ctxSip - Web SIP Client

[Blake]

I wrote an open source web sip client.  It is based on SIP.js and I have it working on my Kazoo installation.  It is designed to be a component in a larger web application.

I just posted it on GitHub and it is open source (MIT license).

Project:

https://github.com/collecttix/ctxSip

Demo:

http://collecttix.github.io/ctxSip/

Features:

• Audio only (SIP.js is video capable, but I don't need it.  It should be trivial to implement.)
• Hold / Resume, Mute, multiple call support.
  
• No plugins required, Works with WebSocket / WebRTC enabled browsers. (Firefox & Chrome, for now Safari 8 when it is released.)
  
• Intuitive interface
  
• Call log is saved to localStorage.
  
• Easy to configure and integrate into your project. (jQuery, Bootstrap)

I am am sure it is rough around the edges.  I would love some input.  Pull requests welcome.

[Te Matau]

You are my hero!

[Dayton Turner]

What are you using for the websockets url for Kazoo installations?

[Blake]

I use 8433 (ssl) on the kamailo server. This is necessary for SSL sites to prevent browser issues when accessing non-SSL content. The default unsecured port is 8080.  

Both are set in:

/etc/kazoo/kamailio/local.cfg

[Dayton Turner]

I may just need to update my kamailio - this is presumably set up using the websockets module?  I havent updated my kamailio in some time.  I'll do that first and look again.

Thanks!

Blake

September 15, 2014 at 2:39 PM

I use 8433 (ssl) on the kamailo server. This is necessary for SSL sites to prevent browser issues when accessing non-SSL content. The default unsecured port is 8080.  

Both are set in:

On Monday, September 15, 2014 3:38:57 PM UTC-5, Dayton Turner wrote:

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Dayton Turner

September 15, 2014 at 1:38 PM

What are you using for the websockets url for Kazoo installations?

On Tuesday, 9 September 2014 14:13:23 UTC-7, Blake wrote:

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[Yumminova Telecom]

Perhaps its nice if u write some doc on which steps to follow to get it to work?

Looks great though!


Op maandag 17 november 2014 01:22:37 UTC+1 schreef G. Arrington:

How do you configure the websocket URL?

[Yumminova Telecom]

Demo gives a 404


Op dinsdag 9 september 2014 23:13:23 UTC+2 schreef Blake:

[Sam Metson]

I was the person who ported it to Kazoo UI. The config file in phone/scripts doesn't do anything (and I probably should have gotten rid of it completely). You can specify the websocket URL in the Kazoo UI config file. with the ws_server key, e.g. winkstart,config={ ws_server: 'ws://ip_of_kamailio:8080' }.

Cheers,

Sam

[Blake]

It is merged in kazoo-ui? Awesome! Thanks Sam!

[Eamon]

Hi

I have enabled websocket in kamailio #!substdef "!MY_WEBSOCKET_DOMAIN!X.X.X.X!g"
added the WSS in the UI , updated config.js in /phone/scripts with device details and WSS
i still get websocket disconnect when i load the app in the UI

[Sam Metson]

To be clear, the config file in phone/scripts does nothing, and shouldn't be used.

As to your issue, I seem to remember having some difficulties with Kamailio. We're running 

You need to add the line #!trydef WEBSOCKETS-ROLE to the enabled roles section in kamailio's local.cfg.

The line you posted causes Kamailio to check the Origin header of the websocket request (and reject non-matching requests), so it might be best to try with that line commented out, at least for testing.

The lines you're probably looking for are the ones below (they may already be present in the default config):

#!substdef "!TCP_WS!tcp:MY_IP_ADDRESS:8080!g"

#!substdef "!UDP_WS_SIP!udp:MY_IP_ADDRESS:8080!g"

#!substdef "!TLS_WSS!tls:MY_IP_ADDRESS:8443!g"

#!substdef "!UDP_WSS_SIP!udp:MY_IP_ADDRESS:8443!g"

Cheers,

Sam

[Eamon]

Hi Sam

I have checked config
#!trydef WEBSOCKETS-ROLE is enabled in the local config

#!substdef "!TCP_WS!tcp:MY_IP_ADDRESS:8080!g"

#!substdef "!UDP_WS_SIP!udp:MY_IP_ADDRESS:8080!g"

#!substdef "!TLS_WSS!tls:MY_IP_ADDRESS:8443!g"

#!substdef "!UDP_WSS_SIP!udp:MY_IP_ADDRESS:8443!g"

are there by default
i still get the below message when i open browser phone

Websocket Disconnected.
An Error occurred connecting to the websocket.

