webrtc2sip asterisk

Hello
i have failed to make asterisk communicate with webrtc2sip
i have installed webrtc2sip and sipML5 and vanilla asterisk 11.2 (all working fine)
but my asterisk is not communicating with webrtc2sip in any way
who do i make them communicate such that when i make a call with asterisk
it goes through webrtc2sip?

That is part of the sipml5 AIP, you need to use the websocket the url of the webrtc2sip, in Outbound proxy the URL of your Asterisk. Then you can call without issue.

At least you need to provide the logs of webrtc2sip, Browser & asterisk for a failed call.

hey …!!

i had the same problem,

) I downloaded the SIPml5 and hardcoded sip stack to avoid registration ,in order to send Invite packet directly ,
2) Set up a call between chrome browser on MAC osX to a PSTN number through Asterisk server (SIP signalling). I can hear the audio with out any problem i can say.
3) when i check the LOG files using tcpdump/wireshark, My shark catches that asterisk is able to connect both ends as a part of signalling , but not direct client to client. So realising that PSTN not consists same profile as webrtc -client (ie., chrome browser) ,i built and installed webrtc2sip . Now ,I can see webrtc2sip is running on server.
and configured as

Webrtc-client (Chrome-OSX) —>> Webrtc Gateway(WEBRTC2SIP) —>> SIP gateway (Asterisk Server) ----->> PSTN

Now , when i called , everything seems broken, links to the log files to trace the call ,

asterisk log ::
gist.github.com/karna41317/6020648

webrtc log ::

gist.github.com/karna41317/6020638

Please guys help here,

Thanks in adavance .

Your logs show a forbidden error. So seems like your authentication data is wrong. I’m guessing about the realm did you changed the asterisk’s realm to match your sipml5 client?

Also re check your webrtc2sip installation there are many errors.

Hello Navaismo and all!!

Please, can you help me?

I have a problem with asterisk 11.9.0 and webrtc2sip and sipml5.
The issue is that :
At the moment to call a sip user I have one ring tone and then the message - Call in progres… Remote ring… and “Not acceptable here”.

  • I can’t do a call never!!

Please help me with some suggestions about this problem! Here the logs.

Thanks !
Regards

[code]*******************************************************************
Copyright © 2012-2013 Doubango Telecom http://www.doubango.org
PRODUCT: webrtc2sip
HOME PAGE: http://webrtc2sip.org
LICENCE: GPLv3 or proprietary
VERSION: 2.6.0
’quit’ to quit the application.


SSL is enabled :slight_smile:
DTLS supported: yes
DTLS-SRTP supported: yes
INFO: transport = udp://:10060
INFO: transport = ws://:10061
INFO: transport = wss://:10062
*INFO: enable-rtp-symetric = yes
*INFO: enable-100rel = no
*INFO: enable-media-coder = yes
*INFO: enable-videojb = yes
*INFO: video-size-pref = vga
*INFO: rtp-buffsize = 65535
*INFO: avpf-tail-length = [100-400]
*INFO: srtp-mode = optional
*INFO: srtp-type = sdes;dtls
*INFO: dtmf-type = rfc4733
*INFO: codecs = vp8
*INFO: UnRegister codec: VP8, VP8 codec (libvpx)
*INFO: codec-opus-maxrates = 48000;48000
*INFO: enable-icestun = yes
*INFO: max-fds = -1
*INFO: ssl-certificates =
/home/webrtc/Escritorio/ssl/private.pem;
/home/webrtc/Escritorio/ssl/webrtc_gbt_tfo_upm_es.pem;
/home/webrtc/Escritorio/ssl/AddTrustExternalCARoot.pem;

INFO: transport = c2c://:10070
INFO: transport = c2cs://:10072
INFO: database = sqlite;
*INFO: sqlite3_threadsafe = 1
*INFO: Database opened = TRUE
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=8
*INFO: Socket added[TCP/IPv4 transport]: fd=8, tail.count=1
*INFO: master fd=3
*INFO: Socket added[TCP/IPv4 transport]: fd=3, tail.count=2
*INFO: Transport::run() - enter
*INFO: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {10070} using fd {3} with type {9}…
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=10
*INFO: Socket added[TLS/IPv4 transport]: fd=10, tail.count=1
*INFO: master fd=4
*INFO: Socket added[TLS/IPv4 transport]: fd=4, tail.count=2
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Transport::run() - enter
*INFO: Starting [TLS/IPv4 transport] server with IP {0.0.0.0} on port {10072} using fd {4} with type {17}…
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER – START
*INFO: SIP STACK::run – START
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=15
*INFO: Socket added[SIP transport]: fd=15, tail.count=1
*INFO: master fd=12
*INFO: Socket added[SIP transport]: fd=12, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=17
*INFO: Transport::run() - enter
*INFO: Socket added[SIP transport]: fd=17, tail.count=1
*INFO: master fd=13
*INFO: Socket added[SIP transport]: fd=13, tail.count=2
*INFO: Starting [SIP transport] server with IP {138.4.10.174} on port {10060} using fd {12} with type {2}…
*INFO: tnet_transport_prepare()
*INFO: Transport::run() - enter
*INFO: pipeR fd=19
*INFO: Socket added[SIP transport]: fd=19, tail.count=1
*INFO: master fd=14
*INFO: Socket added[SIP transport]: fd=14, tail.count=2
*INFO: Starting [SIP transport] server with IP {138.4.10.174} on port {10061} using fd {13} with type {64}…
*INFO: SIP STACK – START
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {138.4.10.174} on port {10062} using fd {14} with type {128}…
*INFO: ioctlt(14), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] – FD_ACCEPT(fd=21)
*INFO: Socket added[SIP transport]: fd=21, tail.count=3
*INFO: WebSocket Peer accepted/connected with fd = 21
*INFO: #1 peers in the ‘SIP transport’ transport
*INFO: NETWORK EVENT FOR SERVER [SIP transport] – TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 21
*INFO: *** Stream Peer destroyed ***
*INFO: #0 peers in the ‘SIP transport’ transport
*INFO: #1 peers in the ‘SIP transport’ transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: webrtc.gbt.tfo.upm.es:10062
Origin: http://webrtc.gbt.tfo.upm.es
Sec-WebSocket-Protocol: sip
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: lBM1/08F3ZFNpJP3NO+7PQ==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits, x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_9_3) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/35.0.1916.114 Safari/537.36
Cookie: __utma=159177788.1081614721.1400851988.1401185982.1401200012.6; __utmb=159177788.4.10.1401200012; __utmc=159177788; __utmz=159177788.1400851988.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none)

