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     Internet Draft                                                 A.Uzelac
     SPEERMINT                                               Global Crossing
     Intended status: Informational                                  Y.Lee
     Expires: Jan 2007                                               Comcast
                                                             Kayote Networks
                                                                     E. Katz
                                                               July 16, 2007
                             VoIP SIP Peering Use Cases
     Status of this Memo
        By submitting this Internet-Draft, each author represents that any
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        This Internet-Draft will expire on Jan 16, 2007.
     Copyright Notice
        Copyright (C) The IETF Trust (2007)
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        This document will capture VoIP use case for SIP Peering.  It is a
        consolidation of Speermint use cases drafts. This documents depicts
        many common VoIP peering use cases. These use cases are categorized
        into three types: Direct, Indirect and Assisted. They are not the
        exhaust set of use cases but the most common use cases deployed in
        production today. This document captures them to provide a reference.
     Table of Contents
        1. Introduction...................................................3
        2. Terminology....................................................3
        3. Use Cases......................................................5
           3.1. Direct Use Cases..........................................6
  Administrative characteristics........................7
  Options and Nuances...................................7
           3.2. Indirect Use Cases........................................8
           3.2.1. Transit PSP.............................................8
  Administrative Characteristics........................9
           3.3. Assisted Use Cases.......................................10
           3.3.1. Assisted PSP (Direct)..................................10
           3.3.2. Assisted PSP (Hub-and-Spoke)...........................11
        4. Federations...................................................12
           4.1. Federation Considerations................................12
           4.2. Federation Examples......................................14
           4.2.1. Trivial Federations....................................14
           4.2.2. Access List based......................................14
           4.2.3. TLS based Federations..................................15
           4.2.4. Central SIP Proxy......................................15
  Architecture, scalability and business scalability...15
           4.2.5. Private Layer 3 Network................................16
           4.2.6. Peer to Peer SIP.......................................16
           4.2.7. DUNDi..................................................16
        5. Security Considerations.......................................17
        6. IANA Considerations...........................................17
           Normative References..........................................18
           Informative References........................................19
           Author's Addresses............................................19
           Full Copyright Statement......................................20
           Intellectual Property.........................................20
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     1. Introduction
        This document attempts to capture VoIP use cases for Session
        Initiation Protocol (SIP)[1] based peering.  These use cases will
        assist in identifying requirements and future works for VoIP Peering
        using SIP.
        Only use cases related to VoIP are considered in this document.
        Other real-time SIP communications use cases, like Instant Messaging
        (IM) and presence are out of scope for this document.  In describing
        use cases, the intent is descriptive, not prescriptive.
        There are existing documents [2][3][4][5][6] that have captured use
        case scenarios.  This draft draws from those documents.  The document
        contains three categories of use cases; Direct, Indirect and
        Assisted.  The use cases contained in this document attempts to be as
        comprehensive as possible, but should not be considered the exclusive
        set of user cases.
     2. Terminology
        Many terms in this document are referenced to the Speermint
        terminology draft [15]. We also use few additional terms to describe
        the VoIP use cases. We define them in this section.
        o Peering Service Provider (PSP): A server prvoider that provides
           peering functions for Indirect and Assisted peerings. PSP serves
           as “Proxy” which will neither originate nor terminate any SIP
           message. It solely serves as a transit domain to bridge the SIP
           messages between two SSPs. SSPs are PSP’s clients. In contrast,
           SSP’s clients are normally SIP endpoints which originates and
           terminates SIP messages.
        o Direct PSP (D-PSP): PSP providing location function or service
           enabling direct peering relationship.
        o Assisting PSP (A-PSP): An Assisting SSP is some entity that
           employs a central SIP proxy (which is not itself a SSP) to bridge
           calls between participating networks.
        o SIP Proxy: A SIP proxy is the entity responsible for routing the
           SIP messages from or to other SIP User Agent and SIP Proxy.
