Speermint Working Group                                         O. Lendl
Internet-Draft                                                   enum.at
Intended status: Informational                          October 28, 2008
Expires: May 1, 2009

                VoIP Peering: Background and Assumptions

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Copyright Notice

   Copyright (C) The IETF Trust (2008).


   This documents provides background for the work on VoIP peering and
   tries to provide guidance on what kind of work is needed to
   facilitate widespread SIP-based peering of telephony networks.  It is
   intended to spur discussion on the work about peering (in the
   SPEERMINT and DRINKS WGs) and should also serve as input to the
   ongoing discussions on reducing Spam for Internet Telephony (SPIT).

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4

   2.  Interconnection Models . . . . . . . . . . . . . . . . . . . .  4
     2.1.  The PSTN model . . . . . . . . . . . . . . . . . . . . . .  4
     2.2.  The email model  . . . . . . . . . . . . . . . . . . . . .  5

   3.  Why is SPEERMINT needed? . . . . . . . . . . . . . . . . . . .  6
     3.1.  Why did the Email Model fail for SIP?  . . . . . . . . . .  6
     3.2.  The PSTN Model does not fit, either  . . . . . . . . . . .  9

   4.  Core Assumptions . . . . . . . . . . . . . . . . . . . . . . . 10
     4.1.  The Real Problem with SPIT . . . . . . . . . . . . . . . . 10
     4.2.  What is a SIP URI? . . . . . . . . . . . . . . . . . . . . 11
     4.3.  Peering vs. Reachability . . . . . . . . . . . . . . . . . 11
     4.4.  The Key to Routing Data  . . . . . . . . . . . . . . . . . 12
     4.5.  Lookups vs. Announcements  . . . . . . . . . . . . . . . . 12
     4.6.  No National Solutions  . . . . . . . . . . . . . . . . . . 14

   5.  A Vision . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     5.1.  The Players  . . . . . . . . . . . . . . . . . . . . . . . 15
     5.2.  Interconnection fabrics  . . . . . . . . . . . . . . . . . 15
       5.2.1.  Old-style TDM based interconnection  . . . . . . . . . 15
       5.2.2.  Private SIP interconnection  . . . . . . . . . . . . . 15
       5.2.3.  Commercial Peering Fabrics . . . . . . . . . . . . . . 16
       5.2.4.  Overlay networks . . . . . . . . . . . . . . . . . . . 16
       5.2.5.  RFC3263-style SIP  . . . . . . . . . . . . . . . . . . 16
     5.3.  An internet of Voice Networks  . . . . . . . . . . . . . . 16
     5.4.  The Lookup Function (LUF)  . . . . . . . . . . . . . . . . 16
     5.5.  The Location Routing Function (LRF)  . . . . . . . . . . . 17

   6.  Design Pitfalls  . . . . . . . . . . . . . . . . . . . . . . . 18
     6.1.  Reliance on fabric internals for multi-hop routing . . . . 19
     6.2.  Mistaking a protocol for softswitch/SBE control for
           LUF/LRF  . . . . . . . . . . . . . . . . . . . . . . . . . 19
     6.3.  Mixing LUF and LRF . . . . . . . . . . . . . . . . . . . . 20
     6.4.  Provisioning vs. routing . . . . . . . . . . . . . . . . . 20
     6.5.  RFC 3263 is part of the problem  . . . . . . . . . . . . . 21
     6.6.  Disregarding enterprises . . . . . . . . . . . . . . . . . 21

   7.  What building-blocks are missing?  . . . . . . . . . . . . . . 21
     7.1.  LUF / I-ENUM provisioning  . . . . . . . . . . . . . . . . 21
     7.2.  LRF Routing announcements  . . . . . . . . . . . . . . . . 21
     7.3.  A call-control protocol  . . . . . . . . . . . . . . . . . 21

   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 22

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   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 22

   10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 22

   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 22
     11.2. Informative References . . . . . . . . . . . . . . . . . . 22

   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22
   Intellectual Property and Copyright Statements . . . . . . . . . . 24

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1.  Introduction

   The Speermint WG is chartered to help with the interconnection of SIP
   based layer 7 networks.  It should not deal with basic IP
   connectivity and SIP protocol issues; those are covered by other
   working groups.

   Speermint focuses on what guidelines (and perhaps protocol elements)
   are needed by service providers and enterprises to move from ad-hoc,
   manual peerings to a fully standardized, secure, easy to implement,
   and thus widespread SIP based peering setup.

   This document aims solely at the telephony network aspects of SIP and
   ignores applications like Instant Messaging or Presence which might
   also be implemented using SIP.  The focus here is on the use of SIP
   in PSTN replacement services.

