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IMS Network Signaling Peering: Challenges and Proposal

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    IMS Network Signaling Peering:Challenges and Proposal

    Jean-Philippe Joseph

    IP Multimedia Subsystem (IMS) network peering is a key enabler that willhelp accelerate deployment of next-generation IMS-based networks. Todays

    early deployments of dispersed IMS networks require public switched

    telephone network (PSTN)/public land mobile network (PLMN) bridges for

    network interconnection between IMS islands. The PSTN/PLMN bridging

    arrangement is inefficient, however, in that it results in unnecessary

    settlements for the carriers. It further impedes the implementation of rich

    multimedia and Voice over IP (VoIP)-related services that require end-to-end

    Internet Protocol (IP) connectivity. Last, it perpetuates the reliance on the

    existing PSTN/PLMN network for voice calls among subscribers served by

    different IMS-based carriers. This paper analyzes in detail the IMS peering

    challenges from the perspective of Session Initiation Protocol (SIP) signaling

    peering. It discusses issues related to the routing of SIP messages, and

    addressing, address resolution, and discovery of peering points for IMS

    signaling peering. It further establishes that a new routing algorithm is

    needed that will allow signaling peering points to dynamically discover the

    best transit network among others for reaching a destination. In closing,

    it presents a high-level IMS signaling routing process that includes, among

    other benefits, support for number portability as a key function for

    inter-carrier IMS peering. 2008 Alcatel-Lucent.

    voice, IMS networks essentially rely on the PSTN to

    route voice calls among the IMS islands. That is, calls

    among IMS subscribers served by different SPs are

    most likely routed through the PSTN, resulting in less

    efficient routing, an inability to support end-to-end

    VoIP related services (e.g., wideband voice), unnec-

    essary settlements, and the undesirable impact of

    perpetuating the PSTN.

    Various standards organizations have recently

    begun to tackle crucial aspects of IMS peering,

    including policy, signaling, network, and security

    peering. The International Telecommunication Union,

    Introduction

    Globally, many service providers (SP), wireline

    and wireless alike, are planning or already starting

    to deploy next-generation IP Multimedia Subsystem

    (IMS) [1] networks to evolve from the embedded

    base public switched telephone network (PSTN). The

    IMS networks will, among other things, support rich

    and robust multimedia services including Voice over

    Internet Protocol (VoIP) and, hence, enhance user

    experience.

    These initial IMS networks, however, are being

    deployed in islands with no direct interconnections

    among themselves. For instance, for narrowband

    Bell Labs Technical Journal 12(4), 3348 (2008) 2008 Alcatel-Lucent. Published by Wiley Periodicals, Inc. Publishedonline in Wiley InterScience (www.interscience.wiley.com) DOI: 10.1002/bltj.20265

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    34 Bell Labs Technical Journal DOI: 10.1002/bltj

    Telecommunication Standardization Sector (ITU-T)

    next-generation network (NGN) defined resourceand admission control functions (RACF) require-

    ments [15] for effective resource management in the

    transport layer. The Internet Engineering Task Force

    (IETF) recently created the Session Peering for

    Multimedia Interworking (SPEERMINT) working

    group to define a framework for layer 5 peering

    [18, 20]. The 3rd Generation Partnership Project

    (3GPP*) defines in Release 7 (R7) a new signalingpeering entity: the interworking border control func-

    tion (IBCF) [2]. The GSM Association (GSMA) defined

    guidelines for IMS roaming and interworking [10].

    The European Telecommunications Standards Institute

    (ETSI) addressed issues related to interconnection

    and routing, and their implications on numbering,

    Panel 1. Abbreviations,Acronyms, and Terms

    3GPP3rd Generation Partnership ProjectAAAAuthentication, authorization and

    accountingBGCFBreakout gateway control functionBGPBorder Gateway Protocol

    CdPCalled partyCSCFCall session control functionDDDSDynamic delegation discovery systemDNSDomain name systemE2UE.164 to URI ResolutionENUMElectronic number mappingGPRSGeneral packet radio serviceGRXGPRS roaming exchangeGSMGlobal System for Mobile

    CommunicationsHSSHome subscriber serverIBCFInterworking border control functionI-CSCFInterrogating CSCF

    IETFInternet Engineering Task ForceIMSIP Multimedia SubsystemINIntelligent networkIPInternet ProtocolISDNIntegrated services digital networkISPInternet service providerISUPISDN user partITUInternational Telecommunication UnionITU-TITU Telecommunication Standardization

    SectorLDLong distanceLERGLocal exchange routing guide

    LFLocation functionLIALocation-info-answerLIRLocation-info-requestLRNLocation routing numberMGCFMedia gateway control functionNAINetwork access identifierNAPTRNaming authority pointerNGNext-generationNGNNext-generation network

    NPNumber portabilityNPANumbering plan areaNPDBNumber portability databaseOPEXOperational expendituresPbUIPublic user identity

    P-CSCFProxy CSCFPDAPersonal digital assistantPFPolicy functionPLMNPublic land mobile networkPPPeering pointPrUIPrivate user identityPSTNPublic switched telephone networkQoSQuality of serviceRACFResource and admission control functionRFCRequest for commentsS-CSCFServing CSCFSCTPStream Control Transmission ProtocolSFSignaling function

    SIPSession Initiation ProtocolSLAService level agreementsSLFSubscriber location functionSMSession managerSPService providerSPEERMINTSession Peering for Multimedia

    InterworkingSPNPService provider number portabilitySRVService recordSS7Signaling System 7TASTelephony application serverTCPTransport Control ProtocolTELTelephoneTHIGTopology hiding interface gatewayTLSTransport layer securityUAUser agentUDPUser Datagram ProtocolURIUniform resource identifierVoIPVoice over IPVPNVirtual private network

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    DOI: 10.1002/bltj Bell Labs Technical Journal 35

    naming, and addressing for NGN Telecommunication

    and Internet Converged Services and Protocols for

    Advanced Networks (TISPAN) [6]. The content of this

    paper reflects the work conducted, thus far, by these

    organizations.

    This paper analyzes in detail the IMS peering

    challenges from the perspective of Session InitiationProtocol (SIP) signaling peering. It focuses primarily

    on voice applications and delves into other critical

    networking issues that further complicate the discov-

    ery of signaling peering points for inter-carrier rout-

    ing. Finally, it presents a high-level signaling routing

    process for IMS signaling routing that includes sup-

    port for number portability as a key function for inter-

    carrier IMS peering, and identifies key areas for

    further investigation.

