Re: [Doubango] Asterisk + Webrtc2sip + Sipml5

[Mamadou DIOP]

1. open expert page
2. if you have your own gateway: fill WebSocket? Server URL field with the IP address and port where your webrtc2sip is listening for incoming connections (e.g ws://192.168.0.1:10060 or wss://192.168.0.1:10062). IMPORTANT:Do not forget the url scheme (ws:// or wss://)
3. check Enable RTCWeb Breaker if you want to call a SIP-legacy endpoint
4. set the SIP outbound proxy url with your asterisk binding address (e.g udp://192.168.0.1:5060)

PS: You should really read the technical guide (http://webrtc2sip.org/technical-guide-1.0.pdf) and Google to find the hundreds of guides on how to configure sipml5+webrtc2sip+asterisk

On Apr 30, 2013, at 12:48 PM, the developper <lamtigu...@gmail.com> wrote:

Hello mamado

i have two virtual machines (servers) One in which installed Asterisk on centOS / and one in which webrtc2sip is well configured on ubuntu 12.04
because i know the version of asterisk that i m using does not support webrtc Sockets ,that  why i decided to use webtc2sip

So my question is how to connect this 2 servers , it is in sipml5 (expert mode) web page , or there is some modifications and adds that must be done in config fils of both asterisk and Webrtc2sip

Sorry if this tread is maybe already discussed but i haven't found the solution yet

Regards

--
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.
 
 

[Mamadou DIOP]

I guess you're very very newbie.

The sipml5 is hosted on the internet but executed by your browser running on your pc. WebSocket, WebRTC… stacks are inside your browser and have access to the LAN if your PC have it.

On Apr 30, 2013, at 5:25 PM, the developper <lamtigu...@gmail.com> wrote:

I m using sipml5 through internet to make call to my asterisk server  . How can i put a local ip address in "WebSocket Server URL" ????
Besides Asterisk server and webrtc2sip are in the same LAN .

Regards

[Mamadou DIOP]

As I always say: "not working" is not bug report.

On May 1, 2013, at 2:32 PM, the developper <lamtigu...@gmail.com> wrote:

I agree mamado i m newbie

But i still work on my LAN  , regardless webrtc2sip  when i try to make a call from sipml5 (loalhost) to another extension of my server asterisk it does'nt work !!!!!

[the developper]

I m working on local but it seems that sipml5 can't register a local user !!!!

YOUR ARE USING DEBUG CODE. PLEASE USE CODE UNDER 'release' FOLDER or check https://code.google.com/p/sipml5/wiki/FAQ#How_to_reduce_the_size_of_the_scripts_before_deploying for more information on how to amalgamate the code. SIPml-api.js:19

User-Agent=Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.31 (KHTML, like Gecko) Chrome/26.0.1410.64 Safari/537.31 tsk_utils.js:110

WebSocket supported = yes tsk_utils.js:110

Navigator friendly name = chrome tsk_utils.js:110

OS friendly name = windows tsk_utils.js:110

Have WebRTC = yes tsk_utils.js:110

Have GUM = yes tsk_utils.js:110

Engine initialized tsk_utils.js:110

s_websocket_server_url=(null) tsk_utils.js:110

s_sip_outboundproxy_url=(null) tsk_utils.js:110

b_rtcweb_breaker_enabled=yes tsk_utils.js:110

b_click2call_enabled=no tsk_utils.js:110

SIP stack start: proxy='ns313841.ovh.net:13062', realm='<sip:192.168.1.12>', impi='2003', impu='"2003"<sip:20...@192.168.1.12>' tsk_utils.js:110

