SIST-TS TS 101 884-1 V4.1.1:2004
(Main)Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) Release 4; Technology Mapping; Part 1: Implementation of TIPHON architecture using SIP
Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) Release 4; Technology Mapping; Part 1: Implementation of TIPHON architecture using SIP
This work item will deliver a mapping of the TIPHON architecture to the SIP protocol.
Harmonizacija telekomunikacij in internetnega protokola prek omrežij (TIPHON), 4. izdaja - 1. del: Tehnologija preslikave - Implementacija arhitekture TIPHON z uporabo SIP
General Information
Standards Content (Sample)
SLOVENSKI STANDARD
SIST-TS TS 101 884-1 V4.1.1:2004
01-april-2004
Harmonizacija telekomunikacij in internetnega protokola prek omrežij (TIPHON), 4.
izdaja - 1. del: Tehnologija preslikave - Implementacija arhitekture TIPHON z
uporabo SIP
Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON)
Release 4; Technology Mapping; Part 1: Implementation of TIPHON architecture using
SIP
Ta slovenski standard je istoveten z: TS 101 884-1 Version 4.1.1
ICS:
33.020 Telekomunikacije na splošno Telecommunications in
general
SIST-TS TS 101 884-1 V4.1.1:2004 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
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SIST-TS TS 101 884-1 V4.1.1:2004
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SIST-TS TS 101 884-1 V4.1.1:2004
ETSI TS 101 884-1 V4.1.1 (2003-08)
Technical Specification
Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON) Release 4;
Technology Mapping;
Part 1: Implementation of TIPHON architecture using SIP
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2 ETSI TS 101 884-1 V4.1.1 (2003-08)
Reference
RTS/TIPHON-03018-1R4
Keywords
architecture, IP, SIP, telephony, VoIP
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ETSI
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3 ETSI TS 101 884-1 V4.1.1 (2003-08)
Contents
Intellectual Property Rights.5
Foreword.5
1 Scope.6
2 References.7
3 Definitions and abbreviations.8
3.1 Definitions.8
3.2 Abbreviations.8
4 SIP environment overview .8
4.1 Introduction.8
4.2 SIP protocol.9
4.2.1 SIP signalling, methods and responses .9
4.2.1.1 SIP signalling.9
4.2.1.2 Methods and responses .9
4.2.2 SIP protocol components .10
4.3 SDP.10
4.4 HTTP/1.1.10
5 Implementation of TIPHON functional architecture using SIP .10
5.1 Introduction.10
5.2 SIP functional architecture .10
6 Registration service.12
6.1 Introduction.12
6.2 Registration functional entities mapping.14
6.3 Registration Messages Mapping.14
6.4 Registration information flow Mapping.15
6.4.1 Relationship ra (RFE1/RFE2).15
6.4.2 Relationship rb (RFE2/RFE3).17
6.4.3 Relationship rc (RFE1/RFE3).18
6.4.4 Relationship rd (RFE2/RFE4).18
6.5 Registration action Mapping .19
6.6 Conclusion.19
7 Simple call application .20
7.1 Introduction.20
7.2 Simple call functional entities mapping .24
7.3 Simple call messages mapping.24
7.4 Simple call information flow mapping.25
7.4.1 Relationship ra (CallingUser/CFE1).26
7.4.2 Relationship rf, ri (CFE3/CFE6/CFE9) .28
7.4.3 Relationship rl (CFE11/CalledUser).30
7.5 Simple call functional entity actions mapping.32
7.6 Timers.33
7.7 Conclusion.34
8 Media Control service .34
8.1 Introduction.34
8.2 Media Control functional entities mapping .34
8.3 Media Control information flow Mapping .34
8.3.1 Relationship ra (CCA/MFE1).35
8.4 Conclusion.36
9 Transport.36
9.1 Introduction.36
10 Supplementary services.36
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11 Control of end-to-end Quality of Service.36
11.1 Introduction.36
11.2 Control of end-to-end Quality of Service functional entities mapping.37
11.3 Control of end-to-end Quality of Service flows mapping .37
11.4 Control of end-to-end Quality of Service information flow data mapping.39
11.4.1 Relationship ra (CallingUser/QFE1).39
11.4.2 Relationship rc, rd (QFE2/QFE8/QFE3) .40
11.4.3 Relationship rf (QFE4/CalledUser) .42
11.4.4 Relationship rg (QFE1/QFE5) .42
11.5 Control of end-to-end Quality of Service functional entity actions mapping.43
11.6 Timers.43
11.7 Conclusion.44
12 Security service.44
Annex A (informative): Bibliography.48
History .49
ETSI
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Intellectual Property Rights
IPRs essential or potentially essential to the present document may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web
server (http://webapp.etsi.org/IPR/home.asp).
Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee
can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web
server) which are, or may be, or may become, essential to the present document.
Foreword
This Technical Specification (TS) has been produced by ETSI Project Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON).
The present document is part 1 of a multi-part deliverable covering implementation, of TIPHON architecture using the
SIP protocol, as identified below:
Part 1: "Implementation of TIPHON architecture using SIP";
Part 2: "Implementation Profile for SIP".
ETSI
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1 Scope
The present document describes how the SIP protocol [9] completed with correlated protocols like SDP [11],
HTTP [12] can be a candidate for TIPHON release 4 according to guidelines given in TS 101 315 Release 4 (see
bibliography) and TS 101 315 Release 3 [2].
The SIP profile is derived from the examination of the following TIPHON Release 4 documents:
• the TIPHON baseline architecture described in TS 101 314 [1];
• the capabilities service required by TS 101 878 (see bibliography);
• the Meta-protocol as defined in multi part document TS 101 882-1 [5], TS 101 882-2 [6], TS 101 882-3 (see
bibliography), TS 101 882-4 (see bibliography) and TS 101 882-5 [7];
• the end-to-end Quality of Service defined in TS 102 024-3 [3];
• the Security service defined in TS 102 165-1 [4].
The mapping of Meta-Protocol to SIP is limited to the following parts, while other parts are not available yet like
supplementary services:
• Registration Meta-Protocol [6];
• Simple Call Meta-Protocol (TS 101 882-3 - see bibliography);
• Media Control Meta-Protocol (TS 101 882-4 - see bibliography);
• the end-to-end Quality of Service defined in TS 102 024-3 [3];
• IETF RFC 3261: "SIP: Session Initiation Protocol" [9];
• IETF RFC 2327: "SDP: Session Description Protocol" [11];
• IETF RFC 2616: "Hypertext Transfer Protocol - HTTP/1.1" [12];
• IETF RFC 2617: "HTTP Authentication: Basic and Digest Authentication" (see bibliography);
Furthermore the following documents have been consulted for information:
• TS 124 229: "IP Multimedia Call Control Protocol based on SIP and SDP" (see bibliograpy);
• TS 124 228: "Signalling flows for the IP multimedia call control based on SIP and SDP" [8].
ETSI
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2 References
The following documents contain provisions which, through reference in this text, constitute provisions of the present
document.
• References are either specific (identified by date of publication and/or edition number or version number) or
non-specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
http://docbox.etsi.org/Reference.
[1] ETSI TS 101 314: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; Abstract Architecture and Reference Points Definition; Network
Architecture and Reference Points".
[2] ETSI TS 101 315: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 3; Functional entities, information flow and reference point definitions;
Guidelines for application of TIPHON functional architecture to inter-domain services".
[3] ETSI TS 102 024-3: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; End-to-end Quality of Service in TIPHON Systems; Part 3: Signalling and
Control of end-to-end Quality of Service".
[4] ETSI TS 102 165-1: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; Protocol Framework Definition; Methods and Protocols for Security; Part 1:
Threat Analysis".
[5] ETSI TS 101 882-1: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; Protocol Framework Definition; Part 1: Meta-protocol design rules,
development method, and mapping guideline".
[6] ETSI TS 101 882-2: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; Protocol Framework Definition; Part 2: Registration and Service
Attachment service meta-protocol definition.".
[7] ETSI TS 101 882-5: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 4; Protocol Framework Definition; Part 5: Transport control service meta-
protocol definition;".
[8] ETSI TS 124 228: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); Signalling flows for the IP multimedia call control based
on SIP and SDP; Stage 3 (3GPP TS 24.228 version 5.3.0 Release 5)".
[9] IETF RFC 3261: "SIP: Session Initiation Protocol".
[10] IETF RFC 3264: "An Offer/Answer Model with Session Description Protocol (SDP)".
[11] IETF RFC 2327: "SDP: Session Description Protocol".
