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

[Joshua Colp]

----- Original Message -----
> 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.

If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet.

--
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