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

[Olle E. Johansson]

John Todd wrote:

>> 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 
Thank you, I've totally missed that.

> 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