[asterisk-dev] 13 November, 2012 New SIP stack update

[Joshua Colp]

Saúl Ibarra Corretgé wrote:
>
> Hi Mark,
>
> Is Sofia out of the picture then? Seems so from the outside.

 From my perspective I consider sofia to be a non-starter as the former 
upstream for it is now abandoned. Freeswitch has forked it and has 
continued to fix issues, but it's not actively being developed with new 
features. Basing our brand new SIP channel driver on that seems like a 
bad decision for the future. We'd most likely end up back where we are - 
being the maintainers (even partially) of an entire SIP stack, including 
being responsible for implementing new RFCs etc. That's not something I 
want for us again. I'd rather we try to put as much of our time and 
effort into the higher level application part. That's not to say we 
won't do stuff with the SIP stack, but it shouldn't be a primary focus.

That's my view.

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