Rfc6405
TitleVoice over IP (VoIP) SIP Peering Use Cases
AuthorA. Uzelac, Ed., Y. Lee, Ed.
DateNovember 2011
Format:TXT, HTML
Status:INFORMATIONAL






Internet Engineering Task Force (IETF)                    A. Uzelac, Ed.
Request for Comments: 6405                               Global Crossing
Category: Informational                                      Y. Lee, Ed.
ISSN: 2070-1721                                            Comcast Cable
                                                           November 2011


               Voice over IP (VoIP) SIP Peering Use Cases

Abstract

   This document depicts many common Voice over IP (VoIP) use cases for
   Session Initiation Protocol (SIP) peering.  These use cases are
   categorized into static and on-demand, and then further sub-
   categorized into direct and indirect.  These use cases are not an
   exhaustive set, but rather the most common use cases deployed today.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc6405.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.




RFC 6405               VoIP SIP Peering Use Cases          November 2011


Table of Contents

   1. Introduction ....................................................2
   2. Terminology .....................................................3
   3. Reference Architecture ..........................................3
   4. Contexts of Use Cases ...........................................4
   5. Use Cases .......................................................4
      5.1. Static Peering Use Cases ...................................5
      5.2. Static Direct Peering Use Case .............................5
           5.2.1. Administrative Characteristics .....................10
           5.2.2. Options and Nuances ................................10
      5.3. Static Direct Peering Use Case - Assisting LUF and LRF ....11
           5.3.1. Administrative Characteristics .....................12
           5.3.2. Options and Nuances ................................12
      5.4. Static Indirect Peering Use Case - Assisting LUF and LRF ..12
           5.4.1. Administrative Characteristics .....................19
           5.4.2. Options and Nuances ................................19
      5.5. Static Indirect Peering Use Case ..........................19
           5.5.1. Administrative Characteristics .....................20
           5.5.2. Options and Nuances ................................21
      5.6. On-Demand Peering Use Case ................................21
           5.6.1. Administrative Characteristics .....................21
           5.6.2. Options and Nuances ................................21
   6. Acknowledgments ................................................22
   7. Security Considerations ........................................22
   8. References .....................................................22
      8.1. Normative References ......................................22
      8.2. Informative References ....................................23

1.  Introduction

   This document describes important Voice over IP (VoIP) use cases for
   SIP-based [RFC3261] peering.  These use cases are determined by the
   Session PEERing for Multimedia INTerconnect (SPEERMINT) working group
   and will assist in identifying requirements and other issues to be
   considered for future resolution by the working group.

   Only use cases related to VoIP are considered in this document.
   Other real-time SIP communications use cases, like Instant Messaging
   (IM), video chat, and presence are out of scope for this document.

   The use cases contained in this document are described as
   comprehensive as possible, but should not be considered the exclusive
   set of use cases.







RFC 6405               VoIP SIP Peering Use Cases          November 2011


2.  Terminology

   This document uses terms defined in [RFC5486].  Please refer to it
   for definitions.

3.  Reference Architecture

   The diagram below provides the reader with a context for the VoIP use
   cases in this document.  Terms such as SIP Service Provider (SSP),
   Lookup Function (LUF), Location Routing Function (LRF), Signaling
   Path Border Element (SBE), and Data Path Border Element (DBE) are
   defined in [RFC5486].

   The Originating SSP (O-SSP) is the SSP originating a SIP request.
   The Terminating SSP (T-SSP) is the SSP terminating the SIP request
   originating from the O-SSP.  The assisting LUF and LRF Provider offer
   LUF and LRF services to the O-SSP.  The Indirect SSP (I-SSP) is the
   SSP providing indirect peering service(s) to the O-SSP to connect to
   the T-SSP.

    +--------------------+------------------------+--------------------+
    |  Originating SSP   |  Assisting LUF and LRF |  Terminating SSP   |
    |     Domain         |    Provider Domain     |      Domain        |
    |                    |                        |                    |
    |  +-----+  +-----+  |    +------+ +------+   |  +-----+  +-----+  |
    |  |O-LUF|  |O-LRF|  |    |A-LUF | | A-LRF|   |  |T-LUF|  |T-LRF|  |
    |  +-----+  +-----+  |    +------+ +------+   |  +-----+  +-----+  |
    |                    |                        |                    |
    | +-------+ +-----+  +------------------------+  +-----+ +-------+ |
    | |O-Proxy| |O-SBE|  |  Indirect SSP Domain   |  |T-SBE| |T-Proxy| |
    | +-------+ +-----+  |                        |  +-----+ +-------+ |
    |                    |    +-----+  +-----+    |                    |
    |    +---+  +-----+  |    |O-SBE|  |O-DBE|    |  +-----+  +---+    |
    |    |UAC|  |O-DBE|  |    +-----+  +-----+    |  |T-DBE|  |UAS|    |
    |    +---+  +-----+  |                        |  +-----+  +---+    |
    |                    |                        |                    |
    +--------------------+------------------------+--------------------+

                             General Overview

                                 Figure 1

   Note that some elements included in Figure 1 are optional.








