SIPPING Working Group                             M. Garcia   / Ericsson
Internet Draft                                    D. Mills    / Vodafone
Document: <draft-garcia-sipping-3gpp-reqs-00.txt> G. Bajko    / Nokia
Network Working Group                             G. Mayer    / Siemens
Date: October 2001                                F. Derome   / Alcatel
Expires: April 2002                               H. Shieh    / AWS
                                                  A. Allen    / Motorola
                                                  S. Chotai   / BT
                                                  K. Drage    / Lucent
                                                  J. Bharatia / Nortel

                         3GPP requirements on SIP

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026 [1].

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that
   other groups may also distribute working documents as Internet-
   Drafts. 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

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   The distribution of this memo is unlimited.

1. Abstract

   The 3rd Generation Partnership Project (3GPP) has selected SIP [3]
   as the session establishment protocol for the 3GPP IP Multimedia
   Core Network Subsystem (IM CN Subsystem).

   Although SIP is a protocol that fulfills most of the requirements to
   establish a session in an IP network, the SIP protocol suite has
   never been evaluated against the specific 3GPP requirements for
   operation in a cellular network.

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   In this document we express the requirements identified by 3GPP to
   support SIP for IM CN Subsystem in cellular networks.

2. Conventions used in this document

   This document does not specify any protocol of any kind. Therefore,
   the use of te key words "MUST", "MUST NOT", "REQUIRED", "SHALL",
   "OPTIONAL" in this document, as described in RFC-2119 [2], does not

3. Table of Contents

   Status of this Memo................................................1
   1. Abstract........................................................1
   2. Conventions used in this document...............................2
   3. Table of Contents...............................................2
   4. Introduction....................................................2
   5. Overview of the 3GPP IM CN Subsystem............................3
   6. 3GPP Requirements on SIP........................................5
   6.1 General requirements...........................................5
   6.2 SIP outbound proxy in the visited network......................6
   6.3 Registration...................................................7
   6.4 De-registration................................................8
   6.5 Compression of SIP signaling...................................9
   6.6 QoS requirements related to SIP...............................11
   6.7 Prevention of theft of service................................11
   6.8 Radio resource authorization..................................12
   6.9 Prevention of denial of service...............................12
   6.10 Identification of users......................................12
   6.11 Identifiers used for routing.................................14
   6.12 Hiding requirements..........................................14
   6.13 Cell-ID......................................................14
   6.14 Release of sessions..........................................15
   6.15 Routing of SIP messages......................................15
   6.16 Emergency sessions...........................................17
   6.17 Identities on session establishment..........................18
   6.18 Charging.....................................................19
   6.19 IPv6.........................................................19
   6.20 General support of additional capabilities...................19
   6.21 Three-way handshake in the session description negotiation...19
   6.22 Security Model...............................................20
   6.23 Access Domain Security.......................................21
   6.24 Network Domain Security......................................25
   7. Security considerations........................................25
   8. Author's Addresses.............................................25
   9. Acknowledgments................................................27
   10. References....................................................27
   Full Copyright Statement..........................................29

4. Introduction

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   3GPP has selected SIP [3] as the protocol to establish and tear down
   multimedia sessions in the IP Multimedia Core Network Subsystem (IM
   CN Subsystem). A description of the IM CN Subsystem can be found in
   [4]. A comprehensive set of session flows can be found in [5].

   This document is an effort to define the requirements applicable to
   the usage of the SIP protocol suite in cellular networks, and
   particularly in the 3GPP IM CN Subsystem.

   The rest of this document is structured as follows:

   Section 5 offers an overview of the 3GPP IM CN Subsystem. Readers
   who are not familiar with it should carefully read this section.

   Section 6 contains the 3GPP requirements to SIP. Requirements are
   grouped by categories. Some requirements include a statement on
   possible solutions that would be able to fulfill the requirement.
   Note also that, as a particular requirement might be fulfilled by
   different solutions, not all the solutions might have an impact on

5. Overview of the 3GPP IM CN Subsystem

   This section gives the reader an overview of the 3GPP IM CN
   Subsystem. It is not intended to be comprehensive. But it provides
   enough information to understand the basis of the 3GPP IM CN
   Subsystem. Readers are encouraged to find a more detailed
   description in [4], [5] and [6].

   For a particular cellular device, the 3GPP IM CN Subsystem network
   is further decomposed in a home network and a visited network.

   An IM CN Subsystem subscriber belongs to his or her home network.
   Services are triggered and may be executed in the home network. One
   or more SIP servers are deployed in the SIP home network to support
   the IP Multimedia Subsystem. Among those SIP servers, there is a SIP
   serving proxy, which is also acting as a SIP registrar.
   Authentication/Authorization servers may be part of the home network
   as well. Users are authenticated in the home network.

   The visited network contains a SIP outbound proxy to support the UA.
   The SIP outbound proxy in the visited network may translate locally
   dialed digits into international format, detect emergency sessions,
   maintain security associations between itself and the terminals, and
   interwork with the resource management in the packet network.

   The SIP outbound proxy is assigned after the mobile has connected to
   the access network. Once this proxy is assigned, it does not change
   while the mobile remains connected to the access network. Thus the
   mobile can move freely within the access network without SIP
   outbound proxy reassignment.

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   The home network may support also a SIP entry proxy. This node may
   act as the first entry point for SIP signaling to the home network
   and may decide (with the help of location servers) which SIP
   registrar server to assign to a particular user. Typically the
   address of the home network SIP entry proxy is configured in DNS in
   the form of a DNS SRV record for SIP.

   Additionally, home and visited networks may deploy, if required, a
   SIP hiding proxy. The main purpose of the SIP hiding proxy is to
   hide the network configuration.

   The 3GPP IM CN Subsystem is designed to be access independent.
   Access is granted from 3GPP cellular terminals or from other
   terminals that use other accesses out of the scope of 3GPP.

   3GPP cellular IP Multimedia terminals use the existing General
   Packet Radio Service (GPRS) [6] as a transport network for IP
   datagrams. The terminals first connect to the GPRS network to get an
   IPv6 address. In order to do this, the terminals must perform a
   (GPRS) Attach procedure followed by a (GPRS) PDP Context Activation
   procedure. These GPRS procedures are required to be completed before
   any IP Multimedia session can be established.

