Input 3rd-Generation Partnership Project (3GPP) Release 5 Requirements on the Session Initiation Protocol (SIP)
RFC 4083
| Document | Type | RFC - Informational (May 2005; No errata) | |
|---|---|---|---|
| Author | Miguel García | ||
| Last updated | 2015-10-14 | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html pdf htmlized bibtex | ||
| Stream | WG state | (None) | |
| Document shepherd | No shepherd assigned | ||
| IESG | IESG state | RFC 4083 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Unknown | ||
| Telechat date | |||
| Responsible AD | Allison Mankin | ||
| Send notices to | <rohan@cisco.com>, <dean.willis@softarmor.com> | ||
Network Working Group M. Garcia-Martin
Request for Comments: 4083 Nokia
Category: Informational May 2005
Input 3rd-Generation Partnership Project (3GPP)
Release 5 Requirements on the Session Initiation Protocol (SIP)
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
The 3rd-Generation Partnership Project (3GPP) has selected Session
Initiation Protocol (SIP) as the session establishment protocol for
the 3GPP IP Multimedia Core Network Subsystem (IMS). IMS is part of
Release 5 of the 3GPP specifications. Although SIP is a protocol
that fulfills most of the requirements for establishing a session in
an IP network, SIP has never been evaluated against the specific 3GPP
requirements for operation in a cellular network. In this document,
we express the requirements identified by 3GPP to support SIP for
Release 5 of the 3GPP IMS in cellular networks.
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Table of Contents
1. Introduction ....................................................4
2. Conventions .....................................................4
3. Overview of the 3GPP IMS ........................................5
4. 3GPP Requirements on SIP ........................................7
4.1. General Requirements .......................................7
4.1.1. Efficient Use of the Radio Interface ................7
4.1.2. Minimum Session Setup Time ..........................7
4.1.3. Minimum Support Required in the Terminal ............8
4.1.4. Roaming and Non-roaming .............................8
4.1.5. Terminal Mobility Management ........................8
4.1.6. IP Version 6 ........................................8
4.2. SIP Outbound Proxy .........................................8
4.2.1. SIP Outbound Proxy ..................................8
4.2.2. Discovery of the SIP Outbound Proxy .................8
4.3. Registration ...............................................9
4.3.1. Registration Required ...............................9
4.3.2. Efficient Registration .............................10
4.3.3. Registration for Roaming and Non-roaming Cases .....10
4.3.4. Visited Domain Name ................................10
4.3.5. De-registration ....................................10
4.4. SIP Compression ...........................................11
4.4.1. Compression Algorithm Independence .................12
4.4.2. Extensibility of the SIP Compression ...............12
4.4.3. Minimal Impact of SIP Compression on the Network ...12
4.4.4. Optionality of SIP Compression .....................12
4.5. QoS Requirements Related to SIP ...........................13
4.5.1. Independence between QoS Signaling and SIP .........13
4.5.2. Coordination between SIP and QoS/Resource
Allocation .........................................13
4.6. Prevention of Theft of Service ............................14
4.7. Radio Resource Authorization ..............................14
4.8. Prevention of Malicious Usage .............................14
4.9. Prevention of Denial of Service ...........................14
4.10. Identification of Users ..................................15
4.10.1. Private User Identity ............................15
4.10.2. Public User Identities ...........................15
4.10.3. Delivery of the Dialed Public User ID ............17
4.11. Identifiers Used for Routing .............................17
4.12. Hiding Requirements ......................................17
4.12.1. Hiding of the Network Structure ..................17
4.12.2. Hiding of IP Addresses ...........................17
4.12.3. SIP Hiding Proxy .................................18
4.13. Cell-ID ..................................................18
4.13.1. Cell-ID in Signaling from the UA to the
Visited and Home .................................18
4.13.2. Format of the Cell-ID ............................18
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4.14. Release of Sessions ......................................18
4.14.1. Ungraceful Session Release .......................19
4.14.2. Graceful Session Release .........................19
4.15. Routing of SIP Messages ..................................20
4.15.1. SIP Outbound Proxy ...............................20
4.15.2. SIP Serving Proxy in the Home Network ............20
4.15.3. INVITE Might Follow a Different Path than
REGISTER .........................................20
4.15.4. SIP Inbound Proxy ................................20
4.15.5. Distribution of the Source Routing Set of
Proxies ..........................................21
4.16. Emergency Sessions .......................................21
4.17. Identities Used for Session Establishment ................21
4.17.1. Remote Party Identification Presentatio ..........21
4.17.2. Remote Party Identification Privacy ..............21
4.17.3. Remote Party Identification Blocking .............21
4.17.4. Anonymity ........................................22
4.17.5. Anonymous Session Establishment ..................22
4.18. Charging .................................................22
4.18.1. Support of Both Prepaid and Postpaid Models ......22
4.18.2. Charging Correlation Levels ......................23
4.18.3. Charging Correlation Principles ..................23
4.18.4. Collection of Session Detailed Information .......24
4.19. General Support of Additional Capabilities ...............24
4.19.1. Additional Capabilities ..........................24
4.19.2. DTMF Signaling ...................................24
4.19.3. Early Media ......................................25
4.20. Exchange of Session Description ..........................25
4.21. Prohibition of Certain SDP Parameters ....................26
4.21.1. Prohibition of Codecs ............................26
4.21.2. Prohibition of Media Types .......................26
4.22. Network-initiated Re-authentication ......................26
4.23. Security Model ...........................................27
4.24. Access Domain Security ...................................28
4.24.1. General Requirements .............................28
4.24.2. Authentication ...................................29
4.24.3. Message Protection ...............................29
4.24.4. Negotiation of Mechanisms ........................31
4.24.5. Verification of Messages .........................31
4.25. Network Domain Security ..................................32
5. Security Considerations ........................................32
6. Contributors ...................................................32
7. References .....................................................32
7.1. Normative References ......................................32
7.2. Informative References ....................................33
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1. Introduction
3GPP has selected SIP [2] as the protocol to establish and tear down
multimedia sessions in the IP Multimedia Subsystem (IMS). 3GPP
Technical Specification 23.228 [28] describes the IMS. 3GPP
Technical Specification 24.228 [29] contains a comprehensive set of
session flows. 3GPP Technical Specification 24.229 [30] describes
the usage of SIP by the various IMS nodes.
