webrtc2sip no audio

Hi.

I am having a trouble in having a sipml5 to call other sipml5.

I have installed and configured asterisk(version 11.10.0) + webrtc2sip(latest) + sipml5(chrome version 30.0.1599.66) to call from one box to other.

I’ve created 2 users on different server(1060 and 1061) and can make a SIP call through and answer from other side but seems like there is no audio/voice packets gets exchanged as is evident by rtp and sip debug log.

asterisk server ip is xxx.xxx.xxx.xxx
sipml5 user 1060 ip is yyy.yyy.yyy.yyy
sipml5 user 1061 ip is zzz.zzz.zzz.zzz

Here is asterisk response with rtp and sip debug enabled when user 1061 was logged in

[code]<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKHZA6qyzLT84xFrezdnbPL1VOgTwxJKho;rport
From: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=FqJgGX0g54uLM3qn8qQd
To: “1061"sip:1061@xxx.xxx.xxx.xxx
Contact: “1061"sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: e3251ac6-4e83-3b89-2735-899b75ef3854
CSeq: 45451 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“xxx.xxx.xxx.xxx”,nonce=””,uri=“sip:xxx.xxx.xxx.xxx”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP yyy.yyy.yyy.yyy:56840;rport;branch=z9hG4bKHZA6qyzLT84xFrezdnbPL1VOgTwxJKho;ws-hacked=WS

<------------->
— (14 headers 0 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKHZA6qyzLT84xFrezdnbPL1VOgTwxJKho;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP yyy.yyy.yyy.yyy:56840;rport;branch=z9hG4bKHZA6qyzLT84xFrezdnbPL1VOgTwxJKho;ws-hacked=WS
From: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=FqJgGX0g54uLM3qn8qQd
To: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=as784ff545
Call-ID: e3251ac6-4e83-3b89-2735-899b75ef3854
CSeq: 45451 REGISTER
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0d28478e"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e3251ac6-4e83-3b89-2735-899b75ef3854’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKHsM0priuEF3JgTosmoZikm5zZge8CZLP;rport
From: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=FqJgGX0g54uLM3qn8qQd
To: "1061"sip:1061@xxx.xxx.xxx.xxx
Contact: "1061"sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e3251ac6-4e83-3b89-2735-899b75ef3854
CSeq: 45452 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“asterisk”,nonce=“0d28478e”,uri=“sip:xxx.xxx.xxx.xxx”,response=“49608097512ac4b902bb1481f646677e”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP yyy.yyy.yyy.yyy:56840;rport;branch=z9hG4bKHsM0priuEF3JgTosmoZikm5zZge8CZLP;ws-hacked=WS

<------------->
— (14 headers 0 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
– Registered SIP ‘1061’ at xxx.xxx.xxx.xxx:8088

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKHsM0priuEF3JgTosmoZikm5zZge8CZLP;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP yyy.yyy.yyy.yyy:56840;rport;branch=z9hG4bKHsM0priuEF3JgTosmoZikm5zZge8CZLP;ws-hacked=WS
From: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=FqJgGX0g54uLM3qn8qQd
To: "1061"sip:1061@xxx.xxx.xxx.xxx;tag=as784ff545
Call-ID: e3251ac6-4e83-3b89-2735-899b75ef3854
CSeq: 45452 REGISTER
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;expires=200
Date: Mon, 23 Jun 2014 01:36:54 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1c7d7d6234e716c759218b5e0e1a5da3@xxx.xxx.xxx.xxx:5060’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:8088:
NOTIFY sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3d882af8;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.xxx.xxx.xxx;tag=as5d3258bb
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Contact: sip:asterisk@xxx.xxx.xxx.xxx:5060
Call-ID: 1c7d7d6234e716c759218b5e0e1a5da3@xxx.xxx.xxx.xxx:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.10.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@xxx.xxx.xxx.xxx
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘e3251ac6-4e83-3b89-2735-899b75ef3854’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK3d882af8
From: "asterisk"sip:asterisk@xxx.xxx.xxx.xxx;tag=as5d3258bb
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Call-ID: 1c7d7d6234e716c759218b5e0e1a5da3@xxx.xxx.xxx.xxx:5060
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1c7d7d6234e716c759218b5e0e1a5da3@xxx.xxx.xxx.xxx:5060’ Method: NOTIFY
Really destroying SIP dialog ‘4478235908b7bf6066f9b5063e45a88d@xxx.xxx.xxx.xxx:5060’ Method: BYE
Really destroying SIP dialog ‘ba475c58-a713-08be-5ee4-9b1afb609341’ Method: REGISTER

