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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