[asterisk-users] RTP Media Path

[Vicky]

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> 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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