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


From: Stuart D Gathman
Subject: Re: [Linphone-users] Unable to call SIP account
Date: Sat, 25 Mar 2023 22:05:30 -0400 (EDT)

On Tue, 21 Mar 2023, Jaroslaw Rafa wrote:

Dnia 21.03.2023 o godz. 17:30:31 Stuart D Gathman pisze:

1) There aren't enough IPv4 addresses.  There are twice as many people
   in the world, not to mention devices.

That's not an issue if someone already *has* an IPv4 address. Don't

Most devices have a private IP4, e.g. 192.168.*.* or 10.*.*.*.  These
cannot be reached from outside, and hence as useless for peer to peer
communication.

understand why you (implicitly) assumed that the target device does not have
an IPv4 address, only IPv6. In fact, if you get an Internet connection from
any ISP, you might not have an IPv6 address, but you certainly do have an
IPv4 one, as otherwise you wouldn't be able to connect to many hosts on the
Internet that are still IPv4-only.

The vast majority of people get a private IP4, and a router uses
Network Address Translation (NAT) to enable outgoing connections.
This forces the use of centralized (or federated) services that have
a public IP they can use.  In the case of SIP, this forces the use
of a SIP server.  Similary, connecting RTP streams between SIP devices
requires the use of TURN or similar centralized service since the
parties can only do outgoing connections.


2) IPv4 has been obsolete at least 2 decades.

By no way you can call IPv4 obsolete. The fact that a newer addressing
scheme exists doesn't mean IPv4 is obsolete. The majority of the Internet is
still running on IPv4 - and probably will be.

The vast majority of centralized services still use IPv4 - of course -
because this locks in their customers to only be able to use centralized
services.

3) IPv4 NAT practices in common use prevent connecting directly
   to the callee because they don't have a stable public IPv4 in
   general.  (Another reason for SIP services - as a sort of TURN server
   for people in IPv4 NAT jail as well as bridging to telco and SMS.)

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.



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