RFC 6405               VoIP SIP Peering Use Cases          November 2011


4.  Contexts of Use Cases

   Use cases are sorted into two general groups: static and on-demand
   peering [RFC5486].  Each group can be further sub-divided into Direct
   Peering and Indirect Peering [RFC5486].  Although there may be some
   overlap among the use cases in these categories, there are different
   requirements between the scenarios.  Each use case must specify a
   basic set of required operations to be performed by each SSP when
   peering.

   These can include:

   o  Peer Discovery - Peer discovery via a Lookup Function (LUF) to
      determine the Session Establishment Data (SED) [RFC5486] of the
      request.  In VoIP use cases, a request normally contains a phone
      number.  The O-SSP will input the phone number to the LUF and the
      LUF will normally return a SIP address of record (AOR) [RFC3261]
      that contains a domain name.

   o  Next-Hop Routing Determination - Resolving the SED information is
      necessary to route the request to the T-SSP.  The LRF is used for
      this determination.  After obtaining the SED, the O-SSP may use
      the standard procedure defined in [RFC3263] to discover the next-
      hop address.

   o  Call setup - SSPs that are interconnecting to one another may also
      define specifics on what peering policies need to be used when
      contacting the next hop in order to a) reach the next hop at all
      and b) 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 Layer 3
      network), which TLS client certificate [RFC5246] to use, and other
      authentication schemes.

   o  Call reception - This step ensures that the type of relationship
      (static or on-demand, indirect or direct) is understood and
      acceptable.  For example, the receiving SBE needs to determine
      whether the INVITE it received really came from a trusted member.

5.  Use Cases

   Please note there are intra-domain message flows within the use cases
   to serve as supporting background information.  Only inter-domain
   communications are germane to this document.







RFC 6405               VoIP SIP Peering Use Cases          November 2011


5.1.  Static Peering Use Cases

   Static peering [RFC5486] describes the use case when two SSPs form a
   peering relationship with some form of association established prior
   to the exchange of traffic.  Pre-association is a prerequisite to
   static peering.  Static peering is used in cases when two peers want
   a consistent and tightly controlled approach to peering.  In this
   scenario, a number of variables, such as an identification method
   (remote proxy IP address) and Quality-of-Service (QoS) parameters,
   can be defined upfront and known by each SSP prior to peering.

5.2.  Static Direct Peering Use Case

   This is the simplest form of a peering use case.  Two SSPs negotiate
   and agree to establish a SIP peering relationship.  The peer
   connection is statically configured and the peer SSPs are directly
   connected.  The peers may exchange interconnection parameters such as
   Differentiated Service Code Point (DSCP) [RFC2474] policies, the
   maximum number of requests per second, and proxy location prior to
   establishing the interconnection.  Typically, the T-SSP only accepts
   traffic originating directly from the trusted peer.

         +--------------------+             +---------------------+
         |        O-SSP       |             |        T-SSP        |
         |       +-----+      |             |       +-----+       |
         |       |O-LUF|      |             |       |T-LUF|       |
         |       |O-LRF|      |             |      /|T-LRF|       |
         |      /+-----+\     |             |     / +-----+       |
         |    (2)     (4,5,6) |             |    /                |
         |    /           \   |             |   /(8,9)            |
         |+-------+     +-----+             +-----+      +-------+|
         ||O-Proxy|-(3)-|O-SBE+-----(7)-----+T-SBE|-(10)-|T-Proxy||
         |+-------+     +-----+             +-----+      +-------+|
         |    |               |             |                |    |
         |   (1)              |             |               (11)  |
         |    |               |             |                |    |
         | +-----+      +-----+             +-----+       +-----+ |
         | | UAC +======|O-DBE+=====(12)====+T-DBE|=======+ UAS | |
         | +-----+      +-----+             +-----+       +-----+ |
         +--------------------+             +---------------------+
              example.com                         example.net


                      Static Direct Peering Use Case

                                 Figure 2





RFC 6405               VoIP SIP Peering Use Cases          November 2011


   The following is a high-level depiction of the use case:

   1.   The User Agent Client (UAC) initiates a call via SIP INVITE to
        O-Proxy.  O-Proxy is the home proxy for UAC.

         INVITE sip:+19175550100@example.com;user=phone SIP/2.0
         Via: SIP/2.0/TCP client.example.com:5060
           ;branch=z9hG4bK74bf9
         Max-Forwards: 10
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=12345
         To: Bob <sip:+19175550100@example.com;user=phone>
         Call-ID: abcde
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@client.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   Note that UAC inserted its Fully Qualified Domain Name (FQDN) in the
   VIA and CONTACT headers.  This example assumes that UAC has its own
   FQDN.

   2.   UAC knows the User Agent Server's (UAS's) TN, but does not know
        UAS's domain.  It appends its own domain to generate the SIP URI
        in the Request-URI and TO header.  O-Proxy checks the Request-
        URI and discovers that the Request-URI contains the user
        parameter "user=phone".  This parameter signifies that the
        Request-URI is a phone number.  So O-Proxy will extract the TN
        from the Request-URI and query the LUF for SED information from
        a routing database.  In this example, the LUF is an ENUM
        [RFC6116] database.  The ENUM entry looks similar to this:

          $ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
          IN NAPTR (
            10
            100
            "u"
            "E2U+SIP"
            "!^.*$!sip:+19175550100@example.net!"
            . )

   This SED data can be provisioned by O-SSP or populated by the T-SSP.