   Version -01 of this draft expands on the implications of the
   interconnection structure on the SPIT problem.  The concepts listed
   in here should thus also be worthwhile for the RUCUS EG.

   Version -02 of this draft provides a rough overview on how the voice
   interconnection landscape might look like in the future and lists
   specific work items for the SPEERMINT and DRINKS WGs.

   This document was written as discussion input.  It is not intended
   for publication as an RFC.

2.  Interconnection Models

   In order to understand the VoIP peering world it is necessary to go
   beyond pure protocol issues and instead talk about the ecosystems in
   which the protocols operate.  This section tries to be purely
   descriptive and makes no recommendations.

2.1.  The PSTN model

   The public switched telephone network (PSTN) is built upon the
   following fundamental assumptions:

   o  When the system was designed, the numbers of operators was
      limited.  Often, there was just a single incumbent per country.
      Building a full mesh for international interconnection was
      possible.  This is no longer true.

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   o  Global reachability is achieved by interconnecting individual
      smaller networks.  There is no global lower-layer connectivity: if
      two networks are not directly interconnected, then calls are
      passed through transit networks on the application layer.

   o  There are no ad-hoc connections between networks: all links are
      manually configured lines (physical, or other transparent bit-

   o  There is a clear separation between network operators and network
      users.  This applies both to protocols (e.g., SS7 ISUP versus
      ISDN), as well as, to regulatory rules.

   o  Routing information is not directly passed from the destination
      network to the source network via some global database.  Instead,
      transit networks communicate to their customers which destinations
      they can handle.

   o  Accounting and settlement are core features.

2.2.  The email model

   SIP according to RFC 3261 [2] and RFC 3263 [3] follows an email alike
   model.  It can be summarized as follows:

   o  Email and SIP addresses are structured as username@domain.  For
      routing purposes, only the domain part is relevant.  The username
      is only interpreted by the machines serving this specific domain.

   o  The global, public DNS is used to map the domain from the address
      to a prioritized set of ingress points that handle incoming
      communication requests for this domain.  As the DNS is agnostic to
      the entity querying data stored there, all senders receive the
      same set of ingress points.

   o  In order to achieve global reachability, these ingress points need
      to accept incoming requests from the open Internet.  If they
      reject, for example, incoming packets from a VoIP provider X from
      country Y then there is no backup path for this communication, and
      the destination just will not be reachable from that VoIP provider

   o  As anybody on the Internet can contact any destination domain, no
      business relationship between sender and destination domain is
      required.  This implies that there is no settlement: No money is
      changing hands because of such a communication.

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   o  There is no inherent distinction between end users and service
      providers hard-coded into the protocol.  Any client can do the DNS
      lookups himself and directly contact the destination servers.

   o  Usually clients do not talk directly to each other: On the source
      side a SIP INVITE is forwarded to the outbound proxy that applies
      the routing algorithm, which then contacts its peer on the
      destination side that also performs additional processing before
      handing off the communication to the destination side.

   The email model has proved to be extremely successful -- for email.

3.  Why is SPEERMINT needed?

   In theory, the Speermint WG is not necessary: The SIP RFCs envision
   global reachability of all SIP devices over the public Internet.
   Source networks just need to resolve the domain from the URI
   according to RFC 3263 and send the INVITE to the SIP proxy in charge
   of that destination domain.

   Telephone number (TN) based calling is also supported: RFC 3761 [1]
   provides TN to URI mapping and thus reduces the call routing problem
   to the already solved case of SIP URI resolution.

   * Apparently, the real world did not choose to implement and deploy
   SIP and public ENUM as initially envisioned by their inventors.  In
   other words: the motivation for Speermint is the failure of the world
   to conform to the original IETF vision of SIP based real-time
   communication. *

3.1.  Why did the Email Model fail for SIP?

   Although SIP has won the protocol war against H.323 (just as SMTP won
   against X.400), it failed to establish the same sort of ecosystem in
   the Internet as SMTP was able to do.  The number of SIP users who are
   reachable via the open Internet using RFC 3263 is minuscule compared
   to the number of SIP based telephones in operation today.

   SIP as a protocol has succeeded; SIP as an ecosystem similar to SMTP
   has failed.

   The need for Speermint arises from that failure: SIP has seen
   widespread deployment within enterprises and service providers, but
   the inter-connection part of SIP has not: current deployments usually
   do not follow RFC 3263, but use either hard-coded IP addresses or
   private DNS to route calls between SSPs, if they use SIP at all and
   not the PSTN.

