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