   3.   O-Proxy examines the SED and discovers the domain is external.
        Given the O-Proxy's internal routing policy, O-Proxy decides to
        use O-SBE to reach T-SBE.  O-Proxy routes the INVITE request to
        O-SBE and adds a Route header that contains O-SBE.



RFC 6405               VoIP SIP Peering Use Cases          November 2011


         INVITE sip:+19175550100@example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP o-proxy.example.com:5060
           ;branch=z9hG4bKye8ad
         Via: SIP/2.0/TCP client.example.com:5060
           ;branch=z9hG4bK74bf9;received=192.0.1.1
         Max-Forwards: 9
         Route: <sip:o-sbe1.example.com;lr>
         Record-Route: <sip:o-proxy.example.com;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=12345
         To: Bob <sip:+19175550100@example.com;user=phone>
         Call-ID: abcde
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@client.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   4.   O-SBE receives the requests and pops the top entry of the Route
        header that contains "o-sbe1.example.com".  O-SBE examines the
        Request-URI and does an LRF for "example.net".  In this example,
        the LRF is a Naming Authority Pointer (NAPTR) DNS query
        [RFC3403] of the domain name.  O-SBE receives a NAPTR response
        from the LRF.  The response looks similar to this:

          IN NAPTR (
            50
            50
            "S"
            "SIP+D2T"
            ""
            _sip._tcp.t-sbe.example.net. )

          IN NAPTR (
            90
            50
            "S"
            "SIP+D2U"
            ""
            _sip._udp.t-sbe.example.net. )

   5.   Given the lower order for TCP in the NAPTR response, O-SBE
        decides to use TCP as the transport protocol, so it sends an SRV
        DNS query for the SRV record [RFC2782] for "_sip._tcp.t-
        sbe.example.net." to O-LRF.






RFC 6405               VoIP SIP Peering Use Cases          November 2011


        ;;     priority  weight   port  target
        IN SRV 0         2        5060  t-sbe1.example.net.
        IN SRV 0         1        5060  t-sbe2.example.net.

   6.   Given the higher weight for "t-sbe1.example.net", O-SBE sends an
        A record DNS query for "t-sbe1.example.net." to get the A
        record:

          ;; DNS ANSWER
          t-sbe1.example.net.   IN A   192.0.2.100
          t-sbe1.example.net.   IN A   192.0.2.101

   7.   O-SBE sends the INVITE to T-SBE.  O-SBE is the egress point to
        the O-SSP domain, so it should ensure subsequent mid-dialog
        requests traverse via itself.  If O-SBE chooses to act as a
        back-to-back user agent (B2BUA) [RFC3261], it will generate a
        new INVITE request in next step.  If O-SBE chooses to act as a
        proxy, it should record-route to stay in the call path.  In this
        example, O-SBE is a B2BUA.

         INVITE sip:+19175550100@example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP o-sbe1.example.com:5060
           ;branch= z9hG4bK2d4zzz
         Max-Forwards: 8
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde-osbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@o-sbe1.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   Note that O-SBE may re-write the Request-URI with the target domain
   in the SIP URI.  Some proxy implementations will only accept the
   request if the Request-URI contains their own domains.

   8.   T-SBE determines the called party home proxy and directs the
        call to the called party.  T-SBE may use ENUM lookup or other
        internal mechanism to locate the home proxy.  If T-SSP uses ENUM
        lookup, this internal ENUM entry is different from the external
        ENUM entry populated for O-SSP.  In this example, the internal
        ENUM query returns the UAS's home proxy.







RFC 6405               VoIP SIP Peering Use Cases          November 2011


         $ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
         IN NAPTR (
           10
           100
           "u"
           "E2U+SIP"
           "!^.*$!sip:+19175550100@t-proxy.example.net!"
           . )

   9.   T-SBE receives the NAPTR record, and following the requirements
        in [RFC3263], queries DNS for the SRV records indicated by the
        NAPTR result.  Not finding any, the T-SBE then queries DNS for
        the A record of domain "t-proxy.example.net.".

          ;; DNS ANSWER
          t-proxy.example.net.   IN A   192.0.2.2

   10.  T-SBE is a B2BUA, so it generates a new INVITE and sends it to
        UAS's home proxy:

         INVITE sip:bob@t-proxy.example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP t-sbe1.example.net:5060
           ;branch= z9hG4bK28uyyy
         Max-Forwards: 7
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@t-proxy.example.net;user=phone>
         Call-ID: abcde-tsbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@t-sbe1.example.net;user=phone;
         transport=tcp>
         </allOneLine>


















RFC 6405               VoIP SIP Peering Use Cases          November 2011


   11.  Finally, UAS's home proxy forwards the INVITE request to the
        UAS.

         INVITE sip:+19175550100@server.example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP t-proxy.example.net:5060
           ;branch= z9hG4bK28u111
         Via: SIP/2.0/TCP t-sbe1.example.net:5060
           ;branch= z9hG4bK28uyyy; received=192.2.0.100
         Max-Forwards: 6
         Record-Route: <sip:t-proxy.example.net:5060;lr>,
           <sip:t-sbe1.example.net:5060;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@t-proxy.example.net;user=phone>
         Call-ID: abcde-tsbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@t-sbe1.example.net;user=phone;
         transport=tcp>
         </allOneLine>

   12.  RTP is established between the UAC and UAS.  Note that the media
        shown in Figure 2 passes through O-DBE and T-DBE, but the use of
        DBE is optional.

