<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2963" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> Thanks a lot.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Dovid</FONT></DIV></BODY></HTML>