Thank you, I've totally missed that.Olle wrote:
STUN is helpful, but as I understand it analyzes the situation and reports
the configuration of a NAT. It doesn't help you keeping the NAT session open,
as SER module nathelper or the FWD/Jasomi solution.
Check here http://www.voip-info.org/wiki-SER+module+nathelper
[snip]
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
There continues to be a great deal of confusion about how Asterisk works with NATs using SIP. It works quite well.
John, since you know the source. Could you write a five-line explanation of what NAT=yes *really* does? I've asked the question many times, without an answer. I've tried to read the source, but there's no comments there either and it wasn't that easy to figure out for me.
The hopefully-soon-to-be-approved ICE RFC's will make things even easier by testing even the RTP ports, but it will be some time before we see clients with that functionality built in.Pointers to ICE RFC's - or drafts?
/O
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