


MARTINI WG                                                     J. Elwell
Internet-Draft                         Siemens Enterprise Communications
Intended status:  Informational                                H. Kaplan
Expires:  January 27, 2011                                   Acme Packet
                                                           July 26, 2010


     Requirements for multiple address of record (AOR) reachability
          information in the Session Initiation Protocol (SIP)
                     draft-ietf-martini-reqs-09.txt

Abstract

   This document states requirements for a standardized SIP registration
   mechanism for multiple addresses of record, the mechanism being
   suitable for deployment by SIP service providers on a large scale in
   support of small to medium sized Private Branch Exchanges (PBXs).
   The requirements are for a solution that can, as a minimum, support
   AORs based on E.164 numbers.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
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   This Internet-Draft will expire on January 27, 2011.

Copyright Notice

   Copyright (c) 2010 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
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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must



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   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.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Problem statement  . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Issues with the REGISTER transaction . . . . . . . . . . .  5
       3.1.1.  Mis-handling by SIP-aware middleboxes  . . . . . . . .  5
       3.1.2.  REGISTER response growth . . . . . . . . . . . . . . .  6
       3.1.3.  Illegal wildcarding syntax . . . . . . . . . . . . . .  6
     3.2.  Issues with routing inbound requests to the SIP-PBX  . . .  6
       3.2.1.  Loss of target information . . . . . . . . . . . . . .  6
       3.2.2.  Inconsistent placement of target URI information
               in inbound requests  . . . . . . . . . . . . . . . . .  6
       3.2.3.  Request-URI mis-routing  . . . . . . . . . . . . . . .  6
     3.3.  Policy-related issues  . . . . . . . . . . . . . . . . . .  7
       3.3.1.  Authorization policy mismatches  . . . . . . . . . . .  7
       3.3.2.  PAI or PPI URI mismatches  . . . . . . . . . . . . . .  7
   4.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Desirables . . . . . . . . . . . . . . . . . . . . . . . . . . 10
   6.  Non-requirements . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  IANA considerations  . . . . . . . . . . . . . . . . . . . . . 11
   8.  Security considerations  . . . . . . . . . . . . . . . . . . . 11
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 11
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 11
     10.2. Informative References . . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13



















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1.  Introduction

   The Session Initiation Protocol (SIP) [RFC3261], together with its
   extensions, supports multiple means of obtaining the connection
   information necessary to deliver out-of-dialog SIP requests to their
   intended targets.  When a SIP proxy needs to send a request to a
   target address of record (AOR) within its domain, it can use a
   location service to obtain the registered Contact Universal Resource
   Identifiers (URIs), together with any associated path information
   [RFC3327], and build a route set to reach each target user agent
   (UA).  The SIP REGISTER method can be used to register Contact URIs
   and path information.  SIP-outbound [RFC5626] enhances this mechanism
   to cater for UAs behind Network Address Translators (NATs) and
   firewalls.  When an entity needs to send a request to a target for
   which it is not authoritative, the entity can follow [RFC3263]
   procedures for using the Domain Name System (DNS) to obtain the next-
   hop connection information.

   In practice, many small and medium-sized businesses use a SIP Private
   Branch Exchange (SIP-PBX) that is authoritative for tens or hundreds
   of SIP AORs.  This SIP-PBX acts as a registrar/proxy for these AORs
   for users hosted by the SIP-PBX.  A SIP Service Provider (SSP)
   provides SIP peering/trunking capability to the SIP-PBX.  The SIP-PBX
   needs to be reachable from the SSP so that the SSP can handle inbound
   out-of-dialog SIP requests targeted at these AORs, routing these
   requests to the SIP-PBX for onward delivery to registered UAs.

   Experience has shown that existing mechanisms are not always
   sufficient to support SIP-PBXs for small/medium businesses.  In
   particular, RFC 3263 procedures are generally inappropriate, except
   for some larger SIP-PBXs.  In current deployments, mechanisms for the
   dynamic provision of reachability information based on the SIP
   REGISTER method are commonly used.  However, implementations of this
   mechanism vary in detail, leading to interoperability issues between
   SIP-PBXs and SSPs, and the need for equipment to support different
   variants.  A more detailed statement of the problem is given in
   section Section 3.

   This document states requirements for a standardized SIP registration
   mechanism for multiple AORs, the mechanism being suitable for
   deployment by SSPs on a large scale in support of small to medium
   sized SIP-PBXs.  The requirements are for a solution that can, as a
   minimum, support AORs based on E.164 numbers.


