFdQBI
a=ice-pwd:NMP9WcQbj49BeA4LXb6sGPkH
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:2043935191 cname:9YLjUZ+QgfTBfRWd
a=ssrc:2043935191 msid:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6 4b6559f6-6a12-429f-8f46-aafb1cb2d505
a=ssrc:2043935191 mslabel:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
a=ssrc:2043935191 label:4b6559f6-6a12-429f-8f46-aafb1cb2d505
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as75ccf8c8
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="1f8890a3",stale=FALSE,algorithm=MD5
tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKgHNUD0sQwFgqFQTMZXkhXuh9TaLtyzrt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as75ccf8c8
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23605 ACK
Content-Length: 0
Max-Forwards: 70
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:6001@192.168.1.105",response="488af2ac260ef24356958095f0b45cc1",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2732090651842997000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
m=audio 44683 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:44683 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 44683 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:xokyb72LGstFdQBI
a=ice-pwd:NMP9WcQbj49BeA4LXb6sGPkH
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:2043935191 cname:9YLjUZ+QgfTBfRWd
a=ssrc:2043935191 msid:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6 4b6559f6-6a12-429f-8f46-aafb1cb2d505
a=ssrc:2043935191 mslabel:U3LAtQLc2XMs2izIxXXJL8xHU7f0FNKQIbx6
a=ssrc:2043935191 label:4b6559f6-6a12-429f-8f46-aafb1cb2d505
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKtDazRM2SUb5OvOKh2dnXLioEV7OcNgma
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 INVITE
Content-Type: application/sdp
Content-Length: 399
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
v=0
o=root 373378232 373378232 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 22550 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
tsk_utils.js?svn=224:116
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js?svn=224:116
setRemoteDescription(answer)
v=0
o=root 373378232 373378232 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 22550 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKu31tJTM5Cr0D092qmUbt;rport
From: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
To: <sip:6001@192.168.1.105>;tag=as7db2a4a3
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 23606 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:6001@192.168.1.105:5060;transport=WS",response="85b438f7d9984c926147554e58bd9fb7",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
onSetRemoteDescriptionError tsk_utils.js?svn=224:116
Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. tsk_utils.js?svn=224:128
==session event = m_early_media tsk_utils.js?svn=224:116
==session event = connected tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=BYE sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK6501c79a
From: <sip:6001@192.168.1.105>;tag=as7db2a4a3
To: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="6002",realm="192.168.1.105",nonce="1f8890a3",uri="sip:192.168.1.105",response="69ccda7dd5a7e67d599e8d557f5c8273",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js?svn=224:116
=== INVITE Dialog terminated === tsk_utils.js?svn=224:116
PeerConnection::stop() tsk_utils.js?svn=224:116
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK6501c79a
From: <sip:6001@192.168.1.105>;tag=as7db2a4a3
To: "amal"<sip:6002@192.168.1.105>;tag=1Tmp2lBCMvsyOTm5DTpE
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws>
Call-ID: 75db4220-9ef3-86c9-8fcf-e7fdf4775892
CSeq: 102 BYE
Content-Length: 0
tsk_utils.js?svn=224:116
==session event = terminated tsk_utils.js?svn=224:116
The FSM is in the final state tsk_utils.js?svn=224:122
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKikApHPC1Gtgl2bUU4fqtgrBP5Mm2baay;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57812 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="3a290ce0",uri="sip:192.168.1.105",response="715632384099d84923a8698184c46be5",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKikApHPC1Gtgl2bUU4fqtgrBP5Mm2baay
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as389399be
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57812 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="745215ab",stale=FALSE,algorithm=MD5
tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfTc0LKHak33wIaKbcKipSai69Z5RKOau;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57813 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="745215ab",uri="sip:192.168.1.105",response="bbe9b541971cab616b0293674a9af37f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKfTc0LKHak33wIaKbcKipSai69Z5RKOau
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as389399be
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57813 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21:06:12 GMT;02
tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js?svn=224:116
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js?svn=224:116
ICE servers:[{"url":"stun:null"}] tsk_utils.js?svn=224:116
==stack event = m_permission_requested tsk_utils.js?svn=224:116
==session event = connecting tsk_utils.js?svn=224:116
onGetUserMediaSuccess tsk_utils.js?svn=224:116
createOffer tsk_utils.js?svn=224:116
onCreateSdpSuccess tsk_utils.js?svn=224:116
==stack event = m_permission_accepted tsk_utils.js?svn=224:116
==session event = m_stream_audio_local_added tsk_utils.js?svn=224:116
onSetLocalDescriptionSuccess tsk_utils.js?svn=224:116
5
onIceCandidate = undefined tsk_utils.js?svn=224:116
ICE GATHERING COMPLETED! tsk_utils.js?svn=224:116
onIceGatheringCompleted tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="6ad3c739",stale=FALSE,algorithm=MD5
tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKDvVLWqTerDmfvU7ntzHJQt1sO5nFm5S2;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as03c5c0eb
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62542 ACK
Content-Length: 0
Max-Forwards: 70
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js?svn=224:116
SEND: INVITE sip:6001@192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=6002;ha1=00b7bc0aba57392cc96d2a266ff7863f;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Type: application/sdp
Content-Length: 1533
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105",response="455fa3f20baf557b4a1bf192fdb1d935",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
v=0
o=- 2577394367678132000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
m=audio 42985 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.1.104
