[asterisk-dev] Regarding SIP performance

[Brian Candler]

On Mon, Oct 16, 2006 at 04:36:04PM +0200, Johansson Olle E wrote:
> >On Mon, Oct 16, 2006 at 03:34:10PM +0200, Johansson Olle E wrote:
> >>Or, set up the call as normal - one call leg to Asterisk, transcode
> >>and another
> >>call leg and media stream to the other device.
> >>
> >>That should be the case here.
> >
> >But the point is - how do you know when you need to do that?
> >
> >At the moment (SVN), Asterisk is passing on an INVITE with SDP  
> >pointing to
> >the first endpoint. That works sometimes, but not other times,  
> >depending on
> >which codecs the second endpoint supports. But you have to know  
> >whether to
> >try this before you send on the INVITE.
> 
> The same rules apply as for the previous re-invites. So i guess we  
> either have
> a bug or you have a bad configuration that makes this happen.

There is no re-INVITE; this is the very first (and only) INVITE which sets
up the call.

I am talking about SVN, which behaves differently from 1.2 in this regard.

Please have a look at the SIP exchange which is attached as
sip-codec-mismatch.txt to http://bugs.digium.com/view.php?id=8152
and see if that makes the problem clearer.

Regards,

Brian.