On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp <jc...@digium.com> wrote:
> ----- Original Message -----
>> On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro <jcas...@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
>
> --
> ___

Joshua

Can you copy and past into a wiki page for everyone's benefit?  Maybe
https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
like page would be good.

-- 
~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~

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