SIST-TP TR 101 308 V1.1.1:2004

Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); Requirements Definition Study; SIP and H.323 Interworking

This document identifies and defines required service mechanisms to ensure service interoperability for TIPHON which are applicable to release 3. It includes, but is not limited to identifying requirements to interwork between SIP and H.323 administrative domain. The approved delivery from this work item should be used as a common base line for TIPHON and should be used during the whole project life cycle regarding phase 3.

Harmonizacija telekomunikacij in internetnega protokola prek omrežij (TIPHON) - Študija definicije zahtev - Medsebojno delovanje SIP in H.323

General Information

• Status: Published
• Publication Date: 31-Mar-2004
• ICS: 33.020 - Telecommunications in general
• Technical Committee: SPN - Services and Protocols for Networks
• Current Stage: 6060 - National Implementation/Publication (Adopted Project)
• Start Date: 01-Apr-2004
• Due Date: 01-Apr-2004
• Completion Date: 01-Apr-2004
• Ref Project: ETSI TR 101 308 V1.1.1 (2001-12) - Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON); Requirements Definition Study; SIP and H.323 Interworking

Standards Content (Sample)

SLOVENSKI STANDARD
SIST-TP TR 101 308 V1.1.1:2004
01-april-2004
Harmonizacija telekomunikacij in internetnega protokola prek omrežij (TIPHON) -
Študija definicije zahtev - Medsebojno delovanje SIP in H.323
Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON);
Requirements Definition Study; SIP and H.323 Interworking
Ta slovenski standard je istoveten z: TR 101 308 Version 1.1.1
ICS:
33.020 Telekomunikacije na splošno Telecommunications in
general
SIST-TP TR 101 308 V1.1.1:2004 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.

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SIST-TP TR 101 308 V1.1.1:2004

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SIST-TP TR 101 308 V1.1.1:2004

ETSI TR 101 308 V1.1.1 (2001-12)
Technical Report


Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON);
Requirements Definition Study;
SIP and H.323 Interworking

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SIST-TP TR 101 308 V1.1.1:2004
 2 ETSI TR 101 308 V1.1.1 (2001-12)



Reference
DTR/TIPHON-01005
Keywords
H.323, Internet, interworking, network, protocol,
SIP
ETSI
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© European Telecommunications Standards Institute 2001.
All rights reserved.

ETSI

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 3 ETSI TR 101 308 V1.1.1 (2001-12)
Contents
Intellectual Property Rights.5
Foreword.5
Introduction .5
1 Scope.6
2 References.6
3 Definitions and abbreviations.6
3.1 Definitions.6
3.2 Abbreviations.7
4 Operating modes.7
4.1 Native H.323 operating modes .7
4.1.1 H.323 peer-to-peer mode.7
4.1.2 H.323 gatekeeper routed call signalling mode.7
4.1.3 H.323 direct call signalling mode .8
4.1.4 H.323 registration.8
4.2 Native SIP operating modes .8
4.2.1 SIP peer-to-peer.8
4.2.2 SIP proxy routed .8
4.2.3 SIP with redirect server.8
4.2.4 SIP registration.8
4.3 Recommended modes of operation .9
4.3.1 H.323 administrative domain.9
4.3.2 SIP administrative domain.9
5 Interworking scenarios.10
5.1 Simple scenarios.10
5.1.1 Dual Stacking in endpoints .10
5.1.2 Interworking between a SIP domain and an H.323 domain.10
5.2 Practical interworking scenarios.11
5.2.1 Interconnection between 3GPP SIP based network and TIPHON H.323 Network .11
5.2.2 Interconnecting PacketCable and TIPHON H.323 Networks.12
5.3 Tandem scenarios.13
5.3.1 Tandem connection of H.323 domains via a SIP intermediary.13
5.3.2 Tandem connection of SIP domains via an H.323 intermediary.13
5.4 Support of administrative domains.13
5.4.1 Administrative control from a SIP domain .14
5.4.2 Administrative control from an H.323 domain.14
5.4.3 Interworking between administrative domains .15
5.4.3.1 Direct peering.15
5.4.3.2 In-direct peering.15
6 Encapsulation of other signalling protocols .15
6.1 Control of tunnels through IWF .16
6.2 Encapsulation of native signalling.16
6.3 Encapsulation of other signalling .17
7 Other considerations.19
7.1 Security considerations.19
7.1.1 Signalling security.19
7.1.2 Media security.19
7.2 Addressing and naming consideration.20
7.3 Quality of service considerations .20
7.4 Management considerations.20
7.4.1 Reporting.20
7.4.2 Diagnostic test.20
ETSI

