[asterisk-dev] Regarding SIP performance

[Kevin P. Fleming]

----- Steve Langstaff <steve.langstaff at citel.com> wrote:
> This is just off the top of my head, so excuse me if I am talking
> rubbish, but perhaps the canreinvite checking could be done *before*
> the
> channels to the endpoints have been established, so that a call that
> will be 'reinvited' does not go though the RTP setup/cleardown
> mechanism
> on Asterisk.

As discussed many times on this list before, that is not possible. The incoming channel is already established before any outbound channels are created, and a single incoming channel can create multiple outgoing channels (with only only finally being connected, of course).

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