5.2.1.  Administrative Characteristics

   The static direct peering use case is typically implemented in a
   scenario where there is a strong degree of trust between the two
   administrative domains.  Both administrative domains typically sign a
   peering agreement that states clearly the policies and terms.

5.2.2.  Options and Nuances

   In Figure 2, O-SSP and T-SSP peer via SBEs.  Normally, the operator
   will deploy the SBE at the edge of its administrative domain.  The
   signaling traffic will pass between two networks through the SBEs.
   The operator has many reasons to deploy an SBE.  For example, the
   O-SSP may use [RFC1918] addresses for their UA and proxies.  These
   addresses are not routable in the target network.  The SBE can
   perform a NAT function.  Also, the SBE eases the operation cost for
   deploying or removing Layer 5 network elements.  Consider the
   deployment architecture where multiple proxies connect to a single
   SBE.  An operator can add or remove a proxy without coordinating with
   the peer operator.  The peer operator "sees" only the SBE.  As long
   as the SBE is maintained in the path, the peer operator does not need
   to be notified.




RFC 6405               VoIP SIP Peering Use Cases          November 2011


   When an operator deploys SBEs, the operator is required to advertise
   the SBE to the peer LRF so that the peer operator can locate the SBE
   and route the traffic to the SBE accordingly.

   SBE deployment is a decision within an administrative domain.  Either
   one or both administrative domains can decide to deploy SBE(s).  To
   the peer network, most important is to identify the next-hop address.
   This decision does not affect the network's ability to identify the
   next-hop address.

5.3.  Static Direct Peering Use Case - Assisting LUF and LRF

   This use case shares many properties with the Static Direct Peering
   Use Case Section 5.2.  There must exist a pre-association between the
   O-SSP and T-SSP.  The difference is O-SSP will use the Assisting LUF/
   LRF Provider for LUF and LRF.  The LUF/LRF Provider stores the SED to
   reach T-SSP and provides it to O-SSP when O-SSP requests it.

                            +-----------------+
                            |LUF/LRF Provider |
                            |                 |
                            |     +-------+   |
                            |   +-+ A-LUF |   |
                            |  /  | A-LRF |   |
       +--------------------+ /  ++-------+   +---------------------+
       |       O-SSP        |/  /             |         T-SSP       |
       |       +------------/(4,5,6)          |        +-----+      |
       |      /             | /               |        |T-LUF|      |
       |    (2)           +-+/                |      +-|T-LRF|      |
       |    /            /  |                 |     /  +-----+      |
       |   /            /   |                 |    /(8,9)           |
       |+-------+     +-----+                 +-----+      +-------+|
       ||O-Proxy|-(3)-|O-SBE+-------(7)-------+T-SBE|-(10)-|T-Proxy||
       |+-------+     +-----+                 +-----+      +-------+|
       |    |               |                 |                |    |
       |   (1)              |                 |              (11)   |
       |    |               |                 |                |    |
       | +-----+      +-----+                 +-----+       +-----+ |
       | | UAC +======|O-DBE+=======(12)======+T-DBE+=======+ UAS | |
       | +-----+      +-----+                 +-----+       +-----+ |
       +--------------------+                 +---------------------+
             example.com                            example.net


             Static Direct Peering with Assisting LUF and LRF

                                 Figure 3




RFC 6405               VoIP SIP Peering Use Cases          November 2011


   The call flow looks almost identical to Static Direct Peering Use
   Case except that Steps 2, 4, 5, and 6 involve the LUF/LRF Provider
   instead of happening in O-SSP domain.

   Similar to Static Direct Peering Use Case, the O-DBE and T-DBE in
   Figure 3 are optional.

5.3.1.  Administrative Characteristics

   The LUF/LRF Provider supplies the LUF and LRF services for the O-SSP.
   Taken together, the LUF/LRF Provider, O-SSP, and T-SSP form a trusted
   administrative domain.  To reach the T-SSP, the O-SSP must still
   require pre-arranged agreements for the peer relationship with the
   T-SSP.  The Layer 5 policy is maintained in the O-SSP and T-SSP
   domains, and the LUF/LRF Provider may not be aware of any Layer 5
   policy between the O-SSP and T-SSP.

   A LUF/LRF Provider can serve multiple administrative domains.  The
   LUF/LRF Provider typically does not share SED from one administrative
   domain to another administrative domain without appropriate
   permission.

5.3.2.  Options and Nuances

   The LUF/LRF Provider can use multiple methods to provide SED to the
   O-SSP.  The most commonly used are an ENUM lookup and a SIP Redirect.
   The O-SSP should negotiate with the LUF/LRF Provider regarding which
   query method it will use prior to sending a request to the LUF/LRF
   Provider.

   The LUF/LRF Providers must be populated with the T-SSP's AORs and
   SED.  Currently, this procedure is non-standardized and labor
   intensive.  A more detailed description of this problem has been
   documented in the work in progress [DRINKS].

