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This is correct, if no NAT is involved anywhere and reinvites are
allowed then Asterisk will stay out of the media path and be used only
as Signaling server. So as for your answer yes, it will be able to
handle more calls than expected because there is no CPU overhead of the
media path.<br>
<br>
It is common strategy to have a single signaling server and have RTP
servers all around the globe for latency and etc, media gateways.<br>
<br>
Vicky wrote:
<blockquote
cite="midff95c3730612030748x1568c315l8d579400fbd035a2@mail.gmail.com"
type="cite">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 .<br>
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 .. <br>
<br>
<div><span class="gmail_quote">On 03/12/06, <b
class="gmail_sendername">Dovid B</b> <<a
href="mailto:asteriskusers@dovid.net">asteriskusers@dovid.net</a>>
wrote:</span>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div 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> </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> </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> </div>
<div><font face="Arial" size="2"> Thanks a lot.</font></div>
<div> </div>
<div><font face="Arial" size="2">Dovid</font></div>
</div>
<br>
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</blockquote>
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