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        o Location Server (LS): A server called upon by originating SSP,
           either Local or Remote, to obtain the Session Establish Data
           (SED). Often, the input to the server is an E.164 number and the
           SED is a SIP URI. The originating SSP's client may call the
           Location Function using ENUM Query/Response, SIP Invite/Redirect,
           or other method depending on orginating SSP's infrastructure and
           methods available for the data being interrogated, with the
           response format being appropriate to the Query format. In the
           case of an ENUM Query, the response should be a NAPTR record
           containing the sip URI that can be resolved by the client. In the
           case of a SIP Invite/Redirect, the response should be a SIP
           Redirect (30X) message containing the URI.
        o Session Manager (SM): A SM is the home registrar of the user
           endpoint. SM is responsible to receive and send SIP messages from
           the peer. If the user endpoint speaks non-SIP, SM will translate
           the non-SIP protocol to SIP protocol and vice versa.
        o User Endpoint (UE): User Endpoint is the client that makes or
           receives calls. UE can be sip based or non-sip based. For non-sip
           based UE, SM acts as a signaling gateway and translates the non-
           sip signaling to sip signaling before sending to SBE.
        In this document, we use “O” to indicate “Originating”, “T” to
        indicate “Terminating”, and “t” to indicate “Transit”. For example,
        O-SBE is the acronym of Originating SBE.
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        |              \          Transit Domain           /             |
        |               \                                 /              |
        |                \       +------+ +------+       /               |
        |                 \      + t-LS + + t-SM |      /                |
        |                  \     +------+ +-----++     /                 |
        |                   \    +------+ +------+    /                  |
        |           +------+ \   | t-SBE| | t-DBE|   /+------+           |
        |     +-----+ O-LS +  \  +------+ +------+  / + T-LS +-----+     |
        |     |     +------+   \                   /  +------+     |     |
        |     |                 \                 /                |     |
        |     |                  \               /                 |     |
        |     |     +------+      \             /     +------+     |     |
        |     |     | O-SBE+       \           /      + T-SBE|     |     |
        |     |     +---+--+        \         /       +------+     |     |
        |     |         |            \       /                     |     |
        |     |         |             \     /                      |     |
        |     |     +---+--+           \   /          +------+     |     |
        |     +-----+ O-SM |            \ /           | T-SM +-----+     |
        |           +-----++             +            ++-----+           |
        |  +----+         |              |             |         +----+  |
        |  |O-UE+---------+              |             +---------+T-UE|  |
        |  +----+         +------+       |      +------+         +----+  |
        |                 | O-DBE|       |      | T-DBE|                 |
        |                 +------+       |      +------+                 |
        |     Originating Domain         |        Terminating Domain     |
        Figure 1 Generalized Overview
        PLEASE NOTE: In figure one – the elements defined are optional in
        many use cases.
     3. Use Cases
        Use cases are sorted into 3 general groupings: Direct, Indirect and
        Assisted. Though there may be some overlap among the use cases in
        these categories, there are different requirements between the
        scenarios and this document serves to help identify the requirements
        for SIP Peering for VoIP.
        Per information in the Speermint terminology draft, the direct use
        cases involve those cases in which two service providers interconnect
        without using an intervening layer 5 network.  This approach is also
        considered a bi-lateral peering agreement.
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        Indirect or transit peering involves a third party proxying both
        signaling and bearer between the Originating and Terminating Domains.
        It is generally required that a trust relationship is established
        between the originating service provider and the transit network on
        one side, and the transit network and the termination network on the
        other side, so there is no requirement for a trust relationship
        directly between the originating and terminating.
        Assisted use cases involve the use of a third party for signaling.
        This third party may or may not have a pre-existing relationship with
        the O-SSP, and/or T-SSP. The A-PSP may only provide next-hop
        discovery for the O-SSP on behalf of the T-SSP and proxy all
        communications, or may be more intimately involved by maintaining
        session state in the signaling plane. Other functions which may be
        provided in Assisted Peering include, peering policies, and
        administrative rules for such sessions (settlement, abuse-handling,
        security requirements) and the specific rules for the technical
        details of the interconnection (signaling, media, layers 1-4 etc.)
