There are basically two ways of doing SIP-through-NAT. The first is to configure the firewall to forward ports 5060 and 10000-10100 or whatever the IP phone uses for SIP and RTP. Then configure the IP phone and set its NAT IP address.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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The second way is to use a RTP Proxy. The way vonage handles SIP-through-NAT is to have their SIP Proxy modify the sdp packets for the INVITE, 183, and 200 OK messages and put the RTP Proxy's IP address and ports into the sdp portion of the message. That way each endpoint sends RTP packets to the RTP Proxy. The RTP Proxy waits for the first packet from each endpoint, then it knows which port to send the RTP packets to.
There is an open source project siproxd at http://sf.net/projects/siproxd which has a basic implementation of RTP Proxy.
Bill
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