[asterisk-dev] [Code Review] WebSocket SIP Support

[Joshua Colp]

----- Original Message -----
> Hi Joshua,
> > Greetings!
> >  
> > That symbol is not being exported. I have a fix but unfortunately
> > can not commit it until a network issue is resolved. I would also
> > like to inform you that currently Google Chrome is not using an
> > RFC compliant ICE implementation yet so you will not be able to
> > achieve media flow using it. The amount of work required to change
> > the ICE implementation we use to be compatible with their
> > implementation is non-trivial, unfortunately.
> >  
> Can you elaborate a bit on this? Is there a bug report somewhere
> describing the problem in detail?

Hola!

The WebRTC issue is viewable at http://code.google.com/p/webrtc/issues/detail?id=345 but it has mostly consisted of state transitions and tag changes.

A list of the various differences that one individual came across in looking at it is available at https://groups.google.com/forum/?fromgroups#!topic/discuss-webrtc/-o2pV5whxsA

Further posts about the subject have all yielded that they are working on it and to just keep checking the issue I gave.

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