*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKnCrFpCa1o6y6ExzwnRlhlTZd6w0Vyg1Q;rport
From: "101"sip:101@138.4.10.174;tag=X5JnnCns1vBib0VGEIg3
To: “101"sip:101@138.4.10.174
Contact: “101"sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: 05c62119-629d-2895-b51b-a0388ec6bdaa
CSeq: 24746 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“101”,realm=“138.4.10.174”,nonce=””,uri=“sip:138.4.10.174”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 138.4.10.100:53299;rport;branch=z9hG4bKnCrFpCa1o6y6ExzwnRlhlTZd6w0Vyg1Q;ws-hacked=WSS

*INFO:

RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKnCrFpCa1o6y6ExzwnRlhlTZd6w0Vyg1Q;received=138.4.10.174;rport=10060
Via: SIP/2.0/TCP 138.4.10.100:53299;rport;branch=z9hG4bKnCrFpCa1o6y6ExzwnRlhlTZd6w0Vyg1Q;ws-hacked=WSS
From: "101"sip:101@138.4.10.174;tag=X5JnnCns1vBib0VGEIg3
To: "101"sip:101@138.4.10.174;tag=as0f3b76cc
Call-ID: 05c62119-629d-2895-b51b-a0388ec6bdaa
CSeq: 24746 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“138.4.10.174”, nonce="4f27e69f"
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKvSh8wlISLkhxaWZZS6zhfI2BuZ2GJhef;rport
From: "101"sip:101@138.4.10.174;tag=X5JnnCns1vBib0VGEIg3
To: "101"sip:101@138.4.10.174
Contact: "101"sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 05c62119-629d-2895-b51b-a0388ec6bdaa
CSeq: 24747 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“101”,realm=“138.4.10.174”,nonce=“4f27e69f”,uri=“sip:138.4.10.174”,response=“8844578bd335673b5cd064e27666ebab”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 138.4.10.100:53299;rport;branch=z9hG4bKvSh8wlISLkhxaWZZS6zhfI2BuZ2GJhef;ws-hacked=WSS

*INFO:

RECV:OPTIONS sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:5060;branch=z9hG4bK2e5c3700;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@138.4.10.174;tag=as4a1aca55
To: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss
Contact: sip:asterisk@138.4.10.174:5060
Call-ID: 7761269621652dc81cc9c2a1091ab407@138.4.10.174:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.9.0
Date: Tue, 27 May 2014 14:19:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKvSh8wlISLkhxaWZZS6zhfI2BuZ2GJhef;received=138.4.10.174;rport=10060
Via: SIP/2.0/TCP 138.4.10.100:53299;rport;branch=z9hG4bKvSh8wlISLkhxaWZZS6zhfI2BuZ2GJhef;ws-hacked=WSS
From: "101"sip:101@138.4.10.174;tag=X5JnnCns1vBib0VGEIg3
To: "101"sip:101@138.4.10.174;tag=as0f3b76cc
Call-ID: 05c62119-629d-2895-b51b-a0388ec6bdaa
CSeq: 24747 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss;expires=200
Date: Tue, 27 May 2014 14:19:54 GMT
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:

SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 138.4.10.174:5060;rport=5060;received=138.4.10.174;branch=z9hG4bK2e5c3700
From: "asterisk"sip:asterisk@138.4.10.174;tag=as4a1aca55
To: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=53299;ws-src-proto=wss
Call-ID: 7761269621652dc81cc9c2a1091ab407@138.4.10.174:5060
CSeq: 102 OPTIONS
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: ioctlt(14), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] – FD_ACCEPT(fd=22)
*INFO: Socket added[SIP transport]: fd=22, tail.count=4
*INFO: WebSocket Peer accepted/connected with fd = 22
*INFO: #2 peers in the ‘SIP transport’ transport
*INFO: NETWORK EVENT FOR SERVER [SIP transport] – TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 22
*INFO: *** Stream Peer destroyed ***
*INFO: #1 peers in the ‘SIP transport’ transport
*INFO: #2 peers in the ‘SIP transport’ transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: webrtc.gbt.tfo.upm.es:10062
Origin: http://webrtc.gbt.tfo.upm.es
Sec-WebSocket-Protocol: sip
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: Co5ppBNZXQ800TOju2XM7w==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits, x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_9_3) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/35.0.1916.114 Safari/537.36
Cookie: wm_alu_sessid=11eorkblk95agjej8unlglj9d2; mailviewsplitterv=165; __utma=159177788.227534629.1401093370.1401139769.1401198939.7; __utmb=159177788.18.10.1401198939; __utmc=159177788; __utmz=159177788.1401093370.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none)

