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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED]
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