     3.1. Direct Use Cases
        There are intra-domain message flows within the use cases to serve as
        supporting background information.  Only inter-domain communications
        is germane to Speermint.
          1. O-UE initiates a call via SIP INVITE
          2. O-SM queries for next-hop information from a routing database.
          3. Routing database entity replies with route to called party
          4. Call sent to terminating domains session manager.
          5. Session manager determines called party status and directs call
             to called party.
          6. RTP is established between O-UE and T-UE.
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         |    Orig Domain   |    Term Domain    |
         |     +--------+   |     +--------+    |
         |     |  O-LS  |   |     |  T-LS  |    |
         |     +--------+   |     +--------+    |
         |  (2) /           |                   |
         |   /(3)           |                   |
         |  +-----+         |          +-----+  |
         |  |O-SM |--------(4)---------|T-SM |  |
         |  +-----+         |          +-----+  |
         |      |           |             |     |
         |     (1)          |            (5)    |
         |      |           |             |     |
         |   +----+         |           +----+  |
         |   |O-UE+===(6)=(RTP)=========+T-UE+  |
         |   +----+         |           +----+  |
        Figure 2 Direct Peering
  Administrative characteristics
        The direct use case is typically implemented in a scenario where
        exists a strong degree of trust between the 2 administrative domains.
        Both administrative domains normally sign a peer agreement which
        state clearly the peering policies and terms.
  Options and Nuances
        In Figure 2, O-SM and T-SM connect directly. An operator may decide
        to deploy a SBE between its SM and the peer network. Normally, the
        operator will deploy the SBE in the edge of its administrative
        domain. The signaling traffic will pass between two networks through
        the SBE. The operator has many reasons to deploy a SBE. For example,
        the SM may use RFC1918 addresses that are not routable in the peer
        operator network. The SBE can perform NAT function. Also, the SBE
        eases the operation cost for deploying old or removing new SM.
        Consider the deployment architecture where multiple SMs connect to a
        single SBE. An operator can add or remove a SM without coordinating
        with the peer operator. The peer operator sees only the SBE. As long
        as the SBE maintain intact, the peer operator does not need to be
        When an operator deploys a SBE, the operator is required to advertise
        the SBE to the peer LS so that the peer operator can locate the SBE
        and route the traffic to the SBE accordingly.
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        SBE deployment is a decision within an administrative domain. Either
        administrative domain or both administrative domains can decide to
        deploy SBE. To the peer network, most important is to identify the
        next-hop address. Whether next-hop is SM or SBE, the peer network
        will not see any difference.
     3.2. Indirect Use Cases
     3.2.1. Transit PSP
        This call flow is similar to the direct approach, but with a Peering
        Service Provider (PSP) providing signaling(i.e. SIP “normalizing”)
        and bearer (i.e. transcoding) services to facilitate communications
        between the originating and terminating domains.  For this call flow
        all signaling and bearer to and from the Originating/Terminating
        domains traverses the Transit Domain, possibly for services like Q0S,
        interoperability and security.
          1. O-UE initiates a call.
          2. The O-SM performs next-hop determination for the called party
             via the LS within the Transit domain.  This can be done via
             ENUM/DNS/Redirect 3XX multiple choices and/or static routing.
          3. The result of the query will be the transit provider’s SBE (t-
             SBE) that is interconnected to the transit domain via the O-SBE.
          4. O-SM signals the t-SBE via the O-SBE.
          5. t-SBE routes call to T-SBE within terminating domain.
          6. T-SBE signals T-SM.
          7. T-SM signals the called party, T-UE.
          8. RTP is established between UEs via DBE path typically
             coordinated by the Transit Domain.