   As a result of the above-mentioned GPRS procedures, the terminal has
   got an IPv6 address. In the case of non-roaming terminals, the IPv6
   address belongs to the home network address space. In the case of a
   roaming terminal, the IPv6 address belongs to the visited network
   address space. The  address does not change as the mobile terminal
   moves while still attached to the same network address space.

   If the terminal moves from a GPRS access to another GPRS access, the
   above-mentioned GPRS procedures needs to start from the beginning to
   allocate an IPv6 address to the terminal.

   Figure 1 shows an overview of the 3GPP architecture for IM CN

             +-------------+  +----------------+   +----------------+
             |             |  |                |   |                |
             |             |  |                |   |                |
             |             |  |                |   |      +------+  |
             |             |  |                |   |      | SIP  |  |
             |             |  |                |   |      |server|  |
       |     |             |  |                |   |      +------+  |
     +-|+    |             |  |                |   |       /        |
     |  |    |             |  |    +------+    |   | +------+       |
     |  |    |             |  |    | SIP  |    |   | | SIP  |       |
     |  | ---|-------------|--|----|server|----|---|-|server|       |
     +--+    |             |  |    +------+    |   | +------+       |
             |             |  |                |   |                |
     SIP     | GPRS access |  | Visited Network|   |  Home Network  |
     dev.    +-------------+  +----------------+   +----------------+

          Figure 1: Overview of the 3GPP IM CN Subsystem architecture

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   Another possible configuration is depicted in Figure 2. In that
   case, a general-purpose computer (e.g., a laptop computer) is
   connected to a GPRS terminal. The computer hosts the Multimedia
   application (comprising SIP, SDP, RTP, etc.). The GPRS terminal
   handles the radio access and the GPRS connectivity. Note that, for
   the sake of clarity, the home network has not been depicted in the

                                  +-------------+  +----------------+
                                  |             |  |                |
                                  |             |  |                |
                                  |             |  |                |
                                  |             |  |                |
                                  |             |  |                |
         +-------+                |             |  |                |
         |       |        +-|+    |             |  |                |
         |       |        |  |    |             |  |    +------+    |
         +-------+        |  |    |             |  |    | SIP  |    |
        /       / --------|  | ---|-------------|-------|server|------
       /-------/          +--+    |             |  |    +------+    |
                                  |             |  |                |
         SIP              GPRS    | GPRS access |  | Visited Network|
        client          terminal  +-------------+  +----------------+

              Figure 2: A computer connected to a GPRS terminal

   Services are typically executed in an application server. The
   interface between the SIP server and the application server is based
   on SIP. However, certain operators may want to reuse the existing
   technology, and therefore, they may need to interoperate SIP with
   protocols like CAMEL/Intelligent-Network or Open services
   Architecture (OSA).

6. 3GPP Requirements on SIP

6.1 General requirements

   This section does not specify any particular requirement to SIP.
   However, it includes a list of general requirements that must be
   considered when developing solutions to particular requirements.

   6.1.1 Efficient use of the radio interface

   The radio interface is a scarce resource. As such, the exchange of
   signaling messages between the UA and the network should be
   minimized. All the mechanisms developed should make an efficient use
   of the radio interface.
   See also the related requirements in section 6.5.

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   6.1.2 Minimum session setup time

   All the procedures and mechanisms should have a minimum impact on
   the session setup time as perceived by the user. When there is
   a choice between performing tasks at session establishment and in
   transactions prior to session establishment, then the tasks should
   be performed prior to session establishment.
   See also the related requirements in section 6.5.

   6.1.3 Minimum support required in the terminal

   As terminals could be rather small devices, memory requirements,
   power consumption, processing power, etc. should be kept to a
   minimum. Mandating support for additional protocols in the terminal
   must meet this requirement.

   6.1.4 Roaming and non roaming

   The developed solutions should work efficiently in roaming and non-
   roaming scenarios.

   6.1.5 Mobility management

   As mobility management is managed by the access network, there is no
   need to support mobility management in SIP.

   6.1.6 IP version 6

   The IP CN Subsystem is solely designed to use IP version 6

6.2 SIP outbound proxy in the visited network

   6.2.1 SIP outbound proxy in the visited network

   A SIP outbound proxy, typically in the visited network, must be
   supported in both roaming and non-roaming case, even when the SIP
   serving proxy in the home network is located in the same network as
   the SIP outbound proxy.

   6.2.2 Discovery of the SIP outbound proxy

   There must be a general mechanism to configure the UA with the
   address of the SIP outbound proxy in the visited network.

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   The Internet Draft "DHCP option for SIP servers" [7] may be a good
   starting point to meet this requirement. However, there is no
   support for IPv6 in this Internet Draft.

   3GPP has another mechanism provided by the GPRS access network that
   meets this requirement, in addition to the above one.

   6.2.3 Removal of headers

   The SIP outbound proxy must be able to remove the network generated
   contents of the Via and Record-Route headers of the SIP requests to
   be sent to the UA. These contents are reinserted in the appropriate
   headers of the responses, as if they would have been included by the
   UA. This increases security and reduces SIP message sizes and thus
   transmission delay and peak bandwidth requirements over the radio

6.3 Registration

   6.3.1 Registration required

   A user must register to the IMS before he/she can initiate or
   terminate any session. The rationale behind this is that:
   1. The user must be reachable for terminating sessions and services;
   2. The user is  authenticated and possibly billed for the resources
   that he/she is authorized to use.

   The procedure should not have a penalty on the session setup time
   (see also requirement 6.1.2).

   6.3.2 Location of the SIP Registrar

   The SIP registrar is located in the home network. The SIP registrar
   authenticates and registers the user.
   Once the terminal is switched on, the UA reads its configuration
   data. This data may be stored in a SIM card or any other memory
   device. The configuration data contains an  identification of the
   home network. The device finds the SIP registrar address from the
   home network domain name. The terminal sends the registration
   through the SIP outbound proxy.
   In order to support the search of the registrar, the home network
   contains one or more SIP servers that are configured in DNS with the
   SRV record of SIP. These are the home network entry proxies. Their
   mission is to serve as a first point of contact in the home network,
   and decide (with the help of location servers) which SIP registrar
   server to assign to a particular user.

   The procedures specified in SIP [3], section 1.4.2, applied to a
   REGISTER message seems to be sufficient to meet this requirement.