This document is an effort to define the requirements applicable to
the usage of the SIP protocol suite in cellular networks,
particularly in the 3GPP IMS for Release 5 of the 3GPP
specifications. Further releases of the 3GPP specifications may
contain additional SIP requirements. This document focuses on the
requirements identified for the 3GPP Release 5 IMS.
The rest of this document is structured as follows:
o Section 3 offers an overview of the 3GPP IMS. Readers who are not
familiar with it should carefully read this section.
o Section 4 contains the 3GPP requirements to SIP. Requirements are
grouped by category. Some requirements include statements on
possible solutions that would be able to fulfill them. Note that,
as a particular requirement might be fulfilled by different
solutions, not all the solutions might have an impact on SIP.
This document is advisory in nature. Its primary purpose is to help
the IETF understand the IMS environment. Given this better
understanding, we expect that the IETF can more effectively evolve
the SIP protocol. The IETF will not respond to the requirements
given in this document on a point-for-point basis. Some requirements
have been and/or will be met by extensions to the SIP protocol.
Others may be addressed by effectively using existing capabilities in
SIP or other protocols, and we expect that individual members of the
SIP community will work with 3GPP to achieve a better understanding
of these mechanisms. Some of the requirements in this document may
not be addressed at all by the IETF, although we believe that the act
of documenting and discussing them is in itself helpful in achieving
a better all-around understanding of the task at hand.
2. Conventions
This document does not specify any protocol of any kind. Therefore,
the usage of the key words "MUST", "MUST NOT", "REQUIRED", "SHALL",
"SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and
"OPTIONAL" in this document, as described in RFC 2119 [1], does not
apply.
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3. Overview of the 3GPP IMS
This section gives the reader an overview of the 3GPP IM CN Subsystem
(IMS). It is not intended to be comprehensive, but it provides
enough information to understand the basis of the 3GPP IMS. Readers
are encouraged to find a more detailed description in the 3GPP
Technical Specifications 23.060 [27], 23.228 [28], and 24.228 [29].
For a particular cellular device, the 3GPP IMS network is further
decomposed in a home network and a visited network.
An IMS 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.
A SIP outbound proxy is provided to support the User Agent (UA). The
SIP outbound proxy is typically located in the visited network,
although it may be located in the home network as well. The SIP
outbound proxy maintains security associations between itself and the
terminals, and interworks with the resource management in the packet
network.
The SIP outbound proxy is assigned after the mobile device 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.
The home network may also support one or more SIP edge proxies.
These nodes may act as the first entry points for SIP signaling to
the home network and may determine (with the help of location
servers) which SIP registrar server to assign to a particular user.
Typically the address of the home network SIP edge proxy is
configured in DNS in the form of a DNS Naming Authority Pointer
(NAPTR) and Service (SRV) records 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.
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3GPP cellular IP Multimedia terminals use the existing General Packet
Radio Service (GPRS) [27] as a transport network for IP datagrams.
The terminals first connect to the GPRS network to get an IPv6
prefix. 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
built an IPv6 address. The IPv6 address belongs to the same network
address space as does the SIP outbound proxy. 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
Subsystem.
+-------------+ +----------------+ +----------------+
| | | | | +------+ |
| | | | | | SIP | |
| | | | | |server| |
| | | | | | +------+ |
+-|+ | | | | | / |
| | | | | +------+ | | +------+ |
| | | | | | SIP | | | | SIP | |
| | ---|-------------|--|----|server|----|---|-|server| |
+--+ | | | +------+ | | +------+ |
| | | | | |
SIP | GPRS access | | Visited Network| | Home Network |
dev. +-------------+ +----------------+ +----------------+
Figure 1: Overview of the 3GPP IMS architecture
Another possible future 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, in this example the home network has not been
depicted in the figure.
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+-------------+ +----------------+
+-------+ | | | | |
| | +-|+ | | | |
| | | | | | | +------+ |
+-------+ | | | | | | 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 such as CAMEL/Intelligent-Network or Open Services
Architecture (OSA).
4. 3GPP Requirements on SIP
4.1. General Requirements
This section does not specify any particular requirement for SIP.
However, it includes a list of general requirements that must be
considered when developing solutions to particular requirements.
4.1.1. Efficient Use of the Radio Interface
The radio interface is a scarce resource. As such, the exchange of
signaling messages between the mobile terminal 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 4.4.
4.1.2. 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 prior to
session establishment, then tasks should be performed prior to
session establishment.
See also the related requirements in Section 4.4.
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4.1.3. 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.
4.1.4. Roaming and Non-roaming
All the requirements must be met for both roaming and non-roaming
scenarios. There should not be a significant change in the signaling
procedures between roaming and non-roaming scenarios.
4.1.5. Terminal Mobility Management
As terminal mobility is managed by the access network, there is no
need to support terminal mobility management in SIP.
4.1.6. IP Version 6
3GPP IMS is solely designed to use IP version 6. As a consequence,
all protocols must support IPv6 addresses.
4.2. SIP Outbound Proxy
4.2.1. SIP Outbound Proxy
A SIP outbound proxy is provided to support both roaming and
non-roaming scenarios. The SIP outbound proxy may be located either
in the home network or in the visited network.
4.2.2. Discovery of the SIP Outbound Proxy
There must be a general mechanism whereby the mobile device (UA)
learns the SIP outbound proxy address.
The DHCPv6 option for SIP servers in RFC 3319 [19] seems to fulfill
the requirement.
In addition to the above-expressed requirement, the 3GPP access
network may provide the SIP outbound proxy address during access
network bearer establishment. This is considered a less general
mechanism, though.
