An Anonymous Coward writes:
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.
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? ]
Not if you want to use VOIP.
At the moment I am contemplating tunnelling to my Asterisk server that has ports forwarded to it. I suspect going straight to my VOIP provider would have less latency, but I can not receive incoming calls while using NAT.
Doesn't port forwarding solve this? Or worst case, DMZ *shudders*
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.
Voip handles non public IPs via STUN servers [voip-info.org].
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.
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.
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
While I've not configured it myself, I believe Asterisk has supported IPv6 since 1.8. PJSIP in Asterisk 11 does support IPv6, and there is information out there on how to set it up and configure it.