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   6.3.3 Efficient registration

   Due to the scarce radio interface resource, a single registration
   must be used to register both with the SIP outbound proxy in the
   visited network and the registrar in the home network.

   A single REGISTER message, addressed to the registrar, may traverse
   the SIP outbound proxy in the visited network. This can install, if
   needed, soft registration states in the SIP outbound proxy.

   6.3.4 Registration for roaming and non roaming cases

   In order to facilitate roaming between different networks, the UA
   must use the same registration procedure(s) within its home and
   visited networks.

   6.3.5 Visited domain name

   The home network must be able to validate that there is a roaming
   agreement between the home and the visited network. The home network
   needs to validate that the user is allowed to roam to such a visited
   network. Therefore, there must be a mechanism so that the visited
   network identity is known at registration time in the home network.
   As such, the visited network identity must be transported from the
   SIP outbound proxy to the home network.

   It is acceptable to represent the visited network identity as a
   visited network domain name.

6.4 De-registration

   6.4.1 De-registration of users

   There must be a procedure for a user to de-register from the
   network. This procedure may be used, e.g., when the user deactivates
   the terminal.

   We believe that a REGISTER with an expiration timer of 0 will meet
   the requirement.

   6.4.2 Types of network initiated de-registrations

   Two types of network initiated de-registrations must be provided:

   - To deal with registration expirations.
   - To allow the network to force de-registrations following any
   possible causes for this to occur.

   6.4.3 Network initiated de-registration, network maintenance

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   The IM CN Subsystem may initiate the network initiated de-
   registration procedure due to forced re-registrations from
   subscribers, e.g. in case of data inconsistency at node failure, in
   case of SIM lost, etc. Canceling the current contexts of the user
   spread among the network nodes at registration, and imposing a new
   SIP registration solves this condition.

   6.4.4 Network initiated de-registration, network/traffic determined

   The system must support a mechanism to avoid inconsistent
   information storage and remove any redundant registration
   information. This case will occur when a subscriber roams to a
   different network. This case occurs in normal mobility procedures
   when the user roams from one access network to another one, or when
   imposing new service conditions to roamers.

   6.4.5 Network initiated de-registration, application layer

   The service capability offered by the system to the application
   layers may have parameters specifying whether all SIP registrations
   are to be removed, or only those from one or a group of terminals
   from the user, etc.

   6.4.6 Network initiated de-registration, administrative

   For different reasons (e.g., subscription termination, lost
   terminal, etc.) a home network administrative function may determine
   a need to clear a user's SIP registration. This function initiates
   the de-registration procedure and may reside in various elements
   depending on the exact reason for initiating the de-registration.

   There must be a procedure for an entity in the network to de-
   register users. The de-registration information must be available at
   all the proxies that keep registration state and the UA.

   We believe that a procedure based on SIP events [15] and a
   registration package will meet the requirement.

6.5 Compression of SIP signaling

   As the radio interface is a scarce resource, the transport of SIP
   messages over the radio interface must be done efficiently.

   Therefore, there must be a mechanism to efficiently transport SIP
   signaling packets over the radio interface, by compressing the SIP
   signaling messages between the UA and the SIP outbound proxy, and by
   compressing the IP and transport layer protocol headers that carry
   these SIP messages.

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   6.5.1 Extensibility of the SIP compression

   The chosen solution(s) must be extensible to facilitate the
   incorporation of new and improved compression algorithms in a
   backward compatible way, as they become available.

   6.5.2 SIP compression and roaming

   The chosen solution(s) for SIP compression must work in roaming

   6.5.3 Minimal impact of SIP compression on the network

   Application specific compression shall minimize impacts on existing
   3GPP network, e.g. the compression must be defined between the UA at
   the SIP terminal and the outbound SIP Proxy in the visited network.

   6.5.4 Optionality of SIP compression

   It must be possible to leave the usage of compression for SIP
   signaling optional. To facilitate mobile terminal roaming between
   networks which are using compression, the mobile terminal should
   always support ability to compress SIP signaling. If compression is
   not supported, communication may continue without compression,
   depending on the local policy of the visited network.

   6.5.5 Default algorithm for SIP compression

   If SIP signaling compression is used, a default algorithm must be
   supported by the UA and the network elements involved for

   6.5.6 Compression Negotiation

   There must be a mechanism to negotiate between the UA and the first
   SIP outbound proxy the compression algorithm to be used. The type of
   negotiation mechanism that should be implemented is that the UAC
   includes a list of compression algorithms and the first SIP outbound
   proxy responds with the selected one. Subsequent SIP messages are
   compressed based on the agreed algorithm.

   Note: 3GPP is investigating if the compression of SIP signaling is
   negotiated on a per call basis, on a per registration basis or
   something completely different. More information will be provided in
   future versions of this document.

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6.6 QoS requirements related to SIP

   6.6.1 Independence between QoS signaling and SIP

   The selection of QoS signaling and resource allocation schemes must
   be independent of the selected session control protocols. This
   allows for independent evolution of QoS control and SIP.

   6.6.2 Coordination between SIP and QoS/Resource allocation Allocation before alerting

   In establishing a SIP session, it must be possible for an
   application to request that the resources needed for bearer
   establishment are successfully allocated before the destination user
   is alerted. Note, however, that it must be also possible for an SIP
   application in a terminal to alert the user before the radio
   resources are established (e.g. if the user wants to participate in
   the media negotiation).

   We believe this requirement is met by [8] and [21]. Destination user participates in the bearer negotiation

   In establishing a SIP session, it must be possible for a terminating
   application to allow the destination user to participate in
   determining which bearers shall be established.

   We believe this requirement is met by the standard SDP negotiation
   described in [3] and the extensions described in [8] and [21]. Successful bearer establishment

   Successful bearer establishment must include the completion of any
   required end-to-end QoS signaling, negotiation and resource

   We believe this requirement is met by the procedures described in
   [8] and [21].

6.7 Prevention of theft of service

   The possibility for theft of service in the 3GPP IM CN Subsystem
   shall be no higher than that for the corresponding GPRS and circuit
   switched services.

   We believe this requirement is met by the procedures described in

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6.8 Radio resource authorization

   As radio resources are very valuable the network must be able to
   manage these in a controlled manner. The network must be able to
   identify who is using these resources and be able to authorize their

   We believe this requirement is met by the procedures described in

6.9 Prevention of denial of service

   The system unavailability due to denial of service attacks in the IM
   CN subsystem shall be no greater than that for the corresponding
   GPRS and circuit switched services.

