[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Joshua Colp]

----- Original Message -----
> On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro <jcastro at instant.com.br>
> wrote:
> > I see no indication of how to do this in sip.conf, and when I start
> > Asterisk, it doesn't wait on port 80.
> >
> 
> Websocket support is being actively worked on.  HTTP support should
> be
> enabled in manager.conf and http.conf first.

Hola!

The above will get the HTTP server portion going, but here's some other items:

1. transport=ws must be added to the peer/friend/user in sip.conf
2. avpf=yes must be set for that peer/friend/user as well.

Depending on what you are testing with this can get you a little further.

If you are using Chrome things will not quite work, yet. While they have made considerable progress becoming compliant with the ICE specification (SDP is now almost proper) it seems as though their STUN implementation is still not there yet. Completely valid packets sent by the library we use just seem to be ignored.

Patience is a virtue really as things are still evolving.

As well I will be working on a wiki page that will describe this stuff in detail. I was holding off until things were a bit more "there" but as people are at least trying it shall appear soon.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org