[asterisk-dev] Regarding SIP performance

[Johansson Olle E]

13 okt 2006 kl. 18.09 skrev 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).
>
>
BUT, that's exactly what we're doing in Asterisk 1.4 - all IFs and  
BUTs regarded, if there's only two SIP endpoints in the
call, we will set up the call with RTP media directly between them  
without a RE-invite.

Thank you, Mark, for that patch!

/O