No audio SIP

SIP Debugging enabled
  == WebSocket connection from '192.168.88.174:49537' for protocol 'sip' accepted using version '13'

<--- SIP read from WS:192.168.88.174:49537 --->
REGISTER sip:192.168.88.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport
From: "888"<sip:[email protected]>;tag=nnfL8hLlvU1QirQeN3rT
To: "888"<sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
CSeq: 6286 REGISTER
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="",uri="sip:192.168.88.251",response=""
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom
Supported: path

<------------->
--- (14 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEVrP6oaHlKMKSVsL8mNqNbUNEyCRLAix;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=nnfL8hLlvU1QirQeN3rT
To: "888"<sip:[email protected]>;tag=as7404acc2
Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
CSeq: 6286 REGISTER
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="09171086"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.88.174:49537 --->
REGISTER sip:192.168.88.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport
From: "888"<sip:[email protected]>;tag=nnfL8hLlvU1QirQeN3rT
To: "888"<sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
CSeq: 6287 REGISTER
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="09171086",uri="sip:192.168.88.251",response="7392735b12a3d08cee2d5aa586a8f225",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom
Supported: path

<------------->
--- (14 headers 0 lines) ---
    -- Registered SIP '888' at 192.168.88.174:49537

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9itTOa16GZuXBxqB0gVrvwIqUupWWEOJ;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=nnfL8hLlvU1QirQeN3rT
To: "888"<sip:[email protected]>;tag=as7404acc2
Call-ID: ef62664f-934f-f862-e3c4-8dd91614c319
CSeq: 6287 REGISTER
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200
Date: Fri, 13 Feb 2015 04:15:38 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.88.174:49537 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26632 INVITE
Content-Type: application/sdp
Content-Length: 1590
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 437287842917936700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.174
a=rtcp:65003 IN IP4 192.168.88.174
a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=ice-ufrag:E3Z2rHZDSTilnu0z
a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
a=ice-options:google-ice
a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
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:9 G722/8000
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:3257448580 cname:gVHxZ+t4jTqxgc7F
a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
<------------->
--- (13 headers 39 lines) ---
Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
Found peer '888' for '888' from 192.168.88.174:49537

<--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>;tag=as29afbf84
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26632 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="5a02570a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)

<--- SIP read from WS:192.168.88.174:49537 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6uvIL7mfS50zv3wPWgmuKKl2fpXS6P6A;rport
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>;tag=as29afbf84
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26632 ACK
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:192.168.88.174:49537 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;impi=888;ha1=5d8dc1eb3104434b361bab5960b4630d;+g.oma.sip-im;language="en,fr"
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26633 INVITE
Content-Type: application/sdp
Content-Length: 1590
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:[email protected]",response="d328cce01e71bf1f662eae00a41ed1aa",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 437287842917936700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
m=audio 65003 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.174
a=rtcp:65003 IN IP4 192.168.88.174
a=candidate:159100432 1 udp 2122194687 192.168.88.174 65003 typ host generation 0
a=candidate:159100432 2 udp 2122194687 192.168.88.174 65003 typ host generation 0
a=candidate:1207456480 1 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=candidate:1207456480 2 tcp 1518214911 192.168.88.174 0 typ host tcptype active generation 0
a=ice-ufrag:E3Z2rHZDSTilnu0z
a=ice-pwd:cNMYG2dhwGJYfd8lcIrt7KOp
a=ice-options:google-ice
a=fingerprint:sha-256 CE:1E:A7:8A:C9:F9:0D:CF:FB:54:C5:97:D0:9D:BE:F8:26:D8:DD:D8:F3:46:70:C1:B8:DB:DC:31:04:EA:A6:08
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:9 G722/8000
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:3257448580 cname:gVHxZ+t4jTqxgc7F
a=ssrc:3257448580 msid:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0 d94562ac-e817-42ea-a235-87799cbc8335
a=ssrc:3257448580 mslabel:EhyZqtzGo6CAQ7yjqcMh1Bjfy6aKaaH2stX0
a=ssrc:3257448580 label:d94562ac-e817-42ea-a235-87799cbc8335
<------------->
--- (14 headers 39 lines) ---
Using INVITE request as basis request - 10add82a-a61b-ce04-6232-186a076dbbb2
Found peer '888' for '888' from 192.168.88.174:49537
  == 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 9
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 audio 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 G722 for ID 9
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|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.88.174:65003
Looking for 889 in default (domain 192.168.88.251)
sip_route_dump: route/path hop: <sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26633 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [889@default:1] Dial("SIP/888-0000005f", "SIP/889") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13566
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.88.187:49625:
INVITE sip:[email protected];rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
Max-Forwards: 70
From: "888" <sip:[email protected]>;tag=as176937a0
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.2.0
Date: Fri, 13 Feb 2015 04:15:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 676

