Move to VOIP an CGNAT

They are not peer to peer. A server of a known address is involved that relays the calls.

The problem with NAT is that it allows outbound connections but makes very little provision for inbound ones.

SIP worked flawlessly on my VOIP equipped router though, and works on the new one too. They *are* the NAT, not behind it!

I suspect that is the way it will go, Phone ports in the router for third party VOIP/SIP or in the fibre modem thingie for locked into BT/ISP s*it There are boxes that will work inside NAT, and there are ways to accept incoming connections behind NAT. PnP is one ghastly one.

As with things like ftp transfers in the early days, the router needs to understand the protocol and accept incoming SYN packets and allow a remote peer to set up a connection.

Or we all go IPV6 for voip and forget NAT

I've never used it, but some SBCs I've installed offer it, either not using NAT or using STUN has always worked for me.

I found this useful in understanding:

TURN acts as a proxy for your packets, so you send all your audio to the TURN server and that forwards it. That's expensive.

ICE tries both STUN and TURN at the same time at each end and the sides then negotiate over which method is best.

This does sound a little like the protocol Skype used in its original P2P implementation, except that clients were also servers. A client would try to decide if it was on the public internet and if so announce itself as a 'supernode', and non-internet clients would funnel their traffic through the nearest supernode. This was a big headache for network managers who suddenly found somebody running Skype on one of their machines would suddenly generate a massive spike in traffic.

Theo

I use it, alongside IPv4. Quite a few of the UK ISPs will give you both either by default, or if you ask. BT, Sky, Zen...

If the telco that provides you SIP knows what they are doing they will set things up so that it works fine through NAT and CGNAT. Nearly all customers are behind NAT now.

You have to set up your telephone or ATA to make an outgoing SIP connection when powered on and send keep alive packets at intervals, but it'd be unusual if you have equipment that can't do that.

Maybe there are still some daft SIP providers that refuse to use the modified versions of SIP and RTP at their end which will work fine through NAT?

Maybe some users of VoIP like to allow anyone who knows their IP address to call them for free without involving any phone company?

No it doesn't. Most SIP telcos now accept SIP connection from any port (so NAT doesn't matter) so you just need to make your equipment keep the SIP connection to the telco open all the time it is powered on.