2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",



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   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   The terms address of record (AOR), proxy, REGISTER, registrar,
   request, response and user agent (UA) are to be interpreted as
   described in [RFC3261].


3.  Problem statement

   A number of other standards organizations have addressed the issue of
   a SIP-PBX registering with its SSP, notably ETSI [ETSI TS 182 025]
   and 3GPP [3GPP TS 24.229].  Also various SSPs have produced
   proprietary specifications for use with their own offerings.  The
   reader is encouraged to review the documents produced by those
   organizations.

   A short summary of the general concept is as follows.  The figure
   below illustrates the scope of the problem.

    +----+
    | UA |----+
    +----+    |             - - - - - - - - - - - - - -
              |            :     SCOPE OF PROBLEM      :
    +----+    |            :                           :
    | UA |--+ |      +---------+                  +---------+
    +----+  | |      |         |                  |         |
            | +------|         |                  |         |
    +----+  +--------| SIP-PBX |------------------|   SSP   |
    | UA |-----------|         |                  |         |
    +----+           |         |                  |         |
                     +---------+                  +---------+
                           :                           :
     ---------------->     :    ------------------>    :
     UAs register with     :    SIP-PBX registers with :
     SIP-PBX on behalf of  :    SSP once on behalf of  :
     individual AORs       :    multiple AORs          :
                           :                           :
                           :    <------------------    :
     <----------------     :    SSP delivers inbound   :
     SIP-PBX forwards      :    requests to SIP-PBX    :
     inbound requests      :                           :
     to appropriate UAs    :                           :
                            - - - - - - - - - - - - - -

   In virtually all models, the SIP-PBX generates a SIP REGISTER request
   using a mutually agreed-upon SIP AOR - typically based on the SIP-
   PBX's main attendant/reception-desk number.  The AOR is often in the



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   domain of the SSP, and both the To and From URIs used for the
   REGISTER request identify that AOR.  In all respects, it appears on
   the wire as a "normal" SIP REGISTER request, as if from a typical
   user's UA.  However, it generally implicitly registers other AORs
   associated with the SIP-PBX.

   For both 3GPP and ETSI mechanisms, the 200 OK response to the
   REGISTER request, sent after a successful authentication challenge,
   contains a P-Associated-URI (PAU) [RFC3455] header field listing the
   other SIP URIs or TEL URIs (i.e., phone numbers) of the SIP-PBX,
   which are implicitly-registered AORs.  The registered reachability
   information from the REGISTER request will be used to reach not only
   the single explicitly-registered AOR but also each of the implicitly-
   registered AORs.  In order to reduce the number of PAU entries, a
   "wildcard" syntax model is defined [3GPP TS 23.003], which uses a
   regular expression syntax in the user field of the URI to express
   multiple AORs in a compressed manner.

   For routing requests for any of the explicitly- or implicitly-
   registered AORs from the SSP to the SIP-PBX, the Request-URI is
   typically replaced with the registered Contact URI.  In the case of
   3GPP and ETSI, the SSP has the option of using loose routing instead,
   and inserting the registered Contact URI as a loose route Route
   header field value, while leaving the Request-URI alone.  This
   decision is made based upon manually provisioned information in the
   registrar's database (i.e., the Home Subscriber Server, HSS).

3.1.  Issues with the REGISTER transaction

3.1.1.  Mis-handling by SIP-aware middleboxes

   None of the currently available mechanisms indicate that the REGISTER
   request or response is any different from a "normal" REGISTER request
   or response.  This has caused issues when SIP-aware middleboxes
   between the SIP-PBX and the registrar serve both SIP-PBXs and normal
   UAs yet need to apply different policies to the two cases.

   Furthermore, some middleboxes expect the registrar to follow normal
   [RFC3261] procedures of Request-URI replacement with the registered
   Contact URI for routing subsequent requests to the SIP-PBX.  If the
   registrar adopts a different practice for requests to SIP-PBXs, this
   can cause the middlebox to fail to route such requests correctly,
   because there is no indication that the registration was any
   different.

   Lastly, lack of an indication of implicit registration makes
   troubleshooting more difficult because the on-the-wire messages are
   indistinguishable from "normal" registrations.  Note that even the



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   existence of a PAU header field in the response does not indicate
   that implicit registration for a SIP-PBX has occurred, since the PAU
   header field is also used for normal UAs with multiple identities.

3.1.2.  REGISTER response growth

   If an SIP-PBX represents many AORs, the PAU list in the response can
   grow the SIP message size beyond the limits for UDP.

