[asterisk-biz] RE: If money is not a considerationwho givesthebest SIP termination??

Steve Kennedy steve-asterisk at gbnet.net
Thu Nov 23 10:42:03 MST 2006


On Thu, Nov 23, 2006 at 11:10:36AM -0600, Mike Hammett wrote:

> I know how the Internet works.  I'm about to get my own AS for 7 sites in 
> the US and 2 in the EU with more to come.
> If both endpoints share a common ISP, its ALMOST as good as having a 
> private IP link.  The data does not cross peering points and your provider 
> is solely responsible for the quality of the traffic.  Better than that is 
> to purchase layer 2 services available from MANY providers.  If you get 
> your own AS, you can then peer with your VoIP partners as well as one or 
> more transit providers.  This then provides a failover if your private link 
> goes down. IP connections can be made much more fault tolerant than TDM 
> connections at a much lower cost.

AS numbers mean nothing, they mean you can run routing protocols such as
BGP and set the metrics for how traffic should pass to other AS numbers
and IP networks (and who you trust).

The only way to guarantee that traffic has QoS is to make sure it goes
into a network that supports QoS and the metrics have been set correctly
so your type of traffic will pass across the network correctly.

You can purchase L2 services, which may be genuine L2 or encapsulated L2
within a L3 network.

If you connect to an ISP at two points, unless they guarantee QoS across
their own network, there's likely to be congestion somewhere in their
network.

Major providers do provide QoS in IP using IP/MPLS (or other protcols).
Currently the only real way to ensure zero congestion at peering points
is to purchase enough capacity at the exchange so congestion can never
occur (and then onward through the exchange point).

Eventually more exchanges supporting QoS will appear, but then everyone
connecting will have to support the same QoS metrics for it to be
meaningfull.

The whole reason IP networks are more resilient is that they support
dynamic routing, which is exactly what you don't want for VoIP type
traffic.

</end rant>


Steve


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