[12] IETF RFC 2616: "Hypertext Transfer Protocol - HTTP/1.1".
[13] IETF RFC 2617: "HTTP Authentication: Basic and Digest Access Authentication".
[14] IETF RFC 1890: "RTP Profile for Audio and Video Conferences with Minimal Control".
[15] IETF RFC 1889: "RTP: A Transport Protocol for Real-Time Applications".
[16] IETF RFC 2806: "URLs for Telephone Calls".
[17] IETF RFC 2748: "The COPS (Common Open Policy Service) Protocol".
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[18] IETF RFC 2326: "Real Time Streaming Protocol (RTSP)".
[19] IETF RFC 3525: "Gateway Control Protocol Version 1".
[20] IETF RFC 3265: "Session Initiation Protocol (SIP)-Specific Event Notification".
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the terms and definitions given in TS 101 314 [1] and TS 101 878 (see
bibliography) apply.
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
B2BUA Back-to-Back User Agent
BC Bearer Control
CC Call Control
COPS Common Open Policy Service
FG Functional Group
ICF Inter-Connect Function
IP Internet Protocol
MC Media Control
NFG Network Functional Group
PCM Pulse Code Modulation
PDP Policy Decision Point
PSTN Public Switched Telephone Network
QoS Quality of Service
RPC Remote Procedure Call
SAP Service Access Point
SC Service Control
SDP Session Description Protocol
SpoA Service point of Attachment
TE Terminal Equipment
UA User Agent
UAC User Agent Client
UAS User Agent Server
URI Uniform Resource Identifier
4 SIP environment overview
4.1 Introduction
The purpose of the present document is not to describe how to implement SIP protocol but how TIPHON protocol can
be represented in a SIP environment. For example parameter mandatory in SIP but without equivalence in TIPHON
information elements are not documented. Mandatory behaviours in SIP that do not correspond to any TIPHON
behaviours are not documented either.
The aim is to identify gap in TIPHON to SIP direction between both protocols. Informative suggestions to fill those
gaps could be given in conclusion.
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4.2 SIP protocol
SIP is a relatively new technology (1995) developed for remote control, establishment and tear-down of multimedia
sessions. The origins of SIP are in the academic and IETF community and assumed in its first incarnation a public
internet although with the interest shown by 3GPP the application to a managed network that uses IP has become
ascendant. SIP is based upon the communication model of HTTP and therefore is broadly viewed as a request-response
protocol. In relation to other well known protocols SIP has close cousins in Remote Procedure Call (RPC) and in the
ITU-T ROSE protocol.
According to RFC 3261 [9], SIP is an application-layer-control protocol to manage multimedia session. But, "SIP is not
a vertically integrated communications system", and will need other IETF protocols to build a complete multimedia
architecture (e.g.: RTP RFC 1889 [15], RTSP RFC 2326 [18], MEGACO RFC 3525 [19], SDP RFC 2327 [11]).
The choice of the protocol for the session description is opened in SIP and appears in SIP only as a parameter value
(Content-Type). The media type descriptions that can be included in the body of a SIP message are Internet Media
Types as in HTPP/1.1. However, in this profile only Session Description Protocol (SDP) defined in RFC 2327 [11] has
been considered. SIP reuses also the authentication mechanism defined in HTTP.
The SIP technology has been considered through the following standards:
• "SIP: Session Initiation Protocol" - RFC 3261 [9].
• "SDP: Session Description Protocol" - RFC 2327 [11].
• "RTP Profile for Audio and Video Conferences with Minimal Control" - RFC 1890 [14].
• "Hypertext Transfer Protocol - HTTP/1.1" - RFC 2616 [12].
SIP does not define services. Rather, SIP provides primitives that can be used to implement different services. For
example, SIP can locate a user and deliver an opaque object to his current location. If this primitive is used to deliver a
session description written in SDP, for instance, the endpoints can agree on the parameters of a session. If the same
primitive is used to deliver a photo of the caller as well as the session description, a "caller ID" service can be easily
implemented. As this example shows, a single primitive is typically used to provide several different services.
SIP does not offer conference control services such as floor control or voting and does not prescribe how a conference
is to be managed. SIP can be used to initiate a session that uses some other conference control protocol. Since SIP
messages and the sessions they establish can pass through entirely different networks, SIP cannot, and does not, provide
any kind of network resource reservation capabilities.
