7 Aug
2012
7 Aug
'12
6:15 p.m.
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(a)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:
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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(a)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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--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
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