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   The same applies to ENUM according to RFC 3761: The technology has
   been successful (as the large number of private ENUM trees
   demonstrates), but the original vision of ENUM proved to be elusive:
   the "golden tree" under e164.arpa contains a fraction of entries
   compared to the numbers found in private trees.

   Speermint is chartered to provide solutions for the interconnection
   problem.  It is thus essential to examine why the current standards
   have failed.  Without this gap-analysis there is little chance that
   Speermint will come up with the missing pieces.

   As mentioned before, this analysis cannot be restricted to pure
   technological aspects, and will thus touch on the business models
   implied by the technical standards.  It is the firm believe of the
   author that the IETF credo "we don't do business models" has been
   implicitly violated by existing standards.  One approach is thus to
   identify these implications and augment the protocols to allow them
   to support a varity of business models and ecosystems.

   Business Model

      The email model hard-codes a "sender-keeps-all" settlement regime.
      As anybody is able to connect to anybody, no business relationship
      is needed between communication partners.  Thus, no termination
      fees can be collected.

      The economic model of the current carrier landscape in most
      countries depends on these charges, and it just does not make
      sense for any single carrier to allow anonymous incoming SIP-based
      interconnection as that means lost income.  If call patterns are
      about symmetrical, switching to sender-keeps-all is revenue-
      neutral for all carriers.  There is no clear path on how such a
      fundamental shift in the bedrock of telco settlement could happen.
      (other than by regulatory fiat)

      The end-state might be a viable business model, but there is no
      incentive for any individual SSP to start the transition.

      This argument applies only to SSPs that are substitutes for PSTN
      carriers, and not to enterprises operating a SIP infrastructure.

   Unwanted Calls

      Spam over Internet Telephony (SPIT) is another concern.  The free
      for all nature of the email ecosystem has led to a barrage of
      unsolicited email (SPAM) which poses a serious threat to the
      usefulness of email.

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      Email is non-interactive: filters can be deployed to detect spam
      by the content of the mail before the recipient is alerted.  That
      is not possible for SPIT; content is only available after the
      recipient has picked up the phone.  A number of SPIT mitigation
      strategies have been proposed over the past few years, their
      effectiveness is yet untested.  See also [6].

      As of 2008, SPIT is not a problem, mainly because the number of
      open reachable SIP devices is so low.  Just as SPAM only started
      to become a problem after open SMTP servers became common, many
      SSPs fear that SPIT will appear if they open up their networks.


      Traditional telecom services provide reasonably reliable caller
      identification.  Telcos trust each other's signaling and end users
      have learned to trust caller-id (even if this trust is somewhat
      unjustified).  Such a trust model is not compatible with the email
      model of open SIP servers: the INVITE message can come from any
      host on the Internet and is thus not trusted.

      Providing a reliable caller identification is also important for
      policing: Harassing and abusive calls are more or less under
      control, as legal and contractual rules can be enforced by tracing
      calls back to the culprit.

      SIP Identity (RFC 4474 [5]) uses a different approach that is
      based on an authentication service cryptographically asserting the
      identity of the caller.  As such, it is different to the current
      practice in the telco space, which is based on transitive trust.

   QoS and Denial of Service

      The email model is not suitable for stringent Quality of Service
      (QoS) deployments.  As there are no pre-arranged relationships
      with between all communicating SIP servers, there are no
      mechanisms to guarantee neither network performance on the IP
      layer for the actual voice transmission, nor can there be
      comprehensive tests on SIP layer compatibility.

      As the ingress points need to be open to anybody on the Internet,
      they are exposed to Denial of Service attacks.

      This combination is at odds with the telco mind-set that thrives
      on predictable quality and stringent service level guarantees.

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   Legal Requirements

      Operators of public telephony services need to observe a range of
      regulatory requirements.  These rules were written for the PSTN
      scenario with clearly defined boundaries between operators and
      users of the telephone network.  Changing the interconnection
      model make these regulations a bad fit for the email model.

      For example, if the user requests CLIR (Calling Line
      Identification Restriction) then its SSPs needs to differentiate
      the call handling, whether the peering partner is another
      commercial SSPs (transmit caller-ID, signal CLIR) or an enterprise
      (suppress caller-ID).  Interconnecting with other SSPs that
      operate under the same rules simplifies compliance.

3.2.  The PSTN Model does not fit, either

   It is of course possible to rebuild the PSTN based on SIP instead of
   SS7.  Some might argue that this is what the IMS and NGN efforts are
   all about.  This is selling SIP and the Internet short: The basic
   infrastructure that the Internet offers allows for far more flexible
   interconnection arrangements than a simple copy of the PSTN

   Shared Layer 3 Infrastructure

      The PSTN is based on trunk lines connecting voice switches.  These
      trunks are manually established between carriers.  Each such link
      needs physical ports, as well as, dedicated bandwidth.
      Establishing direct links between carriers is thus only sensible
      if call volumes can justify the effort.

