[asterisk-dev] Regarding SIP performance

Mon Oct 16 05:48:57 MST 2006

16 okt 2006 kl. 13.52 skrev Morten Isaksen:

>
>
> On 10/16/06, Johansson Olle E <olle at voop.com> wrote:
> 13 okt 2006 kl. 18.09 skrev Kevin P. Fleming:
>
> 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.
>
> Where can I find more information about this patch?
Hmmm. Not very well documented. Mark created it in Pisa earlier this  
year.

>
> Can it be disabled if you for some reason want to keep Asterisk in  
> the media path?

Canreinvite = no - like always.

/O