Asterisk + webrtc2sip + sipml5, Can't LogIn

[Ryan d'Eon]

Hello,
I have followed a few guides (http://autostatic.com/asterisk-and-sipml5-interoperability , http://code.google.com/p/sipml5/wiki/Asterisk , technical-guide-1.0.pdf ) to get :
- A LAN-connected Centos 5 server running Asterisk + webrtc2sip.
- my local OSX machine running Chrome, trying out your sipml5 demo page.
I set up :
Expert Settings
Enable RTCWeb Breaker = ticked
WebSocket Server URL = ws://192.168.1.181:10060

Registration:
Display Name: WebRTCClient
Private Identity: WebRTCClient
Public Identity: sip:WebRTC...@mydomain.ltd
Password : ****
Realm: mydomain.ltd

Everytime I click the LogIn button, with the above settings or with the many variations I've tried, I get this message:
Disconnected: Failed to connet to the server

And this from the javascript console:

s_websocket_server_url=ws://192.168.1.181:10060 SIPml-api.js?svn=179:1

s_sip_outboundproxy_url=(null) 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:10062', realm='<sip:mydomain.ltd>', impi='WebRTCClient', impu='"WebRTCClient"<sip:WebRTC...@mydomain.ltd>' SIPml-api.js?svn=179:1

Connecting to 'ws://192.168.1.181:10060' SIPml-api.js?svn=179:1

==stack event = starting SIPml-api.js?svn=179:1

__tsip_transport_ws_onerror SIPml-api.js?svn=179:1

__tsip_transport_ws_onclose SIPml-api.js?svn=179:1

==stack event = failed_to_start

Which doesn't really seem helpful, so I tried start Chrome with --enable-logging --v=7, but nothing is logged when I click LogIn.

On my Asterisk + webrtc2sip box, I'm looking at the output of ./webrtc2sip with logging set to INFO, and nothing happens when I click the button in chrome.
To make sure it wasn't a problem with closed ports, I tried `echo -n "hello" | nc -4u -v 192.168.1.181 10060` from my OSX, which worked. I can see webrtc2sip trying to handle it:
*INFO:

RECV:X


***ERROR: function: "tsip_message_parse()"
file: "src/parsers/tsip_parser_message.c"
line: "227"
MSG: Failed to parse SIP message:
*INFO:

RECV:hello


***ERROR: function: "tsip_message_parse()"
file: "src/parsers/tsip_parser_message.c"
line: "227"
MSG: Failed to parse SIP message:

Any help I could get to try out this demo I would appreciate.
Thanks.

[Soufiane EL HAMCHI]

hi Ryan :-)

Did you solve this problem ??!!

[Tom Chandler]

I have Asterisk 11.8.0-rc2 and the latest sipml5 connecting.  I can do browser to browser calls, browser to pstn, and pstn to browser.  This is done without using webrtc2sip.  Is the type of problems that your are having, if so I would be glad to share my configurations.

Cheers

Tom C

--
You received this message because you are subscribed to the Google Groups "discuss-doubango" group.
To unsubscribe from this group and stop receiving emails from it, send an email to doubango+u...@googlegroups.com.
For more options, visit https://groups.google.com/groups/opt_out.

[jinxmcg]

Can you please post your configs. I cannot get PSTN to work without the webrtc2sip.

Thank you.

On Saturday, March 1, 2014 8:59:53 PM UTC+2, Tom Chandler wrote:

I have Asterisk 11.8.0-rc2 and the latest sipml5 connecting.  I can do browser to browser calls, browser to pstn, and pstn to browser.  This is done without using webrtc2sip.  Is the type of problems that your are having, if so I would be glad to share my configurations.

Cheers

Tom C

On Sat, Mar 1, 2014 at 4:26 AM, Soufiane EL HAMCHI <soufiane...@gmail.com> wrote:

hi Ryan :-)

Did you solve this problem ??!!

And this from the javascript console:

s_websocket_server_url=ws://192.168.1.181:10060 SIPml-api.js?svn=179:1

s_sip_outboundproxy_url=(null) 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:10062', realm='<sip:mydomain.ltd>', impi='WebRTCClient', impu='"WebRTCClient"<sip:WebRTCCl...@mydomain.ltd>' SIPml-api.js?svn=179:1

[Tom Chandler]

Listed below are notes on how I made it work:

I used the following software and configuration files:

Standard Asterisk-11.5.0.
    ./configure --with-crypto --with-ssl --with-srtp
    make menuselect  (just click on save and exit)
    make
    make install
    make samples

Download via svn the sipml5 code.


Configure sip.conf
    [general]
    udpbindaddr=0.0.0.0:5060
    realm=doubango.org    <= change this for your location
    transport=udp,ws,wss   

Configure http.conf
    enabled=yes
    bindaddr=0.0.0.0
    bindport=8088

Configure rtp.conf
    stunaddr=stun.l.google.com:19302

Create users for testing

[1060]            ;sipml5 browswer 
type=friend
username=1060
host=dynamic
secret=1234
qualify=yes
context=local
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
disallow=all
allow=ulaw

[2250]                ;sipml5
type=friend
secret=1234
host=dynamic
context=local
qualify=yes
transport=udp
directmedia=no
videosupport=no
icesupport=yes
disallow=all
allow=ulaw


extensions.conf

    [local]
    exten => 1060,1,Dial(SIP/1060)
    exten => 1060,2,Hangup

    exten => 2250,1,Dial(SIP/2250)
    exten => 2250,2,Hangup

    exten => 9000,1,Answer        ;I use this for testing each extensions
    exten => 9000,2,MusiconHold()
    exten => 9000,3,Hangup


compile and install Asterisk.  Modify these config files.  Then start Asterisk
load the call.htm demo page set the parameters and it should register.
If it registers, then test with musiconhold.  You should get music.
Do the same for the sip extension.

If both play musiconhold then try a call between the two.  I "should work"

Any comments/question, I will be glad to help.

Tom C

For more options, visit https://groups.google.com/d/optout.

[Tom Chandler]

Followup

I changed directmedia=yes, changed the allow to allow=ulaw,vp8, turned on video in the browser.  I can now make audio/video calls between two call.htm through Asterisk......

Then tested to ensure that I could call sip endpoint, and it works with the above changes. 

So you can make calls via Asterisk, both video and audio.

Tom C

[Cristian Malaia]

Thank you very much for your time and patience. My issue is PSTN related.

As I have read on this group I have to use webrtc2sip.

Thank you again.

You received this message because you are subscribed to a topic in the Google Groups "discuss-doubango" group.
To unsubscribe from this topic, visit https://groups.google.com/d/topic/doubango/os2Ai5kOiiM/unsubscribe.
To unsubscribe from this group and all its topics, send an email to doubango+u...@googlegroups.com.

[anil kumar k]

hi 

I followed u r notes but still not able to login both chrome and google getting "disconnected" after long time. my asterisk cli is == WebSocket connection from '210.89.60.250:55005' for protocol 'sip' accepted using version '13'

please help me in this

[Chris Mcgrath]

How do you install the sipml5 page? What directory do you put it in? How do you tell Asterisk the directory its in?

[Kair]

Chris,

If you already have a running apache on linux, you MAY do

cd /var/www/html
svn checkout http://sipml5.googlecode.com/svn/trunk/ sipml5

then in browser go

http://your-linux-machine-ip/sipml5/call.htm

This sample/demo pages of sipml5 has no relation to Asterisk.