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7.4.3 Fault management.20
8 Prioritized requirements.20
Annex A: Bibliography.21
History .22

ETSI

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 5 ETSI TR 101 308 V1.1.1 (2001-12)
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 Report (TR) has been produced by ETSI Project Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON).
Introduction
The ETSI Project TIPHON is concerned with the interaction between IP based communication devices and circuit
switched networks. The project focuses on voice communication and related multimedia aspects as required for
interoperability between IP based networks and other types of networks.
The project has predicated much of its early work on the use of the ITU-T Recommendation H.323 [1] specification
since this was the most mature and relevant base specification at the time. The IETF's Internet Multimedia
Conferencing Architecture has continued to develop and has started to spawn technologies based upon its signalling and
control component - the Session Initiation Protocol (SIP). A SIP Working Group has since been formed within IETF
and SIP has been adopted by a number of derivative works, including the IPTEL working group in the IETF. It is
therefore appropriate for TIPHON to consider the impact that the introduction of SIP based equipment may have on
large-scale public networks.
ETSI

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1 Scope
The present document identifies and defines required service mechanisms to ensure service interoperability for
TIPHON which are applicable to release 3. It includes, but is not limited to identifying requirements to interwork
between SIP and H.323 administrative domain. The approved delivery from this work item should be used as a common
base line for TIPHON and should be used during the whole project life cycle.
2 References
For the purposes of this Technical Report (TR), the following references apply:
[1] ITU-T Recommendation H.323: "Packet-based multimedia communications systems".
[2] RFC 2543 (1999): "SIP: Session Initiation Protocol".
[3] ETSI TS 101 329-3: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON); End-to-End Quality of Service in TIPHON Systems; Part 3: Signalling and Control of
end-to-end Quality of Service".
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
administrative domain: bounded entity within which all encompassed elements are under common ownership,
operation and management
endpoint: Entity that can originate and terminate both signalling and media streams. An endpoint can both call and be
called. Examples of endpoints include H.323 terminals, SIP User Agents, Gateways, or Multi-party Conference Units.
GateKeeper (GK): H.323 entity on the network which provides address translation and controls access to the network
for H.323 terminals, gateways and MCUs. A Gatekeeper may also provide other services such as bandwidth
management and gateway location to terminals, gateways and MCUs.
InterWorking Function (IWF): function connecting two networks of differing signalling technology or administrative
policies
proxy server: Anetwork element that acts as both a client and server for the purpose of making SIP requests on behalf
of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A
proxy interprets and if necessary rewrites a request message before forwarding it.
redirect server: Server that accepts a SIP request, maps the address into zero or more new addresses and returns these
addresses to the client. Unlike a proxy server, it does not initiate its own SIP request. Unlike a UAS, it does not accept
calls.
registrar: SIP server that accepts REGISTER requests. A registrar is typically co-located with a proxy or re-direct
server and MAY offer location services.
Switched Circuit Network (SCN): Telecommunications network, e.g. Public Switched Telephone Network (PSTN),
Integrated Services Digital Network (ISDN), and General System for Mobile communications (GSM), that uses
circuit-switched technologies for the support of voice calls. The SCN may be a public network or a private network.
telephone call: two-way speech communication between two users by means of terminals connected via network
infrastructure
terminal: endpoint other than a gateway or a multipoint control unit
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User Agent (UA): application which contains both a UAC and UAS
User Agent Client (UAC): client application that initiates the SIP request
User Agent Server (UAS): Server application that contacts the user when a SIP request is received and that returns a
response on behalf of the user. The response accepts, rejects or redirects the request.
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
CMS Call Management Server
CMTS Call Modem Termination System