    The paper is organized as follows: We first pro-

    vide background material on why the PSTN is usedtoday for IMS network interconnection and the

    impacts of PSTN bridging. Next, we describe a generic

    case of IMS peering to discuss the critical issues

    related to address resolution and discovery of peer-

    ing points. We follow by presenting other factors

    such as inter-domain IP infrastructure, identification,

    and address formatthat impact IMS peering. We

    describe a proposal for a signaling routing process for

    IMS peering, summarize our conclusions, and outline

    key areas for further investigation.

    Background

    Today, when placing a Voice over IP call to reach

    a destination endpoint served by a different VoIP

    operator, most likely, the call is routed through the

    PSTN. In some cases, even calls between two VoIP

    subscribers served by the same SP are hairpinned over

    the PSTN. The next section examines why it is so.

    Why Are IMS-to-IMS Calls Routed Through the

    PSTN Today?

    One of the key issues that impede direct peering

    among IMS networks is address resolution in supportof routing of SIP signaling messages, particularly dur-

    ing session/call establishment or setup. In the PSTN

    network, call routing is based on global addressing

    using ITU-T E.164 [13] numbers (e.g., NPA-NXX-

    XXXX in North America) to uniquely identify the

    users, and on point codes to identify switching points.

    In addition, the entire process is supported by com-

    mon databases that are shared among carriers, such

    as the local exchange routing guide (LERG), and by

    pre-defined routing tables that associate switches with

    telephone numbers.

    Call routing in IMS, however, is completely

    different from that in the PSTN. First, the currentIMS systems have not deployed a routing capability

    that is fully equivalent to that of the existing PSTN.

    Second, routing in IMS is based on a different entity:

    SIP uniform resource identifiers (URI) [23]. SIP URIs

    are used to identify IMS users and resources (e.g.,

    servers and databases) and have a format similar to

    an email address with a user name and a domain name

    (e.g., sip:[email protected]). The domain por-

    tion identifies the host responsible for handling the

    request for the user. For instance, the domain portion

    may identify the IMS operator serving the particularuser. Hence, during call processing, SIP URIs need to

    be resolved using domain name system (DNS) proce-

    dures, described in [5], in order to determine the

    IP address and port number of the corresponding

    domain name server for the domain portion of the

    SIP URI.

    In the particular case where an E.164 telephone

    number is used to establish a call, IMS invokes tele-

    phone number mapping (ENUM) [7], an application

    of the DNS, to perform the translation of the E.164

    number into a SIP URI. The ENUM protocol uses aspecific naming authority pointer (NAPTR) DNS

    resource record, the E.164-to-URI resolution (E2U),

    as defined in [19], to identify and map the E.164

    number with associated communications services

    and resources (e.g., call server, email). This mecha-

    nism allows a user to reach a destination by dialing

    a telephone number, regardless of the device (e.g.,

    phone, PDA) the destination party might happen to

    be using. In short, in IMS, ENUM/DNS provides an

    essential location functionequivalent to the LERG

    in PSTNfor reaching a user identified with anE.164 number.

    This location function in IMS is at the heart of

    the problem that forces routing through the PSTN for

    IMS-to-IMS calls. Currently, there are several pro-

    posals for ENUM/DNS implementation in support of

    address resolution in IMS networks. However, there is

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    36 Bell Labs Technical Journal DOI: 10.1002/bltj

    no agreement in the industry on a definitive solution.

    These proposals are as follows:

    Global (public) ENUM is dedicated to making

    ENUM data publicly accessible through the

    E164.arpa domain at the public DNS root.

    Private ENUMis owned and controlled by a single

    SP, which determines whether it wants to shareany information with another SP.

    Infrastructure ENUM, also known as carrier ENUM,

    allows a group of carriers to share routing data

    among themselves and to look-up the destina-

    tion addresses of one anothers customers.

    Note that both the private and the infrastructure

    ENUM solutions are outside the scope of global ENUM

    in that they do not use the E164.arpa domain.

    However, they do share the same technological basis

    and conventions as global ENUM.

    Meanwhile, current implementations of IMSnetworks by early adopters leverage private ENUM

    solutions for the most part. That is, the private

    ENUM database is mainly populated with NAPTR

    records about the SPs own subscribers. Hence, for

    voice applications, if the called party (CdP) belongs to

    a different IMS SP domain for which there is no

    associated NAPTR record in the DNS server of the

    originating network, the SIP URI address resolution

    process results in an ENUM failure indicating that the

    IMS domain does not know how to locate and route

    the call to its destination. Today, when such a situa-tion occurs, by default, the IMS network simply

    routes the call to the PSTN. This default routing ends

    up creating a PSTN bridge in between two IMS

    islands, particularly when the destination party is an

    IP endpoint.

    The technical challenges associated with IMS sig-

    naling peering are explored in more detail in subse-

    quent sections. For now, suffice it to say that the PSTN

    bridging model is not financially sustainable over the

    long term, and it is not a technically viable option for

    non-VoIP-only applications. The next section providesan assessment of the impacts of PSTN bridging.

    Impacts of PSTN Bridging

    PSTN bridging in support of next-generation

    applications will have serious financial implications

    for IMS carriers in terms of both lost revenue oppor-

    tunities and the opportunity to reduce operational

    expenditures (OPEX) by retiring the PSTN more

    quickly.

    First, many IP and multimedia services (e.g., active

    phone book, video telephony) requiring end-to-end

    IP connectivity will not work over the PSTN. As a

    result, an SP might be able to offer a particular serv-

    ice to its own subscriber base; however, it will not beable to guarantee that the service will work if the

    other endpoint belongs to a different SP to which it is

    only connected via the PSTN. Hence, there is a poten-

    tial for lost revenue for the SPs. For instance, a VoIP

    call initiated with a non-E.164 address (e.g., SIP URI)

    to another IP endpoint will not work over the PSTN.

    Support for such a service will require the initial SIP

    INVITE message containing the destination SIP URI

    to be encapsulated in ISDN user part (ISUP) and car-

    ried over to the destination IMS domain via the SS7

    network. SIP message encapsulation in ISUP messagesis not supported today and would require further

    development in the PSTN, which is unlikely to occur.