      In contrast to the point-to-point link world of the PSTN, SIP
      assumes an any-to-any IP based communication model.  This has a
      profound impact on the economics of interconnection: A new peering
      is not a matter of provisioning a new bit-pipe, but just one of
      configuring border elements on both sides.

      Economic theory states that there must be a optimal number of
      peerings per SSP given the costs to establish an interconnection
      versus the costs of transit.  As the cost structure is
      fundamentally different, the mesh density of the optimal SIP based
      network will deviate significantly from the current PSTN.

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   Enterprise Peering

      As a corollary: Peering between TDM-based enterprise telephony
      systems is usually limited to very high traffic cross-links.  As
      enterprise-to-enterprise calls do not require settlement, there is
      a huge potential for additional peering in this space.

   Dynamic Routing

      Worldwide routing in the PSTN is still based mainly on manually
      established routes; these routes reflect business relationships.
      As a consequence, it takes years to a get a new number range
      routed in the PSTN.  The switch from SS7 to SIP must be taken as a
      chance to upgrade the worldwide call routing to a better routing

4.  Core Assumptions

   The author believes that some working assumptions have to be re-
   thought based on the feedback from current deployment.

4.1.  The Real Problem with SPIT

   SPIT and QoS/DoS issues have often been cited as reason why so few
   people (enterprises and commercial SSPs) run an open SIP service
   (i.e. accept SIP calls from the public Internet without pre-
   association).  The IETF has taken on the challenge and tried to
   develop protocol extensions that should help with the SIP adoption.
   These are often based on the following reasoning:

   +------------+          +-------------+          +-------------+
   |We want to  |          |They need to |          |They have a  |
   |interconnect|===(1)===>|run open     |===(2)===>|problem with |
   |SSPs        |          |SIP servers  |          |SPIT and DoS.|
   +------------+          +-------------+          +-------------+

   A lot of time has been spent on step (2).  Protocols and procedures
   have been proposed to mitigate the exposure of open SIP proxies.
   These include the consent framework for SIP, SPIT identification,
   anti-SPIT policy rules, the Identity: header, etc.

   These are all worthwhile proposals that solve certain symptoms.
   Regrettably, they do not remove the roadblocks to widespread SIP-
   based peering.  For that, they tackle the wrong set of problems.
   They assume that the email model can be successful and we just need

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   to make sure that all the associated problems are addressed.

   This assumption is wrong.  The author believes that it is necessary
   to tackle step (1) first.

   The question therefore should not be "How do we deal with the
   unpleasant side effects of universal peering?", but "How can we get
   SSPs to peer at all?".  Instead of "How to keep out the unwanted
   calls?" we should focus on "How can we entice willing partners to a

4.2.  What is a SIP URI?

   SIP URIs are used in various contexts: They can specify contact
   points (sip:user@, they can specify next hop information in
   a private interconnection setting
   (sip:012345678@sbc1.chicago.us.example.net), and they can be public
   SIP URIs (sip:alice@example.com) for which the responsible SIP proxy
   can be determined using RFC 3263.

   There is yet another interpretation of the SIP URI that may be
   relevant for Speermint: The URI as a simple identifier of a telephony
   customer without the commonly implied semantic on how that user can
   be contacted.

   While that is close to the public URI, the difference is important:
   RFC 3263 does not apply.  There is no simple, globally valid set of
   ingress points for calls towards that URI.  The default SIP call
   routing logic just is not applicable to such URIs.

   In other words: It is a very useful concept to use the SIP protocol
   and the URIs from RFC 3261 without also adopting RFC 3263, because
   the latter more or less implies the email model that has not seen a
   lot of deployment yet.  It is thus expected that any SIP URI
   published in a public infrastructure ENUM will not imply the
   applicability of RFC 3263.

   In order to place calls, some alternative to RFC 3263 needs to be
   developed that accommodates the needs of carriers.

4.3.  Peering vs. Reachability

   Whatever the interconnection setup, subscribers of a telephony
   service expect to be able to call all subscribes of all other SSPs.
   When the email model cannot be assumed, this requires the use of
   transit networks and thus some sort of routing mechanism to find a
   path to the destination SSP.

