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Thanks Alex Balashov,<br><br> I am experiencing some one-way audio, that's the reason of the questions! <br><br>Att,<br>
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Flavio Roberto Miranda<br>
MSN:flaviormiranda@hotmail.com<br>Skype: flaviormiranda<br><br><div>> Date: Tue, 26 Jul 2011 09:23:42 -0400<br>> From: abalashov@evaristesys.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] NAT yes<br>> <br>> On 07/26/2011 09:19 AM, Flavio Miranda wrote:<br>> <br>> > In a no natted environment if I letnat=yes on sip.conf it would<br>> > cause some thing bad or it is irrelevant ? Anybody know ?<br>> <br>> There is no harm unless the endpoint you are dealing with does not do <br>> symmetric RTP. The nat=yes option assumes that it is okay to send RTP <br>> back to the source port from which it originated, irrespectively of <br>> what's in the SDP. This will cause one-way audio if the endpoint <br>> happens to want to receive RTP on a different port than the one it is <br>> sending it from.<br>> <br>> Almost all endpoints these days do symmetric RTP, though, so it's not <br>> a huge concern.<br>> <br>> That said, from a methodological and aesthetic perspective, it is <br>> better not to break standard RFC-compliant behaviour unnecessarily. <br>> Thus, I would not enable nat=yes unless there really is no direct <br>> network and transport-layer reachability to the endpoint.<br>> <br>> -- <br>> Alex Balashov - Principal<br>> Evariste Systems LLC<br>> 260 Peachtree Street NW<br>> Suite 2200<br>> Atlanta, GA 30303<br>> Tel: +1-678-954-0670<br>> Fax: +1-404-961-1892<br>> Web: http://www.evaristesys.com/<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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