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@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@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 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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