6364 B 6 Be 97 Dee

Doc 9804
AN/762
Manual on Air Traffic

Services (ATS)
Ground-Ground Voice
Switching and Signalling
Approved by the Secretary General

and published under his authority
First Edition — 2002
International Civil Aviation Organization

Doc 9804
AN/762
Manual on Air Traffic

Services (ATS)
Ground-Ground Voice
Switching and Signalling
Approved by the Secretary General

and published under his authority
First Edition — 2002
International Civil Aviation Organization

AMENDMENTS
The issue of amendments is announced regularly in the ICAO Journal and in the
monthly Supplement to the Catalogue of ICAO Publications and Audio-visual
Training Aids, which holders of this publication should consult. The space below
is provided to keep a record of such amendments.
RECORD OF AMENDMENTS AND CORRIGENDA
AMENDMENTS CORRIGENDA
No. Date Entered by No. Date Entered by
The designations employed and the presentation of the material in this

publication do not imply the expression of any opinion whatsoever on the part
of ICAO concerning the legal status of any country, territory, city or area or
of its authorities, or concerning the delimitation of its frontiers or boundaries.
(ii)
FOREWORD
This manual was developed with the assistance of the ATS requirements for ground-ground communications specified
Voice Switching and Signalling Study Group (AVSSSG) in Annex 11, Chapter 6, Section 6.2.9 Chapter 2 considers
formed by the Air Navigation Commission in 1998 with the the following aspects of ATS communications:
task of updating the ICAO provisions on air traffic services
(ATS) ground-ground voice communications to support the a) network planning, implementation and management;
introduction of digital technology.
b) quality of service (QoS) and network performance
The purpose of this manual is to provide technical issues;
guidance, specifications and reference material to assist in
the engineering of the ground voice communications c) network numbering;
facilities specified in Annex 10, Volume III, Part II,
Chapter 4,8 in order to meet the ground-ground communi- d) details of the signalling systems needed to support
cations requirements specified in Annex 11, Chapter 6, the defined telephone facilities;
Section 6.2.9
e) supervisory tone;
This manual is not a complete guide to the planning and
deployment of networks since there is already ample
f) security aspects; and
relevant material generally available. Rather, the manual
aims to provide general indications of the issues that are
g) migration to digital voice switching and signalling
considered to be important in the context of deploying
technology.
ground voice communications networks to support air
traffic management and to provide guidance on the
Appendix A describes a method for assigning signalling
direction in which such networks should develop.
delay to network elements. Appendix B lists all the
publications referenced in the text.
Historically, voice communications between air traffic
services (ATS) units have been carried out via direct
analogue circuits and, more recently, via switched analogue The manual incorporates, by dated or undated reference,
circuits. The economics of telecommunications service provisions from other publications necessary to assist
provision have now altered in favour of digital technol- administrations with the deployment of digital switched
ogies. This manual, therefore, provides recommendations network technologies as the means of providing the ground
and technical guidance for administrations wishing to voice facilities needed for air traffic management purposes.
migrate their international ATS communications to make
use of switched digital networking. It is intended that the manual be kept up to date. Future
editions will most likely be improved on the basis of
This manual contains two chapters: Chapter 1, dealing with experience gained and of comments and suggestions
Operational Requirements and Chapter 2, dealing with received from users of this manual. Therefore, readers are
Engineering Requirements. Chapter 1 describes in detail invited to give their views, comments and suggestions on
the facilities specified in Annex 10, Volume III, Part II, this edition. These should be directed to the Secretary
Chapter 4,8 and explains how these are used to meet the General of ICAO.
(iii)
TABLE OF CONTENTS
Page Page
Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (vii) 2.2 QoS and network performance . . . . . . . . . . . . . 2-7

2.2.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-7
Acronyms/Abbreviations . . . . . . . . . . . . . . . . . . . . . . (ix) 2.2.2 Traffic engineering . . . . . . . . . . . . . . . . . . 2-7
2.2.3 Transmission planning . . . . . . . . . . . . . . . 2-7
Chapter 1. Operational requirements . . . . . . . . . 1-1 2.2.4 Allocation of signalling delay . . . . . . . . . 2-10
2.2.5 The use of satellite links . . . . . . . . . . . . . 2-10
1.1 Correlation to the requirements 2.2.6 Network synchronization . . . . . . . . . . . . . 2-10
of Annex 11, Section 6.2 . . . . . . . . . . . . . . . . . . 1-1 2.2.7 Alternate routing . . . . . . . . . . . . . . . . . . . 2-11
1.2 Implementation . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1
1.2.1 Means of provision of 2.3 Numbering. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-11
telecommunication services . . . . . . . . . . . 1-1 2.3.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-11
1.2.2 Applicability to domestic 2.3.2 Recommended numbering plan . . . . . . . . 2-11
operational requirements. . . . . . . . . . . . . . 1-2 2.3.3 Future numbering principles. . . . . . . . . . . 2-11
1.3 Basic call types (primary user ground 2.4 Digital signalling systems . . . . . . . . . . . . . . . . . 2-12
telephone facilities) . . . . . . . . . . . . . . . . . . . . . . . 1-2 2.4.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-12
1.3.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2 2.4.2 Recommended signalling system . . . . . . . 2-12
1.3.2 Instantaneous access facility. . . . . . . . . . . 1-2 2.4.3 Use of PSS1 (QSIG) in AGVNs . . . . . . . 2-12
1.3.3 Direct access facility. . . . . . . . . . . . . . . . . 1-3
1.3.4 Indirect access facility . . . . . . . . . . . . . . . 1-3
1.3.5 Performance requirements of primary 2.5 Analogue signalling systems . . . . . . . . . . . . . . . 2-14
user ground telephone facilities . . . . . . . . 1-3
2.6 Supervisory tones . . . . . . . . . . . . . . . . . . . . . . . 2-14
1.4 Supplementary facilities. . . . . . . . . . . . . . . . . . . 1-3
1.4.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-3
2.7 Network management . . . . . . . . . . . . . . . . . . . . 2-14
1.4.2 Indication of calling, called and
2.7.1 Functional areas . . . . . . . . . . . . . . . . . . . . 2-14
connected party identity . . . . . . . . . . . . . . 1-4
2.7.2 Network management standards. . . . . . . . 2-14
1.4.3 Operational implementation —
model for the display of caller identity . . 1-4
1.4.4 Indication of urgent/priority calls . . . . . . . 1-4 2.8 Security aspects . . . . . . . . . . . . . . . . . . . . . . . . . 2-15
1.4.5 Conference capabilities . . . . . . . . . . . . . . . 1-5 2.8.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-15
1.4.6 Automatic recording . . . . . . . . . . . . . . . . . 1-5 2.8.2 Physical security . . . . . . . . . . . . . . . . . . . . 2-15
2.8.3 System security . . . . . . . . . . . . . . . . . . . . . 2-16
Chapter 2. Engineering requirements . . . . . . . . . 2-1 2.8.4 Network security . . . . . . . . . . . . . . . . . . . 2-16
2.1 Network planning, implementation and

2.9 Transition arrangements . . . . . . . . . . . . . . . . . . 2-17
management. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1
2.1.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1
2.1.2 Transmission media and technologies . . . 2-1 Appendix A. Method for assigning signalling
2.1.3 Use of switching elements and delay to network elements . . . . . . . . . . . . . . . . . . . . . A-1
signalling protocols . . . . . . . . . . . . . . . . . . 2-1 A.1 General. . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-1
2.1.4 Speech compression technologies . . . . . . 2-2 A.2 General model for a basic telephone call. . A-1
2.1.5 Recommended basis of ATS ground A.3 Sources of signalling delay . . . . . . . . . . . . . A-3
voice networks (AGVNs) . . . . . . . . . . . . . 2-2 A.4 Integration of signalling to give a
2.1.6 Planning guidelines . . . . . . . . . . . . . . . . . 2-4 specific delay model . . . . . . . . . . . . . . . . . . A-4
(v)
Manual on Air Traffic Services (ATS)
(vi) Ground-Ground Voice Switching and Signalling
Page Page Page
Appendix B. Referenced publications and

