FOOD FOR THOUGHT IPv6 and providers

Randomandy

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I am attempting to configure a system that relies on IPv6 endpoints. Since calls, once setup, communicate peer to peer, I understand all phones must be on IPv6 that want to talk to an IPv6 only phone.

Q1: What about my SIP trunk? If a user endpoint is IPv6 only, does the SIP trunk need to also be IPv6 native? Or do trunks behave differently in a way that the phone-Asterisk link could be on IPv6 while Asterisk-SIP trunk could remain on IPv4? (Somewhat of a media translator.)

Q2: If IPv6 would be required of the SIP provider in the scenario above, does anybody know which providers are already IPv6 native, if any?
 

Porch

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This is all based on my understanding on how native IPv6 works. I have used IPv6 with HE, but beating my ISP with a stick has not given me the desired /64 of IPv6 space, so I have not tried SIP over it.
Since the Asterisk is affectively doing routing anyway, I don't see a problem with having the trunks on IPv4 and the phones on IPv6.
 

Randomandy

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Dynamic IPv4 assignments and limitation to a /29 vs our /48 in IPv6. IPv6 just makes it all a lot simpler than figuring out STUN and PAT and so on.
 

billsimon

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My question is why do you have to do IPv6?
At this point in time, we should be asking, "Why do you have to use IPv4?"

Scoff if you want.

I'm not sure the tone of your question, but if you were in any way implying that IPv6 is inferior, it's time to get with the current technology.

I get frustrated when I see a post on this or any forum where the "solution" is "turn off IPv6! It doesn't work!" I realize that no one has said that in this thread--yet. It's just such a common and wrong sentiment. IPv6 works very well, but like any network stack, you need to configure it.

Back to the topic at hand. Randomandy a default PIAF does not send media peer-to-peer. It all goes through the PBX. So you could cross between IPv6 and IPv4 networks if your Asterisk server is dual-stacked.

I do not know of any IPv6 SIP providers.
 

TwigsUSAN

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So, take the attitude and cram it. I was asking if there is a specific reasons for it. I understand IPv6 is the next thing. However, there are still times it's still a better solution to run a double stack.

My biggest thing is people crying that the end of the world is coming if you don't go to IPv6. You will most likely see it being utilize for certain items at first. For example between ISP networks. IPv4 will still be used in a LAN environment for a very long time.
 

Randomandy

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For me, it's the reverse. I most need IPv6 at the outer realms of the network where lots of devices need routable IP addresses. billsimon's response suggests that my original question is not really a problem. SIP provider can be IPv4 and traffic from IPv6 phones passes through the PBX and translates no problem. I hope he's right.
 

DaveDog

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SIP providers
FreePBX
most phone manufacturers

all seem to be ignoring IPV6 at this time
 

DaveDog

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I should add..

a lot of ISPs

and perhaps "ignoring" is a bit harsh, maybe "not on the roadmap"
 

atsak

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IPV6 is good, but frankly we could continue using IPV4 for quite some time if people would just give up the IPV4 space they are not using and is over allocated to them (Hello Xerox, that's you and others). Anyway, it will happen when it needs to, just like everything else.
 

Randomandy

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I did hope this thread to be about how to do IPv6, rather than whether to do it. Oh well.
 

billsimon

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I did hope this thread to be about how to do IPv6, rather than whether to do it. Oh well.
I have a FreePBX server on public IPv4 and IPv6 space if you'd like to do some testing.

Asterisk is a "back-to-back user agent" when you have "canreinvite=no" set (default in PIAF, as I mentioned). That means that each leg of the call terminates at Asterisk; e.g., IP phone to Asterisk, SIP provider to Asterisk. This is why you can have one leg using IPv6 and another using IPv4. It's also why Asterisk works when it's behind an IPv4 NAT along with a bunch of phones and they use a SIP gateway on the Internet. It could work for the phones-behind-NAT to exchange media directly with a SIP gateway on the Internet, but it would be much more prone to failure than just having Asterisk do this.
 

billsimon

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Code:
pbx*CLI> sip show peers
Name/username            Host                                    Dyn Forcerport Comedia    ACL Port    Status      Description                     
1100/1100                2001:470:8906:0:bae8:56ff:fe30:XXXX      D  Yes        Yes        A  5060    OK (68 ms)                                 
...
5228993386GW1/5228993386  65.254.44.194                              Yes        Yes            5060    OK (26 ms)                                 
5228993386GW2/5228993386  74.81.71.18                                Yes        Yes            5060    OK (49 ms)                                 
5 sip peers [Monitored: 3 online, 2 offline Unmonitored: 0 online, 0 offline]
pbx*CLI> core show channels
Channel              Location            State  Application(Data)           
SIP/1100-00000000    [email protected] Up      Dial(SIP/5228993386GW1/1800724
SIP/5228993386GW1-00 (None)              Up      AppDial((Outgoing Line))     
2 active channels
1 active call
There it is, just for fun. Extension 1100 is connected to Asterisk on IPv6 as you can see, and there are two IPv4 gateways listed. I placed a call to a 1800 number from my IPv6 extension and it connected just fine through the v4 gateway.

For the record, if your IPv6 stack is working on the server, the only thing you need to do to enable it in Asterisk is:

Settings -> Asterisk SIP Settings -> and under Advanced General Settings -> Bind Address, enter ::
 
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Randomandy

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Excellent. Thank you, billsimon. That resolves my issue.

At the risk of not leaving well enough alone, this makes me curious about the details on "canreinvite". Ideally, I would still like reinvite to occur within my local network between devices that are using the same stack. So now I am wondering if I set canreinvite=yes, and the call was first established to the pbx on IPv6, but the destination was not available on IPv6, would the reinvite attempt transparently fail back to a 'non-reinvited' connection, or would the call then drop? I might just try this and see (unless somebody can already tell me what is going to happen.) This would be the best of both worlds for me.

From the RFC, I read,

During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re-INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics.
This sounds like just the behavior I was hoping for. While a respondent using a different IP stack would not be able to respond with the 488 message, it still suggests that the call would continue. Crossing my fingers.
 

billsimon

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I'm trying to test that but so far have not been able to convince Asterisk to get out of the media path between two IPv6 endpoints, even with canreinvite=yes. It might be necessary to explicitly disable the features that would otherwise keep Asterisk in the media path, such as *1 recording. I don't know for sure, but might pick back up on it later. In the meantime, you have a working dual-stack solution. Enjoy!
 

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