    In general, applications requiring end-to-end IP

    fall into these categories:

    Peer-to-peer applications initiated with non-E.164

    identifiers (e.g., initiating a voice call with the email

    address of the called party).

    Applications requiring SIP signaling.

    Client/server applications that require end-to-end

    IP connectivity (e.g., multiparty gaming).

    Applications requiring broadband access andhigher bandwidth than the 64 Kbps narrowband

    used in the PSTN (e.g., video telephony).

    Second, keeping a call between two IP endpoints

    in the IP domain will help SPs avoid unnecessary set-

    tlements and reduce charges associated with PSTN in-

    terconnections. Moreover, relying on the PSTN to

    support calls among IP endpoints ends up deferring

    the decision to retire the PSTN more quickly, and de-

    lays the opportunity for SPs to start realizing OpEx

    savings for managing one packet network.

    Last, PSTN bridging results in quality of service(QoS) degradation caused by the cascading effect of

    bearer conversions at the edges of the network. That

    is, the bearer signal needs to be converted per ITU-T

    G.711 specifications, thus incurring additional pro-

    cessing delays in the media gateways placed on both

    the ingress and egress sides of the PSTN.

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    Need and Scope

    As SPs continue to deploy IMS networks and

    introduce next-generation (NG) applications to enh-

    ance the end user experience, the need for peering

    among IMS networks and across SP boundaries will

    become critical.

    As shown in Figure 1, IMS peering will occur

    at different layers including network, policy, sig-

    naling, and security. Layer 3 peering is well under-

    stoodIP networks are interconnected today and

    protocols such as Border Gateway Protocol (BGP)

    are well defined to support layer 3 peering. Much

    work has been done in the RACF area, which de-

    fines a framework for policy peering in support of

    QoS [15].

    However, peering at the session/signaling layer

    (or layer 5) is not yet standardized. IETF SPEERMINT

    recently defined a peering architecture [21] between

    two SIP SPs, as depicted in Figure 2. The architec-

    ture identifies the following logical functions:

    Managed IPcore

    PSTN/PLMN

    DNS/ENUM

    DNS/ENUM

    IMSapplication

    IMS domain A IMS domain B

    IMSapplication

    L5 peering

    PEP

    PEPMG

    P/I/S-CSCFOther

    core IMSelements

    RACF

    Managed IPcore PEPPEP

    RACF

    P/I/S-CSCFOther

    core IMSelements

    PEP

    Wirelineaccess network

    Wirelessaccess network

    IMS peeringscope

    Application

    Call/sessioncontrol

    Media/access

    CSCFCall session control function

    DNSDomain name system

    ENUMElectronic number mappingIMSIP Multimedia Subsystem

    IPInternet Protocol

    L5Layer 5

    MGMedia gateway

    PEPPolicy enforcement point

    P/I/S-CSCFProxy/interrogating/serving CSCFPLMNPublic land mobile network

    PSTNPublic switched telephone networkRACFResource and admission control function

    Figure 1.IMS peering scope.

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    38 Bell Labs Technical Journal DOI: 10.1002/bltj

    1. Location function (LF) provides the routing data for

    the purpose of the discovery of the signaling and

    policy functions, and reaching the end user. The

    LF, in fact, uses the ENUM/DNS infrastructure to

    support the peering process.

    2. Policy function (PF) provides a framework for

    authentication and the exchange of policy param-

    eters to be used by the signaling function.

    3. Signaling function (SF) handles the routing of SIP

    messages and assists in the discovery of param-

    eters to be used by the media function.4. Media function (MF) performs media-related func-

    tions including media transcoding between two

    SIP-based SPs networks.

    The scope of our work, in what follows, is lim-

    ited to the LF and SF, as described above.

    IMS-to-IMS Signaling Peering for a Basic

    Voice Call

    In general, the inter-carrier signaling routing

    process in IMS can be divided into the following

    major phases:1. Initialization and address analysis and resolution,

    2. Determination of a peering point in the orig-

    inating network,

    3. Discovery of a peering point in the terminating

    network, and

    4. Routing in the terminating domain.

    The next section discusses in detail the steps

    associated with these phases.

    Signaling Steps

    In this example, each carrier implements a pri-

    vate ENUM/DNS solution in its network, and domain

    A and domain B are the home networks for useragent A (UAa) and user agent B (UAb), respectively.

    For simplicity, the registration and authentication

    steps, which would involve the use of an authentica-

    tion, authorization, and accounting (AAA) server, and

    the home subscriber server (HSS) are not shown. We

    further assume that each domain contains only one

    HSS server, and therefore the subscriber location

    function (SLF) is not required. We also combine the

    three call session control functionsinterrogating

    CSCF (I-CSCF), serving CSCF (S-CSCF), and proxy

    CSCF (P-CSCF)into a session manager (SM). Further-more, we do not represent any element at the appli-

    cation layer (e.g., a telephony application server or

    TAS) that might be invoked during the call setup. Last,

    we do not show the breakout gateway control func-

    tion (BGCF) and media gateway control function

    (MGCF) components, which would be invoked if the

    call were destined for the PSTN/PLMN.

    The signaling routing steps for a voice call between

    UAa and UAb, shown in Figure 3, are as follows:

    1. UAa initiates a SIP INVITE with UAbs telephone

    number, as the called party. The CdP number ismapped into the request URI of the SIP INVITE.

    2. Upon receiving the SIP INVITE, SMa queries the

    ENUM server on the CdP: UAb.

    3. Two cases might occur:

    a. If the ENUM server does not contain a NAPTR

    record for the CdP, and the carrier-of-record

    associated with the CdP cannot be found or

    reached via SIP peering, then the call is routed,

    by default, to the PSTN. This case is not shown

    on Figure 3.

    b. Otherwise, the ENUM server returns a NAPTRrecord to SMa with the domain name asso-

    ciated with Uab.

    4. SMa requests support from a local policy infra-

    structure (Pa) to determine the peering point in

    domain A (PPa) where it should forward the SIP

    INVITE. PPa is an IBCF, which provides, among

    OriginatingSPs

    network

    LF

    PF

    SF

    MF

    LF

    PF

    SF

    MF

    LF

    ReceivingSPs

    network

    IETFInternet Engineering Task Force

    LFLocation functionMFMedia function

    PFPolicy function

    SFSignaling function

    SPService providerSPEERMINTSession Peering for Multimedia Interworking

    Figure 2.IETF SPEERMINT reference architecture.

