[plug] VOIP setup for Linux network

[Onno Benschop]

On Tue, 2004-03-23 at 12:05, Chris Caston wrote:
> The company is called Platinum Calling
> (http://www.platinumcalling.com.au)

Never heard of them.


> We are still yet to establish how to technically setup the connection
> but I am considering implementing a SIP proxy to allow machines inside
> the NAT based local address range to send and receive calls.

See below.


> I do have adsl and a staic IP but they suggested I may need *two* IP's.

If you don't do NAT, then you could allocate a static IP to the outside
that is your phone system I suppose, but it seems overkill... (see
below)

> So can you usually use SIP proxies with an upstream provider?
> 
> Examples:
> 
> (http://sourceforge.net/projects/siproxd/)
> 
> (http://sourceforge.net/projects/sarp/)
> 
> (http://sourceforge.net/projects/osipproxy/)

When I have a spare life I'll look at those...


> Any and all insights, ideas and experiences are greatly appreciated.

Let me answer in a side-ways manner...

Big DISCLAIMER:

I'm no VoIP expert, the information in this message is likely to be
completely bogus, I may have misconstrued what I learnt and you're
completely on your own...

(Did you get the idea that I've been madly reading and trying to figure
out how this technology works?)

I've tested VoIP using the Free World Dialup (FWD) service. I can
(finally) connect my ATA186 to their server, my account shows on-line, I
can sometimes pass the echo test, but I've yet to hear the phone ring.

The service provides a proxy that allows you to traverse a NAT/firewall.

VoIP will likely not work over a NAT connection without some external
solution, because you need to support both TCP and UDP connections (the
former to set-up the call, the latter to transport the audio).

The ATA186 supports all manner of protocols, IIRC SIP is supported. You
may need a call manager from CISCO, but I've heard others report that
you can just install Asterix and use that.

I've not yet got a reliable VoIP solution, but I can confirm that the
sound quality across the satellite link to FWD during the echo test was
surprisingly good.

I spent some time playing with ssh tunnels, but putting UDP over TCP was
a waste of time and bandwidth.

In my readings I've come across some explanation that seems to fit how
these NAT traversing proxies work, and I'll attempt to give you my take
on it, rather than subject you to 20 pages of stuff...

AFAIK the actual VoIP conversation is a peer-to-peer kind of thing. You
set up the call with some central device which calls the other phone and
then you start sending voice packets over UDP to the other phone.

The problem is that NAT and Firewalls don't like this very much at
all...

If there are three machines, A: you, B: them, C: call manager, and both
A and B are behind firewalls, and C is on the Internet, both A and B can
talk to C, but not directly to each other.

The proxy system seems to cleverly do the following:
A connects to C and tells it what it's IP address is, C also detects the
actual IP address (after NAT et-al), B does the same. C constructs a
packet to send to A that looks like it came from B and does the same to
A, now the path between A and B is open and UDP packets flow across the
gulf.

I came across this explanation (which I'm sure I've completely
butchered), in a discussion about peer-to-peer gaming and related links
such as p-2-p file sharing etc.

Now I'm sure I missed other stuff I learnt, but aren't you just glad I
was bored?

Onno Benschop 

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