MARTINI WG                                                   A. B. Roach
Internet-Draft                                                   Tekelec
Intended status: Standards Track                          April 19, 2010
Expires: October 21, 2010


   Registration for Multiple Phone Numbers in the Session Initiation
                             Protocol (SIP)
                       draft-ietf-martini-gin-01

Abstract

   This document defines a mechanism by which a SIP server acting as a
   traditional Private Branch Exchange (PBX) can register with a SIP
   Service Provider (SSP) to receive phone calls for terminals
   designated by phone numbers.  In order to function properly, this
   mechanism relies on the fact that the phone numbers are fully
   qualified and globally unique.

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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on October 21, 2010.

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
   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



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Constraints  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Mechanism Overview . . . . . . . . . . . . . . . . . . . . . .  4
   5.  Registering for Multiple Phone Numbers . . . . . . . . . . . .  4
   6.  SSP Processing of Inbound Phone Number Requests  . . . . . . .  6
   7.  Interaction with Other Mechanisms  . . . . . . . . . . . . . .  7
     7.1.  Globally Routable User-Agent URIs (GRUU) . . . . . . . . .  7
       7.1.1.  Public GRUUs . . . . . . . . . . . . . . . . . . . . .  7
       7.1.2.  Temporary GRUUs  . . . . . . . . . . . . . . . . . . .  9
     7.2.  Registration Event Package . . . . . . . . . . . . . . . . 10
       7.2.1.  PBX Aggregate Registration State . . . . . . . . . . . 10
       7.2.2.  Individual Phone Number Registration State . . . . . . 11
     7.3.  Client-Initiated (Outbound) Connections  . . . . . . . . . 11
     7.4.  Non-Adjacent Contact Registration (Path) and Service
           Route Discovery  . . . . . . . . . . . . . . . . . . . . . 11
   8.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     8.1.  Usage Scenario: Basic Registration . . . . . . . . . . . . 13
     8.2.  Usage Scenario: Using Path to Control Request URI  . . . . 14
   9.  Requirements Analysis  . . . . . . . . . . . . . . . . . . . . 16
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
     10.1. New SIP Option Tag . . . . . . . . . . . . . . . . . . . . 20
     10.2. New SIP URI Parameters . . . . . . . . . . . . . . . . . . 21
       10.2.1. 'bnc' SIP URI paramter . . . . . . . . . . . . . . . . 21
       10.2.2. 'sg' SIP URI paramter  . . . . . . . . . . . . . . . . 21
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 21
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 21
     13.2. Informative References . . . . . . . . . . . . . . . . . . 22
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 22














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1.  Introduction

   One of SIP's primary functions is providing rendezvous between users.
   By design, this rendezvous has been provided through a combination of
   the server look-up procedures defined in RFC 3263 [3], and the
   registrar procedures described in RFC 3261 [2].

   The intention of the original protocol design was that any user's AOR
   would be handled by the authority indicated by the hostport portion
   of the AOR.  The users registered individual reachability information
   with this authority, which would then route incoming requests
   accordingly.

   In actual deployments, some SIP servers have been deployed in
   architectures that, for various reasons, have requirements to provide
   dynamic routing information for large blocks of AORs, where all of
   the AORs in the block were to be handled by the same server.  For
   purposes of efficiency, many of these deployments do not wish to
   maintain separate registrations for each of the AORs in the block.
   This leads to the desire for an alternate mechanism for providing
   dynamic routing information for blocks of AORs.

   Although the use of REGISTER to update reachability information for
   multiple users simultaneously is somewhat beyond the original
   semantics defined for REGISTER, this approach has seen significant
   deployment in certain environments.  In particular, deployments in
   which small to medium SIP PBX servers are addressed using E.164
   numbers have used this mechanism to avoid the need to maintain DNS
   entries or static IP addresses for the PBX servers.

   In recognition of the momentum that REGISTER-based approaches have
   seen in deployments, this document defines a REGISTER-based approach
   that is tailored to E.164-addressed terminals in a SIP PBX
   environment.  It does not address registration of SIP URIs in which
   the user portion is not an E.164 number.


2.  Constraints

   The following paragraph is perhaps the most important in
   understanding the solution defined in this document.

   Within the problem space that has been established for this work,
   several constraints shape our solution.  These are being defined in
   the MARTINI requirements document [5].  In terms of impact to the
   solution at hand, the following two constraints have the most
   profound effect: (1) The PBX cannot be assumed to be assigned a
   static IP address; and (2) No DNS entry can be relied upon to



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   consistently resolve to the IP address of the PBX.


3.  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 RFC 2119 [1].