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4.3. Registration
The home network must maintain one or more SIP registrars. The SIP
registrar authenticates the user and registers the IP address where
the user can be located.
Once the terminal is switched on, the mobile device UA reads its
configuration data. This data may be stored in a SIM card or in 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 may be configured in DNS with
the NAPTR/SRV record of SIP. These are the home network edge
proxies. Their mission is to serve as the first points of contact in
the home network, and to decide (with the help of location servers)
which SIP registrar server to assign to a particular user.
The procedures specified in RFC 3263 [10] applied to a REGISTER
message seem to be sufficient to meet this requirement.
4.3.1. Registration Required
A user must register to the IMS before he/she can receive any
invitation to any sessions. In addition, it is desirable for the
user to register before initiating any sessions. The following
factors contribute to the rationale behind this:
1. The SIP serving proxy in the home network needs to know when and
from which terminal the user is available, in order to route
received SIP requests for sessions and services.
2. The user can be pre-authenticated early so that authentication
does not contribute to post-dial delay. The procedure should not
have a penalty on the session setup time (see also the
requirement stated in Section 4.1.2).
3. The user is assigned a particular serving proxy. The serving
proxy downloads the service profile for that user to trigger
services.
Therefore, 3GPP has mandated the mobile device UA to register before
the mobile device UA initiates any session.
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4.3.2. Efficient Registration
Due to the scarce radio interface resource, a single registration
must be sufficient to ensure that the mobile UA is reachable from
both the home and the visited networks.
A single REGISTER message, addressed to the registrar, may traverse
the SIP outbound proxy. This can install, if needed, soft
registration states in the SIP outbound proxy.
4.3.3. Registration for Roaming and Non-roaming Cases
Independent of whether the UA is roaming, it is desirable for the
registration procedure to be the same.
4.3.4. Visited Domain Name
The home network must be able to validate the existence of 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 whereby the visited
network identity is known at registration time at the home network.
It is acceptable to represent the visited network identity either as
a visited network domain name or as a string.
4.3.5. De-registration
4.3.5.1. De-registration of Users
There must be a procedure for a user to de-register from the network.
This procedure may be used, for example, when the user deactivates
the terminal.
We believe that a REGISTER with an expiration timer of 0 will meet
the requirement.
4.3.5.2. Network-initiated De-registration or Re-registration
In a number of situations a network needs to de-register or trigger a
re-registration of a previously registered UA. Examples of usage are
described in sections 4.3.6.3, 4.3.6.4, and 4.3.6.5.
This implies a need for a notification mechanism whereby the UA can
be notified of the de-registration, or of a request for
re-registration.
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We believe that this requirement is met by the SIP-specific event
notification [12] and a registration event package [14].
4.3.5.3. Network-initiated De-registration, Network Maintenance
There might be cases in which the SIP serving proxy has to shutdown;
e.g., due to maintenance operation. Although this situation is not
likely to happen in everyday situations, it is desirable to have a
mechanism to inform the UA that his current registration is being
cancelled. The UA may initiate another registration process that
will lead to the selection of a new SIP serving proxy.
4.3.5.4. Network-initiated De-registration, Network/Traffic Determined
The system must support a mechanism to avoid inconsistent information
storage and to remove any redundant registration information. This
case will occur when a subscriber roams to a different network
without a prior de-registration. This case occurs in normal mobility
procedures when the user roams from one access network to another, or
when new service conditions are imposed on roamers.
4.3.5.5. Network-initiated De-registration, Administrative
For different reasons (e.g., subscription termination, stolen
terminal, etc.) a home network administrative function may determine
a need to clear a user's SIP registration. It is desirable to have a
mechanism whereby the SIP serving proxy can inform the UA that its
registration is being cancelled.
There must be a procedure for the SIP serving proxy 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-specific event notification
[12] and a registration event package [14] will meet this
requirement.
4.4. SIP Compression
The radio interface is a scarce resource, and typically the available
bandwidth over the radio interface is limited. These two factors
seem to limit the transport of possibly large SIP messages over the
air interface. Particularly, the session setup time might be
extended due to the time needed to transport SIP messages over a
limited bandwidth channel.
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On the other hand, the number and size of certain SIP header values,
such as Via or Record-Route, seems not to be limited. A mobile
device UA may present limitations in the available memory to store
this kind of information.
Therefore, there must be a mechanism to efficiently transport SIP
signaling packets over the radio interface, by compressing the SIP
messages between the mobile device UA and the SIP outbound proxy, and
between the SIP outbound proxy and the mobile device UA. Note that
compression of IP and transport layer protocol headers that carry
these SIP messages is also a requirement, although we believe that
this does not have an impact on SIP.
4.4.1. Compression Algorithm Independence
The chosen solution(s) must be able to allow the operation under
several different compression algorithms.
4.4.2. 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.
4.4.3. Minimal Impact of SIP Compression on the Network
Application-specific compression must minimize impacts on existing
3GPP access networks (such as base stations transceivers). On the
other hand, the compression mechanism should be independent of the
access; e.g., the compression must be defined between the mobile
device UA and the outbound SIP proxy.
4.4.4. 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 that are using compression, the mobile terminal should
always support SIP signaling compression. If compression is not
supported, communication may continue without compression, depending
on the local policy of the visited network.
4.4.4.1. Compression Reliability
The compression mechanism should be reliable and able to recover
automatically from errors generated during the decompression.
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4.5. QoS Requirements Related to SIP
4.5.1. 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.
4.5.2. Coordination between SIP and QoS/Resource Allocation
4.5.2.1. 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 that this requirement is met by Integration of Resource
Management and SIP [15].
4.5.2.2. 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 will be established. However, it must be
possible to establish the SIP session without user intervention.
We believe that this requirement is met by the standard SDP
negotiation described in SIP [2], the SDP offer/answer model [11] and
the extensions described in Integration of Resource Management and
SIP
4.5.2.3. Successful Bearer Establishment
Successful bearer establishment must include the completion of any
required end-to-end QoS signaling, negotiation, and resource
allocation.