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                           |   Transit Domain |
                           |                  |
                           |       +------+   |
                           |    +--+ t-SM |   |
                           |   / +-+ t-LS |   |
                           |  / /  +------+   |
        +------------------+ / /              +----------------------+
        |  Orig Domain     |/ /               |      Term Domain     |
        |      +-----------+ /                |         +--------+   |
        |     /            |/                 |         |  T-LS  |   |
        |    /  +----(3)---+                  |         +--------+   |
        |  (2) /           |                  |                      |
        |  /  /            |                  |                      |
        |+-----+     +-----+      +-----+     +-----+         +-----+|
        ||O-SM |-(4)-|O-SBE|------+t-SBE+-(5)-+T-SBE+---(6)---|T-SM ||
        |+-----+     +-----+      +-----+     +-----+         +-----+|
        |    |             |                  |     |            |   |
        |   (1)            |                  |     |           (7)  |
        |    |             |                  |     |            |   |
        | +----+     +-----+      +-----+     +-----+          +----+|
        | |O-UE+=====|0-DBE|=(8)==+t-DBE+=====+T-DBE+==========+T-UE||
        | +----+     +-----+      +-----+     +-----+          +----+|
        Figure 3 Indirect via Transit PSP
  Administrative Characteristics
        The transit peering use case is normally implemented in cases where
        no direct interconnection exists between originating and terminating
        domains due to either business or physical constraints.
        Orig Domain .--. Transit = Relationship O-T
        In the O-T relationship, typical policies, features or functions that
        deem this relationship necessary are NP, Ubiquity of termination
        options, and masquerading of originating VoIP network gear.
        Term Domain .--. Transit = Relationship T-T
        In the T-T relationship, typical policies, features or functions
        observed consist of codec “scrubbing”, anonimizing, and transcoding.
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     3.3. Assisted Use Cases
        Assisted use cases involve an assisting PSP (A-PSP) that facilitates
        direct session establishment between the O-SSP and T-SSP.  There may
        be elements that provide SIP proxy functionality, and are often
        implemented in practice by SBE(s) and DBE(s) which may "filter" or
        "normalize" and provide network-hiding for incoming messages en route
        to their final destination.  Fear and distrust coupled with continued
        interoperability and security concerns have revived the need for the
        neutral central element role enabled by this peering model.
        Popularity of this model can be attributed to the concentration of
        functions provided by A-PSP.  As an external element, A-PSP can
        provide the full set of services for SSPs, and through its own
        relationships with the SSP, eliminate the need of all SSPs for pair-
        wise service relationships.  A-PSP can potentially encompass a large
        namespace of users that is accessible in one query to external SSP
        members (or not -depending on policy).
        In addition there is an interoperability function usually performed
        by an SBE, almost guaranteeing interoperability and protocol
        interchangeability between member SSPs.  As part of the
        interoperability there is also is media sub-function enabling the
        federation to enforce a standard set of codecs or alternatively
        provide transcoding functionality to make sure there is media
        interoperability as well. Finally, A-PSP can implement the routing
        function enabling traffic shaping and throttling across the
     3.3.1. Assisted PSP (Direct)
        This is a direct call flow, as in the minimalist approach, but with
        an A-PSP aiding the originating to terminating domain relationship.
        The A-PSP may have a relationship with the originating and/or
        terminating domain.
          1. O-UE initiates a call.
          2. The O-SM performs next-hop determination for the called party
             via the A-LS within the Assisting domain.  This can be done via
             ENUM/DNS/Redirect 3XX multiple choices and/or static routing.
          3. The result of the query will be the T-SBE that is accessible via
             the O-SBE. There must be a common IP denominator between the
             originating and terminating domains. (i.e. Internet)
          4. Signaling will traverse the O-SM onwards to the O-SBE.
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          5. O-SBE routes call to T-SBE.