   We believe this requirement is met by the procedures described in

6.10 Identification of users

   6.10.1 Private user identity

   To use the 3GPP IM CN Subsystem, a subscriber must have a private
   user identity. The private identity is assigned by the home network
   operator, and used, for example, for registration, authorization,
   administration, and possibly accounting purposes. This identity
   shall take the form of a Network Access Identifier (NAI) as defined
   in RFC 2486 [10].

   The private user identity is not used for routing of SIP messages.

   The private user identity is a unique global identity defined by the
   Home Network Operator, which may be used within the home network to
   uniquely identify the user from a network perspective.

   The private user identity is not accessible by the user. Typically
   this identity is stored in a SIM card.

   The private user identity shall be permanently allocated to a user
   (it is not a dynamic identity), and is valid for the duration of the
   user’s subscription with the home network. Private user ID in registrations

   The UA must deliver the private user identity to the SIP outbound
   proxy and the registrar at registration time.
   The private user identity is used as the basis for authentication
   during registration of the subscriber. The term authentication is
   used in this document with the same meaning as it is defined in

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   The current working assumption is that this requirement is met by
   populating the From: header value of the REGISTER message with the
   private user ID.

   6.10.2 Public user identities

   To use the 3GPP IM CN Subsystem, a subscriber must have one or more
   public user identities. The public user identity/identities are used
   by any user for requesting communications to other users. For
   example, this might be included on a business card.

   Different public user identities may be grouped into a user profile.
   A user may have different profiles, each one containing different
   public user identities. A public user identity can be part of a
   single user profile.

   The current working assumption in 3GPP is that this requirement is
   met by populating the To: header value of a REGISTER message with
   the public user ID. In an outbound call, the From: and/or the
   Remote-Party-ID header values are populated with any of the public
   user identities. Format of the public user identities

   The public user identity/identities must take the form of a SIP URL
   (as defined in SIP [3] and RFC2396 [11]) or the form of a E.164
   number [12].

   We believe this requirement is met by using SIP URLs and telephone
   numbers represented in SIP URLs as described in SIP [3]. In
   addition, tel: URLs as specified in [13] can be used to fulfil the
   requirement. Registration of public user IDs

   It must be possible to register globally (i.e. through one single UA
   request) a subscriber that has more than one public identity that
   belongs to the same user profile, via a mechanism within the IM CN
   Subsystem. In this case, the user will be registered with all the
   public identities associated to a user profile. This must not
   preclude the user from registering individually some of his/her
   public identities if needed. Authentication of the public user ID

   Public user identities are not authenticated by the network. However
   the network authorizes that the public user identity is associated
   to the registered private user identity..

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6.11 Identifiers used for routing

   Routing of SIP signaling within the IM CN Subsystem must use SIP
   URLs as defined in [3]. E.164 [12] format public user identities
   must not be used for routing within the IM CN Subsystem, and session
   requests based upon E.164 format public user identities will require
   conversion into SIP URL format for internal IM CN Subsystem usage.

   We believe that this requirement is achieved by translating E.164
   numbers into SIP URLs. A database, such as ENUM [14] might do the

6.12 Hiding requirements

   We believe that the requirements in this section are met by the
   current SIP protocol [3].

   6.12.1 Hiding of the network structure

   A network operator need not be required to reveal the internal
   network structure to another network (in Via, Route, or other
   headers) that may contain indication of the number of SIP proxies,
   name of the SIP proxies, capabilities of the SIP proxies or capacity
   of the network. Association of the node names of the same type of
   entity and their capabilities and the number of nodes will be kept
   within an operator’s network. However disclosure of the internal
   architecture must not be prevented on a per agreement basis.

   6.12.2 Hiding of IP addresses

   A network need not be required to expose the explicit IP addresses
   of the nodes within the network (excluding firewalls and border

   6.12.3 SIP hiding proxy

   In order to support the hiding requirements, a SIP hiding proxy may
   be included in the SIP signaling path. Such additional proxy may be
   used to shield the internal structure of a network from other

6.13 Cell-ID

   The identity of the cell through which the 3GPP UA is accessing the
   IM CN Subsystem (Cell-ID) may be used by either the visited or the
   home network to provide localized services or information on the
   location of the terminal during an emergency call (see also
   requirement 6.16.3).

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   6.13.1 Cell-ID in signaling from the UA to the visited and home

   Assuming that the cell-ID is obtained by the UA by other mechanisms
   outside the scope or beyond SIP, the cell-ID must be transported at
   least in the following procedures:

   - Registration
   - Session Establishment (Mobile Originated)
   - Session Establishment (Mobile Terminated)
   - Session Release

   6.13.2 Format of the cell-ID

   The cell-ID must be sent in the format of a Cell Global ID, as
   described in [22].

6.14 Release of sessions

   In addition to the normal mechanisms to release a SIP session (e.g.
   BYE), two cases are considered in this section. The ungraceful
   release of the session (e.g., the terminal moves to an out of
   coverage zone) and the graceful session release ordered by the
   network (e.g., pre-paid caller runs out of credit).

   6.14.1 Ungraceful session release

   If an ungraceful session termination occurs (e.g. flat battery or
   mobile leaves coverage), when a call stateful SIP proxy server (such
   as the SIP serving proxy at home) is involved in a session, memory
   leaks and eventually server failure can occur due to hanging state
   machines. To ensure stable proxy operation and carrier grade
   service, a mechanism to handle the ungraceful session termination
   issue must be provided. This mechanism should be at the SIP protocol
   level in order to guarantee access independence for the system.

   6.14.2 Graceful session release

   There must be a mechanism so that an entity in the network may order
   the release of resources to other entities. This may be used, e.g.,
   in pre-paid calls when the user runs out of credit.

   This release must not involve any request to the UA to send out a
   release request (BYE), as the UA might not follow this request. The
   receiving entity needs the guarantee that resources are released
   when requested by the ordering entity.

6.15 Routing of SIP messages

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                       3GPP requirements on SIP           October 2001

   In order to clarify the terminology, we introduce the term vector to
   refer to the set of proxies that the INVITE has to traverse.

