On Wed, 2009-08-26 at 23:27 -0500, Christopher Coleman wrote:
> The overall questions is... What is required to make a SIP to SIP
> phone call over an Internet connection? 

as usual, the answer is "it depends" :-)

> For example: m...@mydomain.com wants to call y...@yourdomain.com. We both
> are registered domains so there isn't really an issue resolving the
> domain down to  an IP. 
> 
> 
> I realize it will have to traverse the NAT but beyond that what are
> the steps? 
> What configurations or hardware is required to place this call? 

You need to have something that you can route the call through to get
past your NAT.  In sipXconfig, this is called an 'SBC' (Session Border
Controller).  You have to add an SBC (under 'Devices'), and the
configure the 'Internet Calling' feature (under 'System') to use that
SBC.

The sipXbridge that's used to connect to an ITSP (SIP Trunk) is a simple
SBC; unfortunately, the 4.0 version of sipXbridge doesn't support this
general Internet Calling mode, so it can't be selected.  In 4.2 (and in
the current 4.1 unstable builds), it is supported.

If you are _not_ behind a NAT, none of this is needed - it just works
(this is the theoretical perfect world that SIP was originally designed
for).

> What gets typed into the phone? If I type in the phone
> y...@yourdomain.com will SipX interpret that as
> y...@yourdomain.com@myDomain.com? (Appending the domain onto whatever
> the phone wants to dial.)

I guess that depends on the phone, but I've never seen one that does
that.

> How does this fit into a dial plan?

'Dial Plan' mostly refers to how to manipulate routing based on digit
strings, so this is mostly orthogonal to it.  The sipXecs dial plans
won't be applied to any call whose domain part (right of the '@') is not
that of the sipXecs system itself - it will just try to route it to that
domain using the DNS.




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