Connecting to 'wss://ns313841.ovh.net:13062' tsk_utils.js:110

==stack event = starting tsk_utils.js:110

__tsip_transport_ws_onopen tsk_utils.js:110

==stack event = started tsk_utils.js:110

State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister tsk_utils.js:110

SEND: REGISTER sip:192.168.1.12 SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKK9cktcqTq18TJWAeqBcadk2JhtfIb3ex;rport From: "2003"<sip:20...@192.168.1.12>;tag=QpEUC43R6imDClupgrz9 To: "2003"<sip:20...@192.168.1.12> Contact: "2003"<sips:20...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: bd2a1851-c370-3edf-ce70-3ba065f8e6b2 CSeq: 5837 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.04.26 Organization: Doubango Telecom Supported: path tsk_utils.js:110

==session event = connecting tsk_utils.js:110

==session event = sent_request tsk_utils.js:110

State machine: tsip_dialog_register_Any_2_Terminated_X_transportError tsk_utils.js:110

=== REGISTER Dialog terminated === tsk_utils.js:110

==session event = transport_error tsk_utils.js:110

==session event = terminated tsk_utils.js:110

The FSM is i

[Mamadou DIOP]

Don't know why you're asking if you don't read the responses:

On Tuesday, April 30, 2013 2:37:47 PM UTC, Mamadou wrote:

1. if you have your own gateway: fill WebSocket? Server URL field with the IP address and port where your webrtc2sip is listening for incoming connections (e.g ws://192.168.0.1:10060 or wss://192.168.0.1:10062). IMPORTANT:Do not forget the url scheme (ws:// or wss://)

1. set the SIP outbound proxy url with your asterisk binding address (e.g udp://192.168.0.1:5060)

You can clearly see that there is a step for step process.

[the developper]

i set the asterisk binding address in SIP outbound proxy SIP but in vain
when i try to register with a local SIP user it does'not work !!!!

s_websocket_server_url=(null) tsk_utils.js:110

s_sip_outboundproxy_url=udp://192.168.1.12:5060 tsk_utils.js:110

b_rtcweb_breaker_enabled=yes tsk_utils.js:110

b_click2call_enabled=no tsk_utils.js:110

SIP stack start: proxy='ns313841.ovh.net:13062', realm='<sip:192.168.1.12>', impi='2003', impu='"2003"<sip:20...@192.168.1.12>' tsk_utils.js:110

Connecting to 'wss://ns313841.ovh.net:13062' tsk_utils.js:110

==stack event = starting tsk_utils.js:110

__tsip_transport_ws_onopen tsk_utils.js:110

==stack event = started tsk_utils.js:110

State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister tsk_utils.js:110

SEND: REGISTER sip:192.168.1.12 SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS1ppsINHJThNUFQKEb595iuTs0w7qGaV;rport From: "2003"<sip:20...@192.168.1.12>;tag=CTgVmBrEFHSNChoBqIRv To: "2003"<sip:20...@192.168.1.12> Contact: "2003"<sips:20...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 71490ac1-a9e1-4e66-5584-f4fd1686c46e CSeq: 63626 REGISTER Content-Length: 0 Route: <sip:192.168.1.12:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.04.26 Organization: Doubango Telecom Supported: path tsk_utils.js:110

==session event = connecting tsk_utils.js:110

==session event = sent_request tsk_utils.js:110

State machine: tsip_dialog_register_Any_2_Terminated_X_transportError tsk_utils.js:110

=== REGISTER Dialog terminated === tsk_utils.js:110

==session event = transport_error tsk_utils.js:110

==session event = terminated v

SEND: REGISTER sip:192.168.1.12 SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKK9cktcqTq18TJWAeqBcadk2JhtfIb3ex;rport From: "2003"<sip:20...@192.168.1.12>;tag=QpEUC43R6imDClupgrz9 To: "2003"<sip:20...@192.168.1.12> Contact: "2003"<sips:2003@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: bd2a1851-c370-3edf-ce70-3ba065f8e6b2 CSeq: 5837 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.04.26 Organization: Doubango Telecom Supported: path tsk_utils.js:110

[Mamadou DIOP]