[/code]

And this is the asterisk response when user 1060 was logged in

[code]<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKZLJbzIq0O4ltptGEo67SOkUzU1w0WXGv;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=KpEK5nCX9GQvr9N23FDT
To: “1060"sip:1060@xxx.xxx.xxx.xxx
Contact: “1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: eaf21dc2-1073-2cea-5dee-8dca70be9866
CSeq: 24416 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“xxx.xxx.xxx.xxx”,nonce=””,uri=“sip:xxx.xxx.xxx.xxx”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56672;rport;branch=z9hG4bKZLJbzIq0O4ltptGEo67SOkUzU1w0WXGv;ws-hacked=WS

<------------->
— (14 headers 0 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKZLJbzIq0O4ltptGEo67SOkUzU1w0WXGv;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56672;rport;branch=z9hG4bKZLJbzIq0O4ltptGEo67SOkUzU1w0WXGv;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=KpEK5nCX9GQvr9N23FDT
To: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=as53c3eb73
Call-ID: eaf21dc2-1073-2cea-5dee-8dca70be9866
CSeq: 24416 REGISTER
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="28f0c236"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘eaf21dc2-1073-2cea-5dee-8dca70be9866’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKrkmPdVNtqXJlFZOSwnDG1AfGBtgGaEwk;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=KpEK5nCX9GQvr9N23FDT
To: "1060"sip:1060@xxx.xxx.xxx.xxx
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: eaf21dc2-1073-2cea-5dee-8dca70be9866
CSeq: 24417 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“28f0c236”,uri=“sip:xxx.xxx.xxx.xxx”,response=“8179f496a8a92d4304297666e08cd9c6”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56672;rport;branch=z9hG4bKrkmPdVNtqXJlFZOSwnDG1AfGBtgGaEwk;ws-hacked=WS

<------------->
— (14 headers 0 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
– Registered SIP ‘1060’ at xxx.xxx.xxx.xxx:8088

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKrkmPdVNtqXJlFZOSwnDG1AfGBtgGaEwk;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56672;rport;branch=z9hG4bKrkmPdVNtqXJlFZOSwnDG1AfGBtgGaEwk;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=KpEK5nCX9GQvr9N23FDT
To: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=as53c3eb73
Call-ID: eaf21dc2-1073-2cea-5dee-8dca70be9866
CSeq: 24417 REGISTER
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws;expires=200
Date: Mon, 23 Jun 2014 01:49:20 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1f62499653269b9040a4cde07d8e8d88@xxx.xxx.xxx.xxx:5060’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:8088:
NOTIFY sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK03eacf28;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.xxx.xxx.xxx;tag=as607e30cb
To: sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws
Contact: sip:asterisk@xxx.xxx.xxx.xxx:5060
Call-ID: 1f62499653269b9040a4cde07d8e8d88@xxx.xxx.xxx.xxx:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.10.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@xxx.xxx.xxx.xxx
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘eaf21dc2-1073-2cea-5dee-8dca70be9866’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK03eacf28
From: "asterisk"sip:asterisk@xxx.xxx.xxx.xxx;tag=as607e30cb
To: sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56672;ws-src-proto=ws
Call-ID: 1f62499653269b9040a4cde07d8e8d88@xxx.xxx.xxx.xxx:5060
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1f62499653269b9040a4cde07d8e8d88@xxx.xxx.xxx.xxx:5060’ Method: NOTIFY
Really destroying SIP dialog ‘05ab452d-201c-8d96-c3f8-a18e0d91e491’ Method: REGISTER[/code]