v=0
o=root 1187332515 1187332515 IN IP4 192.168.88.251
s=Asterisk PBX 13.2.0
c=IN IP4 192.168.88.251
t=0 0
m=audio 13566 RTP/SAVPF 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ice-ufrag:2870a6d64720723e773aa0364395d0f4
a=ice-pwd:1c240a294ddc471a2b5ba61158281a7d
a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 13566 typ host
a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 13567 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
a=sendrecv

---
    -- Called SIP/889

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
From: "888"<sip:[email protected]>;tag=as176937a0
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
From: "888"<sip:[email protected]>;tag=as176937a0
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
Contact: <sip:[email protected];transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
    -- SIP/889-00000060 is ringing

<--- Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>;tag=as4947f672
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26633 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Length: 0


<------------>

<--- SIP read from WS:192.168.88.187:49625 --->
REGISTER sip:192.168.88.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport
From: "889"<sip:[email protected]>;tag=V0GF27XnCoBLdq6gumel
To: "889"<sip:[email protected]>
Contact: "889"<sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
CSeq: 30884 REGISTER
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="889",realm="192.168.88.251",nonce="094b5901",uri="sip:192.168.88.251",response="a635f03607d21873ed8a4a74975b97cc",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (13 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.88.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKuQhVLOJDz5XT2EJwebnyGXT1VPO8EVqI;rport;received=192.168.88.187
From: "889"<sip:[email protected]>;tag=V0GF27XnCoBLdq6gumel
To: "889"<sip:[email protected]>;tag=as0ddfbb67
Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
CSeq: 30884 REGISTER
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="13fe51c6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.88.187:49625 --->
REGISTER sip:192.168.88.251 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport
From: "889"<sip:[email protected]>;tag=V0GF27XnCoBLdq6gumel
To: "889"<sip:[email protected]>
Contact: "889"<sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
CSeq: 30885 REGISTER
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="889",realm="192.168.88.251",nonce="13fe51c6",uri="sip:192.168.88.251",response="a1a4fccb04ffa5cef37a37267ce4cc4e",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (13 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.88.187:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb3EFaRA5eEscWVDyK0pb9N5aZMeePlGP;rport;received=192.168.88.187
From: "889"<sip:[email protected]>;tag=V0GF27XnCoBLdq6gumel
To: "889"<sip:[email protected]>;tag=as0ddfbb67
Call-ID: 2c53a493-007a-22af-b2c6-81a635c44174
CSeq: 30885 REGISTER
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;expires=200
Date: Fri, 13 Feb 2015 04:15:49 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'd49645e7-984f-10b5-863d-bb5631bf9285' Method: REGISTER

<--- SIP read from WS:192.168.88.187:49625 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK13c19a3d
From: "888"<sip:[email protected]>;tag=as176937a0
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
Contact: <sip:[email protected];transport=ws>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 1171
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=- 7710960356793537000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=msid-semantic: WMS 85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
m=audio 50026 UDP/TLS/RTP/SAVPF 0 8 101
c=IN IP4 192.168.88.187
a=rtcp:50027 IN IP4 192.168.88.187
a=candidate:2577307183 1 udp 2122194687 192.168.88.187 50026 typ host generation 0
a=candidate:2577307183 2 udp 2122194686 192.168.88.187 50027 typ host generation 0
a=candidate:3609029343 1 tcp 1518214911 192.168.88.187 0 typ host tcptype active generation 0
a=candidate:3609029343 2 tcp 1518214910 192.168.88.187 0 typ host tcptype active generation 0
a=ice-ufrag:awyAyzrZujG0M9zq
a=ice-pwd:z7uMvoajOVqtt3hHg+yAhqmj
a=fingerprint:sha-256 D8:C4:BF:59:B9:A8:19:A0:4C:31:BA:92:F0:62:A0:3E:27:D4:90:9B:79:33:E3:B6:FC:E9:2A:EB:C3:D3:DF:E6
a=setup:active
a=mid:audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:372692495 cname:pkv1+OHhG1pbR11N
a=ssrc:372692495 msid:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO 4890e5af-7a2d-4d88-a39f-32216cbc0d0d
a=ssrc:372692495 mslabel:85S5W4p6YLClPLcb233yVDhLm3rS47Cg80UO
a=ssrc:372692495 label:4890e5af-7a2d-4d88-a39f-32216cbc0d0d
<------------->
--- (10 headers 25 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/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 192.168.88.187:50026
sip_route_dump: route/path hop: <sip:[email protected];transport=ws>
[Feb 13 06:15:59] ERROR[1055][C-00000031]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Feb 13 06:15:59] WARNING[1055][C-00000031]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
set_destination: Parsing <sip:[email protected];transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (no NAT) to 192.168.88.187:49625:
ACK sip:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3a7ff3e4
Max-Forwards: 70
From: "888" <sip:[email protected]>;tag=as176937a0
To: <sip:[email protected];rtcweb-breaker=yes;transport=ws>;tag=7NfJJeWFCMBeSuhOFJ3T
Contact: <sip:[email protected]:5060;transport=WS>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.2.0
Content-Length: 0