I'm also trying to explain something very basic in vain :(

I've said hundred times you have to set the websocket server url (even in my previous mail):

1. if you have your own gateway: fill WebSocket? Server URL field with the IP address and port where your webrtc2sip is listening for incoming connections (e.g ws://192.168.0.1:10060 or wss://192.168.0.1:10062). IMPORTANT:Do not forget the url scheme (ws:// or wss://)

Sorry but I'm giving up

[the developper]

i m truely truely sorry mamado about my ignorance :(
i just thought that to make local calls between users withing the same LAN ( at least establishement of connection  without audio ) does not require the gateway !!  ,
that is why i assumed that  the issue is from another cause !!

On Wednesday, May 1, 2013 7:21:47 PM UTC, Mamadou wrote:

I'm also trying to explain something very basic in vain :(

I've said hundred times you have to set the websocket server url (even in my previous mail):

1. if you have your own gateway: fill WebSocket? Server URL field with the IP address and port where your webrtc2sip is listening for incoming connections (e.g ws://192.168.0.1:10060 or wss://192.168.0.1:10062). IMPORTANT:Do not forget the url scheme (ws:// or wss://)

Sorry but I'm giving up

On May 1, 2013, at 9:16 PM, the developper <lamtigu...@gmail.com> wrote:

i set the asterisk binding address in SIP outbound proxy SIP but in vain
when i try to register with a local SIP user it does'not work !!!!

s_websocket_server_url=(null) tsk_utils.js:110

s_sip_outboundproxy_url=udp://192.168.1.12:5060 tsk_utils.js:110

b_rtcweb_breaker_enabled=yes tsk_utils.js:110

b_click2call_enabled=no tsk_utils.js:110

SIP stack start: proxy='ns313841.ovh.net:13062', realm='<sip:192.168.1.12>', impi='2003', impu='"2003"<sip:20...@192.168.1.12>' tsk_utils.js:110

Connecting to 'wss://ns313841.ovh.net:13062' tsk_utils.js:110

==stack event = starting tsk_utils.js:110

__tsip_transport_ws_onopen tsk_utils.js:110

==stack event = started tsk_utils.js:110

State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister tsk_utils.js:110

SEND: REGISTER sip:192.168.1.12 SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS1ppsINHJThNUFQKEb595iuTs0w7qGaV;rport From: "2003"<sip:20...@192.168.1.12>;tag=CTgVmBrEFHSNChoBqIRv To: "2003"<sip:20...@192.168.1.12> Contact: "2003"<sips:2003@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 71490ac1-a9e1-4e66-5584-f4fd1686c46e CSeq: 63626 REGISTER Content-Length: 0 Route: <sip:192.168.1.12:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.04.26 Organization: Doubango Telecom Supported: path tsk_utils.js:110

[Mamadou DIOP]

The gateway is not only used for media but also to convert signalling transport from websocket to udp/tcp/tls.

websocket server url = address where to send SIP message over websocket (from browser to webrtc2sip)

sip outbound proxy = address where to send SIP messages received from websocket (from webrtc2sip to asterisk)

This is clearly explained at the bottom of the expert page view.

[the developper]

Thanks you :)

[the developper]

Hi mamadou

I followed all steps you ve pointed out BUT
when i try to login , id does'nt work at all ,just for information the wamp server ,webrtc2sip and asterisk are in he same lan but the test is done from another laptop

OS friendly name = windows tsk_utils.js:110

Have WebRTC = yes tsk_utils.js:110

Have GUM = yes tsk_utils.js:110

Engine initialized tsk_utils.js:110

s_websocket_server_url=ws://192.168.2.104:10060 tsk_utils.js:110

s_sip_outboundproxy_url=udp://192.168.2.101:5060 tsk_utils.js:110

b_rtcweb_breaker_enabled=yes tsk_utils.js:110

b_click2call_enabled=no tsk_utils.js:110

SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:ekiga.net>', impi='nabilpan04', impu='"nabilpan04"<sip:nabil...@ekiga.net>' tsk_utils.js:110

Connecting to 'ws://192.168.2.104:10060' tsk_utils.js:110

==stack event = starting tsk_utils.js:110

__tsip_transport_ws_onclose tsk_utils.js:110

==stack event = failed_to_start tsk_utils.js:110

[Mamadou DIOP]

I don't really know what you're trying to do but I don't think you'll go far.

