John Todd wrote:
You could do this with Asterisk via the existing "qualify=500" syntax or similar in sip.conf to keep a packet going between Asterisk and the SIP device every 45 seconds (or whatever you hacked the timer to use, if you don't like that value.) This keeps the mapping open just fine for any NAT device I've ever seen. It works fine with dynamic hosts, even behind NAT - I just triple-checked and it does do what I expected it to do.
I did not know that "qualify=" caused Asterisk to send a "keep-alive" packet, I thought it was only to set a timeout for the UA to respond when a call needed to be terminated there before moving to the next priority.. If it does what you say then I can definately use it.. Thanks..
My example line will send an "OPTIONS" request every 45 seconds. If the response time to the OPTIONS request is more than 500 milliseconds, the SIP host is tagged as "unavailable" and removed from the operational list.
[snip]
It will be nice when the RTP traffice can go point-to-point and not have to be routed through the proxy (Asterisk) when both UA's are behind NAT.. I still finf it amazing how after the downfall of H.323 and NAT the SIP developers made the exact same mistake.. :)
Later..
It's extremely difficult to get two devices talking to each other that are behind NAT. Almost impossible, actually, due to the nature of NAT. If you read the fine print on Skype, as an example, you'll discover that NAT'ed users can reach each other by using a third party "helper" user, without that third party's explicit knowledge of transit'ed connections.
This difficulty is not a SIP failing; this is an inherent problem with NAT. However, there are some clever ways around this, but the problem has been that there are too many half-baked NAT routers and SIP clients that have had their firmware concocted by cut-rate programming sweatshops, where there has been no understanding of the actual use of the protocols in the real world.
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