          6. T-SBE signals T-SM.
          7. T-SM signals the called party, T-UE.
          8. Bearer path established between O-UE and T-UE through O/A/T DBE.
                        |     Assist Domain      |
                        |                        |
                        |       +--------+       |
                        |       |  A-LS  |       |
                        |       ++---+---+       |
                        |        |   |           |
        +---------------+        |   |           +-----------------+
        |    Orig Domain \       |   |          /   Term Domain    |
        |      +----------+------+   |         /     +--------+    |
        |     /            \         |        /      |  LS-t  |    |
        |    /  +----(3)----+--------+       /       +--------+    |
        |  (2) /             \              /                      |
        |  /  /               +------------+                       |
        |+-----+        +-----+            +-----+         +-----+ |
        ||O-SM |---(4)--|O-SBE+-----(5)----+T-SBE+---(6)---|T-SM | |
        |+-----+        +-----+            +-----+         +-----+ |
        |    |                |            |                  |    |
        |   (1)               | (common IP |                 (7)   |
        |    |                |denominator)|                  |    |
        | +----+        +-----+            +-----+          +----+ |
        | |O-UE+========+O-DBE+=====(8)====+T-DBE+==========+T-UE| |
        | +----+        +-----+            +-----+          +----+ |
        Figure 4 Direct with Assisted PSP
        PLEASE NOTE – elements depicted are optional.
     3.3.2. Assisted PSP (Hub-and-Spoke)
        A-PSP serves as the hub for multiple SSPs. Each SSP is the spoke to
        the A-PSP which does not need to have direct connection among other
        SSPs. To originate a call to a remote number, the SSP will send the
        SIP request to the A-PSP. A-PSP is the default peer for all the
        numbers that are unknown to the SSP. A-PSP can route the call to one
        of its served SSP or to PSTN if A-PSP can’t locate the next-hop for
        that call in its own LS. The routing logic in the A-PSP is hidden to
        the SSP. Figure 5 shows the high-level network setup.
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                               | Assisted|
                               |   PSP   |
                                  | | |
                                  | | |
                                  | | |
                                  | | |
                                  | | |
                                  | | |
                 +----------------+ | +------------------+
                 |                  |                    |
                 |                  |                    |
                 |                  |                    |
                 |                  |                    |
             +---+----+         +---+----+           +---+----+
             |        |         |        |           |        |
             |  SSP1  |         |  SSP2  |  .......  |  SSPx  |
             |        |         |        |           |        |
             +--------+         +--------+           +--------+
        Figure 5 Hub-and-Spoke Assisted PSP
     4. Federations
          This section discusses the federation concept, explains which
          technical parameters make up the foundation of a federation and
          provides examples.
          Contrary to the previous section, this section does not focus on
          specific implementation details like the presence of SBCs or other
          border elements. The aim here is to provide a broader view on what
          kinds of arrangements are possible.
          The concrete implementation details (e.g. "direct with one SBC"
          versus "direct with two SBCs") can involve all the use cases thus
          far described in the document.
     4.1. Federation Considerations
          Each federation has to specify how a few core operations which are
          to be performed by its members.
          These include:
          1. Peer Discovery
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          This specifies how a SSPs discovers that he can place a specify
          call to a peering partner in this federation.
          Possible solution are e.g.: a manually configured list of TN-
          prefixes and domain names, automatically obtained list of reachable
          prefixes/domains by some sort if intra-federation route
          announcements, trial queries to the federation's LS, trial lookups
          in federation-internal databases (e.g. private DNS),public database
          lookups (e.g. I-ENUM).
          2. Location Server
          What methods are used for TN to URI mapping?
          Examples: Public User-ENUM, public Infrastructure ENUM, private
          ENUM tree, SIP Redirect, DUNDi.
          3. Next Hop Domain Resolution
          If the LS did not return an URI of the form sip:user@IP-address,
          then the originating SSP has to translate the domain part of the
          URI to an IP-address (plus perhaps fall-backs) in order to contact
          the next hop.
          Examples: RFC3263 in the public DNS. RFC3263 in a federation
          private DNS. RFC3263 in the public DNS with split-DNS, P2P SIP,
          modified RFC3263 in the public DNS (e.g. a federation-specific
          prefix to the domain name).
          4. Call Setup
          The federation may also define specifics on what SIP features need
          to be used when contacting the next hop in order to a) reach the
          next hop at all and b) to prove that the sender is a legitimate
          peering partner.
          Examples: hard-code transport (TCP/UDP/TLS), non-standard port
          number, specific source IP address (e.g. in a private L3 network),
          which TLS client certificate to use, other authentication scheme.