The nature of the services provided make security particularly important. To that end, SIP provides a suite of security
services, which include denial-of-service prevention, authentication (both user to user and proxy to user), integrity
protection, and encryption and privacy services.
SIP works with both IPv4 and IPv6.
4.2.1 SIP signalling, methods and responses
4.2.1.1 SIP signalling
The SIP protocol client/server machine is very simple: Request is sent and the requestor (client) waits for a response.
The request contains the method and who the method is aimed at, the response contains the status code that informs the
requestor of how the server has dealt with the request.
4.2.1.2 Methods and responses
There are 6 core methods in SIP and these are the basis of the protocol:
• INVITE - starts a session (and modifies it if used as a re-invite).
• ACK - confirms the invite.
• BYE - terminates a sessions.
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• CANCEL - cancel an invite.
• OPTIONS - Querying capability.
• REGISTER - binds a user's address (SIP name) to a network address (IP address).
4.2.2 SIP protocol components
The protocol of SIP is enabled by assigning particular functions to a set of protocol components. A particular SIP
device will contain 1 or more of these components.
• User Agent Client (UAC).
• User Agent Server (UAS).
• Redirect server.
• Proxy server.
• Registrar.
The UAC and UAS exist in a normal terminal device and are termed jointly the User Agent.
The proxy server arises from breaking the assumption that the UACs know the UASs that they want to communicate
with. In anything but the smallest of networks this assumption is inevitably broken so a network resident proxy to the
UA exists to facilitate routing.
• Proxy servers can be configured to perform inter-domain call establishment.
• The registrar server is a special server that attends to REGISTER methods. In most cases the registrar and
proxy server will be co-located.
4.3 SDP
SDP is a session description protocol in text format language. It is used in SIP to define a simple offer/answer model to
a describe unicast session. Mapping in the present document has been based on RFC 2327 [11] overloaded with
RFC 3264 [10].
4.4 HTTP/1.1
Hypertext Transfer protocol provides a scheme description for authentication.
According to RFC 3261 [9], chapter 22, only the "Digest" authentication mechanism described in RFC 2617 [13]
overload by RFC 3261 [9] has to be considered.
5 Implementation of TIPHON functional architecture
using SIP
5.1 Introduction
5.2 SIP functional architecture
The SIP Architecture has the following functional elements, as defined in [9].
User Agent (UA): The user agent is the functional entity that may initiate or respond to a SIP request.
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In a TIPHON compliant system, the SIP User Agent (UA) shall provide the functionality of the terminal functional
group. The terminal functional group performs the roles of the terminal registration functional group, originating
terminal functional group and the terminating terminal functional group. The reference points S1, SC1 and N1 are
regarded as internal to the TE.
Back-to-Back User Agent (B2BUA): B2BUA is a logical entity that receives a request and processes it as a User
Agent Server (UAS). In order to determine how a request should be answered, it acts as a User Agent Client (UAC) and
generates requests. Unlike a proxy server (stateless), it maintains a dialogue state, and must participate in all requests
sent on the dialogues it has established. TIPHON recommends the use of a B2BUA, as network functional groupings
involved in providing a service.
Proxy server: A proxy server acts as both the client and server: It receives a request from an entity, and initiates a
request on behalf of the requesting entity, hence acting as a server for the requesting entity. A proxy server can be
stateless (forgets about the state of a particular session) or statefull (keeps track of the state of the session it is involved
in).
Redirect server: A redirect server receives requests from an entity, and returns the contact address of the destination to
the resquesting entity.
Registrar: The registrar processes registration requests; as a minimum this involves updating the users contact list and
responding to the originator of the request. Typically a registrar is co-located with either the proxy or the redirect server,
and may be adapted to perform location-based services.
SIP gateway: A SIP gateway acts as an interworking medium between the PSTN and a SIP network. It provides an
interworking between SIP and PSTN call control protocols, such as ISUP, as well as interworking between the TDM
and IP media flows. A SIP gateway can be decomposed into a gateway controller (taking cate of the call control
protocol conversion) and a media gateway (taking cate of the TDM to IP media conversion).
Figure 1 shows how the SIP functional elements map onto the functional layers in the IP Telephony Application plane.
Figure 1: Void
The UA maps to Service, Service Control (SC), Call Control (CC), and Bearer Contro
...
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