   6.15.1 SIP outbound proxy in the visited network

   As the SIP outbound proxy in the visited network is supporting the
   UA in terms of limited dialed digits translation (i.e., local to
   international), emergency calls, all sessions initiated in the
   mobile terminal when using IM CN Subsystem, must first route the SIP
   signaling to the SIP outbound proxy in the visited network,
   independently of the destination of the session.

   6.15.2 SIP serving proxy in the home network

   As services are triggered in the home network, all sessions
   initiated in the mobile terminal (except emergency calls) must route
   the SIP signaling to the SIP serving proxy in the home network
   allocated at registration time, independently of the destination of
   the session.

   6.15.3 INVITE might follow a different path than REGISTER

   The path taken by the INVITE need not be restricted to the specific
   path taken by the REGISTER. However, the path taken by the INVITE
   may follow the same path taken by the REGISTER (e.g., the INVITE may
   traverse just the SIP outbound proxy in the visited network and the
   SIP serving proxy in the home network, without passing through any
   other proxies).

   6.15.4 Information of the vector

   There must be some means of dynamically informing the node which
   adds the vector (e.g., the SIP outbound proxy) of what that vector
   should be, in the specific case where the vector is used to find a
   SIP serving proxy in the home network.

   Similarly, there must be some means of dynamically informing the
   node which adds the vector (e.g., the SIP serving proxy) of what
   that vector should be, in the specific case where the vector is used
   to find a SIP inbound proxy in the visited network.

   The hiding requirements expressed in section 6.12 also apply to the

   6.15.5 SIP inbound proxy in the visited network

   The visited network may apply certain local policies to incoming
   sessions. Therefore, there is a need to have an SIP inbound proxy in

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                       3GPP requirements on SIP           October 2001

   the visited network for terminating sessions. In general, the SIP
   inbound proxy and the SIP outbound proxy are the same entity in the
   visited network.

6.16 Emergency sessions

   It must be possible to place an emergency session using the IM CN
   Subsystem. Emergency calls will be routed to the emergency services
   in accordance with national regulations for where the subscriber is

   6.16.1 Registration is not required

   It must be possible to place an emergency session using SIP,
   independently on whether the user is registered to the IM CN
   Subsystem or not. Note, however, that in certain countries, it might
   be possible to reject an emergency call when the user is no
   registered to the IM CN Subsystem.

   6.16.2 SIP outbound proxy support

   Emergency sessions must be handled by the SIP outbound proxy in the
   visited network.

   6.16.3 Cell Global ID in emergency sessions

   It is required that location information including Cell Global ID
   (see also requirement6.13) be made available in the initial INVITE
   and the BYE message for the purpose of locating the user and routing
   to the appropriate Emergency Call Center.

   6.16.4 Types of emergency calls

   It must be possible to initiated emergency calls to different
   emergency call centers, depending on the type of emergency. The
   following types of emergency calls are possible:

   - Police
   - Ambulance
   - Fire brigade
   - Marine guard
   - Mountain rescue
   - Spare, at least three different types

   6.16.4 Default identifier for emergency calls

   In order to support emergency calls in roaming situations, it must
   be allowed to establish an emergency call without the need to dial a

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                       3GPP requirements on SIP           October 2001

   dedicated number or SIP URL. This allows to dial an emergency center
   based on a menu, "red button" or a linkage to a car air bag control.

   Additionally, it is desirable that the user interface for emergency
   calls in 3GPP terminals is similar to the one in other SIP networks.

   3GPP is currently investigating the applicability of the Universal
   Emergency SIP URL described in [36].

6.17 Identities on session establishment

   6.17.1 Remote Party Identification presentation

   It must be possible to present to the caller the identity of the
   party to which he/she may dial back to return a call.

   We believe this requirement is met by the procedures described in

   6.17.2 Remote Party Identification privacy

   In addition to the previous requirement, the called party must be
   able to request that his/her identity not be revealed to the caller.

   We believe this requirement is met by the procedures described in

   6.17.3 Remote Party Identification blocking

   Regulatory agencies, as well as subscribers, may require the ability
   of a caller to block the display of their caller identification.
   This function may be performed by the destination subscriber’s SIP
   serving proxy. In this way, the destination subscriber is still able
   to do a session-return, session-trace, transfer, or any other
   supplementary service.

   Therefore, it must be possible that the caller requests to block the
   display of his/her identity at the callee's display.

   We believe this requirement is met by the procedures described in

   6.17.4 Anonymity

   Procedures are required for an anonymous session establishment.
   However, sessions are not intended to be anonymous to the
   originating or terminating network operators.

   Note: 3GPP is still discussing whether the requirement is needed or

              Network Working Group   Expiration 04/30/02             18

                       3GPP requirements on SIP           October 2001 Anonymous session establishment

   If the caller requests the session to be anonymous, the UAC must not
   reveal any identity information to the UAS.

   If the caller requests the session to be anonymous, the terminating
   network must not reveal any identity or signaling routing
   information to the destination endpoint. The terminating network
   should distinguish at least two cases, first if the caller intended
   the session to be anonymous, and second if the caller’s identity was
   deleted by a transit network.

6.18 Charging

   It must be possible to apply charging, in a flexible manner based on
   any number of different charging models. Specific charging models
   and requirements for charging are under study.

6.19 IPv6

   As the 3GPP architecture is solely based on IP version 6, all
   protocols must support IPv6 addresses.

   We believe SIP [3] and SDP [17] meet this requirement. However, the
   "DHCP option for SIP servers" [7] does not support IPv6.

   6.19.1 Interworking IPv6 with IPv4

   3GPP IM CN subsystem is based on IPv6. As external networks may be
   based on IPv4 addresses, there is a need to interwork  with such a
   external networks. Therefore, interworking between IPv6 and IPv4 at
   the SIP and SDP level (UAs and proxies) must be guaranteed.

6.20 General support of additional capabilities

   3GPP is interested on applying and using additional services, like
   those described in [19], [37] and [38]. Although 3GPP is not going
   to standardize additional services, 3GPP may make sure that the
   capabilities that enable those services are granted in the network.

   As such we believe that the REFER method [18] and the Replaces
   header [20] constitute the enablers in order to meet  the above

6.21 Three-way handshake in the session description negotiation

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                       3GPP requirements on SIP           October 2001

   Typically a session description protocol like SDP is used in SIP to
   describe the media streams and codecs needed to establish the
   session. SIP uses an offer/answer model of the session description
   where one of the parties offers his session description and the
   other answers to that offer.