CSCF Call Serve Control Function
DNS Domain Name Server
GK H.323 GateKeeper
HFC Hybrid Fiber Coax
IP Internet Protocol
ISUP SS7 ISDN User Part
IWF InterWorking Function
MT Mobile Termination
MTA Multimedia Terminal Adapter
SCN Switched Circuit Networks
SIP Session Initiation Protocol
UA SIP User Agent
UAC SIP User Client
UAS SIP User Server
VoIP Voice over IP
4 Operating modes
H.323 and SIP have a number of declared modes of operation that relate the various functions they identify and define
how calls are routed and controlled within their environment. The following clauses identify each mode of operation
that is relevant to a potential network context requiring interworking from one protocol technology to another.
4.1 Native H.323 operating modes
H.323 is an architecture for implementing multimedia conferencing over a packet network. It comprises
application-layer control protocols that can establish, modify and terminate multimedia sessions or calls. These
multimedia sessions may include multimedia conferences, distance learning, Internet telephony and similar
applications. It has essentially three possible modes of operation relevant to possible interworking requirements in the
context of TIPHON networks.
4.1.1 H.323 peer-to-peer mode
H.323 supports a peer-to-peer mode of operation. In a peer-to-peer architecture, endpoints contact each other directly,
without any control or co-ordination from any gatekeeper or intermediate server.
4.1.2 H.323 gatekeeper routed call signalling mode
A gatekeeper may play an active role in mediating call signalling between the calling and called end-points in H.323
networks with gatekeepers. In this environment, a gatekeeper may not only assume responsibility for call routing and
authorization on behalf of served endpoints but may also act as the signalling endpoint for calls entering an
administrative domain.
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4.1.3 H.323 direct call signalling mode
The strict peer-to-peer and gatekeeper routed models may be combined into a hybrid approach referred to as Direct Call
Signalling. In this case, gatekeepers provide call routing and authorization while individual endpoints are responsible
for establishing and disconnecting calls and media streams directly between each other.
4.1.4 H.323 registration
An H.323 zone is the collection of all terminals, gateways, and Multipoint Control Units managed by a single
gatekeeper. Registration is the process by which an endpoint joins an H.323 zone and informs the gatekeeper of its
transport and alias addresses. Once established, the registration of an end-point with a specific gatekeeper may need to
be refreshed on a periodic basis. An end-point must register with a gatekeeper before it can accept any call attempts.
4.2 Native SIP operating modes
As defined in RFC 2543 [2], the Session Initiation Protocol (SIP) is an application-layer control protocol that can
establish, modify and terminate multimedia sessions or calls. These multimedia sessions may include multimedia
conferences, distance learning, Internet telephony and similar applications. The most common SIP operation is the
invitation. Instead of directly reaching the intended destination, a SIP request may be redirected by Redirect Server or
proxied through Proxy Server. Users can also register their location(s) with SIP Registrar.
4.2.1 SIP peer-to-peer
In a peer-to-peer architecture, User Agents (UA) contact each other by sending invitation directly, without any control
or co-ordination from any proxy.
4.2.2 SIP proxy routed
SIP messages may be routed via an intermediary known as a proxy server. In such an environment, proxies not only
assume responsibility for call routing and authorization on behalf of their endpoints, they may also act as the signalling
endpoint for calls into their administrative domain. A proxy server can either be stateful or stateless.
A stateful proxy retains state information concerning both an incoming request and any associated outgoing requests. In
contrast, a stateless proxy does not retain any information concerning a received message or its response once an
outgoing request has been generated.
4.2.3 SIP with redirect server
SIP redirect servers represent an example of a loosely coupled distributed architecture. In this environment, the redirect
server provides the call routing information such that the originating UA first establishes a signalling connection with
the redirect server, before being re-directed to the terminating UA.
4.2.4 SIP registration
The SIP REGISTER method allows a client to let a SIP Registrar know at which address or addresses it can be reached.
A client may also use it to install call handling features at the server. A SIP Registrar may be collocated with either a
proxy or redirect server.
ETSI

