[Asterisk-Users] SER vs STUND with Asterisk..

[WipeOut]

John Todd wrote:

> At 10:22 AM +0100 10/16/03, WipeOut wrote:
>
>> 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.

Just tried it out and mine is sending the OPTIONS request every 60 
seconds, but thats still fine its working like a dream.. My previous 
workaround for NAT closing the ports was to get the phone to re-register 
every 60-120 seconds depending on the router and how quickly it closed 
the ports.. This way will generate far less traffic.. Thanks for the 
tip.. :)

Later..