          5. Filtering Incoming Calls
          On the receiving side, the border element needs to determine
          whether the INVITE it just received really came from a member   of
          the federation. This is the flip side of 4.
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          Example: verify TLS cert, check incoming interface/VLAN,check
          source IP address against a configured list of valid ones.
     4.2. Federation Examples
          This section lists some examples of how federations can operate.
     4.2.1. Trivial Federations
          A private peering arrangement between two SSPs is a special case of
          a federation. These two SSP have agreed to exchange calls amongst
          themselves and they have set up whatever SBC/LS/SBE plus Layer
          3infrastructure they need to route and complete the calls.
          It is thus not needed to treat bi-lateral peerings as conceptually
          different to federation-based peering.
          On the other extreme, the set of all SSPs implementing an open SIP
          service according to RFCs 3261/3263/3761 also fulfills the
          definition of a federation.  In that case, the technical rules are
          contained in these three RFCs, the LS is the public DNS. Whether
          some of these SSPs use SBCs as border elements is not relevant.
          The administrative model of this federation is the "email model":
          There is no "member list", any SIP server operating on the Internet
          which implements call routing according to these RFCs is implicitly
          a member of that federation. No business relationship is needed
          between "members", thus no money is likely to change hands for
          terminating calls. There is no contractual protection against
          nuisance calls, SPIT, or denial of service attacks.
     4.2.2. Access List based
          If running an open SIP proxy is not desired, then a group of SSPs
          which want to allow calls from each other can collect the list of
          IP addresses of all their border elements.
          This list is redistributed to all members which use it to configure
          firewalls in front of their ingress elements.  Thus calls from
          other members of this federation are accepted while calls from
          other hosts on the Internet are blocked.
          Whether SSPs deploy SBCs as border elements is not relevant.  Call
          routing can still be done via standard RFC rules.
          Whenever a new member joins this club every other SSP needs to
          adapt its filter rules.
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     4.2.3. TLS based Federations
          Another option to restrict incoming calls to federation members is
          to use Transport Layer Security (TLS) certificates as access
          control. This works best if the federation runs a certificate
          authority (CA) which signs the TLS keys of each member SSP.  Thus
          the ingress element of a SSP needs to check only whether the client
          certificate presented by the calling SIP proxy contains a proper
          signature from that CA.
          Adding support for Certificate Revocation Lists solves the issue of
          blocking calls from former members of that federation.  The main
          benefit of this model is that no changes need to be made at the
          ingress element of all old members whenever a SSP joins that
     4.2.4. Central SIP Proxy
          One way to simplify the management of these firewall rules is to
          route all SIP messages via a central proxy.
          In that case, all federation members just need to open up their
          ingress elements to requests from that central server. A new SSP
          just triggers a change in the configuration of this box and not at
          all other SSPs.
          While centralized solutions may entail typical hub-and-spoke
          architecture considerations, the added overall federation
          scalability with respect to the number of interconnects required,
          their associated policies and management make this approach quite
          popular today.
          This is an example of Assisted Peering.
  Architecture, scalability and business scalability
          The network architecture which in the case centralized model would
          reflect a hub and spoke model - should be weighed against a
          distributed model. While such a centralized model presents well-
          known network and server scalability challenges, a distributed
          model requires higher interconnection complexity, reflected in
          provisioning and the need for the maintenance of such
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     4.2.5. Private Layer 3 Network
          Federations can also establish a separate layer 3 network for their
          peering traffic. This could be implemented e.g. by creating a new
          VLAN at an Internet exchange point to which all members of that
          federation connect their SBEs.
          Alternatively, a federation can establish a smaller version of the
          Internet to which only members are allowed to connect.  The GRX
          network of the mobile operators is an example of a dedicated layer
          3 infrastructure.
          Such a private layer 3 network can also be implemented using
          virtual private network (VPN) technologies like IPsec.
          In all these cases the SBE can assume that any SIP requests it
          receives via an interfaces located in this L3 network comes from
          legitimate peering partner.