5.4.  Static Indirect Peering Use Case - Assisting LUF and LRF

   The difference between a Static Direct Use Case and a Static Indirect
   Use Case lies within the Layer 5 relationship maintained by the O-SSP
   and T-SSP.  In the Indirect use case, the O-SSP and T-SSP do not have
   direct Layer 5 connectivity.  They require one or multiple Indirect
   Domains to assist with routing the SIP messages and possibly the
   associated media.

   In this use case, the O-SSP and T-SSP want to form a peer
   relationship.  For some reason, the O-SSP and T-SSP do not have
   direct Layer 5 connectivity.  The reasons may vary, for example,




RFC 6405               VoIP SIP Peering Use Cases          November 2011


   business demands and/or domain policy controls.  Due to this indirect
   relationship, the signaling will traverse from the O-SSP through one
   or multiple I-SSPs to reach the T-SSP.

   In addition, the O-SSP is using a LUF/LRF Provider.  This LUF/LRF
   Provider stores the T-SSP's SED pre-populated by the T-SSP.  One
   important motivation to use the LUF/LRF Provider is that the T-SSP
   only needs to populate its SED once to the provider.  Using an LUF/
   LRF Provider allows the T-SSP to populate its SED once, while any
   O-SSP T-SSP's SED can use this LUF/LRF Provider.  Current practice
   has shown that it is rather difficult for the T-SSP to populate its
   SED to every O-SSP who must reach the T-SSP's subscribers.  This is
   especially true in the Enterprise environment.

   Note that the LUF/LRF Provider and the I-SSP can be the same provider
   or different providers.

                            +------------------+
                            | LUF/LRF Provider |
                            |       I-SSP      |
                            |      +-------+   |
                            |   ---+ A-LUF |   |
                            |  /   | A-LRF |   |
       +--------------------+ /    +-------+   +---------------------+
       |       O-SSP        |/     /           |         T-SSP       |
       |      +-------------/     /            |        +-----+      |
       |     /              |(4,5,6)           |        |T-LUF|      |
       |    /               |   /              |   +----+T-LRF|      |
       |  (2)             + +---               |  /     +-----+      |
       |  /              /  |                  | /(9,10)             |
       |+-------+     +-----+     +-----+      +-----+      +-------+|
       ||O-Proxy|-(3)-|O-SBE+-(7)-+I-SBE+-(8)--+T-SBE+-(11)-|T-Proxy||
       |+-------+     +-----+     +-----+      +-----+      +-------+|
       |    |               |                  |                |    |
       |   (1)              |                  |               (12)  |
       |    |               |                  |                |    |
       | +-----+      +-----+     +-----+      +-----+       +-----+ |
       | | UAC +=(13)=|O-DBE+=====+I-DBE+======+T-DBE+=======+ UAS | |
       | +-----+      +-----+     +-----+      +-----+       +-----+ |
       +-------------------------------------------------------------+
            example.com          example.org         example.net


    Indirect Peering via an LUF/LRF Provider and I-SSP (SIP and Media)

                                 Figure 4





RFC 6405               VoIP SIP Peering Use Cases          November 2011


   The following is a high-level depiction of the use case:

   1.   The UAC initiates a call via SIP INVITE to the O-Proxy.  The
        O-Proxy is the home proxy for the UAC.

         INVITE sip:+19175550100@example.com;user=phone SIP/2.0
         Via: SIP/2.0/TCP client.example.com:5060
           ;branch=z9hG4bK74bf9
         Max-Forwards: 10
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=12345
         To: Bob <sip:+19175550100@example.com;user=phone>
         Call-ID: abcde
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@client.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   2.   The UAC knows the UAS's TN, but does not know the UAS's domain.
        It appends its own domain to generate the SIP URI in the
        Request-URI and TO header.  The O-Proxy checks the Request-URI
        and discovers that the Request-URI contains the user parameter
        "user=phone".  This parameter indicates that the Request-URI is
        a phone number.  So, the O-Proxy will extract the TN from the
        Request-URI and query the LUF for SED information from a routing
        database.  In this example, the LUF is an ENUM database.  The
        ENUM entry looks similar to this:

          $ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
          IN NAPTR (
            10
            100
            "u"
            "E2U+SIP"
            "!^.*$!sip:+19175550100@example.org!"
            . )

   Note that the response shows the next-hop is the SBE in I-SSP.
   Alternatively, the O-SSP may have a pre-association with the I-SSP.
   As such, the O-SSP will forward all requests that contain an external
   domain in the Request-URI or an unknown TN to the I-SSP.  The O-SSP
   will rely on the I-SSP to determine the T-SSP and route the request
   correctly.  In this configuration, the O-SSP can skip Steps 2, 4, 5,
   and 6 and forward the request directly to the I-SBE.  This
   configuration is commonly used in the Enterprise environment.





RFC 6405               VoIP SIP Peering Use Cases          November 2011


   3.   Given the O-Proxy's internal routing policy, the O-Proxy decides
        to use the O-SBE to reach the I-SBE.  The O-Proxy routes the
        INVITE request to the O-SBE and adds a Route header that
        contains the O-SBE.

