asterisk users - Oct 2006 - Psst... Top secret information: Codename Pineapple

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@edvina.net
* Asterisk Training http://edvina.net/training/
* Next class: Stockholm, Sweden November 13-17 2006

What are your T.38 plans with this?

On 10/11/06, Olle E Johansson <oej@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@edvina.net
> * Asterisk Training http://edvina.net/training/
> * Next class: Stockholm, Sweden November 13-17 2006
>
>
> _______________________________________________
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>
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On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson
wrote:
> The new channel will have configurations for "trunks",
"services" and
> "phones". It will
Does that mean that it will make a distinction concerning the
difference in administrative span of control between trunks, which go
to the outside world, and stations, which are part of "your PBX" (even
though they may *be* out in the world somewhere, anyway? 

That's a spot that my (admittedly loose) understanding of SIP has
always sort of glossed over...

Cheers,
-- jra
-- 
Jay R. Ashworth                                                jra@baylink.com
Designer                          Baylink                             RFC 2100
Ashworth & Associates        The Things I Think                       
'87 e24
St Petersburg FL USA      http://baylink.pitas.com             +1 727 647 1274

	"That's women for you; you divorce them, and 10 years later,
	  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_

12 okt 2006 kl. 15.51 skrev Jay R. Ashworth:
> On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote:
>> The new channel will have configurations for "trunks",
"services" and
>> "phones". It will
>
> Does that mean that it will make a distinction concerning the
> difference in administrative span of control between trunks, which go
> to the outside world, and stations, which are part of "your PBX"
(even
> though they may *be* out in the world somewhere, anyway?
Right. To explain a bit further:

* Phones = stations, regardless of where they are
* Trunks = trunks to other SIP servers, bilateral
* Services = services you register for, like BroadVoice, Voop or FWD.
    (where asterisk acts as a "phone")

Regards,
/Olle

>>> * Phones = stations, regardless of where they are
> Asterisk = SIP Server, Phone = SIP Client
>
> Is a Media Server a Phone (ie SIP Client) ?
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