[asterisk-users] Psst... Top secret information: Codename Pineapple

[Andrew Joakimsen]

What are your T.38 plans with this?

On 10/11/06, Olle E Johansson <oej at edvina.net> wrote:
> Friends in the Asterisk community,
>
> I've been talking for years about the new version of the SIP channel.
> I've been trying to get funding
> and get going. Well, the funding part remains to be handled, but I
> have other news - if you kan keep
> it to yourself.
>
> ...I've began coding. Finally.
>
> With a happy smile on my face I removed "pedantic=yes" the other day.
> After years of disliking
> that option it's gone! And srvlookup now defaults to yes in the
> source code :-)
>
> So what is the chan_sip3 project (codename pineapple) about?
> ------------------------------------------------------------------------
> --------------
>
> The current SIP channel has many code relationships to the IAX2
> channel. Concepts like
> users, peers and friends doesn't really fit the SIP architecture. The
> channel supports locally
> connected phones very well, but is having severe problems being part
> of a larger SIP
> infrastructure. Forking, branching and such is not handled, as well
> as multiple
> transactions at the same time.
>
> The new channel will have configurations for "trunks", "services" and
> "phones". It will
> be more domain-focused to support multihosting better. It will have a
> proper SIP
> state machine so we can handle TCP and TLS alongside UDP. It will
> have STUN
> support, like the current Google talk channel. And a lot of other
> changes...
>
> Can I test this now?
> --------------------------
> Don't expect this work to be completed yesterday. Right now, I'm
> cleaning up stuff,
> moving around variables, splitting up the code in multiple files and
> grouping variables into
> structures. When all of that is done, the real work will start.
>
> I am expecting to have an experimental version ready for the release
> of Asterisk
> *after* the 1.4 release and a more production-ready version ready for
> the release
> a year from now. As always with Open Source, the final result depends
> a lot on the
> help from the community in testing, providing fixes, development
> time, funding
> and additions.
>
> Is it available for download?
> ---------------------------------------
> The code is hosted in the codename-pineapple branch in the svn server.
> In that branch, there's a chan_sip.c (version 1) and a chan_sip3.c.
>
> As I said: don't expect much yet and don't run this in production!
> Right now,
> downloading it is a good way of wasting the bytes on your hard disk
> drive
> and not much more.
>
> In Q1 2007 I will run an AstriSIPcon developer's meeting to be able
> to meet everyone
> that has interest in Asterisk and SIP to test, discuss and work with
> the new SIP channel.
>
> SIP greetings!
>
> /Olle
>
> PS. A big thank you to Voop AS, who keeps supporting my development
> work with Asterisk
> as well as all the students in my training classes that provide
> development funding
> by attending the classes. Thanks!
>
> ---
> * Olle E. Johansson - oej at edvina.net
> * Asterisk Training http://edvina.net/training/
> * Next class: Stockholm, Sweden November 13-17 2006
>
>
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