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4.3 Recommended modes of operation
4.3.1 H.323 administrative domain
H.323
Gatekeeper
H.323
H.323 H.323 H.323
Terminal Terminal Terminal

Figure 1: An H.323 administrative domain
While a gatekeeper is an optional element for an H.323 network, it is recommended that TIPHON based H.323 centric
networks are deployed with an H.323 Gatekeeper. This should be configured to require registration with all the end
points within its administrative domain. It is further recommended that Gatekeeper Routed Call Signalling is used in
preference to Direct Call Signalling in order to support enhanced calling features such as availability look-ahead for the
called terminal.
4.3.2 SIP administrative domain
As defined in RFC 2543 [2], SIP has no concept of an Administrative Domain. However for practical network
engineering and operational reasons consistent with the TIPHON approach to QoS [3], the cocept of a SIP
Administrative Domain is introduced. This enables the trust boundaries within which all SIP devices are controlled by a
single operator to be delineated. Each SIP Administrative Domain is assumed to contain at least one Registrar. All SIP
UAs within that domain must register with the Registrar in order to allow the user's or terminal's address(es) to be
advertised within the domain.
SIP
Registrar
SIP
SIP UA SIP UA SIP UA

Figure 2: A SIP administrative domain
ETSI

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 10 ETSI TR 101 308 V1.1.1 (2001-12)
5 Interworking scenarios
Since the information elements and elements of procedure offered by both SIP and H.323 are capable of supporting a
wide range of services and applications, connecting a SIP network and H.323 network together will be more or less
complex to achieve depending upon the services being supported in a given scenario. To enable interworking between
two such networks to occur reliably, it is necessary to identify the core the conditions and events that will typically be
involved. The scenarios below are examples defined to facilitate the identification of such conditions and events.
5.1 Simple scenarios
5.1.1 Dual Stacking in endpoints
The simplest approach to achieving interworking between SIP and H.323 is to provide access to both protocol stacks
within the same endpoint. However, while this will provide the means where calls can be originated and terminated
from both types of network, it leaves any additional interworking issues to the responsibility of terminal equipment
manufacturer and will be inherently non-standard. It is therefore not seen as a long term solution.
5.1.2 Interworking between a SIP domain and an H.323 domain
I
H.323 SIP
W
F
H.323 SIP
Endpoint Endpoint

Figure 3: Interworking scenario - between a SIP domain and an H.323 domain
The alternative to providing access to both H.323 and SIP protocols within the same end-point requires the provision of
a suitable Interworking Function (IWF) that allows a call originated using one protocol to be terminated using the other.
This will necessitate the messages provided by SIP and H.323 to be related through a mapping process within the IWF.
Message mapping between protocols may be achieved on either a state-full or state-less basis and will require careful
attention depending upon the overall information flows required. However, before any such call flows can be provided,
it will be necessary for the end-points to be able to addressable outside of their native protocol domain. This will have
to be achieved in association with some form of registration process or procedure. Registration may be made from one
domain to the other depending on the arrangement of the Administration Domains concerned as discussed further in
clause 5.4.
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 11 ETSI TR 101 308 V1.1.1 (2001-12)
5.2 Practical interworking scenarios
5.2.1 Interconnection between 3GPP SIP based network and TIPHON
H.323 Network
TIPHON Network 3GPP Network
I
W
F Mm
SIP CSCF
H.323
(SIP Server)
Gi
GGSN SGSN
Media Media
Gateway Gateway
SIP-MT

Figure 4: Interconnection between TIPHON and 3GPP Network
TIPHON envisages a future network environment where all types of network - whether fixed, mobile, or satellite - are
combined in a manner permitting users to access services regardless of their terminal type, network connection or
geographical location. 3GPP have developed a SIP based mobile network architecture for third generation mobile
networks as part of the 3GPP Release 5 activities. This identifies a multimedia network interface (Mm in figure 4) that
could be realized by a TIPHON compliant fixed network. The TIPHON vision of a fully IP based network interworking
with existing fixed and mobile networks therefore complements the 3GPP Release 5 all-IP network vision. Where the
TIPHON network connecting with the 3GPP network is based upon H.323, there would be the need to establish
interworking between SIP and H.323 across the interface between the two networks, as indicated in figure 4.
ETSI

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5.2.2 Interconnecting PacketCable and TIPHON H.323 Networks
TIPHON System PacketCable Network
I
W
F SIP
Call Management
System
H.323 SIP
(CMS)
NCS
RTP
Media Terminal
Adapter
Media Media
(MTA)
Gateway Gateway

Figure 5: Interconnection between TIPHON and PacketCable Network
PacketCable™ is a project conducted by Cable Television Laboratories, Inc. (CableLabs®) and its member companies.
The PacketCable project is aimed at defining interface specifications that can be used to develop interoperable
equipment capable of providing packet-based voice, video and other high-speed multimedia services over Hybrid Fibre
Coax (HFC) cable systems based upon the DOCSIS protocol.
PacketCable consists of a variety of functional components, to create a mechanism for packet-based services. These
include the:
- Multimedia Terminal Adapter (MTA) which is a PacketCable client device that contains a subscriber-side
interface to the subscriber's CPE (e.g. telephone) and a network-side signalling interface to call control elements
in the network. An MTA provides codecs and all signalling and encapsulation functions required for media
transport and call signalling;
- Call Management Server (CMS) which provides call control and signalling related services for the MTA, Cable
Modem Termination System (CMTS), and PSTN gateways in the PacketCable network. The CMS is a trusted
network element that resides on the managed IP portion of the PacketCable network. The CMTS provides
connectivity between the DOCSIS HFC Access Network and a Managed IP Network;
- Cable Modem Termination System (CMTS) which provides data connectivity and complimentary functionality
to cable modems over the HFC access network. It also provides connectivity to wide area networks. The CMTS
is located at the cable television system head-end or distribution hub.
As shown in figure 5, PacketCable are currently developing a SIP based interface for use between CMS functions. As
this solution is deployed, there will be the opportunity to connect with networks of different types that may include
H.323 based networks, or networks based upon other network protocols. As an illustration of po
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