   In 3GPP IM CN Subsystem, the terminals might have restrictions with
   the memory, DSP capacity, etc. As such, it is required that the
   Session Description negotiation concludes with one out of many
   single codecs per media stream. Both UAC and UAS must know, prior to
   any media is sent or received, which codec is used for each media

   In 3GPP IM CN Subsystem, an efficient use of the network and radio
   resources is an important requirement. As such, the network must
   know in advance which codec is used for a particular media stream.
   The network may use this information to apply the most appropriate
   error correction mechanism depending on the selected codec. The
   network access control may use this information as well.

   Additionally, it is required that the party who pays for the
   resource utilization has the opportunity to decide the codec to use,
   once both end parties are aware of the capabilities supported at the
   remote UA.

   Therefore, it is required a three-way handshake model in the session
   description negotiation within SIP. This follows the model of
   offer/counter-offer/answer of the session description.

6.22 Security Model

   Sections 6.22, 6.23 and 6.24 have been based on the 3GPP documents
   [23], [4], and [24], and the work done by Dirk Kroeselberg in the
   Internet-Draft [31] (now expired).

   The scope for security of the 3GPP IM CN Subsystem is securing the
   SIP signaling between the various SIP entities. Protecting the end-
   to-end media streams may be a future extension but is not considered
   in the first version of the IM CN Subsystem.

   It is expected that security for the underlying GPRS network and the
   IM CN Subsystem will be provided independent of each other.
   Therefore, SIP signaling security must be provided independently of
   underlying access network security mechanisms. In particular, it
   must be possible to access the IM CN Subsystem services securely
   from other accesses than GPRS.

   Each operator providing IM CN Subsystem services acts as its own
   domain of trust, and shares a long-term security association with
   its subscribers. Operators may enter into roaming agreements with
   other operators, in which case a certain level of trust exists
   between their respective domains.

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                       3GPP requirements on SIP           October 2001

   SIP user agents must authenticate to their home network before the
   use of IM CN Subsystem resources is authorized. The current working
   assumption in the 3GPP is to perform authentication during
   registration and re-registrations.

   A hop-by-hop model must be used to protect actual SIP signaling.
   Looking at Figure 1 in Chapter 5, we can distinguish two main areas
   where security is needed:

   - Access Domain: Between the SIP user device and the visited
   - Network Domain: Between the visited and the home networks, or
     inside the home network.

   Characteristics needed in the Access Domain are quite different from
   those of the Network Domain because the terminal’s requirements on
   mobility, computation restriction, battery limit, bandwidth
   conservation and radio interface. SIP entities in the access domain
   should be able to maintain security contexts with a large group of
   users in parallel. Furthermore, Access Domain provides user specific
   security associations while Network Domain provides security
   associations between network nodes. Therefore the weight of
   protocols and algorithms and the compliance of them with compression
   mechanisms are very important to Access Domain Security. It is
   therefore required that the security solutions must allow different
   mechanisms in these two domains.

   Note that authentication, as used in this context, means entity
   authentication that enables two entities to verify the identity of
   the respective peer. This is different from message origin
   authentication, which allows a receiver to verify the origin of a
   single message and is provided by the same means as integrity

6.23 Access Domain Security

   6.23.1 Authentication

   Strong, mutual authentication method must be used.

   It must be possible to support different authentication methods.
   Therefore authentication using an extensible authentication
   framework must be provided.

   Authentication methods must support the secure storage of long-term
   authentication keys and the secure execution of authentication

   The SIP client’s credentials must not be transferred as plain text.

   HTTP Basic Authentication sends the passwords as plain text, also,
   it is neither strong nor does it offer mutual authentication. HTTP
   Digest has an option for mutual authentication. It uses

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                       3GPP requirements on SIP           October 2001

   cryptographic means for authentication, but does not protect against
   man-in-the-middle attacks where attackers modify the request while
   preserving the authentication headers. Lower layer mechanisms allow
   strong and mutual authentication (but do not fulfill other
   requirements). 3GPP intends to reuse UMTS AKA [24], but would prefer
   to a generic authentication framework at SIP level that supports
   UMTS AKA as well as other authentication mechanisms. UMTS AKA
   applies a symmetric cryptographic scheme, provides mutual
   authentication, and is typically implemented on a so-called SIM card
   that provides secure storage on the user’s side.

   Additional requirements related to delegation that apply to the
   authentication method are given in section

   6.23.2 Scalability and Efficiency

   3GPP IM CN Subsystems will be characterized by a large subscriber
   base of up to a billion users, all of which must be treated in a
   secure manner.

   The security solutions must allow global roaming among a large
   number of administrative domains. Bandwidth and Roundtrips

   The wireless interface in 3GPP terminals is an expensive resource
   both in terms of power consumption and maximum utilization of scarce
   spectrum. Furthermore, cellular networks have typically long round-
   trip time delays, which must be taken in account in the design of
   the security solutions.

   Any security mechanism that involves 3GPP terminals should not
   unnecessarily increase the bandwidth needs.

   All security mechanisms that involve 3GPP terminals should minimize
   the number of necessary extra roundtrips. In particular, during
   normal call signaling there should not be any additional security
   related messages.

   The roundtrip requirements are particularly hard to satisfy. It
   seems that IKE [32] adds a number of roundtrips, particularly if run
   together with legacy authentication extensions developed in the
   IPSRA WG. TLS [25] uses less roundtrips, but on the other hand
   doesn't support UDP. Computation

   It must be possible for IM CN Subsystem terminals to provide
   security without requiring public key cryptography and/or
   certificates. There may, however, be optional security schemes that
   employ these techniques.

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                       3GPP requirements on SIP           October 2001

   Current HTTP authentication methods use only symmetric cryptography
   as required here (but do not meet other requirements). Lower-layer
   security mechanisms all require the use of public key cryptography,
   or at least Diffie-Hellman as a mandatory part in their operation.
   HTTP EAP [27] is one candidate method to allow both symmetric
   cryptography and asymmetric cryptography based authentication within
   SIP, though there are probably other candidates as well, such as
   GSS_API [28]. However, definition of UMTS AKA under EAP is already
   in progress [29]. Delegation of Security Tasks

   Performing authentication on all SIP signaling messages would likely
   create bottlenecks in the authentication infrastructure. Therefore,
   a distributed implementation of security functions responsible for
   authentication is required.