a=rtcp:42985 IN IP4 192.168.1.104
a=candidate:1019731727 1 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1019731727 2 udp 2122260223 192.168.1.104 42985 typ host generation 0
a=candidate:1917068287 1 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=candidate:1917068287 2 tcp 1518280447 192.168.1.104 0 typ host generation 0
a=ice-ufrag:fHkUinzS5ohjJhal
a=ice-pwd:6Q37zMp1s4jjos+z63Gy3u+3
a=ice-options:google-ice
a=fingerprint:sha-256 17:AE:2A:F4:FE:8A:C9:AF:E0:32:1C:93:AA:58:C9:91:79:4A:6B:2A:63:5B:67:F2:81:1A:2E:8A:EC:DC:33:D4
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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:1974193999 cname:bB9ulmmnutOJVAtz
a=ssrc:1974193999 msid:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM d16640b3-40b2-4f7d-95cc-5808611d201a
a=ssrc:1974193999 mslabel:HiWXG86M86T3yxkyK5QCAS0zYxPMw3CGJFaM
a=ssrc:1974193999 label:d16640b3-40b2-4f7d-95cc-5808611d201a
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js?svn=224:116
==session event = i_ao_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bK1X2JJVn1ZhXRITLTzwAVwszBoWwztiOn
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: <sip:6001@192.168.1.105:5060;transport=WS>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 INVITE
Content-Type: application/sdp
Content-Length: 401
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
v=0
o=root 2120884642 2120884642 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 21190 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
tsk_utils.js?svn=224:116
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js?svn=224:116
setRemoteDescription(answer)
v=0
o=root 2120884642 2120884642 IN IP4 192.168.1.105
s=Asterisk PBX 11.11.0
c=IN IP4 192.168.1.105
t=0 0
m=audio 21190 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=connection:new
a=setup:active
a=fingerprint:SHA-256 02:9A:35:5F:0C:1B:69:0B:47:1B:B6:FB:BC:D7:96:09:D9:BC:BA:62:12:A9:4F:83:D2:CC:2C:CD:58:97:F1:51
a=sendrecv
tsk_utils.js?svn=224:116
SEND: ACK sip:6001@192.168.1.105:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGQCwpPqNNDIZ481uSBwI;rport
From: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
To: <sip:6001@192.168.1.105>;tag=as6731b4d6
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 62543 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:6001@192.168.1.105:5060;transport=WS",response="9954906e8debce0780742e3b9237ea1a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
onSetRemoteDescriptionError tsk_utils.js?svn=224:116
Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. tsk_utils.js?svn=224:128
==session event = m_early_media tsk_utils.js?svn=224:116
==session event = connected tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=BYE sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="6002",realm="192.168.1.105",nonce="6ad3c739",uri="sip:192.168.1.105",response="4188a9e02c874579f65ebf7b91182d76",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
tsk_utils.js?svn=224:116
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js?svn=224:116
=== INVITE Dialog terminated === tsk_utils.js?svn=224:116
PeerConnection::stop() tsk_utils.js?svn=224:116
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.105:5060;branch=z9hG4bK3fda03bc
From: <sip:6001@192.168.1.105>;tag=as6731b4d6
To: "amal"<sip:6002@192.168.1.105>;tag=Lr6AXBrFObVO4H9emhf3
Contact: <sip:6002@df7jal23ls0d.invalid;transport=ws>
Call-ID: c6e3a7ef-ca6e-41cb-2ef8-a0d39826ccf0
CSeq: 102 BYE
Content-Length: 0
tsk_utils.js?svn=224:116
==session event = terminated tsk_utils.js?svn=224:116
The FSM is in the final state tsk_utils.js?svn=224:122
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKl1MKbTRKWcaQFxEWKxR6KJVXuOvhg1sO;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57814 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="745215ab",uri="sip:192.168.1.105",response="bbe9b541971cab616b0293674a9af37f",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKl1MKbTRKWcaQFxEWKxR6KJVXuOvhg1sO
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as68f45007
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57814 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="321f2a10",stale=FALSE,algorithm=MD5
tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKQmvhBdCeANbzmq5TiKBAGgWTtegsU3RW;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57815 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="321f2a10",uri="sip:192.168.1.105",response="f2bbde081390003688b3211e6b39a325",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
==session event = sent_request tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKQmvhBdCeANbzmq5TiKBAGgWTtegsU3RW
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as68f45007
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57815 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21:07:52 GMT;02
tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb2xiaSxCloGbsthdprWT6ooMhSlRLp8k;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57816 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="321f2a10",uri="sip:192.168.1.105",response="f2bbde081390003688b3211e6b39a325",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKb2xiaSxCloGbsthdprWT6ooMhSlRLp8k
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as7d54f249
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57816 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm="192.168.1.105",nonce="3be23638",stale=FALSE,algorithm=MD5
tsk_utils.js?svn=224:116
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=224:116
SEND: REGISTER sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKwXwTSJ8OyziIgtBcRzJ2Zxy3BHydOHVv;rport
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>
Contact: "amal"<sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57817 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="6002",realm="192.168.1.105",nonce="3be23638",uri="sip:192.168.1.105",response="1905b36637ff0044f263772837a8cbc4",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
tsk_utils.js?svn=224:116
__tsip_transport_ws_onmessage tsk_utils.js?svn=224:116
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.1.104;branch=z9hG4bKwXwTSJ8OyziIgtBcRzJ2Zxy3BHydOHVv
From: "amal"<sip:6002@192.168.1.105>;tag=rzwgKaw5hEdoCSxbFdtg
To: "amal"<sip:6002@192.168.1.105>;tag=as7d54f249
Contact: <sip:6002@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 9f5b3baf-2d7e-6900-494f-c4773252203c
CSeq: 57817 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.11.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 02 Aug 2014 21:09:33 GMT;02
Statistics : Posted by amal_kammoun • on Sun Aug 03, 2014 6:37 am • Replies 1 • Views 131