Mailinglist Archive: opensuse (769 mails)

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Re: [opensuse] VoIP ATA does NOT need to have a foxed address, only the ISP does
On 2016-05-28 14:48, James Knott wrote:
On 05/27/2016 05:48 PM, Carlos E. R. wrote:
On 2016-05-27 23:25, James Knott wrote:
On 05/27/2016 08:36 AM, Carlos E. R. wrote:
On 2016-05-27 14:05, Per Jessen wrote:
Maybe you're just using black magic or maybe you're not behind a NAT
router or maybe you are using STUN or keep-alive, but you don't know.
Yes.

In my case, it is done on a little box placed before the router, and it
uses a 10.*.*.* network.

So, that little box connects to an STUN server or uses other keep alive.
I think neither, because it does not use NAT. It is a 10.*.*.* VPN,
different from the house LAN at 192.168.1.*


Well, that explains it. I don't think you mentioned anything earlier
that would indicate your VoIP wasn't via NAT. You'd have a non NAT
route to the server.

Yes, I did. But the thread is long, easy to miss a point.

However, you still can't talk peer - peer to
someone running VoIP at the other end. You have to keep that server on
the call for the duration.

Assuming I can use a SIP phone on that (I haven't been able to, the
company changes parameters so that it fails, intentionally), I would
have to do NAT on some router on my ISP premises. Or the registrar would
have to do things on it. At least to phone via SIP to people outside
that network. The intention of that network is to replace POTs
transparently, not for people to use VoIP. And charge per call and per
minute.

Notice that Asterisk doesn't behave always "peer to peer". Sometimes it
handles the call, changing to another codec in its CPU, and sends
forward to the destination. Asterisk might thus also handle NAT without
STUN if it sits on both networks.

--
Cheers / Saludos,

Carlos E. R.
(from 13.1 x86_64 "Bottle" at Telcontar)

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