[Sam Metson]

You could try using insecure websockets instead of secure (ws://... instead of wss://...) in Kazoo UI's config.js.

[Eamon]

Hi Sam

switched to port 8080, getting

Registration Error.An Error occurred registering your phone. Check your settings.

log from Kamailio

Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|start|recieved UDP request REGISTER sip:sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|source 95.83.253.115:47242
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|from sip:user...@sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|to sip:user...@sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|originated from external sources
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|this is a websocket request
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|correcting NATed contact in registration
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|end|issued new auth challenge to new registration attempt
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|start|recieved UDP request REGISTER sip:sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|source 95.83.253.115:47242
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|from sip:user...@sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|to sip:user...@sip.kazoo.com
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|originated from external sources
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|this is a websocket request
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|log|correcting NATed contact in registration
Nov 19 20:08:31 kazoo1 kamailio[1485]: INFO: db_kazoo [dbase.c:664]: db_kazoo_query(): query table=subscriber
Nov 19 20:08:31 kazoo1 kamailio[1485]: ERROR: db_kazoo [dbase.c:259]: dbk_creds_build_result(): Json-c error - failed to extract password [{"Defer-Response":"true","Server-ID":"","Node":"whistl...@kazoo1.sip.com","Msg-ID":"3e5228403e14cd17-1485@ip_address","App-Version":"0.4.2","App-Name":"registrar","Event-Name":"authn_err","Event-Category":"directory"}]
Nov 19 20:08:31 kazoo kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|end|failed registration attempt from 95.83.253.115:47242 for user...@sip.kazoo.com

On Wednesday, November 19, 2014 5:56:18 PM UTC, G. Arrington wrote:

Sam, do you have any thoughts on my browserphone inbound call issue?  Does your setup work with inbound?

[Luis Azedo]

i know this may seem a bit cryptic but...

Nov 19 20:08:31 kazoo1 kamailio[1485]: ERROR: db_kazoo [dbase.c:259]: dbk_creds_build_result(): Json-c error - failed to extract password [{"Defer-Response":"true","Server-ID":"","Node":"whistl...@kazoo1.sip.com","Msg-ID":"3e5228403e14cd17-1485@ip_address","App-Version":"0.4.2","App-Name":"registrar","Event-Name":"authn_err","Event-Category":"directory"}]
Nov 19 20:08:31 kazoo kamailio[1485]: INFO: <script>: 7cd2fo1c8qs92ov2vgce96|end|failed registration attempt from 95.83.253.115:47242 for user...@sip.kazoo.com

your password is wrong or there's something wrong on kazoo side, check amqp bus messages, check registrar logs, check ecallmgr logs.

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[Eamon]

I works under user account but not as admin

[Luis Azedo]

does admin have a sip user (device) and you are trying to login with that sip user (device) ?

On Wed, Nov 19, 2014 at 9:54 PM, Eamon <eamon....@gmail.com> wrote:

I works under user account but not as admin

[Eamon]

Hi Luis
yes it does, when i added the app to the UI, a device admin browserphone was created as well.

[Te Matau]

We have seen the following behaviour with browserphone on incoming calls:

1. If browserphone is registered using secure websockets (WSS) then browserphone does not ring. This is probably for the same reason that causes incoming calls to SIP devices registered over TLS to fail. 

2. If browserphone is not using secure websockets then the call can be answered and proceeds for 30 second before being disconnected. This seems to be due to Kamailio sending the ACK message to browserphone over UDP so browserphone doesn't respond appropriately to the ACK. The same problem exists when the far end disconnects the call; the BYE is sent by Kamailio over UDP so browserphone doesn't know the call is disconnected

We're running 3.12 so this may be fixed in a later version.

[Blake]

Is this using Firefox? I Just updated the repo to use SIP.js 0.6.4 which may address those issues.  Try it out and let me know...

