APNIC reminds us that "there are now a large number of ISPs, data centres, cloud services, and software that now support IPv6" and "enabling IPv6 can be as simple as clicking a button on your WiFi router."
I turned it on, with Comcast I received an IPv6 route but no DNS server. Fortunately, Google Public DNS has unmemorable addresses, which I was able to configure manually.
2001:4860:4860::8888
2001:4860:4860::8844
It works. "There's only one thing left for you to do: Turn it on!"
[ ed: What are the alternatives to Google's Public DNS? ]
(Score: 2) by Scruffy Beard 2 on Thursday May 05 2016, @08:13PM
No port-forwarding does not work because I do not get a public IP address from my mobile provider.
I called and asked about IPv6. They will do it if I upgrade to a business plan with a higher minimum monthly commitment.
(Score: 2) by frojack on Friday May 06 2016, @02:25AM
Voip handles non public IPs via STUN servers [voip-info.org].
No, you are mistaken. I've always had this sig.
(Score: 0) by Anonymous Coward on Friday May 06 2016, @07:09AM
We are talking about *incoming* connections. STUN is not going to help you there.
TURN does this, but requires turning the incoming connection into an outgoing one - which means the host receiving the connection needs to know when to expect a connection. That mean always having a different connection open for signalling. I'm guessing that's what he meant by tunneling through his asterisk server.
(Score: 2) by frojack on Friday May 06 2016, @03:07PM
We are talking about *incoming* connections. STUN is not going to help you there.
TURN does this, but requires turning the incoming connection into an outgoing one - which means the host receiving the connection needs to know when to expect a connection.
You are right of course. I linked the wrong page as well.
http://www.voip-info.org/wiki/view/TURN [voip-info.org]
The thing is, MOST VOIP/SIP providers (even free ones) supply TURN services on their systems precisely because such a vast portion of the net is behind NAT routers, and just about always has been. NAT traversal hasn't been a problem for Voip or Sip for some time now, and TURN was in place in one form or another since LONG before the relevant RFCs were formalized.
Most providers have TURN/STUN/ICE all bundled into one server on their network. Incoming calls are just automatically routed to what ever network you happen to be on at the moment. Even cellular networks.
If the OP has a problem its probably because his Asterisk server is behind a double nat (his and his ISPs), but even this is not a problem with any (free) external TURN service configured in his asterisk box. If
No, you are mistaken. I've always had this sig.