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    DOI: 10.1002/bltj Bell Labs Technical Journal 39

    other things, firewall functions for signaling,

    policing of signaling, and topology hiding.

    5. Pa applies its local policy rules to determine the

    appropriate peering point to handle the call, and

    then forwards the SIP INVITE to SMa with the

    URI associated with PPa.

    6. SMa queries the DNS server in domain A to

    retrieve the IP address for PPa.7. The DNS server in domain A returns the PPas IP

    address to SMa.

    8. SMa sends the SIP INVITE toPPa with the domain

    name servingUAb.

    9. PPa queries the DNS in domain A to resolve the

    address of a peering point in domain B (PPb). At

    this stage, resolving the address of UAb results in

    finding the IP address, port number, and transport

    protocol for PPb.

    10. DNS server returns the IP address of PPb to PPa.

    11. PPa , as a gateway to external networks, mayencrypt part of the SIP message to perform the

    topology hiding interface gateway (THIG)

    function and forward the SIP INVITE to PPb.

    12. PPb, upon determining that the domain associated

    withthe request URI isdomainB,needs toforward

    the SIP INVITE to SMb. The following steps, not

    shown on the diagram, might occur in order to

    locate SMb. If the SIP INVITE message does not

    contain a route header field that points directly to

    SMb, PPb sends a location-info-request (LIR) over

    theCxinterfaceto the HSSto locatethe appropriate

    SM. HSS replies with a location-info-answer (LIA)

    command that indicates the SIP URI of the server,

    in thiscaseSMb,thatservesUAb.13. PPb queries the DNS server in domain B to

    obtain the IP address for SMb.

    14. PPb forwards the SIP INVITE request to SMb.

    15. SMb may invoke a TAS to apply further call logic

    and forwards the SIP INVITE request to UAb.

    16. UAb accepts the call and replies with a 200 OK

    message toUAa, also not shown.

    Issues Related to the Signaling Routing Steps

    In this section, we analyze the signaling routing

    steps discussed above to identify the salient rout-ing issues that might occur in IMS network implemen-

    tations. We group these issues under three categories

    and discuss them separately.

    CdP address resolution. As indicated in step 2, the

    CdP address resolution is performed using the ENUM

    service offered on the DNS server. In the example we

    UAa PPa PPb SMbSMa UAb

    E D D E

    PbPa

    IMSdomain A 7

    8

    13

    IMSdomain B

    1

    2 6

    11

    12

    14 15

    1039

    45

    DDNS server

    DNSDomain name systemEENUM server

    ENUMElectronic number mapping

    IMSIP Multimedia Subsystem

    IPInternet ProtocolPPolicy

    PPPeering point

    SMSession managerUAUser agent

    Peering IPnetwork

    Figure 3.Signaling routing steps.

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    40 Bell Labs Technical Journal DOI: 10.1002/bltj

    presented, the SP uses a private ENUM infrastructure,

    which might not contain a NAPTR record for the CdP.

    This might happen if a different SP serves the CdP. In

    this case, the decision to route to the PSTN upon an

    ENUM failure response, as described in step 3a, cre-

    ates a routing dilemma because the CdP could be an

    IP endpoint in domain B, and not a PSTN user. Therouting decision is based on not having the capability

    to map the CdP telephone number with a SIP URI in

    domain A, but does not take into consideration the

    fundamental question as to whether the CdP is an IP

    or a PSTN endpoint. In the latter case, it is appropri-

    ate to route to the PSTN. In the former case, how-

    ever, the call should simply be maintained in the IP

    domain.

    Finally, we believe that private ENUM implemen-

    tation will not satisfy the long term need to remain

    in the IP domain for an inter-domain call served bydifferent SPs. A long term solution should be based

    on a common directory of telephone numbers and

    associated domain names that are restricted to the SPs

    use and not accessible by public Internet users. Such

    a common directory can cover many SPs across a

    geographic area (e.g., regions in country) and be

    administered by a trusted third party on behalf of the

    SPs. This solution is conceptually similar to existing

    administrative domains and database infrastructures

    available in the PSTN, such as the telephone number-

    ing plan or LERG.Determining peering point in originating domain. In

    step 4, we discussed another important aspect of the

    signaling routing process: selection of a peering point

    in the originating IMS domain. The originating IMS

    network will need to make intelligent decisions in the

    selection of its own signaling peering point in support

    of call setup, as it will likely have multiple signaling

    peering points to interconnect with other networks.

    This decision is dynamic in nature and varies with the

    target destination network of the CdP. SPs may employ

    caching methods to minimize call setup delays of sub-sequent calls in the selection of a peering point.

    However, several other conditions in the originating

    IMS network might influence such a selection as well.

    For instance, the choice of a peering point can be

    made based on quality of service policies (e.g., call

    setup delay) defined to meet peering and user service

    level agreements (SLA). Engineering factors such as

    call rate, and current utilization of the proxy server

    performing the peering function can lead to a distri-

    bution of the calls among the available peering points

    in support of load balancing requirements. Last, this is

    an area that requires further investigation and would

    most likely require support from a policy infrastructure.We will discuss further aspects of this selection process

    later in the paper.

    Discovering peering point in terminating domain.

    Step 10 discusses a DNS-based mechanism for the peer-

    ing point in the originating network, which allows the

    discovery of a peering point in the terminating net-

    work. The IETF has addressed in detail the discovery

    process of a proxy server (i.e., the peering point in our

    context) in a different domain in RFC 3263 [22]. In

    essence, the peering point in the originating network

    makes use of DNS procedures, using both servicerecord (SRV) [11] and NAPTR records, to determine

    the IP address, port, and transport protocol (e.g.,

    Transport Control Protocol (TCP), User Datagram

    Protocol (UDP), Stream Control Transmission Protocol

    (SCTP), Transport Layer Security (TLS)), for the peer-

    ing point in the terminating network. Additionally, RFC

    3263 describes the procedures to allow the peering

    point in the terminating network to discover a backup

    peering point in the originating network, in case the

    latter fails in the middle of the call setup transaction.

