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

[John Todd]

>WipeOut wrote:
>
>>One for the gurus..
>Obviously not for me, but I'll dare to give it a shot anyway ;-)
>
>>Anyway, I decided to go and have a quick read through the SER docs 
>>and in the section about NAT they say that the best way to address 
>>NAT is to use STUN or uPNP..
>
>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 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.

STUN is useful and works well for those clients that support it, but 
should not be a part of Asterisk at this time.  The NAT trick that 
Ciscos (and others) use to determine outside NAT address in the Via: 
header is almost always sufficient, and is already part of Asterisk's 
handling of registering agents.  All that is missing is the ability 
for the Asterisk server to implement one or both methods of NAT 
traversal for outbound REGISTER requests, and then (in an optional 
and slightly different functionality mode) to proxy all SIP requests 
outbound through some particular host for those sites that may choose 
an external method of handling SIP NAT translations.

For anyone who wants further information as to Asterisk's use behind 
a NAT or firewall, pleaase check these two bugnotes:

NAT trick: http://bugs.digium.com/bug_view_page.php?bug_id=0000104
Proxy:     http://bugs.digium.com/bug_view_page.php?bug_id=0000359


There continues to be a great deal of confusion about how Asterisk 
works with NATs using SIP.  It works quite well.  Your SIP client 
needs to have some method of understanding that it's behind a NAT, 
but pretty much any modern client written by someone with half a clue 
will do that.  STUN or the Via: header trick have worked in every 
situation that I've come across.  There are still some problems with 
NAT, but they are for the most part overblown - most of the problem 
lies in the confusing explanations and half-understood problems by 
SIP system administrators.  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.

JT