information sources . . . . . . . . . . . . . . . . . . . . . . . . . . B-1
B.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-1
B.2 References. . . . . . . . . . . . . . . . . . . . . . . . . . B-1
B.3 Equivalences for PSS1 (QSIG)
specifications. . . . . . . . . . . . . . . . . . . . . . . . B-3
B.4 Web sites of standardization bodies. . . . . . B-3
GLOSSARY
Address. A string or combination of decimal digits, depending on the capabilities of the signalling system
symbols and additional information that identifies the with which interworking occurs.
specific termination points of a connection in a
network. Equipment impairment factor (Ie). A number allocated to
a network element, in units of “eif”, that indicates the
Administration. An organization responsible for providing anticipated incremental level of impairment that would
international air traffic management communication result when this element is inserted into a connection.
services.
Incoming gateway VCS. A gateway VCS that routes an
Air traffic services (ATS) ground voice network. A
incoming call from a route employing one signalling
telecommunications network providing telecommuni-
system on to an inter-VCS link employing a different
cation services to support operational air traffic
system.
management processes.
Air traffic services unit. A generic term meaning Inter-VCS link (transmission link). A link between two
variously, air traffic control unit, flight information VCSs comprising the totality of the signalling transfer
centre or air traffic services reporting office. means (i.e. a signalling channel) and the user in-
formation transfer (i.e. speech channels) means.
Bilateral agreement. An agreement between two ad-
ministrations to provide a certain capability or act in a Key. A device that not only includes a key but also any
particular manner. means of activating a facility, including push-buttons,
computer mouse, screen icons and touch-sensitive
Circuit. A combination of two transmission channels panels, etc.
permitting bidirectional transmission of signals between
two points to support a single communication. Name. A string of characters used for the name
identification of a party involved in a call.
Class of service. The privileges, priorities, limitations and
restrictions determining a party’s ability to initiate Network. A set of equipment (terminal equipment,
outgoing calls, to receive incoming calls and to switching equipment, call-processing equipment, etc.)
invoke/use relevant supplementary facilities. located at geographically dispersed locations and
interconnected via transmission links to provide tele-
Controller working position (CWP). In the context of this communication services to a defined group of users.
manual, one particular type of terminal equipment used
specifically for the purpose of performing operational
Network performance. The ability of a network or network
duties of air traffic management.
portion to provide the functions related to com-
munications between users.
Digital leased line. A point-to-point digital circuit rented
by an administration from a public network operator or
Number. An address restricted to containing numerical
telecommunications service provider for exclusive use
values, as defined by a numbering plan.
by the administration for the provision of air traffic
management communication services.
Originating VCS. In the context of a call, the VCS to
E1/T1. The primary rates of the plesiochronous digital which the A-party’s terminal equipment is attached.
hierarchy. E1 refers to the 2 048 kbit/s primary rate. T1
refers to the 1 544 kbit/s primary rate. Outgoing gateway VCS. A gateway VCS that routes an
incoming call from an inter-VCS link employing one
End VCS. In the context of a particular call, an originating signalling system on to a route employing another
or terminating VCS. It can also be a gateway VCS; signalling system.
(vii)
(viii) Ground-Ground Voice Switching and Signalling
Party (A-party, B-party and C-party). The users involved, user to one or more services of that network. A
sequentially, in a telephone call as follows: telephone is a typical example of terminal equipment.
A-party: the user who initiates a call (the calling Terminating VCS. The VCS to which the B-party’s
party); terminal equipment is attached.
B-party: the user who first receives the call (the Toll quality (PSTN quality). The average quality of long-
called party and/or the connected party); distance public switched telephone network con-
and nections, i.e. good intelligibility, good speaker identifi-
cation, naturalness, only minor disturbing impairments.
C-party: the user who joins an established call, e.g.
in a conference. Transit VCS. In the context of a call, any VCS through
which the call passes, excluding the originating VCS,
Private integrated services network (PISN). An integrated the terminating VCS, an incoming gateway VCS and an
services digital network providing services to a specific outgoing gateway VCS.
set of users (different from a public network, which
provides services to the general public). User (controller). An air traffic controller or other person
using, via terminal equipment, the services provided by
Private integrated services network exchange (PINX). A ATS ground voice networks to undertake the oper-
PISN nodal entity that provides automatic connection ational duties of air traffic management.
handling functions used for the provision of tele-
communication services. A private automatic branch Note.— This definition does not include personnel
exchange (PABX) is a typical example of a PINX. In carrying out administrative functions. Such persons are
the context of this manual, a voice communication considered to be normal telephone users without
system (VCS) is one particular type of PINX. In some special requirements arising from the function of air
countries a VCS is known as a voice switching and traffic management.
control system (VSCS).
Virtual private network (VPN). A private network based
Private signalling system no. 1 (PSS1). An internationally upon the use of shared public switched network
standardized digital signalling protocol, used between infrastructures where those shared infrastructures take
the nodes of a PISN. PSS1 is also known as QSIG. This the place of dedicated transmission links and transit
document refers to it as PSS1 (QSIG). switching nodes in a private network and incorporate
mechanisms to prevent users in one VPN from
Quality of service. The degree of satisfaction experienced interacting with the users of a different VPN.
by the user of a service, brought about by the collective
effect of the mechanisms employed (support perform- Voice communication system (VCS). See the definition of
ance, operability performance, serviceability per- private integrated services network exchange (PINX)
formance, security performance) to ensure adequate above.
performance of that service.
Voice switching and control system (VSCS). See the
Terminal equipment. An item of equipment attached to a definition of private integrated services network
telecommunications network to provide access for a exchange (PINX) above.
ACRONYMS/ABBREVIATIONS
ACSE Association control service element Ie Equipment impairment factor

ADPCM Adaptive differential pulse code IEC International Electrotechnical
modulation Commission
AGVN ATS ground voice network IP Internetworking protocol
AMR Adaptive multi-rate ISDN Integrated services digital network
ANF Additional network feature ISO International Organization for
APDU Application protocol data unit Standardization
ATM Air traffic management ITU International Telecommunication Union
ATS Air traffic services ITU-T International Telecommunication Union
CC Call control — Telecommunication Standardization
CCA Call control agent Sector
CPE Customer premises equipment LAPD Link access procedure D
CS-ACELP Conjugate-structure LD-CELP Low-delay code excited linear
algebraic-code-excited linear-prediction prediction
CWP Controller working position ms Millisecond
DA Direct access MSC Message sequence chart
DS0 Digital signal level 0 ORIG Originating
DSE Dialogue service element PABX Private automatic branch exchange
ECAC European Civil Aviation Conference PCM Pulse-code modulation
EG ETSI Guide PIN Personal identification number
eif Equipment impairment factor (unit of PSS1 Private signalling system no. 1
measure for) PSTN Public switched telephone network
EN Norme Européenne (European QDU Quantization distortion unit
standard) QoS Quality of service
ES ETSI standard QSIG Signalling at the “Q” reference point
ETR ETSI technical report SS Supplementary service
ETSI European Telecommunications TERM Terminating
Standards Institute TIA Telecommunications Industry
FIC Flight information centre Association (USA)
GFP Generic functional protocol TR Technical report
GoS Grade of service VAD Voice activity detection
GSM Global system for mobile VCS Voice communication system
communications VSCS Voice switching and control system
ICD Interface control document
(ix)
Chapter 1
OPERATIONAL REQUIREMENTS
1.1 CORRELATION TO THE operational requirements, including those specified in

REQUIREMENTS OF ANNEX 11, Annex 11.9 The three basic call types are the following:
SECTION 6.2
a) instantaneous access;
1.1.1 Air traffic services (ATS) requirements for
communications within and between flight information b) direct access; and
regions are set out in Section 6.2 (“Aeronautical fixed
service, ground-ground communications”) of Annex 11.9 c) indirect access.
This section identifies “communications by direct speech
alone . . ., whereby for the purpose of transfer of radar Annex 108 also recommends the provision of sup-
control, the communications can be established in- plementary facilities in order to meet the requirements of
stantaneously and for other purposes the communications Annex 11,9 namely:
can normally be established within fifteen seconds”, as
being the two major components making up the ATS a) means of indicating the calling/called party identity;
ground telephone facilities.
b) means of initiating urgent/priority calls; and
1.1.2 The requirements make it apparent that except
for the purpose of transferring radar control where c) conference capabilities.
“instantaneous” establishment of communications is re-
quired, a switched network in which communications could Sections 1.3 and 1.4, respectively, describe these basic call
normally be established within fifteen seconds is acceptable types and supplementary facilities in detail.
for ground telephone purposes. In the case of transferring
radar control, Annex 119 describes “instantaneous” as
meaning “immediate access between controllers”, i.e. no
appreciable delay between one controller activating a key 1.2 IMPLEMENTATION
and another controller receiving an indication of the call. In
the past this has meant the use of dedicated circuits
between controllers. Today, however, modern switched 1.2.1 Means of provision of
digital networks are sufficiently fast and have the capacity telecommunication services
and the functionality sufficient to meet the “instantaneous”
requirement. Implementation of a switched network should be under-
taken in an evolutionary manner. Where two or more
1.1.3 In addition to the operational requirements administrations are involved, adequate coordination should
contained in Annex 11,9 this manual takes into account be established through the appropriate ICAO regional
current, well-established ground-ground telephone facilities offices. Until all administrations concerned have a high
(such as “hotlines”) as well as the performance criteria level of confidence in a newly implemented network, an
specified for modern air traffic management (ATM) adequate contingency plan should be in place.
procedures, such as the reduced radar separation minima Administrations should agree on a target date for full
applicable in the European Civil Aviation Conference implementation of a proposed network. It should be noted,
(ECAC)7 area. In Annex 10,Volume III, Part II, Chapter 4, 8 however, that even partial implementation of a new digital
three different communication mechanisms, or basic call network will represent an improvement over the existing
types, are stipulated to fulfil the broad spectrum of known arrangements in many cases.
1-1
1-2 Ground-Ground Voice Switching and Signalling
1.2.2 Applicability to domestic Note.— Some administrations may prefer that it be

operational requirements necessary for the A-party to sustain the key operation for
the duration of the call.
Switched networks can also be used to meet domestic
operational requirements, and the concepts and techniques 1.3.2.2 Dial tone and outgoing signalling tones are not
outlined in this manual, although intended to meet given to the A-party. Ringing/ringback tone is not given to
international requirements, may also be applicable to the A-party. Number unobtainable/reorder tone is given to
domestic networks. Adequate security should be in place to the A-party if the call fails for any reason, including any
prevent misuse of the international network by un- busy conditions encountered.
authorized users.
1.3.2.3 The arrival of the call from the A-party to the
B-party causes, simultaneously, the events detailed in a)
to d):
1.3 BASIC CALL TYPES (PRIMARY USER
GROUND TELEPHONE FACILITIES)
a) The A-party identity is indicated to the B-party
either by association with a key assigned and fixed
semi-permanently in the B-party VCS or by means
1.3.1 General of a dynamic display. Due to the usually urgent
nature of instantaneous access calls, any visual
1.3.1.1 As has already been noted in Section 1.1, (and/or audible) alerts should be distinctive from
several different basic call types (telephone facilities) are other types of calls.
needed to meet the operational ground communications
requirements of ATM. These different facilities are
b) An audible alert is generated at the B-party VCS in
generically referred to as the “primary user ground
accordance with the following options:
telephone facilities”. They are the facilities used to conduct
routine and urgent communications between ATS units.
1) no audible alert;
1.3.1.2 Each of the following sections describes, from 2) an alert of fixed duration; or
the user point of view, the operation of one of the primary
ground telephone facilities. To assist the description, the 3) a continuous alert requiring a silencing action
term “user” is used to refer to an air traffic controller or by the B-party.
other operational person using the facility concerned. To
distinguish users who have different roles in the c) The B-party VCS automatically accepts/answers
communication, the terms A-party, B-party and C-party are the incoming call without any intervention required
used. The term “VCS” (voice communication system) is by the user. This occurs regardless of the B-party
used to refer to the equipment providing the facility being engaged on any other type of call. Thus,
concerned. This manual makes no assumptions about what B-party busy is an abnormal situation and should
form a VCS will take, except to say that a VCS normally result in number unobtainable/reorder tone being
has attached to it some kind of terminal equipment (e.g. a given to the A-party (see 1.3.2.2). At this stage the
controller working position (CWP)) through which a user speech channel from the A-party to the B-party is
interacts with the system. established. How speech from the A-party is
managed (i.e. conferenced with other speech at the
Note.— The supervisory tones referred to in 1.3.2, 1.3.3 B-party working position, switched to a
and 1.3.4 are defined in Section 2.6. loudspeaker or to a split-headset) is a matter for the
B-party administration to decide.
1.3.2 Instantaneous access facility d) By bilateral agreement, the establishment of the call
as detailed in c) may also result in the A-party
1.3.2.1 The operation of a single key by the A-party is having some monitoring facilities of the B-party’s
all that is required to initiate a call to the B-party. The working position, including ground-ground and
B-party address is assigned and fixed semi-permanently in air-to-ground radiotelephony. This enables the
the A-party VCS. It is thus uniquely associated with a A-party to exercise discretion before passing the
particular key and each key is labelled as such. message.
Chapter 1. Operational requirements 1-3
1.3.2.4 The B-party may respond to the A-party by Note.— As part of the indirect access facility, some
activation of a key associated with the incoming call. This administrations permit communication to be established
action only enables the return speech path and is not a new with normal telephone users of the wider private telephone
instantaneous access call. The call is cleared by the A-party network and/or the public switched telephone network
only with no effect on other calls in progress at the B-party. (PSTN) as well as to users with operational responsibility,
such as other air traffic controllers.
1.3.3 Direct access facility 1.3.4.2 The design of some VCSs may be such that a
user action equivalent to “off-hook” may be needed prior to
1.3.3.1 The operation of a single key by the A-party is entering the desired B-party address. In this case, dial tone
all that is required to initiate a call to the B-party. The may be given to the A-party. Signalling tones will not
B-party address is assigned and fixed semi-permanently in normally be given. Ringing/ringback tone and busy tone
the A-party VCS. It is, thus, uniquely associated with a shall be given to the A-party, as appropriate. A suitable
particular key and each key is labelled as such. mechanism (i.e. number unobtainable/reorder tone) shall be
provided to inform the A-party, should the call fail for any
1.3.3.2 Dial tone and outgoing signalling tones are not
reason other than because the B-party is busy.
given to the A-party. Ringing/ringback tone is optional by
bilateral agreement between the A-party and B-party 1.3.4.3 The B-party is alerted to the presence of the
administrations. Busy tone shall be given, if appropriate. incoming call by audio and/or visual means, as determined
However, due to either the exclusive, one-to-one assign- by the B-party VCS. The A-party identity may be indicated
ments of the keys between the ‘A’- and ‘B’-parties or the to the B-party (e.g. by means of a dynamic display) but this
reserved capacity in the B-party dynamic display, it is could be limited to an indication of the network over which
abnormal for the A-party to encounter the B-party busy. the call was made. The B-party accepts the incoming call
This is a fundamental attribute of the direct access facility. by means of a user action equivalent to “off-hook”.
1.3.3.3 A suitable mechanism (i.e. number unobtain-
1.3.4.4 The B-party may be able to receive an
able/reorder tone) shall be provided to inform the A-party,
indication of one or more indirect access calls at the same
should the call fail for any reason other than because the
time. By using the A-party identity (if available), the
B-party is busy.
B-party can select which call to answer/handle next. At the
1.3.3.4 The B-party is alerted to the presence of the end of a call, either the A-party or the B-party can
incoming call by audio and/or visual means, as determined deselect/clear the call by an appropriate user action
by the B-party VCS. The A-party identity is indicated to (equivalent to “on-hook”).
the B-party either by association with a key assigned and
fixed semi-permanently in the B-party VCS or by means of
a dynamic display. The B-party accepts the incoming call 1.3.5 Performance requirements of
by means of a single action associated with a key or primary user ground telephone
dynamic display. facilities (see Table 1-1)
1.3.3.5 Under normal conditions the B-party can