    In short, the originating network will find all requiredinformation to establish a call on IP to the terminating

    network, including the border elements of the termi-

    nating network.

    The premise of RFC 3263, however, is based on a

    global address resolution scheme that is accessible

    through the E164.arpa domain at the public DNS root.

    That is, SPs will have to publish SIP URIs of their peer-

    ing points in the public DNS so that they know where

    to signal one another. While such architecture works

    reasonably well today for handling email services over

    the Internet, its implementation in SP networks willend up exposing their critical signaling resources to

    Internet users. Thus, we believe that SPs deploying

    IMS networks will be reluctant to embrace this solu-

    tion until they can be assured that great concerns

    about security risks or attacks on key signaling assets

    are fully addressed.

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    Meanwhile, a way forward is for SPs to create fed-

    erations and use infrastructure ENUM/DNS for peering

    point discovery within a federation. A federation is a

    group of SPs that agree to accept calls from one another

    via SIP, according to a set of administrative rules and

    contractual financial terms established for such calls. In

    the next section, we analyze other networking condi-tions that impact the peering process as well.

    Other Factors Influencing Signaling Peering

    In addition to the issues identified in the signaling

    steps, there are other important factors that will

    influence the signaling peering process. In general,

    these factors are related to the nature of the peering

    infrastructure between the originating and terminat-

    ing domain, as well as the handling of the CdP address

    format and translation at the peering points.

    Inter-Domain IP Infrastructure for SignalingThe inter-domain IP infrastructure between the

    IMS originated and terminated networks can be one

    of the following cases:

    1. The public Internet.

    2. Direct connections over leased lines, IP-virtual

    private network (VPN) or tunneling over the

    public IP network, among the SPs peering points

    to create a fully meshed interconnect signaling

    network.

    3. A managed IP network commonly owned by the

    SPs for the purpose of transferring IMS signaling-related traffic among the different SP networks.

    This case is similar to the proposed GPRS roaming

    exchange (GRX) [9] of the GSM Association for

    PLMN network interconnections.

    4. A third-party-managed IP network similar to an

    existing VoIP clearinghouse, an SP that offers

    termination, financial services, and enhanced

    applications to other SPs (e.g., Internet service

    provider (ISP), VoIP) but serving the IMS SPs

    signaling interconnection needs.

    The inter-domain IP infrastructure need not bean IMS network, but should ensure that key require-

    mentssuch as QoS, reliability, security, and address

    resolutionare met. Issues related to IPv4-to-IPv6

    interwork- ing are outside the scope of this document.

    In what follows, we evaluate each inter-domain archi-

    tecture alternative and its impacts on the signaling

    peering process between the originating and termi-

    nating IMS domains. We evaluate these alternatives

    from the points of view of address resolution and

    discovery of the peering point in the terminating

    network.

    The public Internet. This scenario, as shown

    in Figure 4, leverages an open interconnection

    architecture where the SPs peering points addresses

    are public IP addresses and are directly accessible by

    Internet users. This also implies that inter-carrier IMS

    signaling messages are routed over the public Internetwithout any guarantee that the signaling require-

    ments (e.g., delay, priority of traffic) will be met. This

    scenario has some key advantages. It potentially offers

    a more open solution to the peering issue and enables

    ad-hoc peering among SPs who do not have an

    explicit peering agreement.

    However, this scenario requires support from a

    global ENUM/DNS system with global mapping of

    E.164 numbers to DNS NAPTRs containing the SIP

    URIs of all the SP peering points. For the reasons dis-

    cussed earlier and related to universal connectivitythrough global DNS, we do not believe this scenario

    will address the SPs security concerns, and hence it is

    not recommended.

    Fully meshed interconnect network. This scenario

    provides direct links among the SPs signaling peering

    points and results in an n2 problem, where n is the

    PublicENUM/

    DNS

    Domain A Internet Domain B

    Domain Z

    DNSDomain name systemENUMElectronic number mapping

    Figure 4.Peering via public Internet.

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    42 Bell Labs Technical Journal DOI: 10.1002/bltj

    number of peering points, as illustrated in Figure 5.

    SPs may use public IP addresses for their peering

    points; however such addresses are not accessible

    by public Internet users. The supporting DNS system

    resolves the terminating peering point address into an IPaddress that is routed over a particular link. In general,

    this solution may prove not to be scalable for a large

    number of peering points. However, SPs might decide to

    selectively implement it to support high volume traffic

    in between large communities of interests.

    Common managed IP network. In this scenario,

    depicted in Figure 6, the common managed IP net-

    work is a private network. It may use public IP

    addresses to route SIP requests to the SPs peering

    points; however, these IP addresses are not accessible

    by public Internet users.In addition, this scenario requires an infrastruc-

    ture ENUM/DNS system that is used only to resolve

    SIP URIs associated with SP peering points, which are

    attached to the common managed IP network. In

    order to peer with a public domain (i.e., one not

    attached to the managed IP network), the SP would

    use a public Internet DNS for address resolution.

    Hence, the public Internet DNS system is used to resolve

    the domain names of all destinations not served

    by the common managed IP network.

    Finally, for traffic among the SPs served by the com-mon managed IP network, the discovery of the ter-

    minating peering point always results in finding the IP

    address of an access server in the common managed

    IP network. Once a SIP URI request is sent to the

    common managed IP network, the infrastructure

    ENUM/DNS server resolves the associated domain

    name into the IP address of the terminating peering

    point.

    Clearinghouse model. This scenario, also shown in

    Figure 6, is functionally the same as the one depicted

    for the common managed IP network, except that a

    clearinghouse provides the administration of the

    inter-domain IP infrastructure. A variant of this caseis that the clearinghouse is only used as a trusted third

    party registry administering the infrastructure ENUM/

    DNS hierarchy and providing SPs with the routing

    information needed for address resolution.

    In summary, the nature of the inter-domain IP

    infrastructure impacts the address resolution process

    for the discovery of the terminating peering point. It

    further determines what network knowledge is requi-

    red to reach the terminating domain. For instance, in

    the first scenario, a global directory ENUM/DNS is

    required, whereas in the second case, static routingtables might be sufficient to reach the terminating

    network. Hence, the call processing algorithm for SIP

    peering should take the inter-domain infrastructure

    into consideration.