         INVITE sip:+19175550100@example.org;user=phone SIP/2.0
         Via: SIP/2.0/TCP o-proxy.example.com:5060
           ;branch=z9hG4bKye8ad
         Via: SIP/2.0/TCP client.example.com:5060
           ;branch=z9hG4bK74bf9;received=192.0.1.1
         Max-Forwards: 9
         Route: <sip:o-sbe1.example.com;lr>
         Record-Route: <sip:o-proxy.example.com;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=12345
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@client.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   4.   The O-SBE receives the requests and pops the top entry of the
        Route header that contains "sip:o-sbe1.example.com".  The O-SBE
        examines the Request-URI and does an LRF for "example.org".  In
        this example, the LRF is a NAPTR DNS query of the domain.  The
        O-SBE receives a response similar to this:

          IN NAPTR (
            50
            50
            "S"
            "SIP+D2T"
            ""
            _sip._tcp.i-sbe.example.org. )

          IN NAPTR (
            90
            50
            "S"
            "SIP+D2U"
            ""
            _sip._udp.i-sbe.example.org. )







RFC 6405               VoIP SIP Peering Use Cases          November 2011


   5.   Given the lower order for TCP in the NAPTR response, the O-SBE
        decides to use TCP for transport protocol, so it sends an SRV
        DNS query for the SRV record for "_sip._tcp.i-sbe.example.org."
        to the O-LRF.

        ;;     priority  weight   port  target
        IN SRV 0         2        5060  i-sbe1.example.org.
        IN SRV 0         1        5060  i-sbe2.example.org.

   6.   Given the higher weight for "i-sbe1.example.org", the O-SBE
        sends a DNS query for an A record of "i-sbe1.example.org." to
        get the A record:

          ;; DNS ANSWER
          i-sbe1.example.org.   IN A   192.0.2.200
          i-sbe1.example.org.   IN A   192.0.2.201

   7.   The O-SBE sends the INVITE to the I-SBE.  The O-SBE is the entry
        point to the O-SSP domain, so it should ensure subsequent mid-
        dialog requests traverse via itself.  If the O-SBE chooses to
        act as a B2BUA, it will generate a new back-to-back INVITE
        request in the next step.  If the O-SBE chooses to act as proxy,
        it should record-route to stay in the call path.  In this
        example, the O-SBE is a B2BUA.

         INVITE sip:+19175550100@example.org;user=phone SIP/2.0
         Via: SIP/2.0/TCP o-sbe1.example.com:5060
           ;branch= z9hG4bK2d4zzz
         Max-Forwards: 8
         Route:  <sip:i-sbe1.example.org;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde-osbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@o-sbe1.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   8.   The I-SBE receives the request and queries its internal routing
        database on the TN.  It determines that the target belongs to
        the T-SSP.  Since the I-SBE is a B2BUA, the I-SBE generates a
        new INVITE request to the T-SSP.







RFC 6405               VoIP SIP Peering Use Cases          November 2011


         INVITE sip:+19175550100@.example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP i-sbe1.example.org:5060
           ;branch= z9hG4bK2d4777
         Max-Forwards: 7
         Route: <sip:t-sbe1.example.net;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde-isbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@i-sbe1.example.org;user=phone;
         transport=tcp>
         </allOneLine>

   Note that if the I-SSP wants the media to traverse through the I-DBE,
   the I-SBE must modify the Session Description Protocol (SDP) in the
   Offer to point to its DBE.

   9.   The T-SBE determines the called party home proxy and directs the
        call to the called party.  The T-SBE may use ENUM lookup or
        another internal mechanism to locate the home proxy.  If the
        T-SSP uses ENUM lookup, this internal ENUM entry is different
        from the external ENUM entry populated for O-SSP.  This internal
        ENUM entry will contain the information to identify the next hop
        to reach the called party.  In this example, the internal ENUM
        query returns the UAS's home proxy.

         $ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
         IN NAPTR (
           10
           100
           "u"
           "E2U+SIP"
           "!^.*$!sip:+19175550100@t-proxy.example.net!"
           . )

   Note that this step is optional.  If the T-SBE has other ways to
   locate the UAS home proxy, the T-SBE can skip this step and send the
   request to the UAS's home proxy.  We show this step to illustrate one
   of the many possible ways to locate UAS's home proxy.

   10.  The T-SBE receives the NAPTR record and, following the
        requirements in [RFC3263], queries the DNS for the SRV records
        indicated by the NAPTR result.  Not finding any, the T-SBE then
        queries DNS for the A record of domain "t-proxy.example.net.".





RFC 6405               VoIP SIP Peering Use Cases          November 2011


          ;; DNS ANSWER
          t-proxy.example.net.   IN A   192.0.2.2

   11.  T-SBE sends the INVITE to UAS's home proxy:

         INVITE sip:+19175550100@t-proxy.example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP t-sbe1.example.net:5060
           ;branch= z9hG4bK28uyyy
         Max-Forwards: 6
         Record-Route: <sip:t-sbe1.example.net:5060;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde-tsbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@t-sbe1.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   12.  Finally, the UAS's home proxy forwards the INVITE request to the
        UAS.