---
    -- SIP/889-00000060 answered SIP/888-0000005f
Audio is at 16190
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.88.174:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKn2DWM80i1TTWW7WT2mKSeTTTr5nvsiBg;rport;received=192.168.88.174
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>;tag=as4947f672
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26633 INVITE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 673

v=0
o=root 380639406 380639406 IN IP4 192.168.88.251
s=Asterisk PBX 13.2.0
c=IN IP4 192.168.88.251
t=0 0
m=audio 16190 RTP/SAVPF 0 8 3 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=ice-ufrag:5c8047ce65b257e40fd0093a148878ce
a=ice-pwd:4988ef1c009a1c717d3e1c2444e686ff
a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 16190 typ host
a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 16191 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
a=sendrecv

<------------>
    -- Channel SIP/888-0000005f joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
    -- Channel SIP/889-00000060 joined 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>

<--- SIP read from WS:192.168.88.174:49537 --->
ACK sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKClVdt1n8PFKSu96LXuzf;rport
From: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
To: <sip:[email protected]>;tag=as4947f672
Contact: "888"<sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 26633 ACK
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="888",realm="192.168.88.251",nonce="5a02570a",uri="sip:[email protected]:5060;transport=WS",response="780d8ba6cec222db3fac9c9f8de84e87",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (13 headers 0 lines) ---
       > 0x7fd8389ed7e0 -- Probation passed - setting RTP source address to 192.168.88.187:50026

<--- SIP read from WS:192.168.88.187:49625 --->
BYE sip:[email protected]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport
From: <sip:[email protected]>;tag=7NfJJeWFCMBeSuhOFJ3T
To: "888"<sip:[email protected]>;tag=as176937a0
Call-ID: [email protected]:5060
CSeq: 43548 BYE
Content-Length: 0
Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
--- (15 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.88.187:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4FT3CEEh3yKT7Q2V4MrdeVOgqQPRDice;rport;received=192.168.88.187
From: <sip:[email protected]>;tag=7NfJJeWFCMBeSuhOFJ3T
To: "888"<sip:[email protected]>;tag=as176937a0
Call-ID: [email protected]:5060
CSeq: 43548 BYE
Server: Asterisk PBX 13.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/889-00000060 left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
    -- Channel SIP/888-0000005f left 'simple_bridge' basic-bridge <23ca64ea-9e43-4de3-bf73-2571ac9cb837>
  == Spawn extension (default, 889, 1) exited non-zero on 'SIP/888-0000005f'
Scheduling destruction of SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected];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 192.168.88.174:5060:
BYE sip:[email protected];rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4947f672
To: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.2.0
Proxy-Authorization: Digest username="888", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="5a02570a", response="b05e0a15f788390a4b32dd0b4e0b3ec7"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:192.168.88.174:49537 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK3ec8127b
From: <sip:[email protected]>;tag=as4947f672
To: "888"<sip:[email protected]>;tag=XeSJ7tTx1obTBnxuWVxo
Contact: <sip:[email protected];transport=ws>
Call-ID: 10add82a-a61b-ce04-6232-186a076dbbb2
CSeq: 102 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '10add82a-a61b-ce04-6232-186a076dbbb2' Method: INVITE
Really destroying SIP dialog 'ef62664f-934f-f862-e3c4-8dd91614c319' Method: REGISTER
Really destroying SIP dialog '2c53a493-007a-22af-b2c6-81a635c44174' Method: REGISTER