    Transit network/federation. In the absence of a

    global directory ENUM/DNS, SPs will likely group

    Carrier

    ENUM/DNS

    PublicENUM/

    DNS

    Domain AManaged IP

    networkDomain B

    Domain Z

    Domain X

    DNSDomain name systemENUMElectronic number mapping

    IPInternet Protocol

    Figure 6.Peering via managed IP network.

    Domain A Domain B

    Domain Z

    Fully meshed

    Figure 5.Peering via fully meshed network.

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    themselves into federations to satisfy their intercon-nection needs. Figure 7 illustrates a multi-federation

    arrangement among four SP networks. Moreover, it

    shows that the relationship with a federation might

    not be exclusive. An SP might belong to more than

    one federation for various financial and technical rea-

    sons. For instance, domain A belongs to federations

    1 and 3. Furthermore, an SP which belongs to multi-

    ple federations might offer to become a transit SP for

    other SPs. In our example, domains B and X can be

    such transits for domains A and Z. Likewise, domains

    A and Z can be used as transits for traffic between do-mains X and B. As a result, domain A can use either

    domain X or domain B to reach domain Z. Therefore,

    a key question a routing algorithm needs to address is

    which federation to use to reach a target destination.

    In addition, [16] and [12] discuss the need for fed-

    erations to use fully qualified domain names as their

    identifiers, and that members of a federation might

    need to publish federation membership as domain

    attributes. In other words, to route a call to its destina-

    tion, the originating network might need to retrieve the

    set of federations to which the terminating networkbelongs. If there is a shared federation, which can be

    determined through the domain policy dynamic

    delegation discovery system (DDDS) application as spec-

    ified in [17], it is used to support the call. Otherwise, the

    originating network might route the call to a transit SP.

    Thus, matching a common federation or selecting the

    right transit network to use to reach a destinationbecomes a critical part of the signaling peering routing

    process and must be taken into consideration by the

    routing algorithm during call processing.

    Identification and Address Format

    IMS, as for any type of network, provides mecha-

    nisms to uniquely identify users so that calls can be

    appropriately routed to them. IMS uses public user

    identities (PbUI) and private user identities (PrUI) to

    identify its users. PbUI are SIP URIs or TEL URIs [24]

    and are used to route SIP signaling in IMS. As discussedearlier, SIP URIs are in the form of sip:user@oper-

    ator.com, where the user part can be an E.164 tele-

    phone number. A TEL URI, however, is in the form of:

    tel:1-NPA-NXX-XXXX, to represent, for example, a

    telephone number in the North America Numbering

    Plan. The TEL URI is used when the domain of own-

    ership of the phone number is unknown. SIP, how-

    ever, requires that the URI under registration be a SIP

    URI [3]. Hence, it is not possible to explicitly register a

    TEL URI in SIP. However, PrUIs do not use SIP URI

    or TEL URI formats. Instead, they use a network accessidentifier (NAI) in the form [email protected].

    They are exclusively used for subscription identifica-

    tion and authentication purposes [3].

    The rest of the paper focuses on analyzing the SIP

    URI-based PbUI formats and their impacts on IMS

    signaling peering.

    Domain X

    AXZ

    ABZ

    Domain A Domain BFederation 1 Federation 2

    Fed

    eration3 Federation

    4

    Domain Z

    Figure 7.Peering via transit.

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    SIP URI formats for the PbUI. IMS users will use a

    variety of devices (e.g., PDA, IP phone) to trigger NG

    services and to initiate calls to other users. Likewise,

    SPs may assign more than one PbUI to their customers

    so they can be reached through a variety of means

    (e.g., email, business phone). As a result, SIP URIs as-

    sociated with PbUIs can take various forms, as de-scribed in [6], including the following:

    1. sip:E.164@service provider;user phone

    2. sip:user@service provider

    3. sip:user@domain name

    4. sip:ip address

    Impacts on address resolution for SIP peering. The

    format of the SIP URIs, as presented above, might in

    some cases render impossible the resolution of the

    destination address of the CdP. For instance,

    user@domain name (case 3) might be impos-

    sible to resolve if used to initiate a call. First, assumingthe domain name can be resolved to an associated

    SP domain, as discussed above, user@service

    provider might not be unique in the SP domain.

    Second, different users belonging to a domain name

    might be served by different SPs. For example, johndoe

    @alcatel-lucent.com might be served by a different SP

    than [email protected]. Therefore, the res-

    olution of a user@domain name SIP URI

    should identify the specific SP serving each particular

    user. That is, the domain name server might need

    to be a redirect server pointing to the carrier-of-recordthat serves the user. This can be an implementation

    nightmare in that all domain names will be required

    to support the redirect function. Last, considerations

    should be given to the additional delay such a process

    will introduce on key performance parameters such as

    call setup time.

    In summary, the format of the SIP URI used to

    initiate a call will impact the signaling routing process

    in the originating domain as well as in any transit net-

    work that might be used to support the call. In addi-

    tion, some formats might be impossible to resolve insupport of inter-carrier communications.

    Proposed Signaling Peering Process

    In order to define an appropriate peering and

    routing approach for a given scenario, one needs to

    consider and evaluate the end-to-end process. In light

    of the routing steps and issues discussed above, we pro-

    pose a high level signaling peering process, which

    provides a comprehensive approach to signaling

    peering and takes into account multiple constructs

    (e.g., IP networks, federations, and transits) that might

    exist between the originating and terminating domains.

    Moreover, the proposed signaling peering processaddresses the need to perform address translation in

    support of service portability as an essential step for

    inter-carrier communications. A key assumption, in

    this regard, is that all call query, a method by which

    a number portability query is performed on all origi-

    nating calls as described in [8], is used as the service

    portability implementation option.

    Signaling Process for IMS Peering

    Figure 8 illustrates the signaling peering process

    for a basic voice call initiated with an E.164 number,which we describe as follows:

    1. The calling party initiates the call with a dial

    string, which the originating network converts

    into a fully qualified E.164 number.

    2. Map the CdPs E.164 number into SIP headers

    (e.g., request URI, To ) and send the app-

    ropriate INVITE request to the session manager

    in the originating network.

    3. Analyze the CdP address for service portability

    (e.g., SP portability).