*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK3wd3eXLO2znNmhAc5tvSy51P54CuZ9rS;rport
From: "104"sip:104@138.4.10.174;tag=QS4HsC310PiEq31ToBMT
To: “104"sip:104@138.4.10.174
Contact: “104"sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: 44756f56-aa04-77dc-36e9-a052e50c6e04
CSeq: 38680 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“104”,realm=“138.4.10.174”,nonce=””,uri=“sip:138.4.10.174”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 138.4.10.93:52031;rport;branch=z9hG4bK3wd3eXLO2znNmhAc5tvSy51P54CuZ9rS;ws-hacked=WSS

*INFO:

RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK3wd3eXLO2znNmhAc5tvSy51P54CuZ9rS;received=138.4.10.174;rport=10060
Via: SIP/2.0/TCP 138.4.10.93:52031;rport;branch=z9hG4bK3wd3eXLO2znNmhAc5tvSy51P54CuZ9rS;ws-hacked=WSS
From: "104"sip:104@138.4.10.174;tag=QS4HsC310PiEq31ToBMT
To: "104"sip:104@138.4.10.174;tag=as5b922d2e
Call-ID: 44756f56-aa04-77dc-36e9-a052e50c6e04
CSeq: 38680 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“138.4.10.174”, nonce="6fba683f"
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKIEtuaRW5G7sf7R2pZWTy8LMcar7bgmsM;rport
From: "104"sip:104@138.4.10.174;tag=QS4HsC310PiEq31ToBMT
To: "104"sip:104@138.4.10.174
Contact: "104"sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 44756f56-aa04-77dc-36e9-a052e50c6e04
CSeq: 38681 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“104”,realm=“138.4.10.174”,nonce=“6fba683f”,uri=“sip:138.4.10.174”,response=“1f21cbd9551ef4b20bc0b80132a8df5f”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path
Via: SIP/2.0/TCP 138.4.10.93:52031;rport;branch=z9hG4bKIEtuaRW5G7sf7R2pZWTy8LMcar7bgmsM;ws-hacked=WSS

*INFO:

RECV:OPTIONS sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:5060;branch=z9hG4bK22c27ae2;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@138.4.10.174;tag=as081fb2ca
To: sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss
Contact: sip:asterisk@138.4.10.174:5060
Call-ID: 74b3014b21e75871579713433eea2d64@138.4.10.174:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.9.0
Date: Tue, 27 May 2014 14:19:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:

RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bKIEtuaRW5G7sf7R2pZWTy8LMcar7bgmsM;received=138.4.10.174;rport=10060
Via: SIP/2.0/TCP 138.4.10.93:52031;rport;branch=z9hG4bKIEtuaRW5G7sf7R2pZWTy8LMcar7bgmsM;ws-hacked=WSS
From: "104"sip:104@138.4.10.174;tag=QS4HsC310PiEq31ToBMT
To: "104"sip:104@138.4.10.174;tag=as5b922d2e
Call-ID: 44756f56-aa04-77dc-36e9-a052e50c6e04
CSeq: 38681 REGISTER
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 200
Contact: sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss;expires=200
Date: Tue, 27 May 2014 14:19:55 GMT
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO:

SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 138.4.10.174:5060;rport=5060;received=138.4.10.174;branch=z9hG4bK22c27ae2
From: "asterisk"sip:asterisk@138.4.10.174;tag=as081fb2ca
To: sip:104@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss
Call-ID: 74b3014b21e75871579713433eea2d64@138.4.10.174:5060
CSeq: 102 OPTIONS
Content-Length: 0

*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***

*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO: Add call-id = ‘6a3f788b-d681-f4db-e4f7-e653c9574098’ to peer with local fd = 22
*INFO: State machine: x0500_Current_2_Current_X_iINVITE
*INFO: tnet_ice_ctx_set_remote_candidates
*INFO: tnet_ice_ctx_set_remote_candidates
*INFO: tsk_timer_manager_start
*INFO: tsk_timer_manager_start
*INFO: ICE CTX::run – START
*INFO: State machine: ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: ICE CTX::run – START
*INFO: State machine: ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER – START
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER – START
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports [138.4.10.174:57996]
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports [138.4.10.174:54718]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 138.4.10.174
*INFO: State machine: ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: numb.viagenie.ca:3478
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexiveCandidates
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 138.4.10.174
*INFO: State machine: ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: numb.viagenie.ca:3478
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexiveCandidates
*INFO: ICE reflexive candidates gathering …0,500000
*INFO: ICE reflexive candidates gathering …0,500000
*INFO: Skipping redundant candidate address=138.4.10.174 and port=57996, fd=23, already_skipped(0)=no
*INFO: Skipping redundant candidate address=138.4.10.174 and port=57997, fd=26, already_skipped(1)=no
*INFO: srflx_addr_count_added=0, srflx_addr_count_skipped=2
*INFO: Candidate: zgmvtg92kWtkjWk 1 udp 2130706431 138.4.10.174 57996 typ host
*INFO: Candidate: zgmvtg92kWtkjWk 2 udp 2130706430 138.4.10.174 57997 typ host
*INFO: State machine: ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Success
*INFO: ICE callback: Gathering reflexive candidates succeed
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: State machine: ICE_GatheringCompleted_2_ConnChecking_X_ConnCheck
*INFO: ICE Pair: [zgmvtg92kWtkjWk 1 138.4.10.174 57996] -> [1418740086 1 192.168.1.143 61285]
*INFO: Skipping redundant candidate address=138.4.10.174 and port=54719, fd=24, already_skipped(0)=no
*INFO: ICE reflexive candidates gathering …1,0
*INFO: Skipping redundant candidate address=138.4.10.174 and port=54718, fd=25, already_skipped(1)=no
*INFO: srflx_addr_count_added=0, srflx_addr_count_skipped=2
*INFO: Candidate: FwkTkaOHreDle0x 1 udp 2130706431 138.4.10.174 54718 typ host
*INFO: Candidate: FwkTkaOHreDle0x 2 udp 2130706430 138.4.10.174 54719 typ host
*INFO: State machine: ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Success
*INFO: ICE callback: Gathering reflexive candidates succeed
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: ICE: ignore processing SDP RO because version haven’t changed
*INFO: is_ice_active=1,
is_ro_hold_resume_changed=0,
is_ro_provisional_final_matching=0,
is_ro_media_lines_changed=0,
is_ro_network_info_changed=0,
is_ro_loopback_address=0,
is_media_type_changed=0,
is_ro_codecs_changed=0