3.1.3.  Illegal wildcarding syntax

   The current syntax for "wildcarded" PAUs is illegal for TEL URIs,
   based on the ABNF rules for TEL URIs in [RFC3966].

3.2.  Issues with routing inbound requests to the SIP-PBX

3.2.1.  Loss of target information

   If the proxy-registrar follows [RFC3261] for registration resolution
   of requests targeted at one of the SIP-PBX's AORs, and thus replaces
   the Request-URI with the registered Contact URI, it is not clear
   which AOR is the intended target of the request.  The To URI, for
   example, may not contain the intended target AOR if the request was
   forwarded/retargeted prior to reaching the proxy-registrar.  Some
   middleboxes between the registrar and SIP-PBX will overwrite the
   Request-URI specifically to try to fix this issue.  In some cases, a
   P-Called-Party-ID header field [RFC3455] will contain the intended
   target AOR; and in some cases the History-Info header field [RFC4244]
   will contain it.  The SIP-PBX needs to know where to look to find the
   required information, and in the case of History-Info needs to
   identify the particular element containing the required information.

3.2.2.  Inconsistent placement of target URI information in inbound
        requests

   Even when all information needed by the SIP-PBX is provided, in some
   deployments inbound SIP requests from the SSP have the registered
   SIP-PBX Contact URI in the Request-URI, whereas in other deployments
   inbound SIP requests have the intended target SIP-PBX user (AOR) in
   the Request-URI and the Contact URI in the Route header field.  There
   are other variants too.  Interoperability problems arise when a SIP-
   PBX designed for or configured for one variant attempts to interwork
   with an SSP designed for or configured for another variant.

3.2.3.  Request-URI mis-routing

   Although many SIP-PBXs support registration with an SSP, they do not
   consider themselves authoritative for the explicitly- or implicitly-



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   registered AORs if the domain portion still identifies the SSP's
   domain.  They expect the domain portion to be their own IP Address,
   FQDN, or domain.  Currently middleboxes have to fix that issue.

3.3.  Policy-related issues

   The following are largely policy matters for the SSP, but it should
   be noted the policies described below will not work in some
   situations.  A mechanism for solving the SIP-PBX registration problem
   will not solve these policy issues directly, although when specifying
   the mechanism the opportunity can be taken to highlight the impact of
   such policies.

3.3.1.  Authorization policy mismatches

   Many SSPs perform a first-order level of authorization for requests
   from the SIP-PBX by checking the URI in the From, P-Asserted-
   Identity (PAI), or P-Preferred-Identity (PPI) [RFC3325] header field
   for one matching either an explicitly- or implicitly-registered AOR
   for the same Contact URI and/or Layer-3 IP Address.  However, some
   SIP-PBXs change the Contact URI they use for non-REGISTER requests to
   be different from the one they explicitly-registered.  For example
   they change the user portion of the Contact URI, or even the host
   portion.  This is particularly true for a SIP-PBX operating as a
   proxy and forwarding the Contact URI from the UA behind the SIP-PBX
   (the SIP-PBX typically being identified in a Record-Route header
   field), rather than acting as a B2BUA and substituting its own
   Contact URI.  This can cause an SSP to fail to find an AOR
   corresponding to the Contact URI for non-REGISTER requests, resulting
   in the SSP rejecting such requests or asserting its own PAI value,
   rather than asserting a value based on received header fields.

3.3.2.  PAI or PPI URI mismatches

   Some SSPs expect the PAI or PPI URI in SIP requests received from the
   SIP-PBX to match one of the explicitly- or implicitly-registered
   AORs, whereas some SIP-PBXs generate the URIs using their local IP
   Address, hostname, or domain name.  Some SSPs expect the PAI or PPI
   URI in SIP requests received from the SIP-PBX to be the explicitly-
   registered AOR only, as it is the main billing number, instead of the
   implicitly-registered AOR of the calling party.  In either case, this
   can result in the SSP rejecting requests with values that do not
   match, or asserting its own PAI value.

   Again, these are policy matters for the SSP, but drawbacks should be
   noted.  For example, rejection of requests can rule out requests from
   sources beyond the SIP-PBX (e.g., calls forwarded by the SIP-PBX),
   unless the SIP-PBX changes the PAI or PPI URI to a value acceptable



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   to the SSP (in which case it will no longer identify the calling
   user).  If the SSP changes the PAI or PPI URI, again the request will
   fail to identify the calling user.


4.  Requirements

   The following are requirements of the mechanism.