    4. If the CdP is not ported, proceed with 6.5. If the CdP is ported:

    Perform the portability database lookup to

    retrieve the location routing number (LRN)

    associated with the CdP. Note that this paper

    does not specify an infrastructure for perfor-

    ming a number portability (NP) database (DB)

    query. Options range from reusing existing

    NPDBs in the intelligent network (IN) to ret-

    rieving a NAPTR record from an ENUM/DNS

    query pointing to the LRN.

    Append the npdi yes field to the requestURI header in the subsequent SIP URI request

    created with the LRN.

    6. Perform address resolution of the CdP or LRN

    through the ENUM/DNS infrastructure to de-

    termine which destination network the CdP

    belongs to. The output of this step results in a

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    ID mapping

    CdP addressanalysis

    Perform NP lookup andaddress translation

    PerformNPDB?

    CdP inPSTN

    Terminatingdomain

    Address resolution

    Invoke BGCF Peering scenario

    Discover peering point

    Resolve PP address

    Translate CdP

    Route to next domain

    Route to PSTN

    Select MGCF/MGW

    Perform SIP/ISUPconversion

    ENUM/DNS withlocal policies

    SIP address translationand mapping

    Determine S-CSCF servingCdP and route to CdP

    Route SIP invite to CdPin terminating domain

    IP endpoint

    PSTN endpoint

    BGCFBreakout gateway control functionCdPCalled party

    CSCFCall session control function

    DNSDomain name systemENUMElectronic number mapping

    IAMInitial address message

    IDIdentifierIMIP Multimedia

    IPInternet Protocol

    ISDNIntegrated services digital networkISUPISDN user part

    LERGLocal exchange routing guide

    MGCFMedia gateway control function

    MGWMedia gateway

    NAPTRNaming authority pointerNPNumber portability

    NPANumbering plan area

    NPDBNumber portability databasePPPeering point

    PSTNPublic switched telephone network

    RFCRequest for commentsS-CSCFServing CSCF

    SSFService switching function

    SIPSession Initiation ProtocolSPService provider

    TELTelephone

    URIUniform resource identifier

    RFC 3261Request URI

    PbUl:SIP URI orTEL URI

    Y

    Y

    Y

    N

    N

    NWhats the domain name of CdP SP?Is CdP served by initiating domain?

    obtain ENUM/DNS NAPTR

    Invoke policy peering function forestablishing the nature of peeringnetwork (e.g., federation, transit)

    How to determine that theCdP belongs to the PSTN?

    Selection basedon NPA-NXX andlocal policy rules

    ENUM/DNS queries for NP orquery NPDB via IM-SSF

    Identifier CdP

    All call queryIs CdP open to portability?

    Obtain PSTN routingdata (e.g., from LERG)

    Generate ISUP IAM

    Is RFC 3263 sufficient?Determining/locating PP?(originating/terminating)

    Figure 8.Proposed signaling peering process.

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    46 Bell Labs Technical Journal DOI: 10.1002/bltj

    SIP URI that represents an address of record for

    the CdP.

    7. If the call is destined to the PSTN, invoke the

    BGCF to determine which MGCF to provide

    the ISDN user part and SIP interworking func-

    tion, as defined in [4], and then route the call to

    the PSTN.8. Otherwise, determine whether a common fed-

    eration exists between the originating and ter-

    minating domains:

    If so, consult the policy peering infrastructure

    and apply common federation rules to dis-

    cover the peering point in the terminating

    network.

    If not, determine a transit network to support

    the call and apply corresponding federation

    rules for the transit.

    9. Determine the peering point in the originatingnetwork according to the local policy infrastruc-

    ture and procedures defined in RFC 3263 [22].

    10. Perform the SIP header translation at the

    peering point and according to the SIP URI

    format used, if necessary.

    11. Route the SIP INVITE to the next hop (e.g.,

    proxy/access server in federation) towards the

    destination

    12. When the terminating domain is reached,

    determine the serving CSCF for the CdP, locate

    the CdP, and route the INVITE to the CdP.13. Establish the session and transfer of media.

    Proposal Benefits

    The end-to-end signaling peering process

    described above introduces new decision points and

    routing steps during call setup for inter-carrier com-

    munications. The benefits of such a process include:

    1) support for number portability, 2) more efficient

    routing among IP endpoints, and 3) avoidance of

    unnecessary delays during call setup. The following

    discusses these benefits.Support for number portability. In many countries,

    number portability (NP) is a regulatory requirement

    that SPs must comply with. As presented in step 4

    and step 5 in the section above, support for NP

    requires SPs to perform user identification translation

    for the CdP. The outcome of the translation process

    helps determine which network to route a call to for

    inter-carrier communications. Various options are im-

    plemented for NP. For instance, in the United States,

    SPs implement the all call query option for service

    provider number portability (SPNP), which allows a

    user to change the SP while keeping the same

    telephone number. SPNP leaves the responsibility forperforming the NPDB query to the penultimate net-

    work. This decision is somewhat straightforward

    in the PSTN in that for a long distance (LD) call, the

    carrier of record for LD services is the penultimate

    network, and, hence, it performs the query. For a toll

    local call, the local SP is the penultimate network and,

    as such, does the query.

    IMS, however, introduces new challenges in sup-

    porting NP. First, as discussed above, the CdP identifi-

    cation might not be an E.164 number. Obviously, in

    such cases, existing E.164-based NP databases cannotbe leveraged. Therefore, translation between SIP URIs,

    and IP addresses via DNS in support of NP, becomes a

    critical function in determining the next domain/

    network towards a destination.

    Moreover, the decision as to which SP should per-

    form the NPDB becomes more complex in IMS, if we

    ought to follow the penultimate model discussed

    above. How to determine which network is next to

    the last before performing address translation of the

    CdP? In todays deployment of VoIP networks, these

    demarcation lines are not clearly defined. As a case inpoint, VoIP service offerings assume mobility of end-

    points. For instance, one can be living in California

    and may get assigned a phone number with a New

    Jersey numbering plan area (NPA) (e.g., 732). This

    proposal simplifies the decision process by recom-

    mending that the originating network always perform

    the appropriate translation of the CdP address on all

    calls, once it determines that such an address is open

    to portability.