*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
*INFO: Video ‘zero-artifacts’ option = no
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: ICE enabled on RTP manager
*INFO: Remote party requested bandwidth limitation at 64 using ‘b=AS’ SDP attribute
*INFO: Setting bandwidth_max_upload_kbps=64 according to remote party request
*INFO: Remote SSRC = 4224778704
*INFO: dtls.remote.setup=actpass
*INFO: ICE enabled on RTP manager
*INFO: Remote party requested bandwidth limitation at 512 using ‘b=AS’ SDP attribute
*INFO: Setting bandwidth_max_upload_kbps=512 according to remote party request
*INFO: Remote SSRC = 3075075166
*INFO: dtls.remote.setup=actpass
*INFO: No codec matching for media type = 2
*INFO: dtls.remote.setup=passive
*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO: State machine: ICE_GatheringCompleted_2_ConnChecking_X_ConnCheck
*INFO: ICE Pair: [FwkTkaOHreDle0x 1 138.4.10.174 54718] -> [1418740086 1 192.168.1.143 61285]
*INFO: State machine: x0500_Current_2_Current_X_oINVITE
*INFO: tsk_timer_manager_start
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER – START
*INFO: ICE CTX::run – START
*INFO: State machine: ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports [138.4.10.174:44498]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 138.4.10.174
*INFO: State machine: ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: ICE using STUN server: numb.viagenie.ca:3478
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: ICE_GatheringHostCandidatesDone_2_GatheringReflexiveCandidates_X_GatherReflexiveCandidates
*INFO: ICE reflexive candidates gathering …0,500000
*INFO: Skipping redundant candidate address=138.4.10.174 and port=44499, fd=29, already_skipped(0)=no
*INFO: ICE reflexive candidates gathering …1,0
*INFO: Skipping redundant candidate address=138.4.10.174 and port=44498, fd=28, already_skipped(1)=no
*INFO: srflx_addr_count_added=0, srflx_addr_count_skipped=2
*INFO: Candidate: wD0ziastszVhTLa 1 udp 2130706431 138.4.10.174 44498 typ host
*INFO: Candidate: wD0ziastszVhTLa 2 udp 2130706430 138.4.10.174 44499 typ host
*INFO: State machine: ICE_fsm_GatheringReflexiveCandidates_2_GatheringReflexiveCandidatesDone_X_Success
*INFO: ICE callback: Gathering reflexive candidates succeed
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: State machine: c0000_Started_2_Outgoing_X_oINVITE
*INFO: Video ‘zero-artifacts’ option = no
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: ICE enabled on RTP manager
*INFO: dtls.remote.setup=active
*INFO:

SEND: INVITE sip:101@138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-1126428105;rport
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174
Contact: sip:104@138.4.10.174:10060;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss;transport=udp
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247513 INVITE
Content-Type: application/sdp
Content-Length: 1353
Max-Forwards: 70
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 138.4.10.174
s=-
c=IN IP4 138.4.10.174
t=0 0
a=acap:1 setup:actpass
a=acap:2 connection:new
a=acap:3 fingerprint:sha-256 E9:80:8F:72:B1:9C:09:4F:45:53:05:1A:7D:D1:46:E1:6D:3F:6C:88:F2:8A:93:64:0C:67:4B:3B:41:9B:AD:88
a=acap:4 fingerprint:sha-1 6D:02:0E:33:20:3C:22:BA:67:CB:4C:58:EE:25:92:43:92:C5:86:9E
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/SAVPF RTP/SAVP RTP/AVPF
m=video 44498 RTP/AVP 100
c=IN IP4 138.4.10.174
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack
a=rtcp-fb:* goog-remb
a=label:3
a=content:main
a=rtpmap:100 VP8/90000
a=imageattr:100 recv [x=[128:16:640],y=[96:16:480]] send [x=[128:16:640],y=[96:16:480]]
a=acap:5 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2jdgXsxuyJPk+CqQ9R1jk/WjarUeA9hnuGlF9hcR
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Y+R5EYfrY29ehgMQXbPEC7pyWcsi6LsqXNK0ld91
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3 a=5,6
a=pcfg:4 t=4 a=5,6
a=pcfg:5 t=5
a=sendrecv
a=rtcp-mux
a=ssrc:1976733574 cname:e6b4c9d5f7ad5da73da3bd8a3fc3d83c
a=ssrc:1976733574 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1976733574 label:doubango@video
a=ice-ufrag:oIkAGIpiWmSSSBf
a=ice-pwd:kehEXby5dki3fSlvKdKsWc
a=candidate:wD0ziastszVhTLa 1 udp 2130706431 138.4.10.174 44498 typ host
a=candidate:wD0ziastszVhTLa 2 udp 2130706430 138.4.10.174 44499 typ host

