[asterisk-dev] Potential change to outgoing codec offers (new topic)

[Brian Candler]

On Thu, Oct 19, 2006 at 09:31:28AM +0100, Brian Candler wrote:
> On Wed, Oct 18, 2006 at 12:25:06PM -0500, Kevin P. Fleming wrote:
> > It's pretty late in the game to make a change like this for Asterisk 1.4,
> > but I suspect given the number of people that experience this problem
> > every day it's likely the community would be happy if we did it.
> > 
> > Thoughts?
...
> For this new strategy I think you'll want wide test coverage across a range
> of phones and ATAs. Doing it in a beta will give you this, but may
> disappoint a lot of people if it turns out to be problematic.

Here's a suggestion. Release a tiny test program which just sends an INVITE
with a duff codec offer. Ask people to point it at whatever SIP UAs they
have (VoIP phones, soft phones, ATAs etc) and collect the results, perhaps
on a Wiki page.

Here's one I knocked together in Perl:
http://pobox.com/~b.candler/software/testsiperror

The fake codec it offers is "WIBBLE/8000".

And the results I get when pointing it at various SIP UAs:

  Audiocodes Tulip ATA (2.2.0_build_6)          403 Forbidden
  Ekiga 2.0.1                                   180 Ringing (!!)
  Asterisk trunk r45305 with Zaptel FXS         100 Trying, then rings (!!)

This suggests that there may not be much mileage in this approach :-(

But maybe it means my limited understanding of SIP and SDP has given an
invalid test, so I'd appreciate it if someone with a deeper understanding
than me could look over that program and see if I'm doing something wrong.

(However, I have no idea what these devices would do if you completed the
call, and then they had to use the 'WIBBLE' codec to send audio...)

Regards,

Brian.