   Further, the term "SSP" is meant as an acronym for a "SIP Service
   Provider," while the term "PBX" is used to indicate a SIP Private
   Branch Exchange.


4.  Mechanism Overview

   The overall mechanism is achieved using a REGISTER request with a
   specially-formatted Contact URI.  This document also defines an
   option tag that can be used to ensure a registrar and any
   intermediaries understand the mechanism described herein.

   The Contact URI itself is tagged with a URI parameter to indicate
   that it actually represents a multitude of phone-number-associated
   contacts.

   We also define some lightweight extensions for Globally Routable UA
   URIs (GRUU) to allow the use of public and temporary GRUUs assigned
   by the SSP.

   Aside from these extensions, the REGISTER message itself is processed
   by a registrar in the same way as normal registrations: by updating
   its location service with additional AOR to Contact bindings.

   Note that the list of phone numbers associated with a PBX is a matter
   of local provisioning at the SSP and at the PBX.  The mechanism
   defined in this document does not provide any means to detect or
   recover from provisioning mismatches (although the registration event
   package can be used as a standardized means for auditing such phone
   numbers; see Section 7.2.1).


5.  Registering for Multiple Phone Numbers

   To register for multiple phone numbers, the PBX sends a REGISTER
   message to the SSP.  This REGISTER varies from a typical register in
   two important ways.  First, it must contain an option tag of "gin" in
   both a "Require" header field and a "Proxy-Require" header field.
   (The option tag "gin" is an acronym for "generate implicit numbers".)



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   Second, in at least one "Contact" header field, it must include a
   Contact URI that contains the URI parameter "bnc", and no user
   portion (hence no "@" symbol).  A URI with a "bnc" parameter MUST NOT
   contain a user portion.

   Because of the constraints discussed in Section 2, the host portion
   of the Contact URI will generally contain an IP address, although
   nothing in this mechanism enforces or relies upon that fact.  If the
   PBX operator chooses to maintain DNS entries that resolve to the IP
   address of his PBX via RFC 3263 resolution procedures, then this
   mechanism works just fine with domain names in the Contact header
   field.

   The URI parameter indicates that special interpretation of the
   Contact URI is necessary: instead of representing a single, concrete
   Contact URI to be inserted into the location service, it represents a
   multitude of Contact URIs (one for each associated phone number),
   semantically resulting in a multitude of AOR-to-Contact rows in the
   location service.

   The registrar, upon receipt of a REGISTER message in the foregoing
   form, will use the value in the "To" header field to identify the PBX
   for which registration is being requested.  It then authenticates the
   PBX (using, e.g., SIP Digest authentication, mutual TLS, or some
   other authentication mechanism).  After the PBX is authenticated, the
   registrar updates its location service with a unique AOR to Contact
   mapping for each of the phone numbers associated with the PBX.
   Semantically, each of these mappings will be treated as a unique row
   in the location service.  The actual implementation may, of course,
   perform internal optimizations to reduce the amount of memory used to
   store such information.

   For each of these unique rows, the AOR will be in the format that the
   SSP expects to receive from external parties (e.g.
   "sip:+12145550102@ssp.example.com"), and the corresponding Contact
   will be formed by adding to the REGISTER's Contact URI a user portion
   containing the fully-qualified, E.164-formatted phone number
   (including the preceding "+" symbol) and removing the "bnc"
   parameter.  Aside from the initial "+" symbol, this E.164-formatted
   number MUST consist exclusively of digits from 0 through 9, and
   explicitly MUST NOT contain any visual separator symbols (e.g., "-",
   ".", "(", or ")").  For example, if the "Contact" header field
   contains the URI <sip:198.51.100.3:5060;user=phone;bnc>, then the
   Contact value associated with the aforementioned AOR will be
   <sip:+12145550102@198.51.100.3:5060;user=phone>.

   Aside from the "bnc" parameter, all URI parameters present on the
   "Contact" URI in the REGISTER message MUST be copied to the Contact



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   value stored in the location service.

   Although the SSP treats this registration as a number of discrete
   rows for the purpose of retargeting incoming requests, the renewal,
   expiration, and removal of these rows is bound to the registered
   "bnc" contact.  In particular, this means that REGISTER requests that
   attempt to de-register a single phone number that has been implicitly
   registered MUST NOT remove that phone number from the bulk
   registration.  A further implication of this property is that an
   individual extension that is implicitly registered may also be
   explicitly registered using a normal, non-bulk registration (subject
   to SSP policy).  If such a registration exists, it is refreshed
   independently of the bulk registration, and is not removed when the
   bulk registration is removed.

