[asterisk-dev] Regarding SIP performance

[Steve Langstaff]

> -----Original Message-----
> From: Kevin P. Fleming
> Sent: 16 October 2006 21:12
> 
> ----- Steve Langstaff <steve.langstaff at citel.com> wrote:
> > What you might have to do, if the called phone answers the 
> INVITE with 
> > a codec that the calling phone does not support, is CANCEL the 
> > original INVITE and then make a new INVITE using the IP 
> address of the 
> > server, and then transcode as appropriate.
> 
> It will already be too late at that point; certainly the 
> target endpoint will already be ringing, and may already have 
> answered (if the endpoint never sent us any 1xx responses 
> with SDP attached).

Yes, sorry, my mistake.