[asterisk-dev] Regarding SIP performance

[Brian Candler]

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.

Regards,

Brian.