- you started this thread saying you have a local Asterisk server that you want to use through webrtc2sip. I  this thread you said that webrtc2sip is working

- then, you opened another thread (May 2, 2013 11:18 AM, Title: Running webrtc2sip) saying that webrtc2sip is not working. 5 minutes later you removed the thread before anyone give you a response. The logs shows that you don't have access to internet.

- now, you show logs where you're trying to connect to ekiga.net server using your local Asterisk. The architecture is now: browser <->webrtc2sip<->asterisk<->ekiga.net unless ekiga.net is the domain name used by your Asterisk server.

This said, your logs shows that the webrtc2sip server is closing the connection or ws://192.168.2.104:10060 is not reachable.

Change the DEBUG level to INFO in config.xml (webrtc2sip), try again and share the logs.

PS: if you're working for a company I highly recommend hiring a developer to help you. At Doubango we don't provide such service but other developers here could help.

[the developper]

Before sharing the DEBUGs i want to explain clearly what i m trying to do :

The aim is to develop a web application based on SIPML5 : this application must allow to call Asterisk server ( a ring group or sth like that)  from any SIP account from the web page ( for instance ekiga.net it does'nt matter) it is kinda like clicktocall service !!
So i ve used wamp and i ve downloaded Sipml5 code source and i ve also configured webrtc2sip because i use elastix (which embed asterisk old version)

The problem is  (ASTERISK + WEBRTC2SIP) in a LAN , And the web application may be hosted somewhere else !!! (even though i m developing it in local for the moment)

PS: I m just a student and all this ,it is a part of my internship

[Din Assegaf]

Dear developers i think u need to spend more time trial and errors before asking question.

So many fragmentation like os. Version. And libs. So dig more time before asking question.

sent using android mobiles

[Mamadou DIOP]

I'd change my display name from "the developer" to "the student" or "a developer" :)

There is an open source click-to-call service that you can check: http://click2dial.org/

There are two parts on your project: client (javascript, html) and server (webrtc2sip, asterisk). You should first focus on the client part and use our servers to call public SIP networks (e.g. ekiga.net). Once you get more confident, use your own webrtc2sip then your own Asterisk.

[the student]

Hello
So my question is maybe dumb BUT : if i use sipml5 with an account (ekiga or sip2sip ...) from web regardless my lan . how can i connect this client to my server (webrtc2sip and asterisk ) ???

SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:ekiga.net>', impi='nabilpan04', impu='"nabilpan04"<sip:nabilpan...@ekiga.net>' tsk_utils.js:110

[the student]

I hope someone will give an answer for this simple question !!!

[Mamadou DIOP]

There are already 21 messages in this thread.  I explained several times how to do it (step by step).

You were on the right track when you started to do something with error logs. We asked for webrtc2sip logs then, you went back to "how can i connect this client to my server".

Let's start at day zero. Don't try to be superman by doing something you don't understand. I already told you to focus on the easiest part (the client) first.

To do: connect your ekiga.net accounts using *our* cloud.

how to do?: clear all fields in expert view and try using your settings and report any issue (always attach js console logs) or success.

If you manage to get this working we will move forward with your own servers. 

[the student]

Ok the connexion is established in your cloud with sip2sip.info account and all fieds are blank in the expert mode
I try to call another sip account but there is the ERROR "Media stream permission denied"
What next ??

[Mamadou DIOP]

As this problem doesn't interest other developers, please open a ticket on the tracker (https://code.google.com/p/webrtc2sip/issues/entry).