When I make a call from 1060 to 1061

[code]<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
INVITE sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 INVITE
Content-Type: application/sdp
Content-Length: 1743
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

v=0
o=- 1808848543512580900 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm
m=audio 51742 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 165.228.80.165
a=rtcp:51742 IN IP4 165.228.80.165
a=candidate:1268038522 1 udp 2113937151 zzz.zzz.zzz.zzz 51742 typ host generation 0
a=candidate:1268038522 2 udp 2113937151 zzz.zzz.zzz.zzz 51742 typ host generation 0
a=candidate:85411722 1 tcp 1509957375 zzz.zzz.zzz.zzz 0 typ host generation 0
a=candidate:85411722 2 tcp 1509957375 zzz.zzz.zzz.zzz 0 typ host generation 0
a=candidate:3179594414 1 udp 1845501695 165.228.80.165 51742 typ srflx raddr zzz.zzz.zzz.zzz rport 51742 generation 0
a=candidate:3179594414 2 udp 1845501695 165.228.80.165 51742 typ srflx raddr zzz.zzz.zzz.zzz rport 51742 generation 0
a=ice-ufrag:sTlSXUU350OXrx0A
a=ice-pwd:EAPoYDQRlifezgD9U0Yx03Fk
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:SVD8gyAeovRKa/yhjOO3B7RukXKt8t1/9kx6SQ5P
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VhhqoePtwhe/ZEKDu/TRuKBF+Fg+bmSbfiO96Rim
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:2109387433 cname:ggQEhYVQ8s9B9MDk
a=ssrc:2109387433 msid:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqma0
a=ssrc:2109387433 mslabel:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm
a=ssrc:2109387433 label:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqma0
<------------->
— (13 headers 40 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
Sending to xxx.xxx.xxx.xxx:8088 (no NAT)
Using INVITE request as basis request - c7fdd24f-dba6-582f-c28c-369a02fdaa02
Found peer ‘1060’ for ‘1060’ from xxx.xxx.xxx.xxx:8088

<— Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 INVITE
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="46f5585c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c7fdd24f-dba6-582f-c28c-369a02fdaa02’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
INVITE sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language=“en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 INVITE
Content-Type: application/sdp
Content-Length: 1743
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri="sip:1061@xxx.xxx.xxx.xxx”,response=“44c06fed4be933df30635e07170ba4fe”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;ws-hacked=WS

v=0
o=- 1808848543512580900 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm
m=audio 51742 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 165.228.80.165
a=rtcp:51742 IN IP4 165.228.80.165
a=candidate:1268038522 1 udp 2113937151 zzz.zzz.zzz.zzz 51742 typ host generation 0
a=candidate:1268038522 2 udp 2113937151 zzz.zzz.zzz.zzz 51742 typ host generation 0
a=candidate:85411722 1 tcp 1509957375 zzz.zzz.zzz.zzz 0 typ host generation 0
a=candidate:85411722 2 tcp 1509957375 zzz.zzz.zzz.zzz 0 typ host generation 0
a=candidate:3179594414 1 udp 1845501695 165.228.80.165 51742 typ srflx raddr zzz.zzz.zzz.zzz rport 51742 generation 0
a=candidate:3179594414 2 udp 1845501695 165.228.80.165 51742 typ srflx raddr zzz.zzz.zzz.zzz rport 51742 generation 0
a=ice-ufrag:sTlSXUU350OXrx0A
a=ice-pwd:EAPoYDQRlifezgD9U0Yx03Fk
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:SVD8gyAeovRKa/yhjOO3B7RukXKt8t1/9kx6SQ5P
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VhhqoePtwhe/ZEKDu/TRuKBF+Fg+bmSbfiO96Rim
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:2109387433 cname:ggQEhYVQ8s9B9MDk
a=ssrc:2109387433 msid:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqma0
a=ssrc:2109387433 mslabel:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqm
a=ssrc:2109387433 label:sYbENXojmosbTpSHgZLAmcFPM95JsCT7Ghqma0
<------------->
— (14 headers 40 lines) —
Sending to xxx.xxx.xxx.xxx:8088 (NAT)
Using INVITE request as basis request - c7fdd24f-dba6-582f-c28c-369a02fdaa02
Found peer ‘1060’ for ‘1060’ from xxx.xxx.xxx.xxx:8088
== 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 107
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 107
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 165.228.80.165:51742
Looking for 1061 in default (domain xxx.xxx.xxx.xxx)
list_route: hop: sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 INVITE
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1061@xxx.xxx.xxx.xxx:5060
Content-Length: 0