   It must be possible to perform an initial authentication based on
   long-term authentication credentials, followed by subsequent
   protected signaling that uses short-term authentication credentials,
   such as session keys created during initial registration. The used
   authentication mechanisms must be able to provide such session keys.

   Initial authentication is performed between the SIP UA and the
   authenticating SIP serving proxy in the home network. However, the
   authentication mechanism must not require access to the long-term
   authentication credentials in these nodes. In the home network, the
   authenticating SIP serving proxy must support interaction with a
   dedicated authentication server in order to accomplish the
   authentication task. At the client side a secured (tamper-proof)
   device storing the long-term credentials of the user must perform
   the authentication.

   Additionally, the SIP serving proxy that performed the initial
   authentication must be able to securely delegate subsequent SIP
   signaling protection (e.g. session keys for integrity or encryption)
   to an authorized SIP proxy further downstream. The tamper-proof
   device at the client side must be able to securely delegate the
   session keys to the SIP user agent.

   Initial authentication can be performed with existing mechanisms
   such as HTTP Digest [3], but there exists no method to allow
   subsequent protection of the SIP signaling messages. There are also
   no proposals to allow secure delegation of signaling protection
   task. Currently the use of SIP together with an authentication
   server is not possible, though several proposals are under way to
   extend this [33, 34, 35]. However, the purpose of this document is
   not to discuss AAA requirements. They are discussed somewhere else.

   6.23.3 Negotiation of mechanisms

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                       3GPP requirements on SIP           October 2001

   A method for secure negotiation of security must be provided, to
   negotiate the security services to be used in the access domain.

   This method must at least support the negotiation of different
   security services providing integrity protection and encryption,
   algorithms used within these services and additional parameters they
   require to be exchanged.

   The negotiation mechanism must protect against attackers who do not
   have access to authentication credentials. In particular, it must
   not be possible for man-in-the-middle attackers to influence the
   negotiation result such that services with lower or no security are

   A negotiation mechanism is generally required in all secure
   protocols to decide which security services to use and when they
   should be started. This security mechanism serves algorithm and
   protocol development as well as interoperability. Often, the
   negotiation is handled within a security service. For example, the
   HTTP authentication scheme includes a selection mechanism for
   choosing among appropriate authentication methods and algorithms.
   Note that with the negotiation we mean just the negotiation, not all
   functions in protocols like IKE. For instance, we expect the session
   key generation is to be a part of the initial authentication.

   SIP entities may use the same security mode parameters to protect
   several SIP sessions without re-negotiation. For example, security
   mode parameters may be assumed to be valid within the lifetime of
   one registration.

   Existing lower-layer security mechanisms provide the above
   functionality as a part of them. We do not currently know of any
   mechanism that would allow this also at the SIP layer, [30] might
   perhaps be extended to perform secure negotiation. Note that such a
   mechanism is required not only for negotiation of security
   mechanisms, but for other services as well, e.g. for compression
   (see section 6.5.6). Although negotiation of security mechanisms is
   different due to the need for secure negotiation, all negotiation
   mechanisms could operate in a similar fashion.

   6.23.4 Message protection

   SIP entities (typically a SIP client and a SIP proxy) must be able
   to communicate using integrity and replay protection. By integrity,
   we mean the ability for receiver of a message to verify that the
   message has not been modified in transit. SIP entities should be
   able to communicate confidentially. These protection modes must be
   based on initial authentication. Integrity protection and
   confidentiality must be possible using symmetric cryptographic keys.

   It must be possible to handle also error conditions in a
   satisfactory manner as to allow recovery (see also 6.4.3 and 6.14).

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                       3GPP requirements on SIP           October 2001

   It must be possible to provide this protection between two adjacent
   SIP entities. In future network scenarios it may also be necessary
   to provide this protection through proxies, though at the moment
   3GPP does not require this. .

   The security mechanism should not incur external limitations to any
   transport bearers carrying SIP message.

   All the lower layer security mechanisms offer these services for the
   hop-by-hop case, but currently we do not know of any mechanism that
   would allow also end-to-end operation.

   The security mechanism must be able to protect a complete SIP

   If header compression/removal or SIP compression is applied to SIP
   messages, it must be compatible with message protection.

6.24 Network Domain Security

   Authentication, key agreement, integrity and replay protection, and
   confidentiality must be provided for communications between SIP
   network entities such as proxies and servers.

   Network domain security mechanisms must be scalable up to a large
   number of network elements.

   The 3GPP intends to make it mandatory to have protection discussed
   above at least between two operators, and optional within an
   operator’s own network. Security gateways exist between operator’s

   We believe the above requirements to be fulfilled by applying
   security mechanisms as specified in the current IP Security
   standards [26].

7. Security considerations

   This document does not define a protocol, but still presents some
   security requirements to protocols. The main security requirements
   are in sections 6.22 "Security Model", 6.23 "Access Domain Security"
   and 6.24 "Network Domain Security". Additional security-related
   issues are discussed under 6.7 "Prevention of theft of service", 6.8
   "Radio resource authorization", 6.9 "Prevention of denial of
   service", 6.12 "Hiding requirements" and 6.10 "Identification of

8. Author's Addresses

   Miguel A. Garcia
   FIN-02420, Jorvas, Finland

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                       3GPP requirements on SIP           October 2001

   Tel: +358 9299 3553

   Duncan Mills
   Vodafone UK Ltd.
   The Courtyard, Newbury, Berkshire, RG14 1JX, UK
   Tel: +44 1635 676074
   Fax: +44 1635 234445

   Gabor Bajko
   H-1096 Budapest, Koztelek 6, Hungary
   Tel: +36 20 9849259

   Georg Mayer
   Hofmannstr. 51, 81359 Munich, Germany
   Tel: +49-172-5371233

   Francois-Xavier Derome
   10 rue latecoere, F-78141
   tel: +33 130 773 834

   Hugh Shieh
   AT&T Wireless
   PO Box 97061, Redmond, WA 98073
   Tel: +1 425 580 6898

   Andrew Allen
   1501 W Shure Dr,
   Arlington Hts, IL 60004
   Phone: 847-435-0016

   Sunil Chotai
   Adastral Park, Ipswich, UK.
   Tel: +44 1473 605603

   Keith Drage
   Lucent Technologies
   Tel: +44 1793 776249

   Jayshree Bharatia
   Nortel Networks

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                       3GPP requirements on SIP           October 2001

   2201 Lakeside Blvd.
   Richardson, Texas 75082
   Tel: +1 972 684 5767

9. Acknowledgments

   The authors will like to thank the members of the 3GPP CN1 and SA3
   mailing lists for their collaborative effort.

