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]
Interesting that you bring this topic up - kram was working on this last night (with me testing implementations.)
There is now (thanks, Mark!) an addition in sip.conf called "nat=1" that can flag a sip user/peer/friend as being behind a NAT address translator. The good news is that the REGISTER and INVITE requests seem to work on the ATA-186 from which I was experimenting.
The bad news is that RTP still doesn't work, and so one-sided conversations occur (ATA->NAT->Asterisk->other party where the other party can hear me, but I cannot hear them.) This of course stems from the fact that NAT (in most cases) requires symmetric port usage to pass packets back inside the NAT to the right host. SIP call process information passes quite well between * and my ATA-186, so half of the problem is solved...
Anyone have any ideas? I _know_ it can work, because with the same exact ATA-186, I can connect from behind a NAT to the iconnecthere.com servers, so is this STUN or something else that they're using? I set up the Vocal stund on my * server, but that didn't seem to do the trick.
JT
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