Disconnected/Unautorized running siplm5 live demo with sip2sip recommended

Hello Sir,

I am new to webrtc , I have build webtrc2sip on my ubuntu 12.04 LTS.
Now I am trying to run a live demo with account created on sip2sip.info
Here are my entries--

In Expert mode 
WebSocket Server URL --> ws://192.168.10.127:10060
SIP outbound Proxy URL --> udp://sip2sip.info:5060

In Registration
Private Identity --> nikhil.deore
Public Identity --> sip:nikhil.deore@sip2sip.info
Realm --> sip2sip.info

Here is JS Console log-->

SIPML5 API version = 1.3.203 SIPml-api.js?svn=179:1
location=http://sipml5.org/call.htm# call.htm:147
User-Agent=Mozilla/5.0 (X11; Linux i686) AppleWebKit/537.36 (KHTML, like Gecko) 
Ubuntu Chromium/31.0.1650.63 Chrome/31.0.1650.63 Safari/537.36 
SIPml-api.js?svn=179:1
WebSocket supported = yes SIPml-api.js?svn=179:1
Navigator friendly name = chrome SIPml-api.js?svn=179:1
OS friendly name = linux SIPml-api.js?svn=179:1
Have WebRTC = yes SIPml-api.js?svn=179:1
Have GUM = yes SIPml-api.js?svn=179:1
Engine initialized SIPml-api.js?svn=179:1
event.returnValue is deprecated. Please use the standard event.preventDefault() 
instead. jquery.js:2
s_websocket_server_url=ws://192.168.10.127:10060 SIPml-api.js?svn=179:1
s_sip_outboundproxy_url=udp://sip2sip.info:5060 SIPml-api.js?svn=179:1
b_rtcweb_breaker_enabled=yes SIPml-api.js?svn=179:1
b_click2call_enabled=no SIPml-api.js?svn=179:1
b_early_ims=yes SIPml-api.js?svn=179:1
b_enable_media_stream_cache=no SIPml-api.js?svn=179:1
o_bandwidth={} SIPml-api.js?svn=179:1
o_video_size={} SIPml-api.js?svn=179:1
SIP stack start: proxy='ns313841.ovh.net:11062', realm='<sip:sip2sip.info>', 
impi='nikhil.deore', impu='<sip:nikhil.deore@sip2sip.info>' 
SIPml-api.js?svn=179:1
Connecting to 'ws://192.168.10.127:10060' SIPml-api.js?svn=179:1
==stack event = starting SIPml-api.js?svn=179:1
__tsip_transport_ws_onopen SIPml-api.js?svn=179:1
==stack event = started SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister 
SIPml-api.js?svn=179:1
SEND: REGISTER sip:sip2sip.info SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKHStI1l0txDUbDMmUEy7TTOU4XRRukdvr;rport
From: <sip:nikhil.deore@sip2sip.info>;tag=ys8fkMpSJDnFtsJfy01v
To: <sip:nikhil.deore@sip2sip.info>
Contact: 
<sip:nikhil.deore@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=
200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 21bc4d24-1781-62e6-cda6-0466ee5b08de
CSeq: 58573 REGISTER
Content-Length: 0
Route: <sip:sip2sip.info:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
Supported: path

 SIPml-api.js?svn=179:1
==session event = connecting SIPml-api.js?svn=179:1
==session event = sent_request SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_Any_2_Terminated_X_transportError 
SIPml-api.js?svn=179:1
=== REGISTER Dialog terminated === SIPml-api.js?svn=179:1
==session event = transport_error SIPml-api.js?svn=179:1
==session event = terminated SIPml-api.js?svn=179:1
The FSM is in the final state SIPml-api.js?svn=179:1

PS : I have not installed asterik should i need to install it to run live demo. 
I have attached webrtc2sip log file. Please reply me asap

Thank You
Original issue reported on code.google.com by onlyforu...@gmail.com on 9 Jan 2014 at 5:35
zhzhy86 / webrtc2sip

Disconnected/Unautorized running siplm5 live demo with sip2sip recommended #148