receive one or more direct access calls simultaneously and
by using the A-party identities, together with a defined 1.4 SUPPLEMENTARY FACILITIES
operational procedure or experience, the B-party will deal
with each call appropriately.
1.3.3.6 At the end of a call, either the A-party or the 1.4.1 General
B-party shall be required to deselect/clear the call. Some
implementations may require both parties to deselect/clear 1.4.1.1 Some supplementary facilities are needed to
the call. support the primary user ground telephone facilities (see
Section 1.1). Each of the following sections describes, from
the user point of view, the operation of one of these
1.3.4 Indirect access facility supplementary facilities.
1.3.4.1 By entering the full address of the B-party Note.— Many signalling protocols are not capable of
(e.g. by dialling it on a keypad or by entering a short supporting all of the supplementary facilities described
code/speed call number), the A-party can initiate a call to below. However, they can all be supported by the PSS1
the required B-party. (QSIG) signalling protocol (see Section 2.4.2).
Table 1-1. Performance requirements

for primary user ground telephone facilities
Facility Establishment of communication should occur:
Instantaneous access Within 1 second or less for 99% of call attempts. The interval of 1 second is the delay
between the A-party initiating the call and the A-party to B-party speech path being
established.
Direct access Within 2 seconds or less for 99% of call attempts. The interval of 2 seconds is the delay
between the A-party initiating the call and the B-party being alerted to the presence of the
call.
Indirect accesss Within 15 seconds or less for 99% of call attempts. The interval of 15 seconds is the delay
between the end of dialling by the A-party and the B-party being alerted to the presence of
the call.
1.4.2 Indication of calling, 1.4.3.2 In networks where switching and signalling

called and connected party identity are used, a VCS can use the calling and called party
numbers, transmitted in the inter-VCS signalling protocol,
1.4.2.1 Indications of calling, called and connected to ring a unique direct access key at the CWP. A common
party identity allow the users involved in a call to receive implementation is for the called party number to address a
information concerning the identity of the other users in the particular CWP and the calling party number to address a
call. The identity of a user normally consists of that user’s predefined key at that CWP. Where no key has been defined
telephone number and, optionally, a name. at the addressed position, a call queue or indirect access
indicator lamp can be used to indicate the presence of an
1.4.2.2 When a call is established (either directly via incoming call. An alphanumeric display at the CWP can
a direct access key or indirectly via the telephone keypad), provide either the calling party number or the actual
the identity of the A-party should be indicated at the identity through use of a look-up table within the VCS.
B-party VCS.
1.4.2.3 When a call is alerting, the identity of the

B-party should be indicated at the A-party VCS. 1.4.4 Indication of urgent/priority calls
1.4.2.4 When a call is answered, the identity of the 1.4.4.1 The priority facility is a means of attaching an
B-party should be indicated at the A-party VCS. indicator (or flag) to a telephone call to show that it is
“urgent” as opposed to “routine”. It is intended for use
when it is necessary to make an urgent call concerning the
1.4.3 Operational implementation — safety of aircraft (i.e. an emergency situation) and to
model for the display of caller identity enable, if necessary, the interruption of less urgent calls in
progress at the time. The use of priority is generally agreed
1.4.3.1 In the most basic networks (i.e. those without by bilateral agreement between administrations. The
inter-VCS switching and signalling), the identity of the ultimate decision and responsibility as to whether a call has
caller will be implicit through hardwiring of a particular priority rests with the A-party in accordance with local
key at the CWP to a transmission circuit (line port) of the operational procedures. There are three ways in which
VCS. No addressing information is passed across the line priority can be set:
interface. If the line port detects an incoming call, the key
illuminates and an audible warning is given. The key is a) manually, before the call is made: before making
labelled in some way to indicate the identity of the the call, a priority key is operated on the VCS to
correspondent on the far end of the line. In this instance, set the priority of the call to “urgent”. This method
there will be a one-to-one mapping between the VCS line is used when the call is already known to be
ports and the operator keys. urgent;
Chapter 1. Operational requirements 1-5
b) during call set-up: at any time during call set-up, Add-on conference
the operation of a priority key will change the
priority of the call from “routine” to “urgent”. This 1.4.5.2 The (telephone) add-on conference facility is a
method would be used as a reaction to an urgent means of allowing three or more parties to speak together.
operational situation that has arisen, including a When a call has already been established between two
delay in answering at the far end or on receipt of a parties (A and B), either one may choose to include another
busy tone; party in the call. For example, if the B-party decides to set
up a conference, the B-party first places the A-party on
c) automatic setting of priority: the priority of all calls hold (if required) and then makes a new call to the C-party.
from a particular CWP or set of keys is The B-party can then choose to set up the conference or to
pre-programmed in the VCS to be “urgent”. This speak with the A-party or C-party individually and dis-
method can be used for operational reasons when creetly. The B-party can withdraw from the conference,
calls made from a particular CWP or key are always leaving the other parties in conversation. Alternatively, it
to be treated as urgent. An example of this is to use may be preferred that when the B-party withdraws from the
the priority facility to distinguish between conference, calls to all the parties involved are cleared.
instantaneous access and direct access calls.
1.4.5.3 Conferencing between internal users (i.e. on
1.4.4.2 Equally, the B-party VCS should react to an the same VCS) and external users (i.e. via external lines) is
incoming priority call in the following manner: available with most modern VCSs. Most systems allow
many parties to be included in a single conference.
a) provide some means of indicating that a priority Ultimately, however, it is the responsibility of the ad-
call has been received (e.g. special visual and/or ministration to determine the maximum number of parties
audible indications); and that is acceptable and manageable.
b) allow the priority call to intrude in a call already

established (see Section 1.4.5), preceded by a Conferences resulting from call intrusion
warning indication.
1.4.5.4 Conferences can also arise from the use of the
1.4.4.3 If a priority call cannot proceed due to intrusion facility. In the ATS context, a conference can be
congestion (all available circuits, links or channels are caused by the arrival of an urgent (priority) call (see
busy), the priority call should interrupt an established Section 1.4.4) when the B-party is already engaged (busy)
routine call (should one exist), thus allowing the priority with another call. Subject to the intrusion being permitted
call to proceed. Before the established routine call is by the B-party VCS (due to authorization by the local
interrupted, all parties engaged in that call should receive administration), a conference will be established usually
an interrupt warning tone. after a warning indication is provided to the parties being
intruded upon.
1.4.5 Conference capabilities 1.4.6 Automatic recording
1.4.5.1 There are two kinds of telephone conference: As with all operational telephone services, and in
add-on conference and conferences arising from call accordance with Annex 11, Chapter 6,9 a VCS should
intrusion (e.g. as a result of using the priority facility). Each provide a facility whereby all telephone conversations,
of these is described below. The use of conference is together with call originating and management data, are
generally by bilateral agreement between administrations. automatically recorded.
Chapter 2
ENGINEERING REQUIREMENTS
2.1 NETWORK PLANNING, 2.1.2.2 Many techniques have been developed for
IMPLEMENTATION AND MANAGEMENT processing the information to be sent over these
transmission media in order to maximize usage of the
available bandwidth. Most techniques can be categorized
2.1.1 General into generalized classes, namely:
2.1.1.1 The main task faced by planners of ATS a) frequency division multiplexing;
communications systems is to implement the primary user
ground telephone facilities (basic call types) and sup- b) time division multiplexing;
plementary facilities required by Annex 10, Volume III,8 in
a cost-effective, reliable and flexible manner and to c) statistical multiplexing;
subsequently manage those facilities. To do this in a
manner that ensures interoperability and appropriate levels d) data compression techniques;
of service across national/regional boundaries demands the
use of common technologies and adherence to a set of e) connection-oriented, packet-based techniques; and
general planning rules.
f) connectionless, packet-based techniques.
2.1.1.2 This chapter describes the possible technical
means to implement the facilities and functions described 2.1.2.3 In practice, most real networks employ
earlier. It identifies the switching technologies, trans- combinations of transmission media and several processing
mission systems and signalling protocols that should be mechanisms to create economical and efficient networks
used. Emphasis is given to digital technologies and the that offer high levels of network performance and good
planning rules necessary to avoid degradation of speech quality of service (QoS).
quality, to ensure signalling interoperability and to enable
optimized call routing (e.g. to avoid congested links, to
reduce costs). ATS administrations should also give 2.1.3 Use of switching elements and
consideration to the management of the facilities. signalling protocols
2.1.3.1 To avoid having to fully interconnect each