   Although it is not a special case from a protocol perspective,
   implementors are reminded that this behavior includes any "user"
   parameter on the URI.  For example, if the registering PBX requires
   incoming messages to include a "user=phone" parameter on their
   Request-URI, it will include "user=phone" on the URI in the "Contact"
   header field of its registration.  Because of the requirement on the
   SSP to copy parameters into the location service, it will appear on
   any Request-URIs that are retargeted to the PBX based on that
   registration.  Conversely, if the registering PBX does not include a
   "user=phone" parameter, any SSP using the procedures defined in this
   document would not be at liberty to include one on the Request-URI.

   Any PBX implementing the registration mechanism defined in this
   document MUST also support the Path mechanism defined by RFC 3327
   [6], and MUST include a 'path' option-tag in the Supported header
   field of the REGISTER request (which is a stronger requirement than
   imposed by the Path mechanism itself).  This behavior is necessary
   because proxies between the PBX and the Registrar may need to insert
   Path header field values in the REGISTER request for this document's
   mechanism to function properly, and per RFC 3327 [6], they can only
   do so if the UAC inserted the option-tag in the Supported header
   field.  In accordance with the procedures defined in RFC 3327 [6],
   the PBX is allowed to ignore the Path header fields returned in the
   REGISTER response.

   A Registrar compliant with this document MUST support the Path
   mechanism defined in RFC 3327 [6].


6.  SSP Processing of Inbound Phone Number Requests

   In general, after processing the AOR to Contact mapping described in
   the preceding section, the SSP Proxy/Registrar (or equivalent entity)



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   performs traditional Proxy/Registrar behavior, based on the mapping.
   For any inbound SIP requests whose AOR indicates an E.164 number
   assigned to one of the SSP's customers, this will generally involve
   setting the target set to the registered contacts associated with
   that AOR, and performing request forwarding as described in section
   16.6 of RFC 3261 [2].  An SSP using the mechanism defined in this
   doucument MUST perform such processing for inbound INVITE requests
   and SUBSCRIBE requests to the "reg" event package (see
   Section 7.2.2), and SHOULD perform such processing for all other
   method types, including unrecognized SIP methods.


7.  Interaction with Other Mechanisms

   The following sections describe the means by which this mechanism
   interacts with relevant REGISTER-related extensions currently defined
   by the IETF.

   Currently, the descriptions are somewhat informal, and omit some
   details for the sake of brevity.  If the MARTINI working group
   expresses interest in furthering the mechanism described by this
   document, they will be fleshed out with more detail and formality.

7.1.  Globally Routable User-Agent URIs (GRUU)

   To enable advanced services to work with terminals behind a SIP PBX,
   it is important that the GRUU mechanism defined by RFC 5627 [10] work
   correctly with the mechanism defined by this document -- that is,
   that User Agents services by the PBX can acquire and use GRUUs for
   their own use.

7.1.1.  Public GRUUs

   When a PBX registers a Bulk Number Contact (a Contact with a "bnc"
   parameter), and also invokes GRUU procedures for that Contact during
   registration, then the SSP will assign a public GRUU to the PBX in
   the normal fashion.  Because the URI being registered contains a
   "bnc" parameter, the GRUU will also contain a "bnc" parameter.  In
   particular, this means that the GRUU will not contain a user portion.

   When a terminal registers with the PBX using GRUU procedures for a
   Contact, the PBX adds an "sg" parameter to the GRUU parameter it
   received from the SSP.  This "sg" parameter contains a disambiguation
   token that the PBX can use to route the request to the proper user
   agent.

   So, for example, when the PBX registers with the following contact
   header field:



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   Contact: <sip:198.51.100.3;user=phone;bnc>;
     +sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"

   Then the SSP may choose to respond with a Contact header field that
   looks like this:

   <allOneLine>
   Contact: <sip:198.51.100.3;user=phone;bnc>;
   pub-gruu="sip:ssp.example.com;gr=urn:
   uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6";
   +sip.instance="<urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>"
   ;expires=7200
   </allOneLine>

   When its own terminals register, the PBX can then add whatever device
   identifier it feels appropriate in an "sg" parameter, and present
   this value to its own terminals.  For example, assume the terminal
   associated with the phone number "+12145550102" sent the following
   Contact header field in its register:

   Contact: <sip:line-1@10.20.1.17>;
     +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"

   The PBX will add an "sg" parameter to the pub-gruu it received from
   the SSP with a token that uniquely identifies the device (possibly
   the URN itself; possibly some other identifier); insert a user
   portion containing the fully-qualified E.164 number associated with
   the terminal; and return the result to the terminal as its public
   GRUU.  The resulting Contact header field would look something like
   this:

