asterisk users - Dec 2006 - RTP Media Path

I know this has been asked before and I went over the wiki but I have not been
able to come to a clear answer.

1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA
-----> Asterisk ----> SIP Provider) from what I understand if NO NAT is
being used then asterisk just starts and stops the session however the RTP media
stream will be passed directly from the SIP provider and vice versa. (This is of
course if there is no NAT involved). Now say I had such a set up will the server
be able to handle more calls than "average" if the only responsibility
if the server is to authenticated and pass along the calls ? (There will be an
AGI running in the begining to determine what route to used based on how many
minutes each route has used). Now if the ATA's are behind VOIP and asterisk
is on a public IP then does asterisk have to sit in the media path ? Also can
some one explain exaclty when the RTP session is started and stopped.

Also another set up we are woroking on is SIP Provider (Incoming DID)  ---->
Asterisk (for authentication based on PIN) -----> Back to SIP Provider. The
asterisk server will be on a public IP. Can I have asterisk stay out fo the
media path (here I asume yes. Just wana be 100% sure).

 Thanks a lot.

Dovid
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Asterisk wont sit in media path if both callee and caller agrees on common
codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan (
please correct me if i am wrong ) , no call recording is enabled .
I think asterisk does native bridging even  if one is behind nat  ( i tested
with atleast one party behind nat not sure if it works when both are behind
nat ) and devices should support reinvites ..

On 03/12/06, Dovid B <asteriskusers@dovid.net>
wrote:
>
> I know this has been asked before and I went over the wiki but I have not
> been able to come to a clear answer.
>
> 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa
(ATA
> -----> Asterisk ----> SIP Provider) from what I understand if NO NAT
is
> being used then asterisk just starts and stops the session however the RTP
> media stream will be passed directly from the SIP provider and vice versa.
> (This is of course if there is no NAT involved). Now say I had such a set
up
> will the server be able to handle more calls than "average" if
the only
> responsibility if the server is to authenticated and pass along the calls ?
> (There will be an AGI running in the begining to determine what route to
> used based on how many minutes each route has used). Now if the ATA's
are
> behind VOIP and asterisk is on a public IP then does asterisk have to sit
in
> the media path ? Also can some one explain exaclty when the RTP session is
> started and stopped.
>
> Also another set up we are woroking on is SIP Provider (Incoming DID)
>  ----> Asterisk (for authentication based on PIN) -----> Back to SIP
> Provider. The asterisk server will be on a public IP. Can I have asterisk
> stay out fo the media path (here I asume yes. Just wana be 100% sure).
>
>  Thanks a lot.
>
> Dovid
>
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