2.1.2 Transmission media possible source of information with each possible des-
and technologies tination, switching elements also have to be incorporated in
the network design. Switching elements use either circuit-
2.1.2.1 Developments in electronics, optical trans- oriented switching or packet-/cell-oriented switching
mission techniques, and radio communications have mechanisms, the choice being determined to a large extent
expanded the range of transmission media to include: by the transmission technologies employed in the network.
a) copper cables; 2.1.3.2 The use of switching elements in a network

implies the capability to convey instructions (i.e. signalling)
b) optical fibre; from the calling party to a switching element and between
switching elements to permit calls to be routed via shared
c) satellite transmission; and transmission resources. The capability of the signalling
protocol employed influences the way in which the
d) microwave links. telephone facilities described in Chapter 1 can be provided.
2-1
2.1.4 Speech compression technologies a) AQVNS should use transmission links and core-
switching capabilities based on 64 kbit/s channels
2.1.4.1 Speech signals contain much redundant of the digital transmission hierarchy defined in
information due to the repetitive waveforms and the periods ITU-T Recommendation G.702;30
of silence during which no data need be transmitted. To
remove these inefficiencies, many algorithms (e.g. LD-CELP, Note.— In some regions of the world, these
conjugate structure algebraic-code-excited linear prediction channels are referred to as “DS0” (digital signal
(CS-ACELP), global system for mobile communications level 0) channels.
(GSM) full-rate, GSM adaptive multi-rate (AMR)) have been
developed to compress speech to a much lower bit rate. In all
b) Speech should be encoded according to the rules for
cases, the objective has been to reduce the required
64 kbit/s A-law or µ-law pulse-code modulation
bandwidth without any loss of speech quality. Many of these
(PCM) encoding, with transcoding as appropriate at
sophisticated algorithms introduce additional delay due to the
national/regional boundaries, as specified in ITU-T
large number of speech samples required to process the
Recommendation G.711;32
signal correctly.
2.1.4.2 As well as introducing additional delay, some c) Signalling within AGVNs should be closely based
of these algorithms affect the speech quality; therefore, care on the private signalling system number 1 (PSS1),
has to be taken over the choice of algorithm. For example, also known as “QSIG”, and defined in ISO/IEC
it is not recommended to choose a very low bit rate 1157212 and related ISO/IEC international stan-
algorithm in order to maximize bandwidth usage if the dards. The PSS1 (QSIG) signalling system is
result is only radio-quality speech when “toll- or phone- intended for signalling between nodes of a private
quality” speech is required. If disparate coding algorithms integrated services network (PISN), i.e. between
are adopted at different points within a voice network, the PINXs. The way in which this signalling system
resulting system will be unusable as a network and only should be used in AGVNs is described in detail in
suitable for single hop connections. Section 2.4.2.
The use of alternative technologies is not precluded,

2.1.5 Recommended basis of provided that the required level of performance is achieved.
ATS ground voice networks (AGVNs)
Use of sub-multiplexing and low

bit rate speech encoding schemes
Replacement of analogue technologies
with digital technologies 2.1.5.3 Where a bilateral agreement has been reached
between the administrations, AGVNs may make use of
2.1.5.1 Many administrations currently employ sub-multiplexing and low bit rate speech encoding schemes
analogue technologies and signalling protocols for their in the following circumstances:
AGVNs. While existing implementations of these tech-
nologies are still acceptable for use in ATS operations, the a) where no more than two “hops” (transmission links
technologies are now largely obsolete. New installations in series) are envisaged between the originating
should therefore be based on digital technologies. VCS and the terminating VCS, i.e. one transit VCS;
or
Core technology b) where the use of a satellite is required, e.g. on a link

of transoceanic or transcontinental length.
2.1.5.2 An AGVN will be constructed from
transmission links connected by switching nodes (i.e. 2.1.5.4 In general, a single stage of speech
VCS). Each element influences the QoS provided by the compression and decompression using one of today’s low
network. The greater the number of conversions and the bit rate algorithms has little noticeable effect on the speech
amount of processing introduced into the speech path, the quality perceived by the end-user. However, where several
higher the degradation of the original speech signal and the stages of compression and decompression occur in series
throughput delay. The following general guidelines are (as might be the case in a switched tandem connection), the
recommended for the development of a new AGVN: perceived quality quickly degrades to an unacceptable level.
Chapter 2. Engineering requirements 2-3
Therefore, the combined use of various techniques (transit 2.1.5.7 Network planners should be aware of the
switching, multiple stages of compression, satellite links, impairments introduced by low bit-rate encoders or other
etc.) has to be carefully planned. Furthermore, network voice signal processing and should take these into account
planners are advised to be aware of the potential future use in order that the desired level of speech quality may be met
of low bit rate vocoders in air-ground communications and in the network.
the consequent impact on speech quality that could arise
from the use of these in tandem with low bit rate speech
encoding schemes in ground communications. Subject to Equipment configurations
these considerations, the following arrangements can be
used: 2.1.5.8 There are many different ways in which
switching equipment (VCS), multiplexing equipment,
a) 16 kbit/s sub-multiplexing in conjunction with compression equipment and transmission links can be
LD-CELP speech encoding (ITU-T Recommen- combined to construct an AGVN.
dation G.728);33 and
Note.— Various “scenarios” are described in detail in
b) 8 kbit/s sub-multiplexing in conjunction with a technical report (TR) available from ISO/IEC, T 14475.22
CS-ACELP speech encoding (ITU-T Recommen- This TR is also available from ECMA (an international,
dation G.729)34 speech encoding. Europe-based industry association for standardizing
information and communication systems) as TR/76.2
2.1.5.5 These arrangements are defined in
international standards ISO/IE 1731020 and ISO/IEC 2.1.5.9 In determining the configuration to use,
17311,21 respectively. Compression and decompression network planners have to consider many factors, including:
only take place once. Transit modes complying with the
requirements of these international standards are required to a) whether the transmission links between VCSs are to
switch the compressed bit stream without decompressing be wholly dedicated to voice or whether they will
and recompressing it. The characteristics of the two speech be links shared with other applications (e.g.
encoding algorithms used are shown in Table 2-1, together aeronautical data applications);
with those of the full 64 kbit/s PCM recommended in
Section 2.1.5.2. b) the anticipated level of traffic between VCSs and,
hence, the bandwidth required of the transmission
2.1.5.6 The influence of low bit rate encoding links. For instance, are several 64 kbit/s lines
algorithms on speech transmission quality can be evaluated sufficient or is a wider bandwidth transmission
by associating each type of algorithm with a specific Ie. structure containing many channels (e.g. E1/T1)
The influence of several low bit rate encoding algorithms in necessary?;
tandem is characterized by the sum of the individual
impairment factors for the algorithms in the chain. c) the cost of transmission links and whether it is
Table 2-1 shows the Ie associated with different encoding advantageous to use sub-multiplexing and speech
algorithms. compression techniques;
Table 2-1. Characteristics of speech encoding algorithms
Delay (ms)
Encoding algorithm Rate (kbit/s) (Note 1) Ie
PCM (G.711) 64 0.75 0

LD-CELP (G.728) 16 2 7 (Note 2)
CS-ACELP (G.729) 8 35 10
CS-ACELP (G.729 Annex A and VAD) 8 35 11
Note 1.— All delay values are one-way.

Note 2.— This Ie is the same as for 32 kbit/s adaptive differential pulse code modulation (ADPCM).
d) the need for any overflow arrangements to cope with guidelines should be applied to the design of AGVNs in
traffic overload situations (e.g. routing via alternative order to provision networks that are reliable, of high quality
transmission links, routing via PSTN); and and easy to maintain. These rules are listed below and
elaborated on in subsequent sections.
e) the need for redundancy in transmission links or
other back-up/standby arrangements (e.g. PSTN). a) The configuration of the network should be stable.
That is to say, the network must be able to handle
2.1.5.10 Various configurations are possible and they an increasing volume of traffic and changes in the
are not necessarily mutually exclusive. More than one route pattern without major re-engineering.
configuration can be used in any given network. The matter
can be rendered more complex by other factors, such as: b) Networks should normally be based on the use of
dedicated digital circuits. These should be
a) whether the interface characteristics of the VCS commensurate in number with the operational
match the interface characteristics of the requirements for grade of service (GoS), reliability
transmission link; and and availability. As an alternative to the use of
leased lines, the use of virtual private network
b) whether a sub-multiplexing and compression (VPN) services may be considered.
capability is integrated with a VCS or performed
separately from it. c) VCSs should normally be linked in a polygonal
configuration so that traffic has a choice of routes
2.1.5.11 When the interface characteristics of the by which the destination can be reached. VCSs
VCS match the interface characteristics of the transmission should have the capability for automatically routing
link, the VCS can be directly connected to the transmission a call via an alternative route if the primary route is
link. This configuration is illustrated in Figure 2-1. Points congested or unavailable.
A and B are referred to in Table 2-3.
d) Special care should be taken to maintain a uniform
2.1.5.12 When the interface characteristics of the quality level throughout the length of transmission
VCS do not match the interface characteristics of the links, with special attention being paid to inter-
transmission link, the VCS must be indirectly connected to connections between telecommunications service
the transmission link via another device such as a providers and ATM facilities.
multiplexer. This configuration is illustrated in Figure 2-2.
Points A and B are referred to in Table 2-3. If a e) The overall propagation delay for any specific
sub-multiplexing and compression capability is required, connection through the network should be kept to a
this function can either be performed as an integral function minimum.
of the VCS, or it can be performed by separate equipment,
such as a multiplexer. Table 2-2 shows the possible f) The through-switching (tandem) function should
configurations based on these aspects. normally be limited such that no more than four
VCSs (i.e. the originating VCS, the terminating
2.1.5.13 When sub-multiplexing and compression are VCS and no more than two transit VCSs) may be
performed separately from the VCS (i.e. as part of an connected in tandem for any through-connection.
indirectly connected configuration), restrictions have to be
applied to the network topology to prevent excessive g) When planning for the use of satellite channels,
degradation of speech quality. Table 2-3 gives some care should be taken to maintain the propagation
examples of these configurations and how they determine time within the limits specified for the network as a
the possibilities at each of the interface points A and B in whole.
Figures 2-1 and 2-2. The table identifies the relevant
ISO/IEC international standards and ITU-T recom- h) It should be possible to assign a specific “class of
mendations associated with each configuration. service” to each network resource (e.g. trunk lines,
terminal equipment).
2.1.6 Planning guidelines i) Networks should not be permitted to have a single

point of failure that would impede critical ATS
2.1.6.1 In addition to the general guidelines for core operations or propagate primary failures to end
technology given above, a number of other planning systems.
Table 2-2. Possible configurations based on considerations of

interface matching and compression capability
Interface characteristics Sub-multiplexing and Sub-multiplexing and

of the VCS compression not required compression required
Match interface characteristics of the Directly connected Directly connected

transmission link
No compression capability needed Compression capability integral with
VCS
Do not match interface characteristics Indirectly connected via a multiplexer Indirectly connected via a multiplexer
of the transmission link
No compression capability needed Compression capability integral with
multiplexer
A B
VCS Transmission link
Figure 2-1. Directly connected configuration
A B
VCS Multiplexer Transmission link
Figure 2-2. Indirectly connected configuration