We believe that this requirement is met by the procedures described
in the Integration of Resource Management and SIP [15].
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4.6. Prevention of Theft of Service
Typically, users are allocated QoS resources. There is an admission
control mechanism that prevents users exceeding the limits negotiated
with the network. The network must prevent unauthorized users to
make use of non-authorized resources. For instance, the network must
provide a mechanism to prevent a user from using the resources
allocated to a second user, and for which this second user may be
paying.
We believe that this requirement may be met by the procedures
described in the Private SIP extensions for Media Authorization [16].
4.7. Radio Resource Authorization
As radio resources are very valuable, the network must be able to
manage them in a controlled manner. The network must be able to
identify who is using these resources and to authorize their usage.
For example, a mobile device terminal could execute an unlimited and
uncontrolled resource reservation procedure if the network does not
supervise the usage of radio resources.
We believe that this requirement is met by the procedures described
in the Private SIP extensions for Media Authorization [16].
4.8. Prevention of Malicious Usage
The 3GPP IMS must prevent mobile devices from making malicious use of
the network. For instance, a malicious UA could not obey the
procedures related to the Record-Route header field: when sending
subsequent requests the UA could bypass proxies which inserted a
Record-Route header during the initial transaction.
4.9. Prevention of Denial of Service
The risk that a proxy will receive a denial of service attack should
be minimized. For instance, a malicious mobile device could learn a
SIP proxy IP address and port number (e.g., in a Record-Route header
value) and establish an attack upon that proxy.
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4.10. Identification of Users
4.10.1. Private User Identity
In order to use the 3GPP IMS, a user is assigned a private user
identity. The home network operator assigns the private user
identity, which is used to identify the user uniquely from a network
perspective. The private user identity is used, for example, for
authentication, authorization, administration, and, possibly,
accounting purposes. Note that 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. The identity takes the form of a Network
Access Identifier (NAI) as defined in RFC 2486 [6].
The end user does not have access to the private user identity.
Typically the identity is stored in a Subscriber Identity Module
card.
The private user identity is permanently allocated to a user (it is
not a dynamic identity), and is valid for the duration of the user's
business subscription with the home network.
4.10.1.1. Private User ID in Registrations
The mobile 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 mobile user. The term authentication is
used in this document with the same meaning as it is defined in RFC
2828 [7].
We believe that this requirement is met by populating the username
field of the Authorization: header value of the REGISTER request with
the private user identity.
4.10.2. Public User Identities
In order to use the 3GPP IMS, a user is assigned one or more public
user identities. The user will make use of the public user identity/
identities when requesting communication to other users. For
example, the public user identity might be included on a business
card.
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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 user may need to register one or more public user identities
prior to receiving communications addressed to that public user
identity.
We believe that this requirement is met by populating the From: and
To: header values of a REGISTER message with the public user
identity.
4.10.2.1. Format of the Public User Identities
The public user identity must take the form of a SIP URI (as defined
in RFC 3261 [2] and RFC 2396 [4]) or of a E.164 [34] number.
We believe that 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 RFC 3966 [35] can be used to
fulfill the requirement.
4.10.2.2. Registration of Public User IDs
It must be possible to register globally (i.e., through one single UA
request) a user that has more than one public identity that belongs
to the same user profile, via a mechanism within the IMS. In this
case, the user will be registered with all the public identities
associated to a user profile.
We believe this requirement may be accomplished by external
procedures. For example, the user's profile may contain a list of
alias identities that the registrar considers active if the primary
identity is registered. The user may get informed of the
automatically registered public user IDs by subscribing to its own
registration state.
4.10.2.3. Authentication of the public user ID
Public user identities are not authenticated by the 3GPP IMS.
However, the network authorizes that the public user identity is
associated with the registered private user identity.
There is a list of public user identities associated with each
private user ID within the IMS. IMS will reject attempts to use
other public identities with this private user ID.
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4.10.3. Delivery of the Dialed Public User ID
Typically a UA will be registered under a set of different public
user IDs. As such, sessions destined to the user can be placed with
any of the registered public user IDs. The serving proxy and
application server(s) in the termination network may apply certain
filtering rules or services based on the public user ID contained in
the Request-URI. The UA may also apply certain filtering rules or
services based on the called public user ID.
Therefore, it must be possible for all sessions to deliver the dialed
public user ID to the terminating entities, such as the serving
proxy, application servers, and terminating UA.
4.11. Identifiers Used for Routing
Routing of SIP signaling within IMS must use SIP URLs as defined in
SIP [2]. E.164 [34] format public user identities must not be used
for routing within IMS, and session requests based on E.164 format
public user identities will require conversion into SIP URI format
for internal IMS usage.
We believe that this requirement is achieved by translating E.164
numbers into SIP URIs. A database, such as ENUM [9], might do the
job.
4.12. Hiding Requirements
Although the requirements included in this section are not optional,
the hiding feature is optional to use through configuration. This
means that a network operator can, at his desire, switch the hiding
functionality on or off.
4.12.1. 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,
domain name of the SIP proxies, capabilities of the SIP proxies, or
capacity of the network.
4.12.2. 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
gateways).
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4.12.3. SIP Hiding Proxy
In order to support the hiding requirements, a SIP hiding proxy may
be included in the SIP signaling path. This additional proxy may be
used to shield the internal structure of a network from other
networks.
4.13. Cell-ID
The identity of the cell through which the 3GPP UA is accessing the
IMS (Cell-ID) may be used by the home network to provide localized
services or information on the location of the terminal during an
emergency call (when emergency calls are handled in IMS; see also the
requirement stated in Section 4.16).
