[Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate

[Anton Krall]

I tried doing a sample test with a softphone behind nat and trying to
connect to my asterisk who has ports forwarded, so far, it can connect but
as usual, I can hear the prompt but for example, using the echo test, I
don't hear myself back.

By doing a sip show peers I see the softphone connected but instead of
showing using port 5060, it shows using port 64112 for example.

I have nat=yes and canreinvite=no ... Any ideas? I was thinking about stun
and ser but what do you guys think? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
aka ManxPower
Sent: Martes, 29 de Marzo de 2005 10:28 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate

Anton Krall wrote:
> Any problems with RTP or voice just on one side?
> 
> So as long as you use some STUN server, the RTP packets have the right IP.
> Did you install your own stund or are you using a public one?
> 
> You didn't have to use SER at all right?

Setting nat=yes does pretty much the same as a STUN server.
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