Already said in my previous mail to "always attach js console logs". Do it and continue the discussion using the newly opened thread. 

[the student]

Ok i m gonna open this ticket
anyways thanks a lot :)

[Deekshitha Sundaran]

Hello Mamado,

     We have a private LAN  network(Open IMS) with SIPML5 in one system with a web browser(Firefox nighly and/or chrome).
     We tested SIPML5 with the public network sip2sip.info successfully ,but when tried with open IMS network it was unable to connect giving the error"Disconnected:Failed to connect to the server"

  We are unable to figure out where the problem lies.
 
  Our configuration is:
 
  In Expert settings
   Enable RTC web braker - ticked
   Websocket server URL: ws://IP address of  system where SIPML5 runs :10060
   SIP outbound proxy URL: udp://IP  address of IMS server :4060(P-CSCF port)

In Registration webpage:
 
 Display name :bob(registered with IMS network)
 private identity:b...@open-ims.test
 public identity: sip:b...@open-ims.test
 password : bob
 Realm: sip:open-ims.test

When login is hit, we get the above mentioned failure message;

Please provide us with the solution to check the problem.

The funniest part here is it worked once(only once) by successfully registering with the IMS network(Connected)


 Thanks,
 
 

[Mamadou DIOP]

   Websocket server URL: ws://IP address of  system where SIPML5 runs :10060

Nope. The IP must be where the websocket server (e.g. webrtc2sip.org) is listening.

[Deekshitha Sundaran]

Hello momadou,
        On running webrtc2sip exe I got the following errors.Can u just help us in resolving this issue. We are not aware of the issues in it.

Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>
PRODUCT: webrtc2sip
HOME PAGE: http://webrtc2sip.org
LICENCE: GPLv3 or proprietary
VERSION: 2.5.1
'quit' to quit the application.
*******************************************************************

SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
***ERROR: function: "tnet_getaddrinfo()"
file: "src/tnet_utils.c"
line: "908"
MSG: getaddrinfo(family=2, node=google.com and service=5060) failed: [No address associated with hostname]
***ERROR: function: "tnet_sockaddrinfo_init()"
file: "src/tnet_utils.c"
line: "1279"
MSG: getaddrinfo have failed.
***ERROR: function: "tnet_sockaddrinfo_init()"
file: "src/tnet_utils.c"
line: "1279"
MSG: (SYSTEM)NETWORK ERROR ==>Connection timed out
***ERROR: function: "tsip_stack_start()"
file: "src/tsip.c"
line: "714"
MSG: Failed to get best source [-5].
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "173"
MSG: bind to [192.168.11.254:0]have failed
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "173"
MSG: (SYSTEM)NETWORK ERROR ==>Cannot assign requested address
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "200"
MSG: Invalid socket.
***ERROR: function: "tnet_socket_create_2()"
file: "src/tnet_socket.c"
line: "200"
MSG: (SYSTEM)NETWORK ERROR ==>Cannot assign requested address
***ERROR: function: "tnet_transport_create()"
file: "src/tnet_transport.c"
line: "155"
MSG: Failed to create master socket
***ERROR: function: "_tnet_transport_ssl_init()"
file: "src/tnet_transport.c"
line: "61"
MSG: Invalid parameter
***ERROR: function: "tnet_transport_create()"
file: "src/tnet_transport.c"
line: "160"
MSG: Failed to initialize TLS and/or DTLS caps
***ERROR: function: "tsk_runnable_set_priority()"
file: "src/tsk_runnable.c"
line: "180"
MSG: Invalid parameter
***ERROR: function: "tsip_stack_start()"
file: "src/tsip.c"
line: "755"
MSG: Failed to add new transport
***ERROR: function: "start()"
file: "mp_engine.cc"
line: "560"
MSG: Failed to start SIP stack

Thanks and Regards,
Deeks