   <allOneLine>
   Contact: <sip:line-1@10.20.1.17>;
   pub-gruu="sip:+12145550102@ssp.example.com;gr=urn:
   uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6;sg=00:05:03:5e:70:a6";
   +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
   ;expires=3600
   </allOneLine>

   When an incoming request arrives at the SSP for a GRUU corresponding
   to a bulk number contact ("bnc"), the SSP performs slightly different
   processing for the GRUU than a Proxy/Registrar would.  When the GRUU
   is retargeted to the registered bulk number contact, the SSP MUST
   copy the "sg" parameter from the GRUU to the new target.  The PBX can
   then use this "sg" parameter to determine which user agent the
   request should be routed to.  For example, the first line of an
   INVITE request that has been retargeted to the PBX for the terminal
   shown above would look like this:



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   <allOneLine>
   INVITE sip:+12145550102@198.51.100.3;gr=urn:uuid:f81d4fae-7dec-
   11d0-a765-00a0c91e6bf6;sg=00:05:03:5e:70:a6 SIP/2.0
   </allOneLine>

7.1.2.  Temporary GRUUs

   PBXes have two options for creating temporary GRUUs for use by its
   terminals.

7.1.2.1.  Approach 1 - Self Made GRUUs

   If a PBX wishes to provide temporary GRUUs for its terminals, it may
   do so by producing its own "Self-made GRUUs" (as defined in section
   4.3 of RFC 5627 [10]).  These GRUUs are produced using the PBX's own
   IP address (or domain, if it maintains one in DNS).  The temporary
   GRUUs are then propagated to terminals using normal GRUU mechanism.

   The ability to produce temporary GRUUs in this fashion is predicated
   on the conditions described in section 4.3 of RFC 5627.  In
   particular, it requires PBX to be publicly routable, and willing to
   accept requests destined for its own Self-made GRUUs from sources
   other than the SSP.  If these conditions cannot be satisfied (or the
   PBX operator chooses not to satisfy them for policy reasons), then
   the PBX users will not be able to make use of temporary GRUUs.

   This mechanism is also predicated on the IP address for the PBX being
   relatively stable over a long period of time.  This is generally a
   safe assumption to make, as frequent PBX IP address changes will
   result in intermittent connectivity issues and interruptions to
   ongoing calls.

   On a related note: when used with this extension, the SSP will not
   return a temporary GRUU in the registration response for any contacts
   that include a "bnc" parameter in their URI.

   For example, using the same setup as in the "Public GRUU" section
   above, a terminal registering with the PBX might obtain a temp gruu
   by receiving a Contact header field that looks like:

   <allOneLine>
   Contact: <sip:line-1@10.20.1.17>;
   pub-gruu="sip:ssp.example.com;gr=urn:uuid:f81d4fae-7dec-11d0-a765-
   00a0c91e6bf6;sg=a0471c99573b877b";
   +sip.instance="<urn:uuid:d0e2f290-104b-11df-8a39-0800200c9a66>"
   ;expires=3600
   </allOneLine>




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7.1.2.2.  Approach 2 - Anonymous Public GRUUs

   If a PBX does not satisfy the criteria for producing its own "Self-
   made GRUUs," then it may create temporary GRUUs based on the public
   GRUUs it received from the SSP at registration time.  To create
   Temporary GRUUs of this form, the PBX will add an opaque "sg"
   parameter to the public GRUU it received from the SSP, and will omit
   the user portion.

   Note that, because these GRUUs are temporary GRUUs, a unique "sg"
   parameter will be generated for each successful registration attempt.
   The PBX tracks the various "sg" values associated with each user
   agent, and can retarget to the correct instance when the request
   arrives.

   For this approach to function, the SSP must be able to resolve a GRUU
   based solely on the value of its "gr" parameter, as the user portion
   of the GRUU will not contain an E.164 number.  Further, the SSP will
   not know which actual phone number the request is destined for, only
   that it corresponds to an phone number belonging to the PBX.

   Using the same basic setup as the example for the public GRUU, a
   terminal might receive a temporary GRUU by getting back a Contact
   header field that looks like this:

   <allOneLine>
   Contact: <sip:line-1@10.20.1.17>;
   temp-gruu="sip:ssp.example.com;gr=urn:uuid:f81d4fae-7dec-11d0-a765-
   00a0c91e6bf6;sg=0UYYRV046P";+sip.instance="<urn:uuid:d0e2f290-104b-
   11df-8a39-0800200c9a66>";expires=3600
   </allOneLine>

7.2.  Registration Event Package

   As this mechanism inherently deals with REGISTER behavior, it is
   imperative to consider its impact on the Registration Event Package
   defined by RFC 3680 [8].  In practice, there will be two main use
   cases for subscribing to registration data: learning about the
   overall registration state for the PBX, and learning about the
   registration state for a single PBX phone number.