*INFO:

RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-1126428105;received=138.4.10.174;rport=10060
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174;tag=as0a1ffae0
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247513 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“138.4.10.174”, nonce="3e4e6376"
Content-Length: 0

***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "928"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
*INFO: Sending DNS query to “127.0.0.1”
*INFO: CloseSocket(30)
*INFO: DNS NAPTR (138.4.10.174) query returned zero result
*INFO:

SEND: ACK sip:101@138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-1126428105;rport
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174;tag=as0a1ffae0
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247513 ACK
Content-Length: 0
Max-Forwards: 70

*INFO: State machine: x0000_Any_2_Any_X_i401_407_Challenge
*INFO:

SEND: INVITE sip:101@138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-434024455;rport
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174
Contact: sip:104@138.4.10.174:10060;ws-src-ip=138.4.10.93;ws-src-port=52031;ws-src-proto=wss;transport=udp
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247514 INVITE
Content-Type: application/sdp
Content-Length: 1353
Max-Forwards: 70
Authorization: Digest username=“104”,realm=“138.4.10.174”,nonce=“3e4e6376”,uri="sip:101@138.4.10.174",response=“96c20473cd5aa3531dabef64aa0dfdb5”,algorithm=MD5
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 138.4.10.174
s=-
c=IN IP4 138.4.10.174
t=0 0
a=acap:1 setup:actpass
a=acap:2 connection:new
a=acap:3 fingerprint:sha-256 E9:80:8F:72:B1:9C:09:4F:45:53:05:1A:7D:D1:46:E1:6D:3F:6C:88:F2:8A:93:64:0C:67:4B:3B:41:9B:AD:88
a=acap:4 fingerprint:sha-1 6D:02:0E:33:20:3C:22:BA:67:CB:4C:58:EE:25:92:43:92:C5:86:9E
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/SAVPF RTP/SAVP RTP/AVPF
m=video 44498 RTP/AVP 100
c=IN IP4 138.4.10.174
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack
a=rtcp-fb:* goog-remb
a=label:3
a=content:main
a=rtpmap:100 VP8/90000
a=imageattr:100 recv [x=[128:16:640],y=[96:16:480]] send [x=[128:16:640],y=[96:16:480]]
a=acap:5 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2jdgXsxuyJPk+CqQ9R1jk/WjarUeA9hnuGlF9hcR
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Y+R5EYfrY29ehgMQXbPEC7pyWcsi6LsqXNK0ld91
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3 a=5,6
a=pcfg:4 t=4 a=5,6
a=pcfg:5 t=5
a=sendrecv
a=rtcp-mux
a=ssrc:1976733574 cname:e6b4c9d5f7ad5da73da3bd8a3fc3d83c
a=ssrc:1976733574 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1976733574 label:doubango@video
a=ice-ufrag:oIkAGIpiWmSSSBf
a=ice-pwd:kehEXby5dki3fSlvKdKsWc
a=candidate:wD0ziastszVhTLa 1 udp 2130706431 138.4.10.174 44498 typ host
a=candidate:wD0ziastszVhTLa 2 udp 2130706430 138.4.10.174 44499 typ host

*INFO:

RECV:SIP/2.0 100 Trying
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-434024455;received=138.4.10.174;rport=10060
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247514 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:101@138.4.10.174:5060
Content-Length: 0

*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO:

RECV:SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-434024455;received=138.4.10.174;rport=10060
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174;tag=as4204a7ed
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247514 INVITE
Server: Asterisk PBX 11.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "928"
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "143"
MSG: (SYSTEM)NETWORK ERROR ==>Success
*INFO: Sending DNS query to “127.0.0.1”
*INFO: CloseSocket(30)
*INFO: DNS NAPTR (138.4.10.174) query returned zero result
*INFO:

SEND: ACK sip:101@138.4.10.174 SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:10060;branch=z9hG4bK-434024455;rport
From: sip:104@138.4.10.174;tag=1754705550
To: sip:101@138.4.10.174;tag=as4204a7ed
Call-ID: 9e81eb9c-9526-b7e7-bbb9-454de8dedf79
CSeq: 1743247514 ACK
Content-Length: 0
Max-Forwards: 70

