[asterisk-dev] SIP Interoperability in Asterisk?

[Olle E Johansson]

I am very much aware of that document and thank you for reminding me  
of that pointer.

Asterisk has been participating in two SIPit tests, thanks to  
sponsorship by my company
Edvina, Voop in Norway and Digium. We will participate in a third  
during october
with Asterisk 1.4.

In addition to that, as an Open Source software, the community is  
performing thousands
of interoperability tests every day, every hour around the globe. If  
you visit the bug tracker,
you will see bug reports about Asterisk and all kinds of equipment -  
servers, phones, firewalls.

A big change is that when I started working with changing the SIP  
channel a while back,
most of the bug reports was errors in our support of SIP. Today, it's  
either additional
features that we do not support or problems with the other side's  
implementation.

The biggest problem now is to make the SIP channel more transaction  
and dialog
state aware, so that we can handle TCP connections in a correct way.  
This will also
make it easier to solve some of the rather advanced errors we see in  
the bug tracker.

For the next release of Asterisk, beyond 1.4, this is a priority in  
addition to the
work done by the IPv6 project and the Asterisk Video Task Force.

The will be a Video developer's meeting in France in november and I  
foresee
a Asterisk SIP developer's meeting later on to focus on the Asterisk SIP
implementation. As I refrain from participating in Astricon, due to  
unsolved
business issues, this topics won't be covered there.

Regards,
/Olle

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* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/