          The separation of the peering network from the Internet makes it
          easier to protect the peering arrangement from attacks and to
          ensure QoS.
     4.2.6. Peer to Peer SIP
          P2PSIP replaces the RFC3263 rules by a lookup in a distributed hash
          table (DHT). A federation could use this technology to implement
          call routing between the peers: the border elements of all members
          participate in the DHT algorithm and distribute routing information
          this way.
          Only members of the federation thus can use information stored in
          the DHT which could be the basis of both call routing within the
          federation as well as access control between members.
     4.2.7. DUNDi
          Distributed Universal Number Discovery (DUNDi)
          [http://www.dundi.com/dundi.txt] can also be used to build
          federations: DUNDi itself acts as a distributed LS which can add
          dynamically generated passwords to the URIs it returns.
          This way, the T-SBE can verify that an incoming calls comes from a
          member of this DUNDi cloud.
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     5. Security Considerations
          This document introduces no new security considerations.  However,
          it is important to note that session interconnect, as described in
          this document, has a wide variety of security issues that should be
          considered in documents addressing both protocol and use case
     6. IANA Considerations
          This document creates no new requirements on IANA namespaces
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     Normative References
        [1]   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.
        [2]   Schwartz, David, draft-schwartz-speermint-use-cases-federations
        [3]   Mahy, Rohan, draft-mahy-speermint-direct-peering
        [4]   Lendl, Otmar, draft-lendl-speermint-federations
        [5]   Lee, Yiu, draft-lee-speermint-use-case-cable
        [6]   Uzelac, Adam, draft-uzelac-speermint-use-cases
        [7]   Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
              (SIP): Locating SIP Servers", RFC 3263, June 2002.
        [8]   Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J., and
              T. Wright, "Transport Layer Security (TLS) Extensions", RFC
              3546, June 2003.
        [9]   Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
              "RTP: A Transport Protocol for Real-Time Applications", STD 64,
              RFC 3550, July 2003.
        [10]  Peterson, J., Liu, H., Yu, J., and B. Campbell, "Using E.164
              numbers with the Session Initiation Protocol (SIP)", RFC 3824,
              June 2004.
        [11]  Peterson, J., “Address Resolution for Instant Messaging and
              Presence”,RFC 3861, August 2004.
        [12]  Peterson, J., "Telephone Number Mapping (ENUM) Service
              Registration for Presence Services", RFC 3953, January 2005.
        [13]  ETSI TS 102 333: " Telecommunications and Internet converged
              Services and Protocols for Advanced Networking (TISPAN); Gate
              control protocol".
        [14]  Peterson, J., "enumservice registration for Session Initiation
              Protocol (SIP) Addresses-of-Record", RFC 3764, April 2004.
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     Informative References
        [15]  Meyer, D., "SPEERMINT Terminology", draft-ietf-speermint-
              terminology-06 (work in progress), 2006.
        [16]  Mule, J-F., “SPEERMINT Requirements for SIP-based VoIP
              Interconnection”, draft-ietf-speermint-requirements-00.txt,
              June 2006.
        [17]  Camarillo, G. “Requirements from SIP (Session Initiation
              Protocol) Session Border Control Deployments“, draft-camarillo-
              sipping-sbc-funcs-04.txt, June, 2006.
        [18]  Habler, M., et al., “A Federation based VOIP Peering
              Architecture”, draft-lendl-speermint-federations-03.txt,
              September 2006.
     Author's Addresses
        Adam Uzelac
        Global Crossing
        Email: adam.uzelac@globalcrossing.com
        Rohan Mahy
        Email: rohan@ekabal.com
        Yiu L. Lee
        Comcast Cable Communications
        Email: yiu_lee@cable.comcast.com
        David Schwartz
        Kayote Networks
        Email: david.schwartz@kayote.com
        Eli Katz
        Xconnect Global Networks
        Email: ekatz@xconnect.net
        Otmar Lendl
        enum.at GmbH
        Email: otmar.lendl@enum.at
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     Full Copyright Statement
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