*INFO: State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
*INFO: === INVITE Dialog terminated ===
*INFO: === ICT terminated ===
*INFO: *** ICT destroyed ***
*INFO: === ICT terminated ===
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER – STOP
*INFO: CloseSocket(28)
*INFO: CloseSocket(29)
*INFO: ICE CTX::run – STOP
*INFO: === ICT terminated ===
*INFO: *** tdav_session_video_t destroyed ***
*INFO: tdav_session_video_stop
*INFO: State machine: s0000_Ringing_2_Terminated_X_Reject
*INFO: State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699
*INFO: MPProxyPluginConsumerVideo object destroyed
*INFO: twrap_producer_proxy_video_dtor()
*INFO: MPProxyPluginProducerVideo object destroyed
*INFO: ~ProxyVideoProducer
*INFO: *** RTP manager destroyed ***
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: *** Video session destroyed ***
*INFO: === INVITE Dialog terminated ===
*INFO: State machine: tsip_transac_ist_Any_2_Terminated_X_cancel
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** ICT destroyed ***
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER – STOP
*INFO: Receiving SIP o/ WebSocket message: (null)
*INFO: CloseSocket(24)
*INFO: CloseSocket(25)
*INFO: ICE CTX::run – STOP
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER – STOP
*INFO: CloseSocket(26)
*INFO: CloseSocket(23)
*INFO: ICE CTX::run – STOP
*INFO: MPPeer object destroyed
*INFO: MPSipSessionAV object destroyed
**WARN: function: "tsk_fsm_act()"
file: "src/tsk_fsm.c"
line: "133"
MSG: The FSM is in the final state.
*INFO: *** SIP Session destroyed ***
*INFO: [Stream] Removed call-id = ‘6a3f788b-d681-f4db-e4f7-e653c9574098’ from peer with local fd = 22
*INFO: [Transport] Removed call-id = ‘6a3f788b-d681-f4db-e4f7-e653c9574098’ from transport with type = 128
*INFO: [Transport Layer] Removed call-id = ‘6a3f788b-d681-f4db-e4f7-e653c9574098’ from transport layer
*INFO: MPSipSession object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: MPSipSession object destroyed
*INFO: *** SIP Session destroyed ***
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** tdav_session_video_t destroyed ***
*INFO: tdav_session_video_stop
*INFO: MPProxyPluginConsumerVideo object destroyed
*INFO: twrap_producer_proxy_video_dtor()
*INFO: MPProxyPluginProducerVideo object destroyed
*INFO: ~ProxyVideoProducer
*INFO: *** RTP manager destroyed ***
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: *** Video session destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: *** INVITE Dialog destroyed ***

[/code]

The Chrome Log

[code]{\rtf1\ansi\ansicpg1252\cocoartf1265\cocoasubrtf200
{\fonttbl\f0\fnil\fcharset0 Menlo-Regular;\f1\fnil\fcharset0 Menlo-Bold;}
{\colortbl;\red255\green255\blue255;\red36\green43\blue51;\red255\green255\blue255;\red66\green66\blue66;
\red251\green0\blue7;\red109\green132\blue175;}
{*\listtable{\list\listtemplateid1\listhybrid{\listlevel\levelnfc23\levelnfcn23\leveljc0\leveljcn0\levelfollow0\levelstartat1\levelspace360\levelindent0{*\levelmarker {none}.}{\leveltext\leveltemplateid1’01.;}{\levelnumbers;}\fi-360\li720\lin720 }{\listname ;}\listid1}}
{*\listoverridetable{\listoverride\listid1\listoverridecount0\ls1}}
\paperw11900\paperh16840\margl1440\margr1440\vieww20360\viewh14680\viewkind0
\deftab720
\pard\pardeftab720

\f0\fs22 \cf2 \cb3 SIPML5 API version = 1.4.217
location={\field{*\fldinst{HYPERLINK “http://webrtc.gbt.tfo.upm.es/sipml5/call.htm”}}{\fldrslt \cf4 http://webrtc.gbt.tfo.upm.es/sipml5/call.htm}}
User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_9_3) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/35.0.1916.114 Safari/537.36
WebSocket supported = yes
Navigator friendly name = chrome
OS friendly name = mac
Have WebRTC = yes
Have GUM = yes
Engine initialized
s_websocket_server_url={\field{*\fldinst{HYPERLINK “wss://webrtc.gbt.tfo.upm.es/”}}{\fldrslt \cf4 wss://webrtc.gbt.tfo.upm.es:10062}}
s_sip_outboundproxy_url={\field{*\fldinst{HYPERLINK “udp://138.4.10.174”}}{\fldrslt \cf4 udp://138.4.10.174:5060}}
b_rtcweb_breaker_enabled=yes
b_click2call_enabled=no
b_early_ims=no
b_enable_media_stream_cache=yes
o_bandwidth={}
o_video_size={}
SIP stack start: proxy=‘ns313841.ovh.net:13062’, realm=‘sip:138.4.10.174’, impi=‘101’, impu=’"101"sip:101@138.4.10.174
Connecting to ‘{\field{*\fldinst{HYPERLINK “wss://webrtc.gbt.tfo.upm.es/”}}{\fldrslt \cf4 wss://webrtc.gbt.tfo.upm.es:10062}}’
==stack event = starting
__tsip_transport_ws_onopen
==stack event = started
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKJtTWMtCCNu1P6JdikQEMbvTs65CyPZ9Q;rport
From: "101"sip:101@138.4.10.174;tag=edul930QTPms6w30QH3u
To: “101"sip:101@138.4.10.174
Contact: “101"sips:101@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: 848517fd-13b8-41eb-a73b-49351d8f5c55
CSeq: 23378 REGISTER
Content-Length: 0
Route: sip:138.4.10.174:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“101”,realm=“138.4.10.174”,nonce=””,uri=“sip:138.4.10.174”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

==session event = connecting
==session event = sent_request
__tsip_transport_ws_onmessage
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 138.4.10.174:10060;rport=10060;received=138.4.10.174;branch=z9hG4bKJtTWMtCCNu1P6JdikQEMbvTs65CyPZ9Q
From: "101"sip:101@138.4.10.174;tag=edul930QTPms6w30QH3u
To: "101"sip:101@138.4.10.174;tag=as4e13d479
Call-ID: 848517fd-13b8-41eb-a73b-49351d8f5c55
CSeq: 23378 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 138.4.10.100:50789;rport;branch=z9hG4bKJtTWMtCCNu1P6JdikQEMbvTs65CyPZ9Q;ws-hacked=WSS
Server: Asterisk PBX 11.9.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm=“138.4.10.174”,nonce=“39b56dda”,stale=FALSE,algorithm=MD5

