[Asterisk-Users] Known SIP - NAT Solutions?

[John Todd]

>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.

Asterisk currently sends the RTP packets to the right address (check 
it out with a packet sniffer) but the NAT box doesn't have a mapping 
set up on that return port, so the NAT drops them on the floor.

I know a solution exists here.  My ATA-186 works behind my NAT when I 
have it configured for iconnecthere.com, and they don't have magic 
UDP elves, so it must be able to work for other SIP servers if the 
right trickery can be implemented.  I just don't know yet what that 
trickery is.  :)

JT


>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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