4.13.1. Cell-ID in Signaling from the UA to the Visited and Home
Networks
Assuming that the Cell-ID is obtained by the UA by other mechanisms
outside the scope of SIP, the Cell-ID must be transported at least in
the following procedures:
o Registration
o Session Establishment (Mobile Originated)
o Session Establishment (Mobile Terminated)
o Session Release
The Cell-ID is private information and only of interest in the UA
home network. Therefore, the Cell-ID should be removed prior to
sending the SIP signaling beyond the originating home network.
4.13.2. Format of the Cell-ID
The cell-ID must be sent in any of the formats described in the 3GPP
Technical Specification 23.003 [26].
4.14. Release of Sessions
In addition to the normal mechanisms for releasing a SIP session
(e.g., BYE), two cases are considered in this section: the ungraceful
session release (e.g., the terminal moves to an out-of-coverage zone)
and the graceful session release ordered by the network (e.g.,
prepaid caller runs out of credit).
We believe that this requirement is met by a SIP entity acting as a
so-called transparent back-to-back UA.
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4.14.1. Ungraceful Session Release
If an ungraceful session termination occurs (e.g., a flat battery or
a 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 server operation and carrier grade
service, a mechanism to handle the ungraceful session termination
issue must be provided. We assume that there is a mechanism by which
the SIP outbound proxy is notified, by a mechanism external to SIP,
of the ungraceful session termination. This allows transforming the
ungraceful session release into a graceful session release ordered by
the network (see the next section). For example, upon reception of
the notification of loss of mobile radio coverage, the SIP outbound
proxy could send a BYE request on behalf of the terminal, although
this BYE cannot be authenticated.
4.14.2. Graceful Session Release
There must be a mechanism whereby an entity in the network may order
the release of resources to other entities. This may be used, for
example, in prepaid 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.
The following objectives must be met:
o Accurately report the termination to the charging subsystem.
o Release the associated network resources: bearer resources and
signaling resources.
o Notify other parties to the session, if any, of the session
termination.
When feasible, this mechanism should be at the SIP protocol level in
order to guarantee access independence for the system.
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4.15. Routing of SIP Messages
4.15.1. SIP Outbound Proxy
The 3GPP architecture includes a SIP outbound proxy that is typically
located in the visited network (although it may be located in the
home network). This outbound proxy provides local services such as
compression of SIP messages or security functions. In addition, the
outbound proxy may interact with the media reservation mechanism to
provide authentication and authorization support for media
reservation.
All mobile terminal originated session setup attempts must transit
the outbound proxy so that the services provided by the outbound
proxy can be delivered to the mobile terminal.
4.15.2. SIP Serving Proxy in the Home Network
The serving proxy in the home network allows triggering of user-
customized services that are typically executed in an application
server.
All mobile terminal originated session setup attempts must transit
the serving proxy in the home network so that the proxy can properly
trigger the SIP services allocated to the user (e.g., speed dial
substitution). This implies a requirement for some sort of source-
routing mechanism to ensure these proxies are transited correctly.
4.15.3. INVITE Might Follow a Different Path than REGISTER
The path taken by an INVITE request need not be restricted to the
specific path taken by a mobile terminal originated REGISTER request;
e.g., the INVITE may traverse just the SIP outbound proxy and the SIP
serving proxy, without passing through any other proxies. However,
the path taken by the INVITE may follow the same path taken by the
REGISTER.
4.15.4. SIP Inbound Proxy
The visited network may apply certain services and policies to
incoming sessions (such as establishment of security services or
interaction with the media reservation mechanism). Therefore, the
visited network may contain a SIP inbound proxy for terminating
sessions. In general, the SIP inbound proxy and the SIP outbound
proxy are the same SIP proxy.
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4.15.5. Distribution of the Source Routing Set of Proxies
Sections 4.15.2 and 4.15.4 assume that a source-routing mechanism is
used to effect traversal of the required SIP proxies during session
setup.
There must be some means of dynamically informing the node that adds
the source-routing set of proxies that the INVITE has to traverse
(e.g., the outbound proxy or serving proxy) of what that set of
proxies should be.
The hiding requirements expressed in Section 4.12 also apply to the
said set of proxies.
4.16. Emergency Sessions
3GPP networks already contain alternative procedures for delivering
emergency sessions. Release 5 of the 3GPP specifications does not
add any requirement for SIP emergency sessions.
4.17. Identities Used for Session Establishment
4.17.1. 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 that this requirement is met by the procedures described
in RFC 3325 [17].
4.17.2. 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 that this requirement is met by the procedures described
in RFC 3323 [18].
4.17.3. Remote Party Identification Blocking
Regulatory agencies, as well as subscribers, may require the ability
of callers to block the display of their caller identification. The
destination subscriber's SIP serving proxy may be perform this
function. In this way, the destination subscriber is still able to
do a session-return, session-trace, transfer, or any other
supplementary service.
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Therefore, it must be possible that the caller request to block the
display of his/her identity on the callee's display.
We believe that this requirement is met by the procedures described
in RFC 3323 [18].
4.17.4. Anonymity
Procedures are required for anonymous session establishment.
However, sessions are not intended to be anonymous to the originating
or terminating network operators.
We believe that this requirement is met by the procedures described
in RFC 3323 [18] and RFC 3325 [17].
4.17.5. Anonymous Session Establishment
If the caller requests that the session be anonymous, the User Agent
Client (UAC) must not reveal any identity information to the User
Agent Server (UAS).
If the caller requests that the session 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, whether the caller intended
the session to be anonymous, and second, whether the caller's
identity was deleted by a transit network.
We believe that this requirement is met by the procedures described
in RFC 3323 [18] and RFC 3325 [17].
4.18. Charging
The 3GPP charging implications are described in the 3GPP Technical
Specification 32.225 [31].
4.18.1. Support of Both Prepaid and Postpaid Models
Operators may choose to offer prepaid and/or postpaid services. The
prepaid model is accomplished with the support of the online charging
model. The postpaid model is accomplished with the support of the
offline charging model.
Online charging is the process whereby charging information can
affect, in real-time, the service rendered to the user, such as a
request for a graceful release of an existing session. Online
charging interacts with the SIP signaling.
