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INFORMATIONAL
Internet Engineering Task Force (IETF)                         J. ElwellRequest for Comments: 5947             Siemens Enterprise CommunicationsCategory: Informational                                        H. KaplanISSN: 2070-1721                                              Acme Packet                                                          September 2010Requirements for Multiple Address of Record (AOR) ReachabilityInformation in the Session Initiation Protocol (SIP)Abstract   This document states requirements for a standardized SIP registration   mechanism for multiple addresses of record (AORs), 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 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; seeSection 2 of RFC 5741.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc5947.Elwell & Kaplan               Informational                     [Page 1]

RFC 5947            Multiple AOR reachability in SIP      September 2010Copyright Notice   Copyright (c) 2010 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 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.Table of Contents1. Introduction ....................................................32. Terminology .....................................................33. Problem Statement ...............................................43.1. Issues with the REGISTER Transaction .......................53.1.1. Mishandling by SIP-Aware Middleboxes ................53.1.2. REGISTER Response Growth ............................63.1.3. Illegal "Wildcard" Syntax ...........................63.2. Issues with Routing Inbound Requests to the SIP-PBX ........63.2.1. Loss of Target Information ..........................6           3.2.2. Inconsistent Placement of Target URI                  Information in Inbound Requests .....................63.2.3. Request-URI Misrouting ..............................73.3. Policy-Related Issues ......................................73.3.1. Authorization Policy Mismatches .....................73.3.2. PAI or PPI URI Mismatches ...........................74. Requirements ....................................................85. Desirables .....................................................106. Non-Requirements ...............................................117. Security considerations ........................................118. Acknowledgements ...............................................129. References .....................................................129.1. Normative References ......................................129.2. Informative References ....................................12Elwell & Kaplan               Informational                     [Page 2]

RFC 5947            Multiple AOR reachability in SIP      September 20101.  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 to 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, these mechanisms 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 inSection 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",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this   document are to be interpreted as described in [RFC2119].Elwell & Kaplan               Informational                     [Page 3]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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_182025] and   3rd Generation Partnership Project (3GPP) [3GPP.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   domain of the SSP, and both the To and From URIs used for the   REGISTER request identify that AOR.  In all respects, it appears onElwell & Kaplan               Informational                     [Page 4]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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.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 Transaction3.1.1.  Mishandling 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 theElwell & Kaplan               Informational                     [Page 5]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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 a 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 "Wildcard" 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-PBX3.2.1.  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 or configured for one variant attempts to interwork with   an SSP designed or configured for another variant.Elwell & Kaplan               Informational                     [Page 6]

RFC 5947            Multiple AOR reachability in SIP      September 20103.2.3.  Request-URI Misrouting   Although many SIP-PBXs support registration with an SSP, they do not   consider themselves authoritative for the explicitly or implicitly   registered AORs if the domain portion still identifies the SSP's   domain.  They expect the domain portion to be their own IP Address,   Fully Qualified Domain Name (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 Back-to-Back User Agent (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 registeredElwell & Kaplan               Informational                     [Page 7]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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   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.Elwell & Kaplan               Informational                     [Page 8]

RFC 5947            Multiple AOR reachability in SIP      September 2010   REQ7:   The mechanism MUST provide a means whereby a SIP-PBX knows at           which of its assigned telephone numbers an inbound request           from its SSP is targeted.   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 the registration of              multiple telephone numbers, but the SSP believes a single              AOR is being registered.   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 normalRFC 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 Transport Layer           Security (TLS) as specified in [RFC3261].Elwell & Kaplan               Informational                     [Page 9]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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.   REQ18:  Documentation MUST give guidance or warnings about how           authorization policies may be affected by the mechanism, to           address the problems described inSection 3.3.   REQ19:  The mechanism MUST be extensible to allow a set of assigned           telephone numbers to comprise local numbers as specified in           [RFC3966], which can be in contiguous ranges, discrete, or in           any combination of the two.   REQ20:  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.Elwell & Kaplan               Informational                    [Page 10]

RFC 5947            Multiple AOR reachability in SIP      September 2010   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 onRFC3263 be used.   Provisioning is outside the scope of this work.  In particular, an   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.  Security considerations   The 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.Elwell & Kaplan               Informational                    [Page 11]

RFC 5947            Multiple AOR reachability in SIP      September 20108.  Acknowledgements   The contents of the document have been compiled from extensive   discussions within the MARTINI WG, the individuals concerned being   too numerous to mention.9.  References9.1.  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.   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation              Protocol (SIP): Locating SIP Servers",RFC 3263,              June 2002.9.2.  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.Elwell & Kaplan               Informational                    [Page 12]

RFC 5947            Multiple AOR reachability in SIP      September 2010   [RFC5627]  Rosenberg, J., "Obtaining and Using Globally Routable User              Agent URIs (GRUUs) in the Session Initiation Protocol              (SIP)",RFC 5627, October 2009.   [3GPP.23.003]              3GPP, "Numbering, addressing and identification", 3GPP              TS 23.003 3.15.0, October 2006.   [3GPP.24.229]              3GPP, "IP multimedia call control protocol based on              Session Initiation Protocol (SIP) and Session Description              Protocol (SDP); Stage 3", 3GPP TS 24.229 10.0.0,              June 2010.   [ETSI_TS_182025]              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   EMail: john.elwell@siemens-enterprise.com   Hadriel Kaplan   Acme Packet   71 Third Ave.   Burlington, MA  01803   USA   EMail: hkaplan@acmepacket.comElwell & Kaplan               Informational                    [Page 13]

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