7.2.1.  PBX Aggregate Registration State

   If the PBX (or another interested and authorized party) wishes to
   monitor or audit the registration state for all of the phone numbers
   currently registered to that PBX, it can subscribe to the SIP
   registration event package at the PBX's main URI -- that is, the URI
   used in the "To" header field of the REGISTER message.



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   The NOTIFY messages for such a subscription will contain a body that
   contains one record for each phone number associated with the PBX.
   The AORs will be in the format expected to be received by the SSP
   (e.g., "sip:+12145550105@ssp.example.com"), and the Contacts will
   correspond to the mapped Contact created by the registration (e.g.,
   "sip:+12145550105@98.51.100.3").

   In particular, the "bnc" parameter is forbidden from appearing in the
   body of a reg-event notify.

7.2.2.  Individual Phone Number Registration State

   As described in Section 6, the SSP will generally retarget all
   requests addressed to a phone number owned by a PBX to that PBX
   according to the mapping established at registration time.  Although
   policy at the SSP may override this generally expected behavior,
   proper behavior of the registration event package requires that all
   "reg" event SUBSCRIBE requests are processed by the PBX.  As a
   consequence, the requirements on an SSP for processing registration
   event package SUBSCRIBE requests are not left to policy.

   If the SSP receives a SUBSCRIBE request for the registration event
   package with a Request-URI that indicates a contact registered via
   the "Bulk Number Contact" mechanism defined in this document, then it
   MUST proxy that SUBSCRIBE to the PBX in the same way that is would
   proxy an INVITE bound for that AOR.

   Defining the behavior in this way is important, since the reg-event
   subscriber is interested in finding out about the comprehensive list
   of devices associated with the phone number.  Only the PBX will have
   authoritative access to this information.  For example, if the user
   has registered multiple terminals with differing capabilities, the
   SSP will not know about the devices or their capabilities.  By
   contrast, the PBX will.

7.3.  Client-Initiated (Outbound) Connections

   RFC 5626 [9] -- needs analysis.  Some people think it might "just
   work."

7.4.  Non-Adjacent Contact Registration (Path) and Service Route
      Discovery

   RFC 3327 [6] defines a means by which a registrar and its associated
   proxy can be informed of a route that is to be used between the proxy
   and the registered user agent.  The scope of the route created by a
   "Path" header field is contact-specific; if an AOR has multiple
   contacts associated with it, the routes associated with each contact



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   may be different from each other.

   At registration time, any proxies between the user agent and the
   registrar may add themselves to the Path.  By doing so, they request
   that any requests destined to the user agent as a result of the
   associated registration include them as part of the Route towards the
   User Agent.  Although the Path mechanism does deliver the final Path
   value to the registering UA, UAs typically ignore the value of the
   Path.

   To provide similar functionality in the opposite direction -- that
   is, to establish a route for requests sent by a registering UA -- RFC
   3608 [7] defines a means by which a UA can be informed of a route
   that is to be used by the UA to route all outbound requests
   associated with the AOR used in the registration.  This information
   is scoped to the AOR within the UA, and is not specific to the
   Contact (or Contacts) in the REGISTER request.

   The registrar unilaterally generates the values of the service route
   using whatever local policy it wishes to apply.  Although it is
   common to use the Path and/or Route information in the request in
   composing the Service-Route, registrar behavior is not constrained in
   any way that requires it to do so.

   In considering the interaction between these mechanisms and the
   registration of multiple AORs in a single request, implementors of
   proxies, registrars, and intermediaries must keep in mind the
   following issues, which stem from the fact that GIN effectively
   registers multiple AORs and multiple Contacts.

   First, all location service records that result from expanding a
   single "bnc" Contact will necessarily share a single path.  Proxies
   will be unable to make policy decisions on a contact-by-contact basis
   regarding whether to include themselves in the path.  Second, and
   similarly, all AORs on the PBX that are registered with a common
   REGISTER message will be forced to share a common Service-Route.

   One interesting technique that Path and Service-Route enable is the
   inclusion of a token or cookie in the user portion of the Service-
   Route or Path entries.  This token or cookie may convey information
   to proxies about the identity, capabilities, and/or policies
   associated with the user.  Since this information will be shared
   among several AORs and several Contacts when multiple AOR
   registration is employed, care should be taken to ensure that doing
   so is acceptable for all AORs and all Contacts registered in a single
   REGISTER message.