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Offline charging is the process whereby charging information does not
affect, in real-time, the service rendered to the user.
4.18.2. Charging Correlation Levels
The following levels of correlation for IMS charging are considered:
o Correlation within a session. A session may comprise a number of
media components. It must be possible to correlate the charging
data of the different media components belonging to a session.
o Correlation at media-component level. For a session comprising
several media types (such as audio and video), charging data is
generated for each media type and needs to be correlated between
network elements. For this, a media identifier will be unique and
will clearly identify which media type of a session this charging
information belongs to. This component identifier is not
exchanged between network elements and is based on the ordering of
media flows in the SDP. This ordering is the same as that used in
the binding information passed to the GPRS network.
4.18.3. Charging Correlation Principles
To support the correlation of charging information, the following
principles apply to both offline and online charging:
o The correlation of charging information for an IMS session is
based on the use of IMS Charging Identifiers (ICID).
o The first IMS network entity within the SIP signaling path is
responsible for assigning an ICID. This ICID will then be passed
along the whole session path in an end-to-end manner. However,
this will not preclude further elements (other SIP proxies) along
the session path from generating additional identifiers to be
passed along.
o The ICID is passed to all IMS network entities in the session
signaling path. This is performed using SIP signaling.
o The addresses of the charging functions are passed by the serving
SIP proxy to all IMS network entities in the session signaling
path. This is to provide a common destination for all the
charging records generated by each IMS network entity with the
same ICID.
o For the charging correlation between the GPRS network and the IMS,
one or more GPRS Charging IDs, which identify the PDP contexts of
the session, are passed from the GPRS network to the IMS.
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o The GPRS Charging IDs are passed by the outbound SIP proxy to the
serving SIP proxy and the Application Servers using SIP signaling.
They are not transferred from one home IMS (e.g., caller's home)
to another (e.g., callee's home).
o Inter Operator Identifiers (IOI) are shared between the caller's
home IMS and the callee's home IMS to provide identifiers of the
home originating and home terminating networks.
4.18.4. Collection of Session Detailed Information
The SIP serving proxy or another SIP server in the home network must
be able to log details of all sessions, such as the duration, source,
and destination of a session, to provide to the charging subsystem.
4.19. General Support of Additional Capabilities
4.19.1. Additional Capabilities
3GPP is interested in applying and using additional services, such as
those described in SIP Call Control - Transfer [20], SIP Basic Call
Flow Examples [21], SIP Public Switched Telephone Network (PSTN) Call
Flows [22], and SIP service examples [23]. 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.
Therefore, we believe that the SIP REFER method [24] and the Replaces
header [25] constitute a complement to be used as an enabler in order
to meet the above requirement.
4.19.2. DTMF Signaling
Support for voice calls must provide a level of service similar to
that of the existing circuit-based voice service. This includes the
ability to use DTMF signaling, for example, for control of
interactive voice response systems such as ticket sales lines and
timetable information.
The transport of DTMF tones from the mobile terminal to target
systems that may be in the PSTN, or to SIP-based solutions (i.e., no
PSTN connection), must be supported.
The transport of DTMF signals may be required for the whole call,
just for the first part, or from some later point in the call. The
start time and duration of such signaling is therefore unpredictable.
We believe that the mechanisms specified in RFC 2833 [8] meet the
requirement without impacting SIP.
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4.19.3. Early Media
As mobile terminals will frequently interoperate with the PSTN,
support for early media is required.
4.20. Exchange of Session Description
Typically a session description protocol such as 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,
as described in RFC 3264 [11], in which one of the parties offers his
session description and the other answers that offer.
In the 3GPP IMS, the mobile terminals might have restrictions with
the memory, DSP capacity, etc. As such, a mechanism is required by
which the Session Description negotiation may conclude with one out
of many codecs per media stream. Both UAC and UAS must know, prior
to any media being sent or received, which codec is used for each
media stream.
In the 3GPP IMS, efficient use of the network and radio resources is
an important requirement. As such, the network should know in
advance which codec is used for a particular media stream. The
network access control may use this information to grant access to
the network and to control the resource utilization.
Additionally, it is required that the party who pays for the resource
utilization have the opportunity to decide which codecs to use, once
both end parties are aware of the capabilities supported at the
remote UA.
Therefore, a mechanism is required by which both UAC and UAS have the
ability to negotiate and trim down the number of codecs used per
media stream, so that at the end of the negotiation there can be a
reduced set of agreed codecs per media stream.
We believe that the mechanism specified in RFC 3264 [11] meets the
requirement.
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4.21. Prohibition of Certain SDP Parameters
4.21.1. Prohibition of Codecs
The SIP outbound proxy may contain local policy rules with respect
the codecs allowed in the network. For instance, certain networks
may disallow high-bandwidth-consuming audio codecs. There has to be
a mechanism whereby the SIP outbound proxy can reject a session
establishment attempt when a codec is prohibited in the network due
to local policy.
4.21.2. Prohibition of Media Types
Certain users' subscriptions may include restrictions on certain
media types. For instance, a user may not be allowed to establish a
video session. The SIP serving proxy in the home network downloads
the user profile, which contains the rules for these kinds of
restrictions.
As the establishment of sessions traverse the SIP serving proxy in
the home network, the proxy can prohibit an attempt to establish a
session that includes a non-allowed media type for the user.
Therefore, there has to be a mechanism whereby the SIP serving proxy
can reject a session establishment attempt when the session includes
a forbidden media type.
4.22. Network-initiated Re-authentication
Network operators need to authenticate users to ensure that they are
charged appropriately for the services they use. The
re-authentication done when the user initiates a message will not
suffice for this purpose, as described below.
If the duration of the authentication period is set to a relatively
low value to ensure that the user cannot incur a high amount of
charges between two authentications, it may create a lot of
unnecessary authentications of users that have remained largely
inactive, and therefore it may use unnecessary air interface
resources.
If the duration of the authentication period is set to a relatively
high value to avoid these unnecessary authentications, the risk is
then that some users may incur high charges between authentications.