Table 2-3. Example configurations
Directly Indirectly
Directly connected, Indirectly connected,
connected, with integral connected, with separate
no compression compression no compression compression
Bit rate of voice channels
– at point A 64 kbit/s 16 kbit/s 64 kbit/s 64 kbit/s

– at point B 64 kbit/s 16 kbit/s 64 kbit/s 16 kbit/s
Speech encoding
– at point A A/µ-law PCM 16 kbit/s LD-CELP A/µ-law PCM A/µ-law PCM

– at point B A/µ-law PCM 16 kbit/s LD-CELP A/µ-law PCM 16 kbit/s LD-CELP
Decompression required
for tandem switching No No (Note 2) No Yes
Interface at point A 6 kbit/s G.703 64 kbit/s G.703 Fractional E1/T1 Fractional E1/T1
or E1/T1
(Note 1)
Interface at point B 64 kbit/s or E1/T1 64 kbit/s G.703 E1 or T1 64 kbit/s G.703

(type of transmission link) digital leased lines digital leased line digital leased line digital leased line
(Note 1) (Note 3)
Bit rate of signalling

channel
– at point A 64 kbit/s 16 kbit/s 64 kbit/s 64 kbit/s

– at point B 64 kbit/s 16 kbit/s 64 kbit/s 16 kbit/s
Relevant specifications ITU-T Rec. G.703 ITU-T Rec. G.703 ITU-T Rec. G.703 ITU-T Rec. G.703
ITU-T Rec. G.711 ITU-T Rec. G.728 ITU-T Rec. G.711 ITU-T Rec. G.711
ISO/IE 11474 ISO/IEC 17310 ISO/IEC 11474 ITU-T Rec. G.728
Note 1.— Several possibilities are available based on the expected level of traffic. It is possible to use a single 64 kbit/s
digital leased line, several 64 kbit/s digital leased lines or an E1 or T1 digital leased line.
Note 2.— No decompression is required because the compression and sub-multiplexing capability is integral to the
VCS. In this case the VCS is able to determine whether or not it should perform an end VCS role or a transit VCS role
for each particular call. For calls for which it acts as a transit VCS, the compressed bit stream is switched transparently
from the input port to the output port without the need for decompression and recompression. The compressed bit stream
is rate-adapted to, and switched at, the normal channel rate of the VCS.
Note 3.— Only a fraction of the channels available in this type of leased line are allocated for use as voice channels
between VCSs. The remaining channels are allocated to other applications.
j) Network availability is dependent upon the 2.2.1.3 As a general principle, ATS networks have to
availability values of individual circuits which, be carefully planned to take account of the expected levels
typically, could be in the range of 98 per cent to of telephone traffic while at the same time maintaining an
99.5 per cent (although there may be significant acceptable QoS and minimizing network operating costs.
variations from these values in some areas). The Specifying the appropriate QoS and delivering that level of
overall target availability of an AGVN should be service is the responsibility of the network planner. QoS
99.98 per cent or better if the prevailing influences the route that voice traffic will take over a
circumstances permit. network. Routing strategies should be specified so as to
minimize impairments to speech quality and cost. Other
k) When planning an AGVN, network planners should determinants, such as priority and security levels, should
give consideration to how the network is to be also be considered.
managed on a day-to-day basis.
2.1.6.2 These guidelines are in addition to any tele- 2.2.2 Traffic engineering
communication regulatory requirements applicable to
private telephone networks in the State of operation. Normal methods of traffic calculation for telephone networks
can be used for the design of AGVNs. It should be noted,
however, that telephone calls used for the purpose of ATM
are normally of a shorter duration than ordinary telephone
2.2 QoS AND calls. To avoid over-provisioning (and hence, excessive cost)
NETWORK PERFORMANCE or under-provisioning (and hence, overload) of resources
(switching equipment and trans-mission links), networks are
usually dimensioned to a chosen GoS. GoS is defined as the
2.2.1 General probability that a call will be lost during the busy hour due
to the shortage of switching resources or transmission links.
2.2.1.1 The QoS offered by a network is the degree of Mathematically, it is equal to the proportion of calls lost. In
satisfaction experienced by the user of a service, brought the case of AGVNs, the recommended GoS value is 0.001. It
about by the collective effect of the mechanisms employed is recommended that traffic statistics be used regularly to
to ensure adequate performance of that service. In any ensure that this GoS be achieved.
telephone network there are many performance
mechanisms that have to be considered to ensure an
acceptable QoS, i.e.: 2.2.3 Transmission planning
a) rapid call set-up times (in order to meet the

performance requirements specified for the primary Recommended planning methodology
user ground telephone facilities (Section 1.3.5);
2.2.3.1 One of the most important aspects of QoS in
b) intelligibility of speech; and ATS voice networks is the intelligibility of speech. It can
affect the ability of ATS personnel to accurately judge the
c) minimum delay in the voice path.
disposition (e.g. mood, level of stress) of the other party in
a conversation. Intelligibility depends on speech quality
2.2.1.2 This section covers the following mechanisms,
which in turn is determined by bandwidth, lowest usable
each of which contribute to the overall QoS:
audio frequency, linear frequency response and other
transmission characteristics.
a) traffic engineering (GoS, availability) aspects;
b) transmission planning; 2.2.3.2 To ensure optimum speech quality, planners

must pay careful attention to the speech encoding
c) allocation of signalling delay; algorithm, other transmission parameters and the values
(limits) allowed for these parameters in given network
d) the use of satellite links; configurations. A brief introduction to some of the terms is
provided in this section.
e) network synchronization; and
2.2.3.3 Administrations are strongly recommended to
f) alternate routing. use a modern transmission planning methodology such as
that described in ITU-T Recommendations G.10826 and other networks, should result in an upper value of fifteen
G.109.27 This methodology, addressing the overall trans- for the total impairment. Planned values above this limit
mission plan aspects for telephony in a private network, has should be subject to careful analysis to ensure that the
been adopted by ITU-T as the basis for transmission operational requirements will be met. For exceptional
planning in private networks with global scope. It was configurations (e.g. satellite links to rural areas), higher
developed jointly by the European Telecommunications values (not exceeding 45) are acceptable.
Standards Institute ETSI and the America-based Telecom-
munications Industry Association (TIA). It is also described
Transmission levels
in ETSI guide EG 201 050,3 and TIA/EIA publication
TIA/EIA-TSB-32-A.47 This methodology uses a model (the
2.2.3.6 Transmission levels are not as relevant in the
E-model) for assessing the mouth-to-ear performance of
digital environment as they are in the analogue
voice telephony across networks of all types. The E-model
environment. In a wholly digital environment, signals
is described in ITU-T Recommendation G.107.25 It is also
traverse a network without any loss. However, speech is
described in ETR 250.6
still delivered to the user in analogue format. Therefore,
even in a modern digital network, transmission levels must
2.2.3.4 ITU-T Recommendation G.113 gives
still be considered where an analogue-to-digital or
guidance on the effect of impairments on end-to-end speech
digital-to-analogue conversion occurs at the edges of the
quality. The Ie method allocates a value of distortion to
network. For an interim period, some networks may contain
each network element and then allows the simple addition
a mixture of digital and analogue transmission links or
of these impairments to determine the overall impairment
switches. In these cases, transmission levels must be given
introduced by all the elements of a connection. An
more careful consideration. Today, relative rather than
expectation factor (A) is then subtracted from the overall
absolute transmission levels form the basis of network
impairment value (Itot) to generate the calculated planning
transmission planning.
impairment factor (Icpif). The expectation factor (A)
represents the “access advantage”, i.e. the effect on the
overall transmission quality (change in Icpif), as perceived Delay
by the user, caused by the ease or difficulty with which a
connection can be established. For example, satellite links 2.2.3.7 Each element (transmission link, processing
confer an advantage in that they allow the provision of element, switching element) involved in a networked
service to a remote location, and as a result, the user may telephone call introduces an amount of delay into the call.
discount the speech impairments resulting from the satellite Such delays can be attributed to the processing and
system. Typically, a satellite link will result in a value of transmission (propagation) of the speech signal and to the
twenty for A. In normal wired connections, A has a value of processing and transmission (propagation) of the signalling
zero. ITU-T Recommendation G.113 also provides rec- information required to control the call’s progress through
ommendations for upper limits for the total impairment the network. The former can be taken account of in
value with regard to different perceived levels of speech transmission planning for the network (this section). The
quality, as shown in Table 2-4. latter is not catered for by normal transmission planning
rules and is dealt with in Section 2.2.4.
2.2.3.5 For practical planning purposes, it is
recommended that normal connections within the planned 2.2.3.8 Speech delay manifests itself in two ways:
network, or between the planned network and the public or first, in the forward path where the speech from the caller
Table 2-4. Quality levels as a function of the total impairment value (Icpif)
Upper limit for total impairment value (Icpif) Speech quality
5 Very good
10 Good
20 Adequate
30 Limiting case
takes a significant time to reach the called party, and locally generated (see Figure 2-3). Most voice systems
second, through echoed signals returning from the far end. provide a local side-tone to callers to provide confidence
The greater the delay, the more difficult it becomes to that their speech is being transmitted. This is added to the
sustain normal conversation. ITU-T Recommendation echo return signal to provide a composite side-tone.
G.11429 gives guidance on the acceptable end-to-end,
one-way transmission time delays for international 2.2.3.11 Reflection of the signal can occur at any
telephone connections as follows: point in the transmission path. This can result, for example,
from a four-wire to a two-wire conversion, from cross-talk
0–150 ms = acceptable for most applications; in the electronics or from the remote user’s microphone
picking up the output of the speaker or headset. Where the
150–400 ms = acceptable, but with an impact on returned level is particularly high, a feedback loop can be
transmission quality, depending on the established with the resultant whistling and howling
degree of user interactivity required in characteristic of the feedback condition. Echo loss has to be
the transmission; and carefully controlled where there is an appreciable round trip
delay in the speech path. If the echo loss is low, the round
>400 ms = unacceptable for general planning trip delay must also be low to ensure that the caller does not
purposes. perceive an echo. Callers will only tolerate very low levels
of echo before it starts to affect the flow of conversation.
2.2.3.9 As a planning value, the maximum one-way The higher the echo loss, the longer the round trip delay
transmission time for 64 kbit/s and E1/T1 leased lines in an that can be tolerated by the user. For many long-distance
AGVN can be assumed to be (10 + 0.01 × G) ms, where G links, echo cancellers have to be installed to cancel out the
is distance in kilometres between the endpoints. This value returned echo signal if the voice path is to be usable.
is a maximum delay budget and will rarely be met in
practice. It takes account of the fact that many leased lines
are routed over fibre, which introduces slightly more delay Quantization distortion
than other transmission media. It is based on geographical
rather than routed distance between the endpoints and 2.2.3.12 Quantization distortion results from
includes an allowance for delay introduced by intermediate differences between the actual signal level and the
transmission and switching equipment. This allowance is granularity of the sampling levels in the processing of
sufficient to also cater for the initial delay encountered with analogue-to-digital and digital-to-analogue conversions. It
those speech compression algorithms where a finite period also arises in the transcoding of a digitally encoded signal
is needed to gather the initial samples of the caller’s speech from one coding scheme to another. Various speech
before onward transmission commences. Delay on satellite encoding/transcoding algorithms have differing amounts of
links is covered in Section 2.2.5. quantization distortion, expressed in terms of quantization
distortion units (QDUs), associated with them.
Echo loss and stability 2.2.3.13 For a particular speech path, the quantization
distortion associated with the processing elements forming
2.2.3.10 Echo loss and stability relate to how much of the path are totalled to give a quantization distortion figure
the originator’s speech is heard by the originator, reflected for the route. A high number of QDUs implies that the
from the remote end. In other words, it is the level of the received speech will be of poor quality. However, the QDU
user’s voice (side-tone) heard in the headset that is not totalling method, while providing a good approximation for
Calling party speech path
Calling party echo return path
Called party speech path
Figure 2-3. Echo return path