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8.  Examples

8.1.  Usage Scenario: Basic Registration

   This example shows the message flows for a basic bulk REGISTER
   transaction, followed by an INVITE addressed to one of the registered
   terminals.  Example messages are shown after the sequence diagram.

   Internet                        SSP                              PBX
   |                                |                                 |
   |                                |(1) REGISTER                     |
   |                                |Contact:<sip:198.51.100.3;bnc>   |
   |                                |<--------------------------------|
   |                                |                                 |
   |                                |(2) 200 OK                       |
   |                                |-------------------------------->|
   |                                |                                 |
   |(3) INVITE                      |                                 |
   |sip:+12145550105@ssp.example.com|                                 |
   |------------------------------->|                                 |
   |                                |                                 |
   |                                |(4) INVITE                       |
   |                                |sip:+12145550105@198.51.100.3    |
   |                                |-------------------------------->|


   (1) The PBX registers with the SSP for a range of phone numbers.

   REGISTER sip:ssp.example.com SIP/2.0
   Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7
   Max-Forwards: 70
   To: <sip:pbx@ssp.example.com>
   From: <sip:pbx@ssp.example.com>;tag=a23589
   Call-ID: 843817637684230@998sdasdh09
   CSeq: 1826 REGISTER
   Proxy-Require: gin
   Require: gin
   Supported: path
   Contact: <sip:198.51.100.3:5060;user=phone;bnc>
   Expires: 7200
   Content-Length: 0










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   (3) The SSP receives a request for a phone number assigned
       to the PBX.

   INVITE sip:+12145550105@ssp.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
   Max-Forwards: 69
   To: <sip:2145550105@some-other-place.example.net>
   From: <sip:gsmith@example.org>;tag=456248
   Call-ID: f7aecbfc374d557baf72d6352e1fbcd4
   CSeq: 24762 INVITE
   Contact: <sip:line-1@192.0.2.178:2081>
   Content-Type: application/sdp
   Content-Length: ...

   <sdp body here>


   (4) The SSP retargets the incoming request according to the
       information received from the PBX at registration time.

   INVITE sip:+12145550105@198.51.100.3;user=phone SIP/2.0
   Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
   Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50
   Max-Forwards: 68
   To: <sip:2145550105@some-other-place.example.net>
   From: <sip:gsmith@example.org>;tag=456248
   Call-ID: 7ca24b9679ffe9aff87036a105e30d9b
   CSeq: 24762 INVITE
   Contact: <sip:line-1@192.0.2.178:2081>
   Content-Type: application/sdp
   Content-Length: ...

   <sdp body here>

8.2.  Usage Scenario: Using Path to Control Request URI

   This example shows a bulk REGISTER transaction with the SSP making
   use of the "Path" header field extension [6].  This allows the SSP to
   designate a domain on the incoming Request URI that does not
   necessarily resolve to the PBX from when the SSP applies RFC 3263
   procedures to it.










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   Internet                        SSP                              PBX
   |                                |                                 |
   |                                |(1) REGISTER                     |
   |                                |Path:<sip:pbx@198.51.100.3;lr>   |
   |                                |Contact:<sip:pbx.example;bnc>    |
   |                                |<--------------------------------|
   |                                |                                 |
   |                                |(2) 200 OK                       |
   |                                |-------------------------------->|
   |                                |                                 |
   |(3) INVITE                      |                                 |
   |sip:+12145550105@ssp.example.com|                                 |
   |------------------------------->|                                 |
   |                                |                                 |
   |                                |(4) INVITE                       |
   |                                |sip:+12145550105@pbx.example     |
   |                                |Route:<sip:pbx@198.51.100.3;lr>  |
   |                                |-------------------------------->|


   (1) The PBX registers with the SSP for a range of phone numbers.
       It includes the URI it expects to receive in the Request-URI
       in its "Contact" header field, and includes information that
       routes to the PBX in the "Path" header field.

   REGISTER sip:ssp.example.com SIP/2.0
   Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7
   Max-Forwards: 70
   To: <sip:pbx@ssp.example.com>
   From: <sip:pbx@ssp.example.com>;tag=a23589
   Call-ID: 843817637684230@998sdasdh09
   CSeq: 1826 REGISTER
   Proxy-Require: gin
   Require: gin
   Supported: path
   Path: <sip:pbx@198.51.100.3:5060;lr>
   Contact: <sip:pbx.example;user=phone;bnc>
   Expires: 7200
   Content-Length: 0












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   (3) The SSP receives a request for a phone number assigned
       to the PBX.