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A user's authentication is automatically invalidated when a certain
threshold for charges (or number, or duration of sessions) is reached
without giving the user a chance to re-authenticate, even if a valid
registration exists. This would not provide an adequate level of
service.
Consequently, it must be possible for the network to initiate a
re-authentication process at any time. The triggers must be set
within the network and may include charging thresholds, number of
events, session duration, etc.
4.23. Security Model
Sections 4.23, 4.24, and 4.25 have been based on the 3GPP Technical
Specifications 33.203 [32], 23.228 [28], and 33.210 [33].
The scope for security of the 3GPP IMS is the SIP signaling between
the various SIP entities. Protecting the end-to-end media streams
may be a future extension, but it is not considered in the Release 5
version of the IMS specifications.
Each operator providing IMS services acts as its own domain of trust
and shares a long-term security association with its subscribers
(e.g., pre-shared keys). Operators may enter into roaming agreements
with other operators, in which case a certain level of trust exists
between their respective domains.
SIP UAs must authenticate to their home network before the use of IMS
resources is authorized. In Release 5 of the 3GPP IMS
specifications, authentication is performed during registration and
re-registrations.
Portions of the SIP signaling must be protected hop by hop. Looking
at Figure 1 in Section 3, we can distinguish two distinct zones where
the required security is unique:
o Access Domain: Between the SIP user device and the visited
network.
o Network Domain: Between the visited and home networks, or inside
the home network.
Characteristics needed in the Access Domain are quite different from
those of the Network Domain because of the terminal's requirements
for 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
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security associations, whereas Network Domain provides security
associations between network nodes. Therefore, the weight of
protocols and algorithms and their compliance with compression
mechanisms are very important to Access Domain Security. It is
therefore required that the security solutions allow different
mechanisms in these two domains.
4.24. Access Domain Security
4.24.1. General Requirements
4.24.1.1. Scalability and Efficiency
3GPP IMS is 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.
4.24.1.2. Bandwidth and Round-trips
The wireless interface in 3GPP terminals is an expensive resource
both in terms of power consumption and maximum use of scarce
spectrum. Furthermore, cellular networks typically have 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 round-trips. In particular, during
normal call signaling there should not be any additional security-
related messages.
4.24.1.3. Computation
It must be possible for mobile device terminals to provide security
without requiring public key cryptography and/or certificates. 3GPP
IMS may, however, include optional security schemes that employ these
techniques.
Current HTTP authentication methods use only symmetric cryptography,
as required here. Lower-layer mechanisms (IKE, TLS) require
implementation of public-key cryptography e.g., Diffie-Hellman. If
these lower-layer mechanisms were used, the mobile terminal would
authenticate and negotiate session keys with the visited network
using only symmetric methods.
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4.24.1.4. Independence of the Transport Protocol
The selected security mechanism should work with any transport
protocol allowed by SIP (e.g., TCP, UDP).
4.24.2. Authentication
Authentication, as used in this context, means entity authentication
that enables two entities to verify the identity of the respective
peer.
4.24.2.1. Authentication Method
A strong, mutual authentication must be provided.
The authentication method must be able to work when there are zero or
more SIP proxies in the SIP path between the authenticator and the
authenticated user.
It must be possible to support extensible authentication methods.
Therefore, authentication using an extensible authentication
framework is strongly recommended.
Authentication methods based on the secure storage of long-term keys
used for authentication and the secure execution of authentication
algorithms must be supported.
The SIP client's credentials must not be transferred as plain text.
3GPP intends to reuse UMTS AKA [13]. 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 message protection that apply to
the authentication method are stated in Section 4.24.3.
4.24.3. Message Protection
4.24.3.1. Message Protection Mechanisms
SIP entities (typically a SIP client and a SIP proxy) must be able to
communicate using integrity. By integrity, we mean the ability for
the receiver of a message to verify that the message has not been
modified in transit. SIP entities should be able to communicate
confidentially. In 3GPP IMS, these protection modes must be based on
initial authentication. Integrity protection and confidentiality
must be possible using symmetric cryptographic keys.
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It must also be possible to handle error conditions in a satisfactory
manner as to allow recovery (see also sections 4.3.6.3 and 4.14).
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 the 3GPP Release 5
IMS does not require this.
The security mechanism must be able to protect a complete SIP
message.
If header compression/removal or SIP compression is applied to SIP
messages, it must be compatible with message protection.
4.24.3.2. Delegation
3GPP IMS implements distributed security functions responsible for
authentication and message protection.
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 authentication. The
authentication mechanism used is able to provide such session keys.
It must be possible to apply subsequent message protection as soon as
possible, even during the initial authentication period.
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-resistant) 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 delegate subsequent SIP signaling
protection (e.g., session keys for integrity or encryption) securely
to an authorized SIP proxy further downstream. The tamper-resistant
device at the client side must be able to delegate the session keys
securely to the SIP UA.
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4.24.4. Negotiation of Mechanisms
A method must be provided to negotiate the security mechanisms to be
used in the access domain securely.
This method must at least support the negotiation of different
security mechanisms providing integrity protection and encryption,
algorithms used within these mechanisms, and additional parameters
that they require in order to be exchanged.
The negotiation mechanism must protect against attackers who do not
have access to authentication credentials. In particular, the
negotiation mechanism must be able to detect a possible
man-in-the-middle attacker who could influence the negotiation result
so that services with weaker security or with none are negotiated.
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 algorithms. Note that when referring to
negotiation we mean just the negotiation, not all functions in
protocols such as IKE. For instance, we expect that the session key
generation is to be a part of the initial authentication.
SIP entities must be able to 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 a registration. Note that it is necessary to amortize
the cost of security association setup and parameter negotiation over
several INVITEs.
4.24.5. Verification of Messages
4.24.5.1. Verification at the SIP Outbound Proxy
The SIP outbound proxy must be able to guarantee the message origin
and to verify that the message has not been changed (e.g., it is
integrity protected).
