[asterisk-dev] Regarding SIP performance

Mon Oct 16 04:52:47 MST 2006

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?

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

-- 
> Morten Isaksen
> http://www.misak.dk/blog/
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