State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SEND: REGISTER sip:138.4.10.174 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKsrukcMQMF2ZrB4LQM7AvbWqQbtuMgWT6;rport
From: "101"sip:101@138.4.10.174;tag=edul930QTPms6w30QH3u
To: "101"sip:101@138.4.10.174
Contact: "101"sips:101@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 848517fd-13b8-41eb-a73b-49351d8f5c55
CSeq: 23379 REGISTER
Content-Length: 0
Route: sip:138.4.10.174:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“101”,realm=“138.4.10.174”,nonce=“39b56dda”,uri=“sip:138.4.10.174”,response=“88b06e90fe9d8c68cf2c5acf1abdcd03”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom
Supported: path

==session event = sent_request
__tsip_transport_ws_onmessage
recv=OPTIONS sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=wss;ws-src-ip=138.4.10.100;ws-src-port=50789;ws-src-proto=wss SIP/2.0
Via: SIP/2.0/UDP 138.4.10.174:5060;rport=5060;received=138.4.10.174;branch=z9hG4bK1bec55c6
From: "asterisk"sip:asterisk@138.4.10.174;tag=as4b9137d0
To: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=50789;ws-src-proto=wss
Contact: sips:asterisk@138.4.10.174:10062;transport=wss
Call-ID: 060c08902796b7bf72b57a4c40d397ad@138.4.10.174:5060
CSeq: 102 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.9.0
Date: 27 May 2014 10:49:57 GMT;27
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

\pard\tx220\tx720\pardeftab720\li720\fi-720\sl240
\ls1\ilvl0\cf5 {\listtext . }Not implemented\cf2
\pard\pardeftab720
\cf2 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 138.4.10.174:5060;rport=5060;received=138.4.10.174;branch=z9hG4bK1bec55c6
From: "asterisk"sip:asterisk@138.4.10.174;tag=as4b9137d0
To: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=50789;ws-src-proto=wss
Call-ID: 060c08902796b7bf72b57a4c40d397ad@138.4.10.174:5060
CSeq: 102 OPTIONS
Content-Length: 0

__tsip_transport_ws_onmessage
recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 138.4.10.174:10060;rport=10060;received=138.4.10.174;branch=z9hG4bKsrukcMQMF2ZrB4LQM7AvbWqQbtuMgWT6
From: "101"sip:101@138.4.10.174;tag=edul930QTPms6w30QH3u
To: "101"sip:101@138.4.10.174;tag=as4e13d479
Contact: sip:101@138.4.10.174:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=138.4.10.100;ws-src-port=50789;ws-src-proto=wss;expires=200
Call-ID: 848517fd-13b8-41eb-a73b-49351d8f5c55
CSeq: 23379 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP 138.4.10.100:50789;rport;branch=z9hG4bKsrukcMQMF2ZrB4LQM7AvbWqQbtuMgWT6;ws-hacked=WSS
Server: Asterisk PBX 11.9.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Date: 27 May 2014 10:49:57 GMT;27

State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
==session event = connected
State machine: c0000_Started_2_Outgoing_X_oINVITE
PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] }
Video Contraints:{“mandatory”:{},“optional”:[]}
ICE servers:[]
==stack event = m_permission_requested
==session event = connecting
onGetUserMediaSuccess
createOffer
==stack event = m_permission_accepted
onCreateSdpSuccess
==session event = m_stream_video_local_added
==session event = m_stream_audio_local_added
onSetLocalDescriptionSuccess
\pard\pardeftab720