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4.4.  The Key to Routing Data

   Currently, the PSTN side is using telephone numbers (TN) as the key
   to the routing information, whereas the RFC 3263 SIP uses the domain

   The TN used to be the perfect identifier for routing as the
   hierarchical structure of the number corresponded to the network
   topology in the PSTN.  The emergence of alternative carriers, number
   porting, and service numbers (free-phone, premium rate, ...) changed
   that: This is a form of "locator" / "identifier" split.

   Prefix-based routing used to be the way to aggregate routes to
   telephone numbers in order to keep the routing tables and their
   updates manageable.  While that is still useful to encode policies
   like "Route all of +43 to Carrier XYZ", within a country number
   portability made prefix-based routing increasingly inefficient.

   Looking at the routing information from a database design point of
   view, it does not make sense to store the set of possible routes
   (incl. all meta-data like prices, capacity, quality,...) for every
   individual number, as these will be identical for at least all
   numbers operated by a single carrier in some area.

   Any routing protocol will thus scale by several orders of magnitude
   better if it is based on some sort of "Destination-Group" that
   comprises a carrier identification plus optionally a service or
   region-specific tag.

   While the telephone number is the starting point of the routing
   information lookup, it is not a good identifier to use as the key for
   storing routes.

4.5.  Lookups vs. Announcements

   Generally speaking, there are two ways how to distribute routing

   On Demand

      Whenever routing information is needed some external database is

      Example: DNS (including ENUM)

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   Pro-active Distribution

      The information for all possible destinations is distributed
      before the first routing decision is made.

      Example: BGP, OSPF

   There are of mixed models as well, e.g., when an organization gathers
   routing information pro-actively to load an internal database that is
   then queried on an "on-demand" basis by network elements.
   Alternatively, some systems might pro-actively fetch the fraction of
   the global routing information covering the most likely destinations,
   and only fall back to on-demand queries for the rest.

   The on-demand model requires a lower level transport infrastructure
   to contact the external database.  It's thus clear that Layer 3
   routing cannot use that model as this leads to a chicken-and-egg
   problem.  However, for VoIP peering basic Internet connectivity can
   be assumed and the same constraints do not apply.

   The pro-active model on the other hand operates under a different
   constraint: Distributing all information needed for the routing
   decisions to all carriers requires that the aggregate dataset size of
   these routing information tables does not exceed sensible size
   limits.  For example, it is not feasible to replace the MX record
   lookup of mail-servers by a routing protocol which replicates the
   domain-name to mail-server mapping information to all ISPs and
   enterprise mail-servers.  There are just too many domains in use and
   thus the "mail-routing tables" would exceed all practical limits.

   With regards to TN based calls, both options are possible.  ENUM
   according to RFC 3761 is a clear on-demand approach.  On the PSTN
   side, downloads of database dumps are a common method to distribute
   routing information.

   A scalable routing system is needed, up to the set of all active TN.
   They number in the billions.  Installing a "full routing table" into
   a core telephony (soft-)switch is thus not feasible.  Current PSTN
   implementations cope by crude aggregation of routes to foreign

   The number of reachable IP addresses is roughly of the same order of
   magnitude, but the aggregation properties of IP addresses reduces to
   global routing table to under 500000 entries without any impact on
   the quality of the routing decisions.  Telephone numbers do not
   aggregate as well and therefore makes a TN-based protocol in the
   style of BGP infeasible (that is one reason why TRIP [4] failed).

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   The obvious solution is to add an on-demand mapping step ahead of the
   routing protocol.  That on-demand mapping should include the option
   to seed a cache with the most likely destination TNs.

4.6.  No National Solutions

   Telecom regulation, especially concerning number assignments and
   interconnection rules, is a national matter.  Calling patterns favor
   local destinations: local and national calls make up the majority of
   all calls.  It is thus not surprising that number based PSTN (and
   some of the emerging VoIP-based) routing exchanges only deal with
   numbers from a single country code.

   On the other hand, international voice termination markets deal
   usually not with individual numbers, but with routes to number

   Given the increasingly international footprint of voice operators the
   country-specific ways of handling inter-carrier routing is an
   anachronism.  Just as the Internet routing does not care about
   national borders, there is no inherent reason why a single set of TN
   mapping and voice routing protocols cannot be seamlessly deployed in
   an international setting.  There should be no need for special
   handling per country-code in the routing logic.

   Consider the case of a pan-European mobile operator Foo. If Foo has
   signed a peering agreement with a local Austrian VoIP operator Bar,
   then Bar should pass all calls over this link that terminate in any
   GSM network that Foo operates.  Ideally, Bar should notice when a
   number was ported to Foo's network in Germany and adapt the routing.
   If Foo acquires a new network in, say Bulgaria, then Bar should
   automatically route all calls to that set of numbers over the peering
   with Foo. All this should happen without Bar having to participate in
   Germany- or Bulgaria-specific TN database exchanges.

