[Asterisk-Users] Known SIP - NAT Solutions?

[T Aksoy]

Finally someone has hit the same problems that we have. Everyone on this
newsgroup seems to have static IPs!

The problems you get can manifest in 2 ways:

1) you cannot get through to the phone at all

2) one-way audio - you can hear the other end but they can't hear you.

The problem is a combination of things:

1) router port forwarding - you have to set udp port 5060 (default sip
signalling port) to be forwarded to the sip phone. This will enable the
initial port can take place i.e. to make the phone ring etc.

2) the router also has to allow symmetrical nat (I think that's what they
call it) so that when your phone opens the relevant rtp port the other end
can talk to your phone along the same temporarily open port connection.

3) asterisk has to support STUN (or something similar). This will enable the
mapping of a phone's internal private address to the router's external
address, so that asterisk knows where to actually send the packets to. At
present it isn't supported.

As an example, the snom phones work from behind nat because they have a stun
client which talks to the snomag.de stun server. So as long as port
forwarding it correctly configured then snom (behind nat) to snom (behind
nat) works. When asterisk gets in the way then it doesn't.

Does anyone know if stun will be implemented within asterisk? We're quite
desperate for this functionality.

Thanks
Tan


----- Original Message -----
From: "Matthew Farley" <asterisk at wheatstate.net>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, March 05, 2003 9:08 PM
Subject: [Asterisk-Users] Known SIP - NAT Solutions?


     I have recently begun experimenting with Asterisk, and have been
mightily impressed by its capabilities and flexibility. I have run
across one problem, however, that challenges my ability to use it as a
production system.

     My Asterisk box has a public Internet IP, and works great with SIP
(ATA 186) clients that also have public IP addresses. Unfortunately,
most of the locations that I would like to put these SIP phones into are
behind NAT. Calls placed from behind NAT are consistantly unsuccessful.
I have read in several places that there are software solutions to this
problem, though I have found no specific references to precisely what
software to use, or how it should be configured to hand these calls off
to Asterisk.

     Has anyone on the list successfully overcome this limitation? If
so, any advice you might be able to provide would be greatly
appreciated.

Thanks!

Sincerely,
Matthew Farley
asterisk at wheatstate.net

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