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 <[EMAIL PROTECTED]> 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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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