older processing techniques, does not work well with more delays of 12 ms for low earth orbit (1 400 km) (LEO)
modern coding algorithms (e.g. low bit rate encoding satellites and 260 ms for geostationary satellites (36 000
algorithms of the ITU-T G.72x series of recommendations). km). These figures are for the space segment only. For
Such algorithms contribute distortions (resulting in a planning purposes, they should be increased to 95 ms and
decrease of the perceived voice quality) that cannot be 350 ms, respectively, to include ground station delays.
readily quantified using a QDU as the measure. To deal
with these distortions, a new measure, the Ie, has been 2.2.5.2 The link delay increases the call establishment
introduced (ref. ITU-T G.113,28 EG 201 050,3 delay in addition to the degradation of voice quality. For
TIA/EIA-TSB-32-A).47 geostationary satellites, the delay may cause the call set-up
time to exceed the one second maximum allowed for
2.2.4 Allocation of signalling delay instantaneous access, especially if a large terrestrial
segment is also involved in reaching the satellite ground
2.2.4.1 The amount of signalling delay acceptable in station. It is clear that two or more satellite hops should be
AGVNs is constrained by the performance requirements of avoided in an ATS environment.
the primary user ground telephone facilities (see Section
1.3.5). The objective, therefore, should be to establish each 2.2.5.3 Geostationary satellite links should not be
call in less than one second. This is the requirement planned for normal ATS communications unless there is no
deriving from the instantaneous access facility, as it is economic alternative and the resulting degraded perform-
generally not possible to distinguish from the signalling ance is judged to be acceptable in the planned context (e.g.
whether a call results from the use of this facility or from low traffic, remote locations). This does not preclude the
the use of another facility with less stringent performance use of these links as emergency back-up facilities, where
requirements. the degraded performance is of secondary importance. Low
earth orbit satellites have delays comparable to terrestrial
2.2.4.2 Clearly, there will be circumstances where links; therefore, for network planning purposes, they may
some calls take longer than one second to establish, and be treated as normal circuits.
some circumstances where the call fails to be established at
all. For network planning purposes, therefore, the following
guidelines can be stated:
2.2.6 Network synchronization
a) 99 per cent of calls during the busiest hour of a
typical day should be established within one 2.2.6.1 Proper synchronization within a network is
second; important for the control of bit slip. Nodes in a network can
derive their timing from a variety of sources:
b) one per cent of calls during the busiest hour of a
typical day can take longer than one second to a) another node in the network;
establish; and
b) another network; or
c) of the one per cent of calls taking longer than one
second to establish, one in ten can fail to be
c) an internal clock source.
established at all.
These guidelines support both the specified instantaneous 2.2.6.2 As a general rule, network nodes should be
access performance requirement (see Section 1.3.5) and the synchronized to an external clock source provided via an
recommended GoS value of 0.001 (see Section 2.2.2). interface to a digital network. The clock source can be
derived from the AGVN or from an external source, such as
2.2.4.3 To assist with the correct provisioning of
the PSTN. In the event that no external clock source is
equipment to meet these signalling performance re-
available, a node may take its system timing from an
quirements, Appendix A describes a method that can be
internal clock source. However, network planners should be
used to plan the assignment of the signalling delay budget
aware that use of an inaccurate clock source can lead to a
to network elements.
high level of slips between the node and the rest of the
network. In particular, clock accuracies of ±1.10–6,
2.2.5 The use of satellite links commonly used in commercially available equipment,
normally would not meet an administration’s requirements.
2.2.5.1 ITU-T Recommendation G.11429 details the Further guidance on synchronization strategies is given in
various delays to be expected in the network and gives ISO/IE 11573.13
2.2.7 Alternate routing c) the area identifier may be used to identify either a
single country or a group of countries.
Alternate routing provides a path for a call when the
primary path is out of service or congested. The capability
for alternate routing may or may not exist in the network, 2.3.3 Future numbering principles
depending on its configuration. In some cases it may not be
economical to provide alternate routing capabilities, or it 2.3.3.1 The recommended numbering plan is adequate
simply may not be possible. The PSTN may be used as a to meet current and near-future network requirements. In
means of providing an alternative routing capability the longer term, the development of a numbering plan
(depending on the class of service required). allowing addressing of a greater number of users will be
required. In view of the expected future requirement for
additional digits, it is recommended that when new
equipment is to be procured, it should be capable of
2.3 NUMBERING accommodating these changes.
2.3.3.2 Worldwide, AGVNs operate collectively to

2.3.1 General
provide communication services for ATM. They are,
therefore, a federation of individual networks, usually
The numbering plan is an essential element in a switched
separated along national or regional boundaries. This
communication system. It identifies all users and provides
structure imposes several requirements on future global
necessary information to the switching equipment for the
numbering schemes as follows:
routing of the traffic. Numbering plans in general have to
balance the desire to keep the number of digits dialled for
a call to the minimum while including the possibility of a) a future global numbering scheme should be based
expansion beyond the planned capacity without changing on internationally approved standards for
the basic structure of the plan. numbering and addressing. In particular, it should
meet the requirements for PISN numbering plans
specified in ISO/IE 11571;11
2.3.2 Recommended numbering plan
b) each AGVN is self-standing but has to cooperate
The following characteristics are recommen-ded for the with other AGVNs (i.e. route calls to or accept calls
numbering plan, but it is advisable to check that there are from another AGVN). Therefore, a coordinated
no overriding regional plans in force: numbering scheme is required. Such a numbering
scheme should permit the identification and
a) numbers should consist of six digits, whereby the
addressing of individual networks as well as the
first two digits identify the area, the third and fourth
identification and addressing of entities within
digits, the ATS unit, and the fifth and sixth digits,
networks;
the CWP or correspondent within the ATS unit.
This format is illustrated in Figure 2-4;
c) it should not be necessary to use the full global
b) up to two additional digits may be added following number in order to address another entity within the
the sixth digit to allow a larger number of CWPs same network. Thus, the global numbering scheme
within an ATS unit to be uniquely addressed; and should be hierarchical;
A A C C N N
AA Area identifier
CC Unit identifier
NN CWP identifier
Figure 2-4. Format of numbers in the recommended numbering plan

d) each AGVN is independently administered. features of those communication systems (e.g. call transfer)
Therefore, a mechanism is needed to control the in a network-wide manner. In particular, PSS1 (QSIG)
allocation of blocks of numbers to the admin- allows products from different vendors to interoperate with
istrations in order to avoid numbering conflicts. one another.
ICAO should be the responsible authority;
2.4.2.2 In addition to the ability of the PSS1 (QSIG)
e) backward compatibility with the current rec- to set up a basic call, an extensive range of SSs and
ommended ICAO numbering plan is a desirable additional network features (ANFs) specify how telephony
objective; and features, such as call transfer and call forwarding, operate
in a networked multi-vendor environment. This range of
f) the maximum number length in a future global capabilities is enabled by the generic functional protocol
numbering scheme should be fixed (i.e. closed (GFP), a mechanism that supports the addition of new
numbering scheme) and should not exceed 15 digits functionality (either standardized or proprietary) in a
(as recommended in ITU-T Recommendation straightforward and backward compatible manner. PSS1
E.164);23 (QSIG) is defined by a number of international standards
published by ISO/IEC.
g) due to the increasing growth in internetworking
protocol (IP) technology, it is highly probable that a Note.— Standards for PSS1 (QSIG) have been mainly
CWP will need an IP address even though a developed by ECMA, an international, Europe-based
numbering plan may also be in use. industry association for standardizing information and
communication systems, and are available as ECMA
publications. Subsequently, they were adopted as a series of
international standards by ISO/IEC and as a series of ENs
2.4 DIGITAL SIGNALLING SYSTEMS by ETSI. Appendix B contains a table of equivalence for the
publications available from the three organizations.
2.4.1 General 2.4.2.3 The manner in which PSS1 (QSIG) should be

