[Asterisk-Users] SIP Testing

[Mark Spencer]

We're on track for a release of Asterisk 0.4.0 soon.  I'd like to try to
see to it that we have squared away our SIP implementation by then, and
after that point, try to keep it in tip top shape.

In general, I find that SIP is extremely fragile, and every time I try to
fix one bug, I end up creating another somewhere.  What I need are
strategies for verifying that the SIP implementation is correct, either
via some sort of SIP test suite or even just a collection of users who
will sign off on things.

Anyway I'm soliciting for ideas from the list.  I'd be happy to get some
feedback.

Mark