[Asterisk-Users] How do you handle NAT?

[OMS]

Does MyStun take care of RTP issues?

> The STUN server was extremely easy to set up.  Just check ou the MyStun
> sources (you have to use CVS), compile and run the server executable.
That's
> about all there is to it.

People have been posting  this question before, but did not get any clear
cut response, or probably there is'nt one?
How Vonage and Road Runner is doing it then?

I am able to get my inside Sonicwall SIP phones registered to the Asterisk
(on public IP).

The scenario is
SIP ATA (NAT) <---> Asterisk <----> AS5400 <---> PSTN

ATA to PSTN calls are working great.
But when I call from PSTN to SIP ATA I am getting ....

a) no audio when I have canreinvte=no in SIP.conf
b) one way audio from SIP to PSTN only if I have canreinvte=yes

Again do you think  MyStun will take care of RTP issues in this case?  Is ok
to install it on same machine.

I know this can be resolve using SIP enabled firewall or VPN, but it is not
practical for us.

Obaid.

----- Original Message ----- 
From: "Ray Van Dolson" <rayvd at digitalpath.net>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Tuesday, June 28, 2005 4:23 PM
Subject: Re: [Asterisk-Users] How do you handle NAT?


> The STUN server was extremely easy to set up.  Just check ou the MyStun
> sources (you have to use CVS), compile and run the server executable.
That's
> about all there is to it.
>
> Ray
>
> On Tue, Jun 28, 2005 at 11:07:48AM -0700, hank wrote:
> > how easy is it to set up a stun server? with asterisk amd will this fix
> > part of the nat problem?
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