used in AGVNs to provide the facilities identified in
Telecommunications networks evolve in response to the Chapter 1 is described in Section 2.4.3.
demand for new features and services. The essential
components in current voice communication network
evolution are digital transmission technology and the use of 2.4.3 Use of PSS1 (QSIG) in AGVNs
common channel signalling systems. Digital technology can
offer many benefits including the use of digital voice
compression techniques to save bandwidth and reduce the PSS1 (QSIG) capabilities required to
number of circuits, improved voice quality, well- support the facilities defined in Chapter 1
standardized services, vendor independence, flexible
connection and routing options, integration of voice and 2.4.3.1 Provision of the facilities described in
data, and easier transition in the future. The greatest Chapter 1 can be achieved by supporting a subset of the
advantage of digital technology, however, is likely to be the complete PSS1 (QSIG) signalling system in conjunction
flexibility and range of supplementary services (SSs) that with one or more of the configurations described in Section
can be provided to the individual user by the use of 2.1.5.8. Tables 2-5 and 2-6 relate the facilities described in
sophisticated common channel signalling systems. Chapter 1 with the capabilities of PSS1 (QSIG) that have
to be supported.
2.4.2 Recommended signalling system 2.4.3.2 The use of additional capabilities of PSS1
(QSIG) over and above those identified in Tables 2-5 and
2.4.2.1 The use of PSS1 is recommended in AGVNs. 2-6 is not precluded.
PSS1 (QSIG) is an internationally standardized signalling
system for use in corporate voice networks. It is an
ISDN-based common channel system suitable for Implementation of PSS1 (QSIG) in AGVNs
peer-to-peer signalling. It provides the ability to connect
separate communication systems together in a way that 2.4.3.3 To implement PSS1 (QSIG) in AGVNs, the
allows users to share network resources and to use the following capabilities are required:
Table 2-5. PSS1 (QSIG) capabilities required to support

primary user ground telephone facilities
Primary user ground telephone facilities PSS1 (QSIG) capabilities required
Instantaneous access (see Section 1.3.2)
Direct access (see Section 1.3.3) Basic call, transit counter
Indirect access (see Section 1.3.4)
Table 2-6. PSS1 (QSIG) SSs required to support

supplementary user ground telephone facilities
Supplementary user ground telephone facilities PSS1 (QSIG) SSs required (Note 1)
Indication of calling, called and connected party identity Number identification, name identification
(see Section 1.4.2)
Indication of urgent/priority calls (see Section 1.4.4) Call priority interruption and protection, call intrusion
Add-on conference (see 1.4.5.2 and 1.4.5.3) Basic call with appropriate VCS functionality
Conferences arising from call intrusion (see 1.4.5.4) Call intrusion
Automatic recording (see Section 1.4.6) (Note 2)
Note 1.— Support of basic call and the GFP is required for the use of PSS1 (QSIG) SSs.
Note 2.— Support of this facility is unrelated to PSS1 (QSIG) capabilities.
a) A-law/µ-law PCM speech encoding in accordance association control service element (ACSE)
with ITU-T Recommendation G.711;32 protocol handling and DSE protocol handling);
b) E1/T1 interface structures in accordance with g) PSS1 (QSIG) transit counter ANF in accordance
ITU-T Recommendation G.703;31 with ISO/IEC 15056;18
c) static circuit-mode inter-PINX connections in h) PSS1 (QSIG) call priority interruption and call
accordance with ISO/IE 14474;16 priority interruption protection SSs in accordance
with ISO/IEC 15992;19 and
d) symmetrical LAPD procedures in the data link
i) PSS1 (QSIG) call intrusion SS in accordance with
layer in accordance with Amendment 1 to ITU-T
ISO/IEC 14846.17
Recommendation Q.921;46
e) PSS1 (QSIG) basic call procedures in accordance Regional variants of PSS1 for ATS use
with ISO/IE 11572;12
2.4.3.4 Some regional specifications for the use of
Note.— Basic call procedures include PSS1 (QSIG) in ATS voice ground networks exist, in
procedures for number identification. particular:
f) PSS1 (QSIG) GFP in accordance with ISO/IE a) profile standard EN 301 8464 (also published by
1158214 (excluding procedures for connectionless ECMA as ECMA-312),1 which has been adopted
application protocol data unit (APDU) transport, by EUROCONTROL for use in the ECAC area; and
b) regional ICD10 for ATS speech digital signalling and its environment. Typically, it is concerned with failure
system, adopted for use in the Asia/Pacific Region. monitoring (alarm surveillance), fault reporting, fault
isolation and service restoration.
Note.— Both of the above specifications make use of
sub-multiplexing and voice compression techniques. Configuration management
2.7.1.3 Configuration management consists of the

functions that are necessary to exercise control over,
2.5 ANALOGUE SIGNALLING SYSTEMS identify, collect data from and provide data to equipment in
the network for the purpose of controlling the arrangement
Signalling systems No. 5 and R2 are defined in the ITU-T of network resources. Typically, configuration management
Recommendations of the Q.1xx series36,37,38,39 and of the involves provisioning, installation, commissioning, pro-
Q.4xx series,40,41,42,43 respectively. There are still many gramming of network specific data, identification of parts,
instances of installations that use analogue signalling reporting and record-keeping.
systems. In particular, a special variant of signalling system
R248 is in widespread use in some AGVNs. Signalling Accounting
system No. 5 is also used but to a lesser extent. It is
expected that existing installations of both signalling 2.7.1.4 Accounting is the function necessary for
systems No. 5 and R2 will continue to be used for an tracking, allocating and controlling the costs of using
extended period of time, at least until they have been network resources. Typically, it is concerned with call
completely replaced by digital signalling installations. logging and call accounting.
However, the deployment of new installations of either
No. 5 or R2 or other analogue technologies (e.g. SS1, SS3) Performance management
is no longer recommended.
2.7.1.5 Performance management is the measurement,
reporting and evaluation of equipment and network be-
haviour to permit assessment of the long-term operational
effectiveness of the network (QoS) and to aid future
2.6 SUPERVISORY TONES network planning. Typically, it is concerned with perform-
ance monitoring (data collection), traffic measurement,
The characteristics of supervisory tones (ringing, busy, monitoring of resource status and QoS monitoring and
number unobtainable, etc.) should conform to the control. Performance management also includes planning,
requirements of ITU-T Recommendation E.180/Q.35,24 procedures and mechanisms for network back-up and
except where indicated in Table 2-7. disaster recovery.
Note.— In the case where the tones are provided locally Security management
by the end system, administrations have the flexibility of
changing them without affecting other users of the network. 2.7.1.6 Security management is the function
necessary to ensure adequate security of the installation and
the prevention of both malicious and inadvertent damage
that compromises the operational integrity of the network.
2.7 NETWORK MANAGEMENT Typically, it is concerned with physical security, system
security and network security. Security is treated separately
in Section 2-8.
2.7.1 Functional areas
2.7.1.1 Network management is classified into five 2.7.2 Network management standards
functional areas briefly described in 2.7.1.2 to 2.7.1.6.
There are currently no network management standards that
are explicitly accepted by the telecommunications industry
Fault management for the management of private telephone networks.
Although some standard protocols are used, almost all
2.7.1.2 Fault management is the detection, isolation private telephone network management is performed in a
and correction of the abnormal operation of the network proprietary manner with proprietary products.
Table 2-7. Supervisory tones
Tone Purpose Frequency (Hz) Period
Dial Returned to a user attempting to call without 425 Continuous tone

using a direct access key when that user
indicates to the system readiness to dial (e.g.
pressing a dedicated key to select a line or
placing the telephone set off-hook).
Ringing/ringback Returned to the A-party after successful call 425 (1 s on, 4 s off),
establishment and prior to call acceptance. repeated
Terminal Returned to the A-party if all available signalling 425 (0.5 s on, 0.5 s off),
busy/engaged means to the B-party are occupied. repeated
Congestion Returned to the A-party if a call cannot be 425 (0.5 s on, 0.5 s off),
completed to the required B-party due to all repeated
appropriate network links being occupied or
otherwise unavailable.
Number Returned to the A-party if a terminal is “out of 1 000 (0.5 s on, 0.5 s off),
unobtainable/ service” or the B-party address is unassigned. repeated
reorder (Note 1)
Interrupt warning Injected into the voice path to warn a party of 1 000 (40 ms on, 0.5 s off)
(Note 1) the imminent priority interruption of an repeated for approximately
established call. 5–15 s, until disconnection
Forced conference Injected into the voice path to warn a party of the 1 000 (1 s on)
warning (Note 1) imminent priority conferencing of an established
call.
Note 1.— Not specified in ITU-T Recommendation E.180.
Note 2.— The recommendation for congestion tone is the same as for busy tone. If users wish to distinguish between the
two tones, a dual frequency congestion tone can be used.
2.8 SECURITY ASPECTS 2.8.1.2 Security can be considered from the

perspective of three general categories: physical security,
system security and network security. They are discussed in
2.8.1 General Sections 2.8.2, 2.8.3 and 2.8.4.
2.8.1.1 The emplacement of adequate security

