mohacsi at niif
Mar 4, 2011, 2:50 AM
Post #16 of 16
On Thu, 3 Mar 2011, Marc Blanchet wrote:
Re: Curious choices made by Cisco/Tandberg
[In reply to]
> Le 11-03-03 11:38, Hannigan, Martin a écrit :
>>>> ^^^^ ^^ ^^^^^^^^
>>>> Yes, that's right, you can have IPv6, but only if you're willing to
>>>> IPv4. On the one hand, I want to applaud their optimism. But I'm too
>>>> by the cluelessness. . .
>>> Almost the same at Polycom. Your system can registered either via IPv4
>>> or via IPv6. Exclusive OR!
>> I wouldn't be too fast to assume that this is lack of clue. I would guess
>> that it's a processor limit = cost. Why would you need to dual stack your
>> phone regardless?
> - another phone vendor also does either v4 or v6.
> - we have helped phone manufacturers and PBX for porting to IPv6.
> - the main reason for the exclusive OR was simplicity for end users: as plug
> and play as possible, simpler scenarios (don't need to tackle complex
> IPv4-IPv6 scenarios), given the fact that the typical deployment scenario is
> the phone only talks to the PBX and using a single IP is what is needed.
> - not related to processor or memory constraints.
> - I'm not trying to excuse anyone, because they shall be at the end really
> support dual-stack, but I'm giving some context.
In real environment tou might expect, that some end point will be IPv4
only, dual-stacked, IPv6-only. For communication between IPv4 only and IPv6
only UA the signaling can be done with dual-stack SIP proxy, but there is
a need for a rtp proxy media gateway - a kind of protocol
translator. A dual-stacked UA should be able to communicate IPv4 only
only and IPv6 only devices.
For simplicity the user should just register its end point to gatekeeper
or SIP registrar with both IPv4 and IPv6 address. The user just
specifying the server (gatekeeper/ SIP proxy) via DNS name or via DHCP. The
registrar should associate the SIP URI both IPv4 and IPv6 address (and
ports) - dual SDR.
Then in SIP INVITE the client automatically select based on the collected
information form SIP proxy, initiate the connection over IPv4 or over IPv6
to the remote UA or proxy.
Alternatively the ICE protocol can negotiate the common protocol version
and global addresses (in case of IPv4 NAT) to be used.
The most elegant solution would be to use a fairly recent SIP Outbound
standard: setup more than one SDP session (one for IPv4, one for IPv6,
and probably sam fall-back sessions), and use only one for RTP stream and
switch to other session in case of failure.
Clearly for IP telephony and IP videoconferencing the next steps toward
IPv6 is a dual stack operation (dual registration in the SIP registrar,
and proper selection between IPv4 and IPv6 at the UA level or in ICE
protocol). IPv6 only SIP operation is only useful for green-field or
isolated IPv6 networks.
For H.323 it is much more complex....
I doubt H.323 endpoints ever correctly support IPv6.
> Regards, Marc.
> IPv6 book: Migrating to IPv6, Wiley. http://www.ipv6book.ca
> Stun/Turn server for VoIP NAT-FW traversal: http://numb.viagenie.ca
> DTN Implementation: http://postellation.viagenie.ca
> NAT64-DNS64 Opensource: http://ecdysis.viagenie.ca