   All this has been standard in Internet-based communication: both BGP,
   as well as, application layer protocols like SMTP or HTTP do not care
   about national borders.  The protocol to resolve a .com name is the
   same as the one to resolve a domain under .cn.  BGP speakers announce
   routes without any regard for national borders.  Speermint should
   strive to achieve the same level of universality.

   This does not preclude local optimizations.  For example, if the
   mapping from TN to some sort of routing identifier is done by
   Infrastructure ENUM, then it makes sense to pro-actively prime the
   SSP name-servers with the data for all local numbers.

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5.  A Vision

   This section provides a "view from 20,000 feet" on how the
   interconnection of voice networks might look like in the future.

5.1.  The Players

   Historically, voice interconnection used to be the domain of
   commercial voice network operators.  Direct interconnection between
   corporate PBX systems was the exception as such links required
   dedicated circuits.  With the move to IP based phone networks, this
   is bound to change.

   Right now, there is no protocol support for controlled, secure, and
   dynamic peering between enterprises.  Once this is made possible, the
   voice interconnection landscape is bound to change dramatically.

   The future might look like a mixture between the email and the IP
   world: The majority of end-users will still contract voice service
   from large operators (either the same providers where they buy basic
   connectivity, or independent, voice-only operators).  Smaller
   companies might run their own PBX (which buy upstream "minutes" from
   operators) or just buy hosted voice services.  Larger PBX
   installations will be (from a technology point of view)
   indistinguishable from smaller SSP.

5.2.  Interconnection fabrics

   How will these voice networks interconnect?  An "interconnection
   fabric" will be any physical or logical arrangement where two or more
   voice operators interconnect.  There will be a number of very
   different such fabrics.  No single one will be dominant.

5.2.1.  Old-style TDM based interconnection

   The SS7 technology will be here to stay for quite some time.

5.2.2.  Private SIP interconnection

   Whenever two SSPs use a dedicated IP link between their networks to
   exchange calls, this is a trivial interconnection fabric.  Such links
   will be used for PBX to Upstream-SSP connections, as well as for
   carrier to carrier peering, just like private peering links in the
   Layer 3 world.

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5.2.3.  Commercial Peering Fabrics

   Just as Internet Exchange Points have sprung up to simplify
   interconnection on layer 3, similar dedicated SIP interconnection
   facilities have appeared.  These provide optimal conditions for
   secure, QoS enabled L3 connections, coupled with support
   infrastructure like directory services or SIP scrubbing.

5.2.4.  Overlay networks

   Instead of building a physical peering fabric, the public Internet
   can be used to build a logical peering fabric.  That can been done
   with VPN technologies or just by implementing suitable access control
   on SBEs.

5.2.5.  RFC3263-style SIP

   The set of all SSPs implementing SIP according to RFC 3263 also forms
   a peering fabric.

5.3.  An internet of Voice Networks

   Just as the Layer 3 Internet is built by linking individual networks
   together, the world-wide voice network will similarly emerges as the
   meta-network that comprises all the member networks.

   Such a meta-network (or a lower-case 'I' internet) needs as

   o  a common addressing scheme
   o  an inter-domain routing protocol

5.4.  The Lookup Function (LUF)

   The first step in all call processing is the mapping from telephone
   number to the Destination-Group (see Section 4.4).

   This mapping step is performed not only by commercial voice
   operators, but also by PBX installations, if they have more than just
   one upstream interconnection provider.  As argued in Section 4.5,
   this needs to be an on-demand lookup into an external database.  That
   database needs to accept queries from such a large number of
   legitimate questioners that it is unfeasible to keep that data

   Regarding who can provision entries into that database, some more
   restrictions seem plausible, e.g. all registered service providers
   who directly provide service to the customer using that number.

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   The logical choice for the LUF is thus public Infrastructure ENUM.

   In the case of URI-based dialing, ENUM cannot be used.  We thus also
   need some sort of LUF which maps domain-names to Destination-Group.
   This assumes that all SIP URIs with the same domain-name will be
   routed the same path.  That is a reasonable design decision.

   A trivial approach would be just to re-use the domain-name as the
   Destination-Group.  That works fine for carrier-owned domains, but
   raises significant scaling issues if customer-owned domains are used
   in AoRs.  It is thus more sensible to define a simple DNS Resource
   Record which indicates the Destination-Group for SIP URIs with this

   As a corollary, the I-ENUM lookup could just as well return AoR SIP
   URIs and only a second lookup into the DNS yields the Destination-

5.5.  The Location Routing Function (LRF)

   The second step in the call routing algorithm is the next-hop
   selection based on the Destination-Group the LUF generated.

