[sr-dev] Websocket module

Carlos Ruiz Díaz carlos.ruizdiaz at gmail.com
Tue Aug 7 21:46:27 CEST 2012


Ok, now I understand.

I changed parts of the code starting at line 298 in the file *call.htm* of
sipml5. The modified code looks something like this:

*//i_port = 4062 + (((ew Date().getTime()) % 5) * 1000);^M*
*i_port = 5060;*
*// s_proxy = "sipml5.org";^M*
*s_proxy = "127.0.0.1";*
*
*
Having done that, sipml5 started to try to authenticate against my Kamailio
and the next step is to make SIP calls which is where I'm heading right now.

Thanks a lot for your help.

Carlos.


On Tue, Aug 7, 2012 at 12:15 PM, Peter Dunkley <
peter.dunkley at crocodile-rcs.com> wrote:

> **
> Hi Carlos,
>
> You do need a SIP over WebSockets client (for example, an HTML5 SIP phone)
> to use the WebSocket transport.  sipml5, running on Google Chrome Beta (or
> Canary) with the PeerConnection API enabled, works just fine with the
> Kamailio WebSocket implementation without any additional software being
> needed.
>
> OverSIP is alternative SIP proxy that you could use instead of (or in
> addition to Kamailio), but you do not need this at all if you just want to
> connect a WebSocket client to Kamailio.
>
> You can make calls between WebSocket clients, or you can make calls
> between WebSocket clients and non-WebSocket clients using Kamailio (you
> just need to make sure that the non-WebSocket clients support the correct
> set of media options).
>
> If you look in the history of the sr-dev mailing list there is a thread
> between 07/07/2012 and 09/07/2012 about using the WebSocket module.
>
> Regards,
>
> Peter
>
>
> On Tue, 2012-08-07 at 11:25 -0400, Carlos Ruiz Díaz wrote:
>
> Hi Peter,
>
>
>
>  Could you please explain a little bit more about how to test the module
> without using a HTML5 SIP phone?
>
>
>
>  AFAIK, I need a HTML5 SIP client (on a browser that supports websocket)
> plus a media stack capable of transporting RTP packets, such webRTC. So far
> I couldn't be able to get this components to work at all.
>
>
>
>  Thanks for your help.
>
>
>
>  Carlos
>
>  On Tue, Aug 7, 2012 at 5:41 AM, Peter Dunkley <
> peter.dunkley at crocodile-rcs.com> wrote:
>
>  Hi,
>
> You don't need to use OverSIP to use the WebSocket module in Kamailio.
> The Kamailio implementation will allow you connect one or more WebSocket
> clients directly to Kamailio and make calls between them.  It can also be
> used to convert calls from the WebSocket transport to SCTP/TCP/UDP for
> routing to other proxies.
>
> Although Kamailio doesn't support the full set of outbound features needed
> for WebSockets (yet) it is possible to use the same NAT traversal
> techniques that are used for TCP clients that connect through a NAT.  These
> are pretty trivial to use/set-up and there is an example Kamailio
> configuration file in the WebSockets module directory that does this.
>
> Regards,
>
> Peter
>
>
>
>
> On Tue, 2012-08-07 at 09:30 +0200, Muhammad Shahzad wrote:
>
> For WS client, you can try SIPML5,
>
>
> http://code.google.com/p/sipml5/
>
>
> Just download source code to some web server's root and edit call.html to
> point to your web sockets server (Kamailio or OverSIP).
>
>
> You can install OverSIP as follows (below instructions are for Debian 6.x
> / Ubuntu 11.x)
>
>
> apt-get install build-essential ruby1.9.1-full libev-dev
> gem1.9.1 install oversip
> ln -s /var/lib/gems/1.9.1/gems/oversip-1.0.5/etc /etc/oversip
>
>
> And then finally edit /etc/oversip/oversip.conf for your needs. Your web
> sockets address and port should be same as what you have mentioned in
> sipml5/call.html page.
>
>
> You can start oversip as, (there is an init.d script in sources, but its
> not installed by gem1.9.1)
>
>
> oversip -P /var/run/oversip.pid
>
>
> The advantage of OverSIP is that it supports PATH and outbound support, so
> you can create chain of SIP proxies.
>
>
> Thank you.
>
>
>
> On Tue, Aug 7, 2012 at 12:40 AM, Carlos Ruiz Díaz <
> carlos.ruizdiaz at gmail.com> wrote:
>
> Hello list,
>
>
> I'm trying to test the websocket module on Kamailio but I currently lack
> of a working web SIP phone that makes use of the websocket transport
> protocol. I'm trying with OverSIP <https://github.com/versatica/OverSIP> but
> there's no documentation on how to try it (and I don't do Ruby).
>
>
> Is there another HTML5 SIP client that I can use or at least a page where
> I can find documentation about how to configure OverSIP?
>
>
> Regards.
>
>
> Carlos.
>
> _______________________________________________
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>
>
>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
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>
>
>
>     --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
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>
>
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> Technical Director
> Crocodile RCS Ltd
>
>
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