\f1\b \cf3 \cb6 9
\pard\pardeftab720

\f0\b0 \cf2 \cb3 onIceCandidate = undefined
ICE GATHERING COMPLETED!
onIceGatheringCompleted
SEND: INVITE sip:104@138.4.10.174 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKsC0IRL01IFfxt3fJRbKQZxLeWa9cLLVv;rport
From: "101"sip:101@138.4.10.174;tag=pNNZfZcsUA2dWRnTUqm9
To: sip:104@138.4.10.174
Contact: "101"sips:101@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss;impi=101;ha1=8c3dc0afbd42a4e4268cb25f5cb13d3e;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: b7f0a0b4-3dd4-d77a-f6ef-e213ede60d1d
CSeq: 39189 INVITE
Content-Type: application/sdp
Content-Length: 2767
Route: sip:138.4.10.174:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 5084049374086740000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS 9VjtQcrK8I7IdAvkS4pq1fljJ26olzfe2JBZ
m=audio 59052 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 138.4.10.100
a=rtcp:59052 IN IP4 138.4.10.100
a=candidate:3316158352 1 udp 2122194687 138.4.10.100 59052 typ host generation 0
a=candidate:3316158352 2 udp 2122194687 138.4.10.100 59052 typ host generation 0
a=candidate:2334880608 1 tcp 1518214911 138.4.10.100 0 typ host generation 0
a=candidate:2334880608 2 tcp 1518214911 138.4.10.100 0 typ host generation 0
a=ice-ufrag:0LHhoPSHvl82u8o0
a=ice-pwd:3oJAKriwqkS9OMBV2lp+v+Ae
a=ice-options:google-ice
a=fingerprint:sha-256 C6:6F:0F:8F:44:82:3F:00:67:54:87:59:99:18:FE:7D:11:D4:9B:5A:B7:6B:49:43:DD:0A:E9:96:B4:4C:24:32
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 {\field{*\fldinst{HYPERLINK “http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time”}}{\fldrslt \cf4 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:3143178063 cname:JOZwGyH7wwvYVuJA
a=ssrc:3143178063 msid:9VjtQcrK8I7IdAvkS4pq1fljJ26olzfe2JBZ 58f22680-3c24-46bb-a419-3a3f918e1c25
a=ssrc:3143178063 mslabel:9VjtQcrK8I7IdAvkS4pq1fljJ26olzfe2JBZ
a=ssrc:3143178063 label:58f22680-3c24-46bb-a419-3a3f918e1c25
m=video 59052 UDP/TLS/RTP/SAVPF 100 116 117
c=IN IP4 138.4.10.100
a=rtcp:59052 IN IP4 138.4.10.100
a=candidate:3316158352 1 udp 2122194687 138.4.10.100 59052 typ host generation 0
a=candidate:3316158352 2 udp 2122194687 138.4.10.100 59052 typ host generation 0
a=candidate:2334880608 1 tcp 1518214911 138.4.10.100 0 typ host generation 0
a=candidate:2334880608 2 tcp 1518214911 138.4.10.100 0 typ host generation 0
a=ice-ufrag:0LHhoPSHvl82u8o0
a=ice-pwd:3oJAKriwqkS9OMBV2lp+v+Ae
a=ice-options:google-ice
a=fingerprint:sha-256 C6:6F:0F:8F:44:82:3F:00:67:54:87:59:99:18:FE:7D:11:D4:9B:5A:B7:6B:49:43:DD:0A:E9:96:B4:4C:24:32
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 {\field{*\fldinst{HYPERLINK “http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time”}}{\fldrslt \cf4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time}}
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:103153660 cname:JOZwGyH7wwvYVuJA
a=ssrc:103153660 msid:9VjtQcrK8I7IdAvkS4pq1fljJ26olzfe2JBZ 3d80d35f-092a-4887-9f45-2bd892851e7c
a=ssrc:103153660 mslabel:9VjtQcrK8I7IdAvkS4pq1fljJ26olzfe2JBZ
a=ssrc:103153660 label:3d80d35f-092a-4887-9f45-2bd892851e7c
__tsip_transport_ws_onmessage
recv=SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport;branch=z9hG4bKsC0IRL01IFfxt3fJRbKQZxLeWa9cLLVv
From: "101"sip:101@138.4.10.174;tag=pNNZfZcsUA2dWRnTUqm9
To: sip:104@138.4.10.174
Call-ID: b7f0a0b4-3dd4-d77a-f6ef-e213ede60d1d
CSeq: 39189 INVITE
Content-Length: 0

State machine: x0000_Any_2_Any_X_i1xx
==session event = i_ao_request
__tsip_transport_ws_onmessage
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport;branch=z9hG4bKsC0IRL01IFfxt3fJRbKQZxLeWa9cLLVv
From: "101"sip:101@138.4.10.174;tag=pNNZfZcsUA2dWRnTUqm9
To: sip:104@138.4.10.174;tag=47759686
Contact: sips:104@138.4.10.174:10062;transport=wss;ws-src-ip=138.4.10.100;ws-src-port=50789;ws-src-proto=wss
Call-ID: b7f0a0b4-3dd4-d77a-f6ef-e213ede60d1d
CSeq: 39189 INVITE
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

State machine: x0000_Any_2_Any_X_i1xx
==session event = i_ao_request
__tsip_transport_ws_onmessage
recv=SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport;branch=z9hG4bKsC0IRL01IFfxt3fJRbKQZxLeWa9cLLVv
From: "101"sip:101@138.4.10.174;tag=pNNZfZcsUA2dWRnTUqm9
To: sip:104@138.4.10.174;tag=47759686
Call-ID: b7f0a0b4-3dd4-d77a-f6ef-e213ede60d1d
CSeq: 39189 INVITE
Content-Length: 0
Reason: text=“Not Acceptable Here”;cause=488;text="Not Acceptable Here"

SEND: ACK sip:104@138.4.10.174 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKsC0IRL01IFfxt3fJRbKQZxLeWa9cLLVv;rport
From: "101"sip:101@138.4.10.174;tag=pNNZfZcsUA2dWRnTUqm9
To: sip:104@138.4.10.174;tag=47759686
Call-ID: b7f0a0b4-3dd4-d77a-f6ef-e213ede60d1d
CSeq: 39189 ACK
Content-Length: 0
Route: sip:138.4.10.174:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70

State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
=== INVITE Dialog terminated ===
PeerConnection::stop()
==session event = i_ao_request
==session event = terminated
The FSM is in the final state
}[/code]

and the asterisk info

Connected to Asterisk 11.9.0 currently running on webrtc-System-Product-Name (pid = 7695) [May 27 12:19:44] NOTICE[7727]: chan_sip.c:23645 handle_response_peerpoke: Peer '101' is now Reachable. (26ms / 2000ms) [May 27 12:20:25] WARNING[7727][C-00000003]: channel.c:940 ast_best_codec: Don't know any of (vp8) formats [May 27 12:20:25] NOTICE[7839][C-00000003]: chan_sip.c:29842 sip_request_call: Asked to get a channel of unsupported format (vp8) while capability is (gsm|ulaw|alaw|h263|vp8|testlaw) [May 27 12:20:25] WARNING[7839][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available) [May 27 12:20:25] WARNING[7839][C-00000003]: channel.c:940 ast_best_codec: Don't know any of (vp8) formats webrtc-System-Product-Name*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups