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

[Chris Albertson]

--- John Todd <jtodd at loligo.com> wrote:
> >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.


"Overblown?"  I've get to see one example of Asterisk placing
and acepting SIP call through a NAT firewall.

Here s the problem:

I have an Asterisk server in back of a NAT
fire wall and I want users to be able to dial a SIP number at
fwd.pulver.com. by first dialing "97"

So in extension.conf
I have something like this:
exten => _97.,3,Dial(SIP/${EXTEN:2}@fwd.pulver.com,r)

What goes in sip.conf?

The problem I see is that the SER server at pulver.com complains
about the 192.168.x.x address it is getting from me.

I know you CAN get SIP calls through my firewal because X-Lite
can do it just fine.  But Asterisk can't, or I can't get it to.

I'm about ready to hack Asterisk so that it sends _all_ SIP
requiests to some fixed, (set at compile time) IP address.
I'll run SER at that address and have SER "mangle" the
packets.


=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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