[asterisk-users] RTP Media Path

[Tomer Horn]

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.

It is common strategy to have a single signaling server and have RTP 
servers all around the globe for latency and etc, media gateways.

Vicky wrote:
> 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 at dovid.net 
> <mailto:asteriskusers at 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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