4.24.5.2. Verification at the SIP Serving Proxy
The serving SIP proxy needs to receive an indication if the outbound
proxy was able to verify the message origin and, in the case of a
REGISTER request, whether or not it was integrity protected.
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4.25. Network Domain Security
Message authentication, key agreement, integrity and replay
protection, and confidentiality must be provided for communications
between SIP network entities such as proxy servers.
Network domain security mechanisms must be scalable up to a large
number of network elements.
3GPP intends to make having the protection discussed above mandatory
at least between two operators, and optional within an operator's own
network. Security gateways exist between operator's networks.
We believe that the above requirements are fulfilled by applying
security mechanisms as specified in the current IP Security standards
in RFC 2401 [5].
5. Security Considerations
This document does not define a protocol, but still presents some
security requirements to protocols. The main security requirements
are stated in sections 4.23, 4.24, and 4.25. Additional
security-related issues are discussed under sections 4.6, 4.7, 4.8,
4.9, 4.10, and 4.12.
6. Contributors
The following people contributed to this document:
Duncan Mills (Vodafone), Gabor Bajko (Nokia), Georg Mayer (Siemens),
Francois-Xerome Derome (Alcatel), Hugh Shieh (AWS), Andrew Allen
(dynamicsoft), Sunil Chotai (mmO2), Keith Drage (Lucent), Jayshree
Bharatia (Nortel), Kevan Hobbis (Huthison 3G UK), Dean Willis
(dynamicsoft), Krisztian Kiss (Nokia), Vesa Torvinen (Ericsson), Jari
Arkko (Ericsson), and Sonia Garapaty (Nortel).
7. References
7.1. 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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7.2. Informative References
[3] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0",
RFC 2246, January 1999.
[4] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifiers (URI): Generic Syntax", RFC 2396,
August 1998.
[5] Kent, S. and R. Atkinson, "Security Architecture for the
Internet Protocol", RFC 2401, November 1998.
[6] Aboba, B. and M. Beadles, "The Network Access Identifier",
RFC 2486, January 1999.
[7] Shirey, R., "Internet Security Glossary", RFC 2828, May 2000.
[8] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals", RFC 2833, May 2000.
[9] Faltstrom, P., "E.164 number and DNS", RFC 2916, September
2000.
[10] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol
(SIP): Locating SIP Servers", RFC 3263, June 2002.
[11] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[12] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[13] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext Transfer
Protocol (HTTP) Digest Authentication Using Authentication and
Key Agreement (AKA)", RFC 3310, September 2002.
[14] Rosenberg, J., "A Session Initiation Protocol (SIP) Event
Package for Registrations", RFC 3680, March 2004.
[15] Camarillo, G., Marshall, W., and J. Rosenberg, "Integration of
Resource Management and Session Initiation Protocol (SIP)", RFC
3312, October 2002.
[16] Marshall, W., "Private Session Initiation Protocol (SIP)
Extensions for Media Authorization", RFC 3313, January 2003.
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[17] Jennings, C., Peterson, J., and M. Watson, "Private Extensions
to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks", RFC 3325, November 2002.
[18] Peterson, J., "A Privacy Mechanism for the Session Initiation
Protocol (SIP)", RFC 3323, November 2002.
[19] Schulzrinne, H. and B. Volz, "Dynamic Host Configuration
Protocol (DHCPv6) Options for Session Initiation Protocol (SIP)
Servers", RFC 3319, July 2003.
[20] Sparks, R., "Session Initiation Protocol Call Control -
Transfer", Work in Progress, February 2005.
[21] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K.
Summers, "Session Initiation Protocol (SIP) Basic Call Flow
Examples", BCP 75, RFC 3665, December 2003.
[22] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K.
Summers, "Session Initiation Protocol (SIP) Public Switched
Telephone Network (PSTN) Call Flows", BCP 76, RFC 3666,
December 2003.
[23] Johnston, A. and R. Sparks, "Session Initiation Protocol
Service Examples", Work in Progress, February 2005.
[24] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[25] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) 'Replaces' Header", RFC 3891, September 2004.
[26] 3GPP, "TS 23.003 Numbering, addressing and identification
(Release 5)", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/23_series/23.003/>.
[27] 3GPP, "TS 23.060:General Packet Radio Service (GRPS); Service
Description; Stage 2", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/23_series/23.060/>.
[28] 3GPP, "TS 23.228: IP Multimedia Subsystem (IMS) (Stage 2) -
Release 5", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/23_series/23.228/>.
[29] 3GPP, "TS 24.228: Signaling flows for the IP Multimedia call
control based on SIP and SDP", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/24_series/24.228/>.
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[30] 3GPP, "TS 24.229: IP Multimedia Subsystem (IMS) (Stage 3) -
Release 5", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/24_series/24.229/>.
[31] 3GPP, "TS 32.225: Telecommunication Management; Charging
Management; Charging Data Description for IP Multimedia
Subsystem; (Release 5)", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/32_series/32.225/>.
[32] 3GPP, "TS 32.203: 3G Security; Access security for IP based
services; (Release 5)", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/33_series/33.203/>.
[33] 3GPP, "TS 32.210: 3G Security; Network Domain Security; IP
network layer security (Release 5)", September 2002,
<ftp://ftp.3gpp.org/Specs/archive/33_series/33.210/>.
[34] ITU-T, "Recommendation E.164 (05/97): The international public
telecommunication numbering plan", May 1997,
<http://www.itu.int/rec/recommendation.asp?
type=folders&lang=e&parent=T-REC-E.164>.
[35] Schulzrinne, H., "The tel URI for Telephone Numbers", RFC 3966,
December 2004.
Author's Address
Miguel A. Garcia-Martin
Nokia
P.O. Box 407
NOKIA GROUP, FIN 00045
Finland
EMail: miguel.an.garcia@nokia.com
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RFC 4083 3GPP R5 Requirements on SIP May 2005
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