Psst. Top Secret... 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 setting 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!