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

[Michael L. Young]

----- Original Message -----
> From: "Joshua Colp" <jcolp at digium.com>
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Sent: Wednesday, November 14, 2012 8:43:04 AM
> Subject: Re: [asterisk-dev] 13 November, 2012 New SIP stack update
> 
> 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.

I agree with your view.  +1

Michael
(elguero)