    Determining the nature of CdP endpoint. Another

    important decision point during call processing,summarized in step 7 and step 8 above, is for the

    originating network to determine whether the CdP

    is a PSTN or IP endpoint, so the call is handled appro-

    priately. Otherwise, SPs will continue to use PSTN

    bridging for IP-to-IP inter-carrier calls. At this point, it

    is not clear whether the mechanism to identify the

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    nature of an endpoint (i.e., PSTN or IP) would exist

    outside the ENUM framework. Hence, as discussed

    earlier, we recommend that a common directory of

    telephone numbers and associated domain names

    (e.g., carrier ENUM) be in place to support that

    decision point. Such an infrastructure should be

    restricted to SP use and would be inaccessible topublic Internet users.

    This proposal takes advantage of the recom-

    mended common infrastructure to help determine

    whether a call terminates in the PSTN or in another

    IMS domain and to route it accordingly.

    Avoiding additional call setup delay. A net benefit of

    determining the nature of the CdP endpoint is to avoid

    the additional call setup delay, as defined in ITU-T

    E.127 [14], which might occur in the PSTN when

    bridging two IMS islands. For instance, for calls in the

    PSTN, E.127 recommends an average call setup delayof no more than 3 seconds and 5 seconds, respectively,

    for local and toll calls. The 95th percentiles are

    6 seconds and 8 seconds, respectively.

    Conclusions and Areas for Further Investigations

    IMS network signaling peering is a very complex pro-

    blem to tackle for the deployment of next-generation

    IMS networks. Unlike the embedded base PSTN/PLMN,

    where common databases and established routing prin-

    ciples allow SPs to securely share call routing data and

    identify one anothers signaling points, there are newchallenges facing the IMS SPs. First, SPs need to per-

    form address resolution of users identifiers (e.g., SIP

    URI) and other resources (e.g., call servers) at almost

    every step during call setup. Second, the format of the

    identifiers may vary depending on what service is being

    initiated. Accordingly, the address resolution process

    requires different mechanisms to identify the users and

    their serving domains. Third, the nature of the infra-

    structure between two SP networks impacts the peer-

    ing process in that it influences the discovery of the

    peering points for a given call. Last, SPs will create fed-erations and transit networks for network intercon-

    nect, as achieving universal connectivity is an

    unrealistic goal. Hence, the selection of what federa-

    tion/transit network to use to reach a destination be-

    comes an integral part of the signaling peering decision

    process.

    This paper discusses these new challenges and

    proposes a high-level signaling process that identifies

    key peering decision points and provides a compre-

    hensive approach to solving them. Such an approach

    is necessary for accelerating the deployment of IMS

    networks and reducing the reliance on PSTN/PLMN

    network for inter-carrier communications among IPendpoints.

    Finally, it is clear that more detailed work is

    needed to address some specific technical challenges

    identified in this paper. However, in priority order,

    the following areas are critical for further investiga-

    tions. First, more work is needed to define the re-

    quirements for a policy peering infrastructure in

    support of intelligent selection of peering points in an

    SP IMS network. This will invoke the PF in both the

    originating and terminating IMS networks and apply

    policy rules for the selection of the peering points.Second, a new routing algorithm for the selection of

    federation/transit in a multi-transit/federation sce-

    nario is needed. This will invoke the LF and PF in

    both originating and terminating networks.

    Acknowledgments

    The author would like to acknowledge several

    colleagues for sharing their insights on various aspects

    of this work. In particular, he is grateful to Deirdre

    Doherty, Mohamed El-Sayed, Igor Faynberg, Paul

    Gagen, Vijay Gurbani, Volker Hilt, Markus Hofmann,

    Vanita Katkar, Eunyoung Kim, Lyle Kipp, Humberto

    La Roche, Tom Morawski, Bob Moul, Morgan Stern

    and Carlos Urrutia-Valds.

    *Trademarks

    3GPP is a trademark of the European Telecommunica-

    tions Standards Institute.

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    Resource and Admission Control Functions in

    Next Generation Networks, ITU-T Rec. Y.2111,

    Sept. 2006,http://www.itu.int.

    [16] O. Lendl, Publishing SIP Peering Policy,

    IETF Internet Draft, Dec. 2005,

    http://www.ietf.org

    .[17] O. Lendl, The Domain Policy DDDS

    Application, IETF Internet Draft, Feb. 2006,

    http://www.ietf.org.

    [18] R. Mahy, A Minimalist Approach to Direct

    Peering, IETF Internet Draft, June 2006,

    http://www.ietf.org.

    [19] M. Mealling, Dynamic Delegation Discovery

    System (DDDS) Part Three: The Domain

    Name System (DNS) Database, IETF RFC 3403,

    Oct. 2002,http://www.apps.ietf.org/

    rfc/rfc3403.txt.

    [20] J.-F. Mule, SPEERMINT Requirements for SIP-

    Based VoIP Interconnection, IETF InternetDraft, June 2006,http://www.ietf.org.

    [21] R. Penno (ed.), SPEERMINT Peering

    Architecture, IETF Internet Draft, Aug. 2006,

    http://www.ietf.org.

    [22] J. Rosenberg and H. Schulzrinne, Session

    Initiation Protocol (SIP): Locating SIP Servers,

    IETF RFC 3263, June 2002, http://www.apps.

    ietf.org/rfc/rfc3263.txt.

    [23] J. Rosenberg, H. Schulzrinne, G. Camarillo,

    A. Johnston, J. Peterson, R. Sparks, M. Handley,

    and E. Schooler, SIP: Session Initiation

    Protocol, IETF RFC 3261, June 2002,

    http://www.apps.ietf.org/rfc/rfc3261.txt.[24] H. Schulzrinne, The tel URI for Telephone

    Numbers, IETF RFC 3966, Dec. 2004,

    http://www.apps.ietf.org/rfc/rfc3966.txt.

    (Manuscript approved August 2007)

    JEAN-PHILIPPE JOSEPH is a member of technical staff in

    the Advanced Network Planning and

    Modeling group at Alcatel-Lucent in Murray

    Hill, New Jersey. He holds a B.S. in

    electronics engineering from La Facult des

    Sciences de lUniversit dEtat dHati in

    Port-au-Prince, and an M.S. degree in electrical

    engineering from Polytechnic University of New York in

    New York City. Mr. Josephs current research interests

    include IMS-based architectures and solutions for

    network evolution and service integration, IMS

    signaling peering, network modeling, and

    optimization.

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