   INVITE sip:+12145550105@ssp.example.com;user=phone SIP/2.0
   Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
   Max-Forwards: 69
   To: <sip:2145550105@some-other-place.example.net>
   From: <sip:gsmith@example.org>;tag=456248
   Call-ID: f7aecbfc374d557baf72d6352e1fbcd4
   CSeq: 24762 INVITE
   Contact: <sip:line-1@192.0.2.178:2081>
   Content-Type: application/sdp
   Content-Length: ...

   <sdp body here>


   (4) The SSP retargets the incoming request according to the
       information received from the PBX at registration time.
       Per the normal processing associated with "Path," it
       will insert the "Path" value indicated by the PBX at
       registration time in a "Route" header field, and
       set the request URI to the registered Contact.

   INVITE sip:+12145550105@pbx.example;user=phone SIP/2.0
   Via: SIP/2.0/UDP foo.example;branch=z9hG4bKa0bc7a0131f0ad
   Via: SIP/2.0/UDP ssp.example.com;branch=z9hG4bKa45cd5c52a6dd50
   Route: <sip:pbx@198.51.100.3:5060;lr>
   Max-Forwards: 68
   To: <sip:2145550105@some-other-place.example.net>
   From: <sip:gsmith@example.org>;tag=456248
   Call-ID: 7ca24b9679ffe9aff87036a105e30d9b
   CSeq: 24762 INVITE
   Contact: <sip:line-1@192.0.2.178:2081>
   Content-Type: application/sdp
   Content-Length: ...

   <sdp body here>


9.  Requirements Analysis

   The document "Requirements for multiple address of record (AOR)
   reachability information in the Session Initiation Protocol (SIP)"
   [5] contains a list of requirements and desired properties for a
   mechanism to register multiple AORs with a single SIP transaction.
   This section evaluates those requirements against the mechanism
   described in this document.



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   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 deemed to have been assigned to the SIP-PBX.

      The requirement is satisfied.

   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.

      The requirement is satisfied; the DIDs associated with a
      registration is established by bilateral agreement between the SSP
      and the PBX, and is not part of the mechanism described in this
      document.

   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.

      The requirement is satisfied.

   REQ4 - The mechanism MUST NOT prevent UAs attached to a SIP-PBX
   registering with the SIP-PBX on behalf of 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.

      The requirement is satisfied; in the presumed architecture, PBX
      terminals register with the PBX, an require no interaction with
      the SSP.

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

      The requirement is satisfied; PBXes may recognize their own DID
      and their own GRUUs, and perform on-PBX routing without sending
      the 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 PBX can route those requests
   onwards to its UAs, as it would for internal requests to those
   telephone numbers.

      The requirement is satisfied

   REQ7 - The mechanism MUST provide a means whereby a SIP-PBX knows



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   which of its assigned telephone numbers an inbound request from its
   SSP is targeted at.

      The requirement is satisfied.  For ordinary calls and calls using
      Public GRUUs, the DID is indicated in the user portion of the
      Request-URI.  For calls using Temp GRUUs constructed with the
      mechanism described in Section 7.1.2.2, the "sg" parameter
      provides a correlation token the PBX can use to identify which
      terminal the call should be routed to.

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

      The requirement is satisfied through the 'gin' option tag and the
      'bnc' Contact parameter.

   REQ9 - The mechanism MUST observe SIP backwards compatibility
   principles.

      The requirement is satisfied through the 'gin' option tag.

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

      The requirement is satisfied through the use of the Path mechanism
      defined in RFC 3327 [6]

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

      The requirement is satisfied by allowing the PBX to use an IP
      address in the Bulk Number Contact URI contained in a REGISTER
      Contact header field.

   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.

      The requirement is satisfied by allowing the PBX to use an IP
      address in the Bulk Number Contact URI contained in a REGISTER
      Contact header field.

   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.



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      The requirement is satisfied by allowing the domain name in the
      Request URI used by external entities to resolve to the SSP's
      servers via normal RFC 3263 resolution procedures.

   REQ14 - The mechanism MUST be able to operate over a transport that
   provides integrity protection and confidentiality.

      The requirement is satisfied; nothing in the proposed mechanism
      prevent the use of TLS between the SSP and the PBX.

   REQ15 - The mechanism MUST support authentication of the SIP-PBX by
   the SSP and vice versa.

      The requirement is satisfied; PBXes may employ either SIP digest
      authentication or mutually-authenticated TLS for authentication
      purposes.

   REQ16 - The mechanism MUST allow the SIP-PBX to provide its UAs with
   public or temporary Globally Routable UA URIs (GRUUs) [10].