   REQ1 - The mechanism MUST allow a SIP-PBX to enter into a trunking
   arrangement with an SSP whereby the two parties have agreed on a set
   of telephone numbers assigned to the SIP-PBX.

   REQ2 - The mechanism MUST allow a set of assigned telephone numbers
   to comprise E.164 numbers, which can be in contiguous ranges,
   discrete, or in any combination of the two.

   REQ3 - The mechanism MUST allow a SIP-PBX to register reachability
   information with its SSP, in order to enable the SSP to route to the
   SIP-PBX inbound requests targeted at assigned telephone numbers.

   REQ4 - The mechanism MUST allow UAs attached to a SIP-PBX to register
   with the SIP-PBX for AORs based on assigned telephone numbers, in
   order to receive requests targeted at those telephone numbers,
   without needing to involve the SSP in the registration process.

   REQ5 - The mechanism MUST allow a SIP-PBX to handle requests
   originating at its own UAs and targeted at its assigned telephone
   numbers, without routing those requests to the SSP.

   REQ6 - The mechanism MUST allow a SIP-PBX to receive requests to its
   assigned telephone numbers originating outside the SIP-PBX and
   arriving via the SSP, so that the SIP-PBX can route those requests
   onwards to its UAs, as it would for internal requests to those
   telephone numbers.

   REQ7 - The mechanism MUST provide a means whereby a SIP-PBX knows
   which of its assigned telephone numbers an inbound request from its
   SSP is targeted at.

   REQ8 - The mechanism MUST provide a means of avoiding problems due to
   one side using the mechanism and the other side not.

      In other words, the mechanism is required to avoid the situation
      where one side believes it is using the mechanism and the other
      side believes it is not, e.g., the SIP-PBX believes it is
      performing registration of multiple telephone numbers, but the SSP
      believes a single AOR is being registered.



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   REQ9 - The mechanism MUST observe SIP backwards compatibility
   principles.

      In other words, the mechanism is required to provide a graceful
      means of recovery or fall-back if either side does not support the
      mechanism.  For example, this might involve the use of an option
      tag.

   REQ10 - The mechanism MUST work in the presence of a sequence of
   intermediate SIP entities on the SIP-PBX-to-SSP interface (i.e.,
   between the SIP-PBX and the SSP's domain proxy), where those
   intermediate SIP entities indicated during registration a need to be
   on the path of inbound requests to the SIP-PBX.

      These intermediate SIP entities can be edge proxies, session
      border controllers, etc..

   REQ11 - The mechanism MUST work when a SIP-PBX obtains its IP address
   dynamically.

   REQ12 - The mechanism MUST work without requiring the SIP-PBX to have
   a domain name or the ability to publish its domain name in the DNS.

   REQ13 - For a given SIP-PBX and its SSP, there MUST be no impact on
   other domains, which are expected to be able to use normal RFC 3263
   procedures to route requests, including requests needing to be routed
   via the SSP in order to reach the SIP-PBX.

   REQ14 - The mechanism MUST be able to operate over a transport that
   provides end-to-end integrity protection and confidentiality between
   the SIP-PBX and the SSP, e.g., using TLS as specified in [RFC3261].

   REQ15 - The mechanism MUST support authentication of the SIP-PBX by
   the SSP and vice versa, e.g., using SIP digest authentication plus
   TLS server authentication as specified in [RFC3261].

   REQ16 - The mechanism MUST allow the SIP-PBX to provide its UAs with
   public or temporary Globally Routable UA URIs (GRUUs) [RFC5627].

   REQ17 - The mechanism MUST work over any existing transport specified
   for SIP, including UDP.

   REQ19 - Documentation MUST give guidance or warnings about how
   authorization policies may be affected by the mechanism, to address
   the problems described in Section 3.3.

   REQ20 - The mechanism MUST be extensible to allow a set of assigned
   telephone numbers to comprise local numbers as specified in



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   [RFC3966], which can be in contiguous ranges, discrete, or in any
   combination of the two.

   REQ21 - The mechanism MUST be extensible to allow a set of
   arbitrarily assigned SIP URI's as specified in [RFC3261], as opposed
   to just telephone numbers, without requiring a complete change of
   mechanism as compared to that used for telephone numbers.


5.  Desirables

   The following are desirable properties of the mechanism.

   In addition to the desirables below, the general aim is to require
   only relatively modest changes to a substantial population of
   existing SSP and SIP-PBX implementations, in order to encourage a
   fast market adoption of the standardized mechanism.  Ease of market
   adoption is paramount here.  Many SIP-PBXs and SSPs have implemented
   mechanisms based on the REGISTER method, and the need for substantial
   changes to those implementations will discourage convergence on a
   single standard in the foreseeable future.

