[asterisk-dev] Regarding SIP performance

[Brian Candler]

On Mon, Oct 16, 2006 at 10:29:52AM +0100, Brian Candler wrote:
> I note Asterisk offers extra codecs in the second leg, because it could
> transcode if necessary; presumably if the second handset chose a codec not
> supported by the first, Asterisk would need to re-INVITE to set up a two leg
> call. I've not tested this though.

I've tested it now, and in my SVN build, which is about 10 days out of date,
it appears to be broken.

Full description posted at http://bugs.digium.com/view.php?id=8152

In summary: the two endpoints think they can both use different codecs at
the same time, with the RTP stream running directly between them :-(

Regards,

Brian.