      The requirement is satisfied via the mechanisms detailed in
      Section 7.1.

   REQ17 - The mechanism MUST NOT preclude the ability of the SIP-PBX to
   route on-PBX requests directly, without hair-pinning the signaling
   through the SSP.

      The requirement is satisfied; PBXes may recognize their own DID
      and their own GRUUs, and perform on-PBX routing without sending
      the requests to the SSP.  (Note that this requirement duplicates
      REQ5, and will probably be removed in a future version of the
      requirements document.)

   REQ18 - The mechanism MUST work over any existing transport specified
   for SIP, including UDP.

      The requirement is satisfied to the extent that UDP can be used
      for REGISTER requests in general.  The application of certain
      extensions and/or network topologies may exceed UDP MTU sizes, but
      such issues arise both with and without the mechanism described in
      this document.  This document does not exacerbate such issues.

   DES1 - The mechanism SHOULD allow an SSP to exploit its mechanisms
   for providing SIP service to ordinary subscribers in order to provide
   a SIP trunking service to SIP-PBXes.






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      The desired property is satisfied; the routing mechanism described
      in this document is identical to the routing performed for singly-
      registered AORs.

   DES2 - The mechanism SHOULD scale to SIP-PBX's of several thousand
   assigned telephone numbers.

      The desired property is satisfied; nothing in this document
      precludes DID pools of arbitrary size.

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

      The desired property is satisfied; nothing in this document
      precludes an arbitrary number of PBXes from attaching to a single
      SSP.

   DES4 - The mechanism SHOULD require 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.

      The desired property is difficult to evaluate in the context of
      any solution.  The mechanism proposed in this document uses the
      REGISTER method, which is the method preferred by many existing
      PBX deployments.  The handling of request routing logic is nearly
      identical to that of RFC 3261 proxy/registrars, allowing
      implementors to leverage existing proxy/registrar code.


10.  IANA Considerations

   This document registers a new SIP option tag to indicate support for
   the mechanism it defines, plus two new SIP URI parameters.

10.1.  New SIP Option Tag

   This section defines a new SIP option tag per the guidelines in
   Section 27.1 of RFC 3261[2].
   Name:  gin
   Description:  This option tag is used to identify the extension that
      provides Registration for Multiple Phone Numbers in SIP.  When
      present in a Require or Proxy-Require header field of a REGISTER
      request, it indicates that support for this extension is required
      of registrars and proxies, respectively, that are a party to the
      registration transaction.





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   Reference:  RFCXXXX (this document)

10.2.  New SIP URI Parameters

   This specification defines two new SIP URI parameters, as per the
   registry created by RFC 3969 [4].

10.2.1.  'bnc' SIP URI paramter

   Parameter Name:  bnc
   Predefined Values:  No (no values are allowed)
   Reference:  RFCXXXX (this document)

10.2.2.  'sg' SIP URI paramter

   Parameter Name:  sg
   Predefined Values:  No
   Reference:  RFCXXXX (this document)


11.  Security Considerations

   There are certainly security implications associated with the
   mechanisms described in this document, mostly dealing with the
   unprecedented semantic inclusion of multiple AORs in a single
   REGISTER request.  This section will be formulated following an
   analysis of the security impact of GIN on Path, Service-Route, and
   Outbound.


12.  Acknowledgements

   Thanks to John Elwell for his requirements analysis of the mechanism
   described in this document, and to Dean Willis for his analysis of
   the interaction between this mechanism and the Path and Service-Route
   extensions.


13.  References

13.1.  Normative References

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

   [2]   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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   [3]   Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
         (SIP): Locating SIP Servers", RFC 3263, June 2002.

   [4]   Camarillo, G., "The Internet Assigned Number Authority (IANA)
         Uniform Resource Identifier (URI) Parameter Registry for the
         Session Initiation Protocol (SIP)", BCP 99, RFC 3969,
         December 2004.

13.2.  Informative References

   [5]   Elwell, J. and H. Kaplan, "Requirements for multiple address of
         record (AOR) reachability information in the Session Initiation
         Protocol (SIP)", draft-ietf-martini-reqs-03 (work in progress),
         March 2010.

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

   [7]   Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
         Extension Header Field for Service Route Discovery During
         Registration", RFC 3608, October 2003.

   [8]   Rosenberg, J., "A Session Initiation Protocol (SIP) Event
         Package for Registrations", RFC 3680, March 2004.

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

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


Author's Address

   Adam Roach
   Tekelec
   17210 Campbell Rd.
   Suite 250
   Dallas, TX  75252
   US

   Email: adam@nostrum.com






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