   DES1 - The mechanism SHOULD allow an SSP to exploit its mechanisms
   for providing SIP service to normal UAs in order to provide a SIP
   trunking service to SIP-PBXs.

   DES2 - The mechanism SHOULD scale to SIP-PBXs of several thousand
   assigned telephone numbers.

      This will probably preclude any mechanism involving a separate
      REGISTER transaction per assigned telephone number.

      In practice, the mechanism is more likely to be used on SIP-PBXs
      with up to a few hundred telephone numbers, but it is impossible
      to give a precise limit, and hence the desire to be able to
      support several thousand.

   DES3 - The mechanism SHOULD scale to support several thousand SIP-
   PBXs on a single SSP.


6.  Non-requirements

   The means by which a third domain can route a request to the SSP for
   onward delivery to the SIP-PBX is outside the scope of this work.
   This is related to REQ13, which requires normal routing based on RFC
   3263 to be used.




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   Provisioning is outside the scope of this work.  In particular, a SSP
   will need to assign a set of telephone numbers to a SIP-PBX, and a
   SIP-PBX will need to be aware of the set of assigned numbers and
   allocate those numbers to its users.  Automated means for a SIP-PBX
   to obtain, from its SSP, the set of assigned telephone numbers is
   considered to be a provisioning topic.


7.  IANA considerations

   This document requires no IANA actions.


8.  Security considerations

   Security of signaling between the SIP-PBX and the SSP is important.
   Some of the requirements above already address this.

   In particular, it is important that an entity acting as a SIP-PBX
   cannot register with an SSP and receive inbound requests to which it
   is not entitled.  The SSP is assumed to have procedures for ensuring
   that a SIP-PBX is entitled to use a set of E.164 telephone numbers
   prior to entering into agreement with that SIP-PBX for using those
   telephone numbers with this mechanism.  Furthermore, by
   authenticating the SIP-PBX when it provides reachability information,
   the SSP can be sure that it delivers inbound requests only to the
   correct destination.


9.  Acknowledgements

   The contents of the document have been compiled from extensive
   discussions within the MARTINI WG, the individuals concerned being
   too numerous to mention.


10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.




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

10.2.  Informative References

   [RFC3325]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", RFC 3325,
              November 2002.

   [RFC3327]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Registering Non-Adjacent
              Contacts", RFC 3327, December 2002.

   [RFC3455]  Garcia-Martin, M., Henrikson, E., and D. Mills, "Private
              Header (P-Header) Extensions to the Session Initiation
              Protocol (SIP) for the 3rd-Generation Partnership Project
              (3GPP)", RFC 3455, January 2003.

   [RFC3966]  Schulzrinne, H., "The tel URI for Telephone Numbers",
              RFC 3966, December 2004.

   [RFC4244]  Barnes, M., "An Extension to the Session Initiation
              Protocol (SIP) for Request History Information", RFC 4244,
              November 2005.

   [RFC5626]  Jennings, C., Mahy, R., and F. Audet, "Managing Client-
              Initiated Connections in the Session Initiation Protocol
              (SIP)", RFC 5626, October 2009.

   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User
              Agent URIs (GRUUs) in the Session Initiation Protocol
              (SIP)", RFC 5627, October 2009.

   [3GPP TS 23.003]
              "3GPP TS 23.003 "3rd Generation Partnership Project;
              Technical Specification Group Core Network and Terminals;
              Numbering, addressing and identification"".

   [3GPP TS 24.229]
              "3GPP TS 24.229 "3rd Generation Partnership Project;
              Technical Specification Group Core Network and Terminals;
              IP multimedia call control protocol based on Session
              Initiation Protocol (SIP) and Session Description Protocol
              (SDP); Stage 3"".

   [ETSI TS 182 025]



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              "ETSI TS 182 025 "Telecommunications and Internet
              converged Services and Protocols for Advanced Networking
              (TISPAN); Business trunking; Architecture and functional
              description"".


Authors' Addresses

   John Elwell
   Siemens Enterprise Communications

   Phone:  +44 1908 855608
   Email:  john.elwell@siemens-enterprise.com


   Hadriel Kaplan
   Acme Packet
   71 Third Ave.
   Burlington, MA  01803
   USA

   Email:  hkaplan@acmepacket.com





























Elwell & Kaplan         Expires January 27, 2011               [Page 13]