[Blake]

I think I found the problem.  Registrations over wss had "transport=wss" in the registration.  RCF 7118 says it should be "ws".  I submitted a pull request to kazoo-configs with a fix.  There is probably a more elegant solution, but this seems to fix it.

https://github.com/2600hz/kazoo-configs/pull/38/files

Refs:

http://tools.ietf.org/html/rfc7118#section-5.2

[Darren Schreiber]

I believe using WS:// is deprecated in Chrome and maybe others and you are required to use WSS. 

From: Blake <blakem...@gmail.com>
Reply-To: "2600h...@googlegroups.com" <2600h...@googlegroups.com>
Date: Monday, March 23, 2015 at 6:59 PM
To: "2600h...@googlegroups.com" <2600h...@googlegroups.com>
Subject: Re: ctxSip - Web SIP Client

I think I found the problem.  Phones registered over wss had "transport=wss" on the registration.  RFC 7118 says it should be "ws".  I created a pull request to kazoo-configs with a fix.

https://github.com/2600hz/kazoo-configs/pull/38/files

I am way outside my lane here, can anyone test it out and see if it breaks anything?

[Blake McSwain]

I think your right on that, and that is the problem being addressed.

There is a problem with websocket registrations made over WSS,  Kamailio had wss as the transport on the contact.  Registrations worked fine.  Outbound worked, but inbound Invites were coming in via UDP instead of over the websocket.  

My small patch changes the registration to show as "ws" on the contact transport, even if it was made over wss.  Fixed the problem in my case.

I am admittedly outside my lane with kamailio, but I am learning.  Is this the right approach?

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[Darren Schreiber]

Weird. Ok we will try to find some time to look

[Blake McSwain]

It is.  I have been looking for that one for a while.  

Thanks for checking on it.

[Blake]

Pull request was merged is working for me.  Big thanks to lazedo and k-anderson.

Fixed in : kazoo-configs-3.19-16.el6.noarch.rpm 

[Vitalicus]

21.04.2015 .... Script is not working. Demo error 404.

[SH]

Hi

I wan to use this web based sip phone but need some changes

so do you offer any paid customization support?


On Wednesday, 10 September 2014 02:43:23 UTC+5:30, Blake wrote:

I wrote an open source web sip client.  It is based on SIP.js and I have it working on my Kazoo installation.  It is designed to be a component in a larger web application.

I just posted it on GitHub and it is open source (MIT license).

Project:

https://github.com/collecttix/ctxSip

Demo:

http://collecttix.github.io/ctxSip/

Features:

• Audio only (SIP.js is video capable, but I don't need it.  It should be trivial to implement.)
• Hold / Resume, Mute, multiple call support.
  
• No plugins required, Works with WebSocket / WebRTC enabled browsers. (Firefox & Chrome, for now Safari 8 when it is released.)
  
• Intuitive interface
  
• Call log is saved to localStorage.
  
• Easy to configure and integrate into your project. (jQuery, Bootstrap)

I am am sure it is rough around the edges.  I would love some input.  Pull requests welcome.

[Darren Schreiber]

Please email people directly for support of this nature, or contact 2600hz’s support team

From: <2600h...@googlegroups.com> on behalf of SH <sumith...@gmail.com>
Reply-To: "2600h...@googlegroups.com" <2600h...@googlegroups.com>
Date: Thursday, August 13, 2015 at 2:14 PM
To: 2600hz-dev <2600h...@googlegroups.com>
Subject: Re: ctxSip - Web SIP Client

--

[sumit hembrom]

Hi

thanks for your kind response

is there any mail address to direct contact to main developer of this webphone?

Thanks

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[Darren Schreiber]

On Google Groups it should let you click the thread and show the email address I believe.

From: <2600h...@googlegroups.com> on behalf of sumit hembrom <sumith...@gmail.com>
Reply-To: "2600h...@googlegroups.com" <2600h...@googlegroups.com>
Date: Thursday, August 13, 2015 at 2:19 PM
To: "2600h...@googlegroups.com" <2600h...@googlegroups.com>
Subject: Re: ctxSip - Web SIP Client

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