   For each individual call, this is simply a lookup in a routing-table
   which maps Destination-Group to one (or more) sets of "session
   establishment data (SED)" records which describe the next SIP hop for
   this call.  This SED might contain such elements like IP address of
   an SBE, TLS parameters to use, QoS parameters, transcoding
   instructions, bandwidth limits and whatnot else.

   This is straight-forward, and very much like the IP routing lookup
   into the forwarding routing table.  The tricky question is how this
   routing table is constructed.  There are multiple possibilities:

   Static / Manual routing

      A human operator can configure individual routing rules into the

   A dynamic routing protocol spoken between two connected networks

      Here the human operator just configures basic rules concerning
      which neighbors his systems should talk to and under which
      constraints they should accept and prioritize routing information,
      as well as what routes they should announce.  This would be the
      voice meta-network's equivalent to the BGP of the Layer 3

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      Transit Routing is possible with such a protocol.

   Route registries, e.g. run by Peering Fabrics

      Support systems built into fabrics might act as a routing
      clearinghouse which can reduce the n-squared complexity of a full
      routing mesh on peering exchanges.  The Layer 3 analogy would be
      route-reflectors run by Internet exchange points.  Such
      clearinghouses work fine for pure peering relationships, but have
      a hard time handling transit routing.

   A dynamic routing protocol is vastly preferable to manual routing.
   The data exchanged in such a protocol could to be structured roughly
   in the following way:


      The key into the routing table is the Destination-Group (the
      result of the LUF).


      This is the information the device handling the call needs to
      generate SIP messages.

   Route metadata

      This information is needed to determine whether this route is
      acceptable for the SSP receiving the route as well as for
      priorizing alternative routes.  It might contain:

      *  Network path (equivalent to the BGP AS-path)
      *  Re-distribution restrictions
      *  What media-types are supported
      *  Capacity-related data
      *  Regulatory data (what kind of networks are involved?)
      *  QoS offered by the route
      *  Trust (e.g. does the destination require some reputation
         network score for the caller?  SIP-Identity?)
      *  Commercial terms

   Poor-man's multihoming of an enterprise PBX could be implemented by
   allowing the LUF to return multiple Destination-Groups.

6.  Design Pitfalls

   Reading documents from various sources, a few common design mistakes

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   need to mentioned and cautioned against:

6.1.  Reliance on fabric internals for multi-hop routing

   The cardinal mistake here is to assume that a single peering /
   interconnection fabric will be the sole means of interconnection
   between SSPs.  A good example is the design of the IPX (see
   http://www.gsmworld.com/documents/ireg/ir3444.pdf) and the idea of
   transparent SIP proxies within that network.  From the scarce public
   documentation available it seems like the L3 paths selected via BGP
   routing should determine which (transparent) SIP proxies need to take
   care of transit agreements.

   This can work in theory as long as *all* possible destination
   networks partake (i.e. the IP addresses associated with SBEs of the
   target network) in the BGP routing cloud of the IPX.  Even simple
   non-IPX cross-links between two carriers mess up that scheme, let
   alone the idea that enterprises will also want to do their own call

   Generally speaking, a routing protocol within an interconnection
   fabric cannot replace a routing protocol which finds the right fabric
   over which to route the call.

6.2.  Mistaking a protocol for softswitch/SBE control for LUF/LRF

   The LUF and LRF steps need to happen somewhere in the SSPs network.
   They don't necessarily need to be done directly on the device which
   actually generates SIP messages.  Neither need the inter-domain
   routing data exchanges be done between the actual SBEs.  There
   certainly are good arguments for that, but this is not a given.  (In
   the Layer 3 Internet, this is the common practice: the actual routers
   which do the IP packet routing speak BGP.)

   For voice routing, it may well be a good choice to centralize the
   routing protocol handling and call routing decision processing to
   dedicated servers which are not in the SIP call path themselves.  In
   such a setup, the soft-switch / SBE needs to query this central
   instance on what to do with each individual call.

   That protocol is neither LUF, nor LRF.  It is something completely

   What does this protocol need?  The device needs to pack up all
   information it has on that call (source URI, source "trunk", media
   requested, destination TN / URI, softswitch ID) into a query, send
   that to the central routing instance.  The answer needs to contain
   basically the SED (which set of attribute-value pairs) which tells

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   the softswitch exactly what do to.

