[asterisk-users] SIP - no audio behind nat problem

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of A J Stiles
Sent: Friday, October 15, 2010 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP - no audio behind nat problem

On Friday 15 Oct 2010, Zarko Zivanovic wrote:
> Hello,
>
> We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
> natted network.
>
> We have the issue with calls to these SIP phones - no audio.
>
> It is probably the problem with port forwarding on router - but I am not
> sure how can I forward same sip ports (5004 to 5100) to two phones (nat
> addresses?)?

Simple answer, don't run SIP through NAT.  Have another Asterisk server on
the 
outside and run IAX2 through NAT instead.  Much cleaner  :)

If you ARE going to run SIP through a NAT, you're going to need to designate
a chunk of ports in rtp.conf and poke those in your firewall.  We did UDP
pokes for 10001-10004 to use 1 line.  When you do an Asterisk communication,
the "handshake" occurs on 5060; the actual call occurs on 2-4 RTP ports
usually in the 10000 range.