<p>Thanks,</p>
<p>But if i open rtp ports from 10000-20000 how would you ping ports from both sides to not loose rtp or having one way audio if the ports are choosen randomly between 10.000-20.000 in every call?</p>
<p>The keep alive works for signalling (Asterisks sends Options to the contact), but not for RTP. For RTP i think it is mandatory to have an STUN server ir RTP proxy. Right?</p>
<div class="gmail_quote">El 27/04/2012 12:15, <<a href="mailto:isrlgb@gmail.com">isrlgb@gmail.com</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
The asterisk side has to have the router ports 5060 and 10000-20000 forwarded to asterisk these are the standard ports but you could cut way down on the rtp ports in rtp.conf then you have to tell asterisk what's the external ip of your nat and most of the times this should work today no problem lots of us here have it working that way (of course you have to take care of security fail2ban etc )<br>
On the phone side you might have to use stun but it depends on the firewall also you should set the phone to send a nat keep alive each 30 seconds (asterisk also sends a options packet to keep the nat open but doesn't always work ok )<br>
<br>
-----Original Message-----<br>
From: Danny Dias <<a href="mailto:ing.diasdanny@gmail.com">ing.diasdanny@gmail.com</a>><br>
Sender: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
Date: Fri, 27 Apr 2012 10:22:38<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Subject: Re: [asterisk-users] Asterisk + Phones behind different Nat<br>
Firewalls<br>
<br>
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