   Good role models are access servers for dial-in application and the
   RADIUS protocol used there.  ENUM is not suited for this.  A number
   of recent I-Ds proposing extensions to the basic ENUM algorithm (e.g.
   source variability, encoding calling-party-ID in the query) and a
   long list of URI parameters show that people have tried to force ENUM
   to do something it wasn't design to do.  One should use Radius,
   Diameter, or even MGCP as inspiration on how this can be implemented.

6.3.  Mixing LUF and LRF

   There is a good reason for the LUF/LRF separation.  The two functions
   solve different problems and require different approaches (simple
   lookup vs. routing protocol).

   Just as with normal network layering, it is sometimes tempting to
   violate the layering principles.  For example, when the focus is not
   so much on full-scale transit-capable VoIP routing as on very simple
   SIP peering, then folding the LRF into the LUF can seem like a good

   As with all layering-violations, the perceived simplification quickly
   turns into a mess of workarounds once the scenario expands beyond the
   trivial case.  In this case, just consider the effect of number
   portability on the routing table size.

6.4.  Provisioning vs. routing

   With the establishment of the DRINKS working group, data-exchange and
   "provisioning" in the context of VoIP peering are now subject to IETF
   work.  The data-exchanges needed for LUF and LRF are very different:

   To provision and implement the LUF, we need:
   o  a provisioning protocol which adds TN->Destination-Group mappings
      to a registry
   o  a query protocol into that registry
   o  a database replication protocol to make local copies of LUF
      registries possible

   The data-exchange for the LRF is:
   o  Routing data updates between peering SSPs or a fabric's route
      aggregator (a routing protocol)

   These are two completely different types of data exchanges.  The LUF
   side is close to what needed to run the DNS, the second is a routing
   protocol like BGP.  Trying to find a unified solution to these two
   problems is futile.

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6.5.  RFC 3263 is part of the problem

   RFC 3263 hardcodes the email model of interconnection.  With respect
   to speermint, it is part of the problem, and not part of the

6.6.  Disregarding enterprises

   As mentioned above (Section 3.2), PBX interconnection used to be rare
   and utilized completely different technology than carrier to carrier

   This used to make sense in the TDM world, but in a VoIP landscape it
   is essential that enterprises also get the tools to simplify peering.
   As this is pure settlement-free peering, there is a huge potential
   for ad-hoc, dynamic peering if the protocols coming out of speermint
   and DRINKS support that in a secure manner.

   The IETF should take great care that these two workinggroups do not
   intentionally exclude the requirements from enterprise peering from
   their work.

7.  What building-blocks are missing?

   A mentioned before, the LUF lookup could be implemented via
   Infrastructure ENUM, so no new protocol work is needed in that case,
   only a new enumservice-type needs to be defined.  More work is needed
   for other parts of the framework:

7.1.  LUF / I-ENUM provisioning

   EPP combined with the ENUM extension is a possible solution.  Some
   enhancements (e.g. block provisioning) and better support for the
   typical number portability operations might be needed, though.  There
   has been good input into the DRINKS WG on that topic.

7.2.  LRF Routing announcements

   There currently is no IETF protocol suitable for this data-exchange.

7.3.  A call-control protocol

   As mentioned above (Section 6.2), Radius or similar protocol might be

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8.  Security Considerations

   Not applicable at this stage of the discussion.

9.  IANA Considerations

   Not applicable.

10.  Acknowledgements

   The ideas expressed in this draft evolved during discussions with a
   large number of people.  Version -01 includes significant input from
   Hannes Tschofenig.

11.  References

11.1.  Normative References

   [1]  Faltstrom, P. and M. Mealling, "The E.164 to Uniform Resource
        Identifiers (URI) Dynamic Delegation Discovery System (DDDS)
        Application (ENUM)", RFC 3761, April 2004.

   [2]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
        Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
        Session Initiation Protocol", RFC 3261, June 2002.

   [3]  Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
        (SIP): Locating SIP Servers", RFC 3263, June 2002.

11.2.  Informative References

   [4]  Rosenberg, J., Salama, H., and M. Squire, "Telephony Routing
        over IP (TRIP)", RFC 3219, January 2002.

   [5]  Peterson, J. and C. Jennings, "Enhancements for Authenticated
        Identity Management in the Session Initiation Protocol (SIP)",
        RFC 4474, August 2006.

   [6]  Rosenberg, J. and C. Jennings, "The Session Initiation Protocol
        (SIP) and Spam", RFC 5039, January 2008.

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Author's Address

   Otmar Lendl
   enum.at GmbH
   Karlsplatz 1/9
   Wien  A-1010

   Phone: +43 1 5056416 33
   Email: otmar.lendl@enum.at
   URI:   http://www.enum.at/

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