Subject: [Speermint] Some remarks on peering/interconnection models

From: Otmar Lendl <[email-masked]>
Date: Tue, 17 Oct 2006 17:13:19 +0200

<rant>

I've been mulling about whether we have got some of the basics
of SIP peering correct: What kind of basic interconnection scheme
will emerge out of the symbiosis of the Internet and PSTN worlds?

What models do we see right now?

A) The point-to-point model: 

   Incarnations are the PSTN and the Layer 3 Internet: 
   The overall Network is built upon the physical interconnection
   of smaller networks.

   In both cases, commercial factors have resulted in a multi-tier
   network architecture, with global players, regional networks,
   and small access networks. Any call (or packet) traversing
   that network is passed over pre-established interconnection
   links between provider networks.

   Routing information is always next-hop based: The switches
   (or routers) which make up the network need to know which
   outgoing trunk they need to forward the call/packet too.

   The routing protocols differ: the PSTN is more database
   driven and uses manual configuration, whereas the Internet relies
   on a dynamic routing protocol (BGP).

   [The commercial aspects vary, too. But that's not important here.]

B) The email model:

   The basic premise here is that we have a functional and
   worldwide L3 network. On the email layer, we thus don't
   have hard-coded point-to-point links. Everybody can talk
   to everybody else if a single email demands it. 

   There is thus no need for pre-configured "email trunks". 

   All that is needed is a global directory system (the DNS)
   which maps domains to L3 addresses of ingress points.

   Email ingress points are a simple set of addresses,
   which are identical for whoever wants to send email to
   that domain.

   No prior arrangement of any kind are necessary before
   emails can be transmitted. Email is accepted from any
   source on the Internet, and not only from pre-screened
   email partners.

   SIP as defined by RFC 3261/3263 follows this model.

---------------------

How does SIP peering as envisioned by the speermint fit into
these models? In my humble opinion: somewhere in the middle.

* It won't be a pure point-to-point model.

  On the commercial side, a lot depends on the sustainability
  of the current PSTN-style settlement regime. As long as
  money flows for each call, bilateral agreements will be
  the norm. Authentication based on VPN or TLS keys can supplant 
  authentication based on a physical connection. Thus even
  for logical point-to-point links, there is no need to
  actually use dedicated physical point-to-point links.

  From a technical point of view, n:n peerings are simple to 
  establish using a shared L3 network. 

  For small VSPs (especially enterprises), widespread peering
  may be desirable to avoid transit charges to larger
  operators. In that cases, the main criteria for entering
  a peering relationship is the question of how to establish
  trust between the parties.

  Some people may even continue to run completely open SIP
  proxies. 

  The number of domains to route is way to large to establish
  a routing protocol (similar to BGP, OSPF, ...) based on domains.

* It won't be the email model.

  Settlement, trust and SPIT are an issue. Networks will be 
  selective regarding from whom they will accept calls.

  Selective ingress points: The will not be a single set
  of ingress points which are open for all other networks to
  send calls to. Some might be on private networks, some might
  involve geographic restrictions, some might depend on codecs.

  While there is the ubiquitous global L3 network, peerings
  will make extensive use of private L2/3 infrastructure.

* It will be a wild mixture:

  - some networks will have strict rules making them close to
    the point-to-point model. Other will be more open and
    accept calls from almost anyone.
  
    I expect the large networks to rely primarily on pre-configured links.
  
    Smaller networks and especially enterprises will employ
    a mixture of both statically configured links as well as dynamically
    established peerings.
  
  - the commercial models will vary a lot. An enterprise will take
    a different approach than an old school carrier.
  
  - even a singe VSP will have different types of peerings at
    the same time. E.g. one or two "upstream" connections which
    connects it to the carrier network and the PSTN, some
    peering relationships at a local exchange with similar players
    and perhaps dynamically peerings based on TLS.

  - The peering decision is not a question of establishing connectivity,
    but one of establishing trust, commercial relations (if needed), 
    and QoS. 

---------------

Right now, all this is happening in an ad-hoc matter: The PSTN
is still the default network for voice calls. SIP peering are
configured manually, various services have sprung up to solve
bits and pieces of the problem space, but right now there is
no basic architecture for such an hybrid interconnection world.

Another complication is the fact that call-routing is both
based on E.164 numbers and SIP URIs.

What approaches do exist?

* User-ENUM and open SIP servers.

  Nice in theory, but the last few years have shown that this
  isn't the winner.

* IMS / NGN

  The telco approach. Ain't going to be popular in the
  enterprise sector.

* Individual peering fabrics

  Usually local solutions for a specific market. Competitive
  business. Usually number-based. World dominination ambitions abound.

What's needed:

* Get the number vs. URI stuff sorted out once and for all.

  We can't go on indefinitely using two different addressing
  schemes for same routing decisions. The extra complexity
  will kill us.

* A call routing algorithm which

  - works with static peerings

  - dynamic peer discovery

  - enables path selection based on a wide range of (real-time)
    parameters.

  - can be used by the big international carriers as well as
    small enterprise SIP installations.

* Get the bilateral technical issues sorted out

  - SIP feature set
  - caller-id/privacy/...
  - codecs/transcoding
 
</rant>

/ol
-- 
< Otmar Lendl ([email-masked]) | nic.at Systems Engineer >

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