Re: [Linphone-users] Unable to call SIP account

[Jaroslaw Rafa]

Dnia 25.03.2023 o godz. 22:05:30 Stuart D Gathman pisze:
> >In typical home/small office network scenarios the VoIP endpoint is your
> >router, that connects you to your ISP. The router has a VoIP gateway built
> >in and one or more of typical analog phones connected to it. The NAT issue
> >does not exist in that case, as the router almost always has a public IP
> >address. This address may change because it may be dynamically assigned by
> >the provider (but not necessarily - for example I have a static one: it is
> >assigned by DHCP but I always get the same address), but services like
> >DynDNS exist to solve this issue.
> 
> That does not work for peer to peer unless you have exactly one SIP
> device behind said router and a port forward.  A small office might
> install a SIP server (e.g. asterisk) - but that is a federated solution,
> and not peer to peer.

I didn't mean any SIP device behind the router.
I did mean a VoIP-enabled router, which is a SIP device by itself. You don't
need any port forward then.
They are quite common.
The phones (normal, analog phones) are connected directly to the router on
RJ-11 ports.
Most of the router types support two phones. Some cheaper models support
only 1 phone, but there are also the ones like Fritz!Box series, that
support a whole bunch of phones...

You can use these routers with some provider's SIP server, but you can also
call them directly at their IP address. Each phone will use a different SIP
port. If you are using a SIP server, then each phone can be configured to
use different server and/or account.
-- 
Regards,
   Jaroslaw Rafa
   raj@rafa.eu.org
--
"In a million years, when kids go to school, they're gonna know: once there
was a Hushpuppy, and she lived with her daddy in the Bathtub."