         INVITE sip:+19175550100@server.example.net;user=phone SIP/2.0
         Via: SIP/2.0/TCP t-proxy.example.net:5060
           ;branch= z9hG4bK28u111
         Via: SIP/2.0/TCP t-sbe1.example.net:5060
           ;branch= z9hG4bK28uyyy; received=192.2.0.100
         Max-Forwards: 5
         Record-Route: <sip:t-proxy.example.net:5060;lr>,
           <sip:t-sbe1.example.net:5060;lr>
         From: Alice <sip:+14085550101@example.com;user=phone>
           ;tag=54321
         To: Bob <sip:+19175550100@example.net;user=phone>
         Call-ID: abcde-tsbe1
         CSeq: 1 INVITE
         <allOneLine>
         Contact: <sip:+19175550100@t-sbe1.example.com;user=phone;
         transport=tcp>
         </allOneLine>

   13.  In Figure 4, RTP is established between the UAC and UAS via the
        O-DBE, I-DBE and T-DBE.  The use of DBE is optional.








RFC 6405               VoIP SIP Peering Use Cases          November 2011


5.4.1.  Administrative Characteristics

   This use case looks very similar to the Static Direct Peering Use
   Case with Assisting LUF and LRF.  The major difference is the O-SSP
   and T-SSP do not have direct Layer 5 connectivity.  Instead, O-SSP
   connects to the T-SSP indirectly via the I-SSP.

   Typically, an LUF/LRF Provider serves multiple O-SSPs.  Two O-SSPs
   may use different I-SSPs to reach the same T-SSP.  For example,
   O-SSP1 may use I-SSP1 to reach T-SSP, but O-SSP2 may use I-SSP2 to
   reach T-SSP.  Given the O-SSP and T-SSP pair as input, the LUF/LRF
   Provider will return the SED of I-SSP that is trusted by O-SSP to
   forward the request to T-SSP.

   In this use case, there are two levels of trust relationship.  The
   first trust relationship is between the O-SSP and the LUF/LRF
   Provider.  The O-SSP trusts the LUF/LRF to provide the T-SSP's SED.
   The second trust relationship is between the O-SSP and I-SSP.  The
   O-SSP trusts the I-SSP to provide Layer 5 connectivity to assist the
   O-SSP in reaching the T-SSP.  The O-SSP and I-SSP have a pre-arranged
   agreement for policy.  Note that Figure 4 shows a single provider to
   supply both LUF/LRF and I-SSP, but O-SSP can choose two different
   providers.

5.4.2.  Options and Nuances

   Similar to the Static Direct Peering Use Case, the O-SSP and T-SSP
   may deploy SBE and DBE for NAT traversal, security, transcoding, etc.
   The I-SSP can also deploy the SBE and DBE for similar reasons (as
   depicted in Figure 4).

5.5.  Static Indirect Peering Use Case

   This use case shares many properties with the Static Indirect Use
   Case with Assisting LUF and LRF.  The difference is that the O-SSP
   uses its internal LUF/LRF to control the routing database.  By
   controlling the database, the O-SSP can apply different routing rules
   and policies to different T-SSPs.  For example, the O-SSP can use
   I-SSP1 and Policy-1 to reach T-SSP1, and use I-SSP2 and Policy-2 to
   reach T-SSP2.  Note that there could be multiple I-SSPs and multiple
   SIP routes to reach the same T-SSP; the selection process is out of
   scope of this document.









RFC 6405               VoIP SIP Peering Use Cases          November 2011


      +--------------------+-------------------+---------------------+
      |       O-SSP        |       I-SSP       |         T-SSP       |
      |      +-----+       |                   |        +-----+      |
      |     -+O-LUF|       |                   |        |T-LUF|      |
      |    / |O-LRF+\      |                   |   +----+T-LRF|      |
      |   /  +-----+ \     |                   |  /     +-----+      |
      |  /(2)         \(4,5,6)                 | /(9,10)             |
      |+-------+     +-----+      +-----+      +-----+      +-------+|
      ||O-Proxy|-(3)-|O-SBE+--(7)-+I-SBE+-(8)--+T-SBE+-(11)-|T-Proxy||
      |+-------+     +-----+      +-----+      +-----+      +-------+|
      |    |               |                   |                |    |
      |   (1)              |                   |               (12)  |
      |    |               |                   |                |    |
      | +-----+      +-----+      +-----+      +-----+       +-----+ |
      | | UAC +=(13)=+O-DBE+======+I-DBE+======+T-DBE+=======+ UAS | |
      | +-----+      +-----+      +-----+      +-----+       +-----+ |
      +--------------------------------------------------------------+
           example.com          example.org          example.net


                Indirect Peering via I-SSP (SIP and Media)

                                 Figure 5

5.5.1.  Administrative Characteristics

   The Static Indirect Peering Use Case is implemented in cases where no
   direct interconnection exists between the originating and terminating
   domains due to either business or physical constraints.

   O-SSP <---> I-SSP = Relationship O-I

   In the O-I relationship, typical policies, features, or functions
   that deem this relationship necessary are number portability,
   ubiquity of termination options, security certificate management, and
   masquerading of originating VoIP network gear.

   T-SSP <---> I-SSP = Relationship T-I

   In the T-I relationship, typical policies, features, or functions
   observed consist of codec "scrubbing", anonymizing, and transcoding.
   The I-SSP must record-route and stay in the signaling path.  The
   T-SSP will not accept any message sent directly from the O-SSP.