10. References

   1.   Bradner, S., "The Internet Standards Process -- Revision 3", BCP
     9, RFC 2026, October 1996.

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

   3.   Handley M, Schulzrinne H, Schooler E, Rosenberg J., "SIP, Session
     Initiation Protocol", draft-ietf-sip-rfc2543bis-04.txt, Work in

   4.   3GPP TS 23.228 "IP Multimedia (IM) Subsystem (Stage 2) - Release
     5". Version 5.1.0 is available at

   5.   3GPP TS 24.228: "Signaling flows for the IP Multimedia call
     control based on SIP and SDP". Version 1.5.0 is available at

   6.   3GPP TS 23.060: "General Packet Radio Service (GRPS); Service
     Description; Stage 2". Version 4.1.0 is available at

   7.   H. Schulzrinne, G. Nair. "DHCP Option for SIP Servers", draft-
     ietf-sip-dhcp-04.txt, Work in progress.

   8.   W. Marshall et al. "Integration of Resource Management and SIP",
     draft-ietf-sip-manyfolks-resource-02.txt, Work in progress.

   9.   W. Marshall et al. "SIP Extensions for Media Authorization",
     draft-ietf-sip-call-auth-02.txt, Work in progress.

   10.  B. Aboba, M. Beadles, "The Network Access Identifier", RFC
     2486, January 1999.

   11.  T. Berners-Lee, R. Fielding, L. Masinter, "Uniform Resource
     Identifiers (URI): Generic Syntax", RFC 2396, August 1998.

   12.  ITU-T Recommendation E.164 (05/97): "The international public
     telecommunication numbering plan".

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                       3GPP requirements on SIP           October 2001

   13.  A. Vaha-Sipila, "URLs for Telephone calls", RFC 2806, April

   14.  P. Faltstrom, "E.164 number and DNS", RFC 2916, September 2000.

   15.  A. Roach, "SIP-Specific Event Notification", draft-ietf-sip-
     events-00.txt, Work in progress.

   16.  W. Marshall et al, "SIP Extensions for Caller Identity and
     Privacy", draft-ietf-sip-privacy-02.txt, Work in progress.

   17.  M. Handley, V. Jacobson, C. Perkins: "SDP: Session Description
     Protocol", draft-ietf-mmusic-sdp-new-03.txt, Work in progress.

   18.  R. Sparks: "The REFER method", draft-ietf-sip-refer-01.txt,
     Work in progress.

   19.  R. Sparks: "SIP Call Control - Transfer", draft-ietf-sip-cc-
     transfer-05.txt, Work in progress.

   20.  B. Biggs and R. Dean, "The SIP Replaces Header", draft-sip-
     replaces-00.txt, Work in progress.

   21.  J. Rosenberg, H. Schulzrinne: "Reliability of Provisional
     Responses in SIP", draft-ietf-sip-100rel-03.txt, Work in progress.

   22.  3GPP TS 23.003, "Numbering, addressing and identification
     (Release 5)". Version 5.0.0 is available is available at

   23.  3GPP TS 33.203 "Access Security for IP-Based Services",
     Version 0.5.0 is available at

   24.  3GPP TR 33.210 "Network Domain Security", Version 0.6.0.

   25.  T. Dierks, C. Allen. "The TLS Protocol Version 1.0", RFC 2246,
     January 1999.

   26.  S. Kent, R. Atkinson. "Security Architecture for the Internet
     Protocol", RFC 2401, November 1998.

   27.  V. Torvinen, J. Arkko, A. Niemi. "HTTP Authentication with
     EAP", draft-torvinen-http-eap-00.txt, Work In Progress, June 2001.

   28.  J. Linn. "Generic Security Service Application Program
     Interface Version 2, Update 1". RFC 2743, IETF. January 2000.

   29.  J. Arkko, H. Haverinen. "EAP AKA Authentication", draft-arkko-
     pppext-eap-aka-00.txt, Work In Progress, May 2001.

   30.  S. Parameswar, B. Stucker. "The SIP NEGOTIATE Method", draft-
     spbs-sip-negotiate-00.txt, Work In Progress, IETF, September 2001.

              Network Working Group   Expiration 04/30/02             28

                       3GPP requirements on SIP           October 2001

   31.  D. Kroeselberg. "SIP security requirements from 3G wireless
     networks", draft-kroeselberg-sip-3g-security-req-00.txt. Work In
     Progress, IETF, January 2001.

   32.  D. Harkins, D. Carrel: "The Internet Key Exchange (IKE)", RFC
     2409, November 1998.

   33.  Srinivas, Chan, Sengodan, Costa-Requena: "Mapping of Basic and
     Digest Authentication to DIAMETER AAA Messages", draft-srinivas-
     aaa-basic-digest-00.txt, Work in progress, July 2001.

   34.  B. Sterman: "Digest Authentication in SIP using RADIUS", draft-
     sterman-sip-radius-00.txt, Work in progress, February 2001.

   35.  P.R. Calhoun, W. Bulley, A.C. Rubens, J. Haag, G. Zorn:
     "Diameter NASREQ Application", draft-ietf-aaa-diameter-nasreq-
     07.txt, Work in progress, July 2001.

   36.  H. Schulzrinne: "Universal Emergency Address for SIP-based
     Internet Telephony", draft-schulzrinne-sipping-sos-00.txt, Work in
     progress, July 2001.

   37.  A. Johnston, S. Donovan, R. Sparks, C. Cunningham, D. Willis,
     J. Rosenberg, K. Summers, H. Schulzrinne: "SIP Call Flow
     Examples", draft-ietf-sip-call-flows-05.txt, Work in progress,
     June 2001.

   38.  A. Johnston, R. Sparks, C. Cunningham, S. Donovan, K. Summers:
     "SIP Service Examples", Work in progress, draft-ietf-sip-service-
     examples-02.txt, June 2001.

   39.  R. Shirey: "Internet Security Glossary", RFC 2828, May 2000.

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                       3GPP requirements on SIP           October 2001


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