[asterisk-dev] Regarding SIP performance

[Kevin P. Fleming]

----- 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).

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.