<------------>
– Executing [1061@default:1] Dial(“SIP/1060-00000004”, “SIP/1061”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14338
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:8088:
INVITE sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK45187c7c;rport
Max-Forwards: 70
From: “New User” sip:1060@xxx.xxx.xxx.xxx;tag=as171e4586
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Contact: sip:1060@xxx.xxx.xxx.xxx:5060
Call-ID: 3a181ab81f239e8125ddb79b0e9fe83c@xxx.xxx.xxx.xxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.10.0
Date: Mon, 23 Jun 2014 01:44:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322

v=0
o=root 978353590 978353590 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk PBX 11.10.0
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 14338 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hs3ljT07zhHzFD5hEo15ZWk+rifQ3Zd6cJiKJxh9


-- Called SIP/1061

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK45187c7c
From: "New User"sip:1060@xxx.xxx.xxx.xxx;tag=as171e4586
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Call-ID: 3a181ab81f239e8125ddb79b0e9fe83c@xxx.xxx.xxx.xxx:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK45187c7c
From: "New User"sip:1060@xxx.xxx.xxx.xxx;tag=as171e4586
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;tag=prA5e35h5IbTmSNtn2GF
Contact: sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Call-ID: 3a181ab81f239e8125ddb79b0e9fe83c@xxx.xxx.xxx.xxx:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
– SIP/1061-00000005 is ringing

<— Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 INVITE
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1061@xxx.xxx.xxx.xxx:5060
Content-Length: 0

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘e3251ac6-4e83-3b89-2735-899b75ef3854’ Method: REGISTER

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK45187c7c
From: "New User"sip:1060@xxx.xxx.xxx.xxx;tag=as171e4586
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;tag=prA5e35h5IbTmSNtn2GF
Contact: sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
Call-ID: 3a181ab81f239e8125ddb79b0e9fe83c@xxx.xxx.xxx.xxx:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 1290
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

v=0
o=- 6409220059572838000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=msid-semantic: WMS ROWkktTW7rXqWV8wfjn092hOdN7esGmOvNoI
m=audio 58362 RTP/SAVPF 0 101
c=IN IP4 165.228.80.165
a=rtcp:59168 IN IP4 165.228.80.165
a=candidate:2313719679 1 udp 2113937151 yyy.yyy.yyy.yyy 58362 typ host generation 0
a=candidate:2313719679 2 udp 2113937150 yyy.yyy.yyy.yyy 59168 typ host generation 0
a=candidate:3345707919 1 tcp 1509957375 yyy.yyy.yyy.yyy 0 typ host generation 0
a=candidate:3345707919 2 tcp 1509957374 yyy.yyy.yyy.yyy 0 typ host generation 0
a=candidate:2147022507 1 udp 1845501695 165.228.80.165 58362 typ srflx raddr yyy.yyy.yyy.yyy rport 58362 generation 0
a=candidate:2147022507 2 udp 1845501694 165.228.80.165 59168 typ srflx raddr yyy.yyy.yyy.yyy rport 59168 generation 0
a=ice-ufrag:FsuF2JpMDViryyO3
a=ice-pwd:mbu9SKrPaIw8+saE2ZQ9RzPO
a=mid:audio
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vGvEdU+oPdwIUZfVZ0Xxbo9yisn511vytHvmxm4N
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:3562592649 cname:UyTmbuly7H5XSHFd
a=ssrc:3562592649 msid:ROWkktTW7rXqWV8wfjn092hOdN7esGmOvNoI ROWkktTW7rXqWV8wfjn092hOdN7esGmOvNoIa0
a=ssrc:3562592649 mslabel:ROWkktTW7rXqWV8wfjn092hOdN7esGmOvNoI
a=ssrc:3562592649 label:ROWkktTW7rXqWV8wfjn092hOdN7esGmOvNoIa0
<------------->
— (10 headers 25 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 165.228.80.165:58362
list_route: hop: sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws
set_destination: Parsing sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws for address/port to send to
set_destination: set destination to xxx.xxx.xxx.xxx:8088
Transmitting (NAT) to xxx.xxx.xxx.xxx:8088:
ACK sip:1061@xxx.xxx.xxx.xxx:8088;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK44b4021b;rport
Max-Forwards: 70
From: “New User” sip:1060@xxx.xxx.xxx.xxx;tag=as171e4586
To: sip:1061@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;transport=udp;ws-src-ip=yyy.yyy.yyy.yyy;ws-src-port=56840;ws-src-proto=ws;tag=prA5e35h5IbTmSNtn2GF
Contact: sip:1060@xxx.xxx.xxx.xxx:5060
Call-ID: 3a181ab81f239e8125ddb79b0e9fe83c@xxx.xxx.xxx.xxx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.10.0
Content-Length: 0