RFC 6405               VoIP SIP Peering Use Cases          November 2011


5.5.2.  Options and Nuances

   In Figure 5, we show an I-DBE.  Using an I-DBE is optional.  For
   example, the I-DBE can be used when the O-SSP and T-SSP do not have a
   common codec.  To involve an I-DBE, the I-SSP should know the list of
   codecs supported by the O-SSP and T-SSP.  When the I-SBE receives the
   INVITE request, it will make a decision to invoke the I-DBE.  An
   I-DBE may also be used if the O-SSP uses Secure Real-time Transport
   Protocol (SRTP) [RFC3711] for media and T-SSP does not support SRTP.

5.6.  On-Demand Peering Use Case

   On-demand peering [RFC5486] describes how two SSPs form the peering
   relationship without a pre-arranged agreement.

   The basis of this use case is built on the fact that there is no pre-
   established relationship between the O-SSP and T-SSP.  The O-SSP and
   T-SSP do not share any information prior to the dialog initiation
   request.  When the O-Proxy invokes the LUF and LRF on the Request-
   URI, the terminating user information must be publicly available.
   When the O-Proxy routes the request to the T-Proxy, the T-Proxy must
   accept the request without any pre-arranged agreement with the O-SSP.

   The On-demand Peering Use Case is uncommon in production.  In this
   memo, we capture only the high-level descriptions.  Further analysis
   is expected when this use case becomes more popular.

5.6.1.  Administrative Characteristics

   The On-demand Direct Peering Use Case is typically implemented in a
   scenario where the T-SSP allows any O-SSP to reach its serving
   subscribers.  The T-SSP administrative domain does not require any
   pre-arranged agreement to accept the call.  The T-SSP makes its
   subscribers information publicly available.  This model mimics the
   Internet email model.  The sender does not need an pre-arranged
   agreement to send email to the receiver.

5.6.2.  Options and Nuances

   Similar to the Static Direct Peering Use Case, the O-SSP and T-SSP
   can decide to deploy the SBE.  Since the T-SSP is open to the public,
   the T-SSP is considered to be a higher security risk than static
   model because there is no trusted relationship between the O-SSP and
   T-SSP.  The T-SSP should protect itself from any attack launched by
   an untrusted O-SSP.






RFC 6405               VoIP SIP Peering Use Cases          November 2011


6.  Acknowledgments

   Michael Haberler, Mike Mammer, Otmar Lendl, Rohan Mahy, David
   Schwartz, Eli Katz and Jeremy Barkan are the authors of the early
   individual documents.  Their use cases are captured in this document.
   Also, Jason Livingood, Daryl Malas, David Meyer, Hadriel Kaplan, John
   Elwell, Reinaldo Penno, Sohel Khan, James McEachern, Jon Peterson,
   Alexander Mayrhofer, and Jean-Francois Mule made many valuable
   comments related to this document.  The editors would also like to
   extend a special thank you to Spencer Dawkins for his detailed review
   of this document.

7.  Security Considerations

   Session interconnect for VoIP, as described in this document, has a
   wide variety of security issues that should be considered.  For
   example, if the O-SSP and T-SSP peer through public Internet, the
   O-SSP must protect the signaling channel and accept messages only
   from an authorized T-SSP.  This document does not analyze the threats
   in detail.  [RFC6404] discusses the different security threats and
   countermeasures related to VoIP peering.

8.  References

8.1.  Normative References

   [RFC1918]  Rekhter, Y., Moskowitz, R., Karrenberg, D., Groot, G., and
              E. Lear, "Address Allocation for Private Internets",
              BCP 5, RFC 1918, February 1996.

   [RFC2782]  Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
              specifying the location of services (DNS SRV)", RFC 2782,
              February 2000.

   [RFC3261]  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.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation
              Protocol (SIP): Locating SIP Servers", RFC 3263,
              June 2002.

   [RFC3403]  Mealling, M., "Dynamic Delegation Discovery System (DDDS)
              Part Three: The Domain Name System (DNS) Database",
              RFC 3403, October 2002.





RFC 6405               VoIP SIP Peering Use Cases          November 2011


   [RFC5486]  Malas, D. and D. Meyer, "Session Peering for Multimedia
              Interconnect (SPEERMINT) Terminology", RFC 5486,
              March 2009.

   [RFC6116]  Bradner, S., Conroy, L., and K. Fujiwara, "The E.164 to
              Uniform Resource Identifiers (URI) Dynamic Delegation
              Discovery System (DDDS) Application (ENUM)", RFC 6116,
              March 2011.

   [RFC6404]  Seedorf, J., Niccolini, S., Chen, E., and H. Scholz,
              "Session PEERing for Multimedia INTerconnect (SPEERMINT)
              Security Threats and Suggested Countermeasures", RFC 6404,
              November 2011.

8.2.  Informative References

   [DRINKS]   Channabasappa, S., "Data for Reachability of Inter/
              tra-NetworK SIP (DRINKS) Use cases and Protocol
              Requirements", Work in Progress, August 2011.

   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474,
              December 1998.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

Authors' Addresses

   Adam Uzelac (editor)
   Global Crossing
   U.S.A.

   EMail: adam.uzelac@globalcrossing.com
   URI:   http://www.globalcrossing.com


   Yiu L.Lee (editor)
   Comcast Cable
   U.S.A.

   EMail: yiu_lee@cable.comcast.com
   URI:   http://www.comcast.com