-- SIP/1061-00000005 answered SIP/1060-00000004

Audio is at 27902
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:8088 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;received=xxx.xxx.xxx.xxx;rport=8088
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKULrw3tl0rUOWd5k7okVkxTQRmy09pXug;ws-hacked=WS
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 INVITE
Server: Asterisk PBX 11.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1061@xxx.xxx.xxx.xxx:5060
Content-Type: application/sdp
Content-Length: 324

v=0
o=root 1784975298 1784975298 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk PBX 11.10.0
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 27902 RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:NljhMLgOBBOEk954eFggVY/vUn3s7J+ShwpvYxva

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘05ab452d-201c-8d96-c3f8-a18e0d91e491’ Method: REGISTER

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as621a5054
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34912 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;ws-hacked=WS
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;branch=z9hG4bKbyzQGKo9s6Adau9chLDSkCpFtKzPZNgh;rport;ws-hacked=WS

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS

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

<— SIP read from UDP:xxx.xxx.xxx.xxx:8088 —>
ACK sip:1061@xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:8088;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport
From: "1060"sip:1060@xxx.xxx.xxx.xxx;tag=mcoeoWqkiPxA1lPsVFMK
To: sip:1061@xxx.xxx.xxx.xxx;tag=as477a9ae3
Contact: "1060"sip:1060@xxx.xxx.xxx.xxx:8088;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=zzz.zzz.zzz.zzz;ws-src-port=56647;ws-src-proto=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c7fdd24f-dba6-582f-c28c-369a02fdaa02
CSeq: 34913 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“asterisk”,nonce=“46f5585c”,uri=“sip:1061@xxx.xxx.xxx.xxx:5060”,response=“80ba94dd23d8ac890068c8066f1b5eed”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;rport;branch=z9hG4bKqZTGi0aT2obwN4QbATef;ws-hacked=WS
Via: SIP/2.0/TCP zzz.zzz.zzz.zzz:56647;branch=z9hG4bKqZTGi0aT2obwN4QbATef;rport;ws-hacked=WS

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

[/code]

This is what I see when make a call from 1061 to 1060

rtp set debug on RTP Debugging Enabled *CLI> == Spawn extension (default, 1060, 1) exited non-zero on 'SIP/1061-00000000' == Using SIP RTP CoS mark 5 -- Executing [1060@default:1] Dial("SIP/1061-00000002", "SIP/1060") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1060 -- SIP/1060-00000003 is ringing -- SIP/1060-00000003 answered SIP/1061-00000002

Asterisk settings are…

sip.conf

[general] port=5060 bindaddr=0.0.0.0 context=default transport=ws,wss,udp srvlookup=yes

users.conf

[code][1060]
type=peer
username=1060
host=dynamic
secret=1234
context=default
disallow=all
allow=ulaw
transport=udp,ws,wss
encryption=yes
avpf=yes
icesupport=yes

nat=yes,force_rport

[1061]
type=peer
username=1061
host=dynamic
secret=1234
context=default
encryption=yes
avpf=yes
icesupport=yes
nat=yes
disallow=all
allow=ulaw
transport=udp,ws,wss
[/code]

extensions.conf

[code][general]
static=yes
writeprotect=no

[default]
exten=>1060,1,Dial(SIP/1060)
exten=>1061,1,Dial(SIP/1061)

[/code]

rtp.conf

[general] icesupport=yes stunaddr=stun.l.google.com:19302 strictrtp=no rtcpinterval=6000 rtpchecksums=no

I can hear dialling sound on one end and ringing on the other end but as soon as a call is connected, can’t hear anything.

Could anyone kindly help please… I am very very desperate… :cry:

Thanks in advance!