measures is an obvious duty of administrations, and as 2.8.2 Physical security
international networks evolve toward an open digital
architecture, administrations will need to establish and 2.8.2.1 Appropriate physical security measures,
implement a robust security policy. Safeguards are similar to those usually employed throughout facilities
necessary to protect their own ATM as well as other associated with the management of air traffic, should be
(external) users connected to the network. This section taken to protect the areas and systems used to provide
relates to the security of all systems and services that communication services.
contribute to the provision of ground telephone services,
although there will be commonality with the security of 2.8.2.2 Guidance on safeguarding against physical
other services. risks is outside the scope of this manual, but in protecting
communication services, consideration also needs to be location are included, such as VCS and line transmission
given where there is a dependency on the services of third equipment. System security includes ensuring that man-
parties, such as telecommunications service providers. It is agement and maintenance interfaces are not vulnerable to
unlikely that the safeguarding of these can be assured; unauthorized use. VCSs can be attacked through their
therefore, independent contingency measures need to be maintenance and network management interfaces as well as
considered and put in place as required. The scenario that through interfaces to the outside world. Such attacks may
should be considered is a total and sustained loss of one or manifest themselves as blockage of port accessibility,
all third-party services. unusual traffic patterns or modification/corruption of
configuration files.
2.8.2.3 All areas of the communications system
(including VCS, transmission links, PSTN links and the 2.8.3.2 Where many items of equipment are
related power supply, network management and main- physically interconnected, steps need to be taken to prevent
tenance centres) should be considered as being vulnerable either deliberate or inadvertent unauthorized access to
to physical tampering, including an electrical or electronic operational VCSs and external links. VCSs should have
attack. Where it is possible for an administration to do so, very tightly controlled call-barring and class-of-service
the following is a list of measures that can be taken to mechanisms to prevent such occurrences while still
reduce the effect of physical damage: permitting flexibility in their use. The use of personal
identification number (PIN) techniques to restrict access to
a) duplicate equipment;
operational services is not usually considered to be
b) physically split systems; and adequate.
c) separate environments/locations for equipment with 2.8.3.3 VCSs should, ideally, provide support for
independent power and other services. identification and authentication of authorized users, in
particular, management and maintenance personnel.
2.8.2.4 Of particular importance are those areas where Mechanisms to prevent misuse of the trunk network by
the external communications services are provided, unauthorized users (i.e. call barring) should also be
including patch bays, line transmission equipment and provided.
external cable ducts. In cases where there is no alternative
to the use of a single telecommunications service provider,
consideration should be given to connecting to completely 2.8.4 Network security
separate network access points via completely separate
physical routes. The ideal is to mitigate against the effects 2.8.4.1 AGVNs, each under different administrative
of all common points of potential failure. control, can be interconnected at national or regional
boundaries. These networks will make use of facilities
2.8.2.5 The provision of standby links is a common
leased/rented from telecommunications service providers.
practice to reduce the effects of physical damage and other
They may also have interfaces to the PSTN to permit calls
causes of loss of service. From the point of view of
to be made to destinations outside the network to provide
achieving an improved overall availability, however,
link back-up or simply as an alternative to the leased-line
standby links must be provided with as much physical
scenario.
separation and independence from the main links as is
practicable.
2.8.4.2 Attacks on the network could come from
2.8.2.6 Consideration of the consequences of loss of within the network itself, via the infrastructure rented from
service due to physical effects should also be extended to telecommunications service providers or via interfaces to
the operational level and planned in conjunction with the the PSTN. Such attacks can be physical, logical or both and
users. It might be possible for one user or working position could affect the availability, integrity or confidentiality of
to provide contingency services for another user or working voice transport.
position, but this could be severely impaired if the
contingency was due to the loss of a link they both shared. 2.8.4.3 Where a VCS provides external (public)
access, this functionality must be restricted to ensure that
external callers cannot gain access to trunk lines used for
2.8.3 System security operational ATM. Where a VCS does not support all of the
functionality listed above, consideration must be given to
2.8.3.1 Within this category, all items of an alternative means of providing the same level of
communications equipment within a particular operational protection.
2.8.4.4 AGVNs should provide accountability, access functionality, the transition is more complex. The options
control and assurance features through the use of for transition include:
mechanisms such as authentication of authorized users and
a) modifying the external interfaces of the old
the maintenance of an audit log of all actions.
switches to coexist with digital VCSs, which may
or may not be possible or economically feasible;
b) using the new digital VCS with analogue interfaces
2.9 TRANSITION ARRANGEMENTS to emulate the analogue-routing functionality until
enough digital VCSs are commissioned to start
2.9.1 The degree of difficulty experienced in the transitioning to digital trunks and routing
transition of the analogue network to the digital network functionality; and
will depend on the size and complexity of the analogue
c) reverting to dedicated point-to-point circuits during
network being replaced. If the existing network is based on
transition until enough digital VCSs are
an analogue VCS and dedicated point-to-point circuits with
commissioned.
no routing capability, then the digital VCS may be
introduced with temporary analogue interfaces that are 2.9.2 Because of the variety of different analogue
converted to digital when enough digital VCSs are VCSs and routing capabilities in existence, careful
commissioned. The routing functionality can be utilized for operational and economic analysis will be required to
parts of the network as the number of digital VCSs grow, determine the best transition strategy, and a high degree of
thus benefiting from the reductions in network costs. In coordination will be required with other administrations for
cases where analogue VCSs are providing routing international links.
Appendix A
METHOD FOR ASSIGNING SIGNALLING
DELAY TO NETWORK ELEMENTS
A.1 GENERAL (CC) functional entities for the provision of the service
requested by the user. CCAs are located within the CWP
This appendix describes a method that can be used to plan involved in the basic call, i.e. the CWP of the user who
the assignment of a signalling delay budget to elements of requests the service (the A-party) and the CWP of the
an AGVN in order to achieve the correct provisioning of destination user (the B-party). Specific forms of CCA exist
equipment to meet the signalling performance require- for the A-party and the B-party.
ments. It also describes a general model for apportioning
call establishment and clearing delays to a call through a A.2.2 The CC functional entities cooperate to provide
telephone network and identifies potential sources of the service requested by a CCA. Specific forms of CC are
signalling delay. It then shows how this general model can located at the network nodes (typically, VCSs) through
be refined by integrating the signalling for a particular which the call is routed, including the node serving the
telephone facility (in this example, the direct access A-party (originating CC), the node serving the B-party
facility) to deliver a specific delay model. Network planners (terminating CC) and any intermediate nodes (transit CC).
can develop similar models for real network configurations. With specialization as described, the model becomes
similar to that shown in Figure A-2.
A.2.3 A notional transmission system (indicated by

A.2 GENERAL MODEL FOR A the horizontal lines) is considered to appear between each
BASIC TELEPHONE CALL pair of functional entities. Such a transmission system may
or may not actually exist, depending on the realization of
the model in the physical world. The model is independent
Functional elements of a basic telephone call of the network or networks involved in the call, i.e. it
(see Figure A-1) applies equally to situations where interworking between
networks occurs. In this case there are other specific forms
A.2.1 The call control agent (CCA) functional entity of CC (incoming gateway CC, outgoing gateway CC)
acts on behalf of the user and calls upon the call control located at the interworking point.
User User
CCA CC CC CC CCA
Note.— The number of CCs can vary from one upwards.
Figure A-1. Functional entities involved in a basic call
A-1
A-2 Ground-Ground Voice Switching and Signalling
Use of a functional model for TccORIG In the case of call establishment, the delay
apportioning call establishment/clearing delays caused by the time taken by the originating
CC to process the A-party’s request for a
A.2.4 Each component of the functional model service. In the case of call clearing, the
introduces an element of delay into the processing of a call. delay caused by the time taken to process
Delays due to the processing and transmission of signalling the A-party’s request to clear a call.
information are of particular interest. Elements of delay can
be apportioned to each component of the functional model, TtxnY The delay for the transmission of signalling
as shown in Figure A-3. The definition of each delay information between two CCs. Where a
element is as follows: connection involves one or more transit
CCs, there will be multiple transmission
TccaA In the case of call establishment, the delay segments represented by TtxnY where Y = 1
caused by the time taken by CC “A” to to N + 1 (N = number of transit CCs). The
process the A-party’s request for a service. delay imposed by all connections (sum of
In the case of call clearing, the delay caused all TtxnY) is represented by TtxnTOT.
by the time taken to process the A-party’s
request to clear a call. TccTRANSIT In the case of call establishment, the delay
caused by the time taken by a transit CC to
TlocalA, TlocalB The delay for the transmission of signalling process a transit call (i.e. an incoming call
information between a CCA and its serving and onward-routed outgoing call). In the
CC. case of call clearing, the delay caused by the
A-party B-party
CCA ORIG Transit TERM CCA

“A” CC CC CC “B”
Figure A-2. Functional model for a basic call
A-party B-party
CCA ORIG Transit TERM CCA

“A” CC CC CC “B”
TccaA TlocalA TccORIG TtxnY TccTRANSIT TtxnY TccTERM TlocalB TccaB
Figure A-3. Apportionment of delay elements

Appendix A. Method for assigning signalling delay to network elements A-3
time taken to process a request to clear a intended to guide the design of public exchange equipment.
transit call. Where a connection involves Table A-1, which is based on this ITU-T Recommendation,
more than one transit CC, the delay summarizes the basic design objectives for public exchange
imposed by all transit CCs is represented by equipment in integrated services digital networks (ISDN).
N × TccTRANSIT.
A.3.3 As can be seen from the table, the design
TccTERM In the case of call establishment, the delay objectives for public exchanges are quite relaxed. Typical
caused by the time taken by the terminating modern customer premises equipment (CPE) has little
CC to present the incoming call to the difficulty in meeting or exceeding objectives twice as
B-party. In the case of call clearing, the stringent as those shown. By selecting and manipulating the
delay caused by the time taken to process appropriate delay elements (TccaA, TccORIG, TccTRANSIT,
the A-party’s request to clear a call. etc.) in the delay model, performance parameters similar to
those in Table A-1 can be determined for connections in
TccaB In the case of call establishment, the delay AGVNs.
caused by the time taken by CC “B” to
present the incoming call to the B-party. In
the case of call clearing, the delay caused
by the time taken by CCA “B” to process Signalling delay resulting from
the A-party’s request to clear a call. the transmission system
A.3.4 Similarly, it is difficult to assign accurate

one-way delay values for the TtxnY parameters. This is due
A.3 SOURCES OF SIGNALLING DELAY to the general unavailability of accurate information for
specific connection segments. Network planners are thus
forced to use values that apply to the general case and, as
Signalling delay arising in CCAs and CCs such, include conservative allowances. Although primarily
intended for calculating delay in speech and data
A.3.1 Delays arising in CCAs and CCs can only be transmission, the values specified in Section 2.2.3.7 can also
calculated accurately for specific signalling systems im- be applied to the transmission of signalling information.
plemented on specific items of equipment (e.g. CWPs,
VCSs) from particular manufacturers. It is difficult to calcu- A.3.5 As already noted, ITU-T Recommendation
late generally applicable values because much depends on Q.54344 gives maximum values for processing signalling
the overall system architecture, the microprocessors used information in digital exchanges. Based on this, it is
and the efficiency of the software design. For network reasonable to expect that intermediate signalling equipment
planning purposes, therefore, typical values that are maxima in a transmission link might introduce some additional
that must not be exceeded have to be chosen. delay. However, in the context of leased lines, there is little
dynamic switching overhead associated with the line itself
A.3.2 ITU-T Recommendation Q.54344 deals with and consequently, the effect of signalling processing delay
performance design objectives for digital exchanges and is is minimal. No additional allowance needs to be made.
Table A-1. Q.543 performance design objectives for public exchanges
Performance parameter Design objective
User signalling acknowledgement delay <400 ms

Signalling transfer delay <200 ms
Call set-up delay (overlap sending) <400 ms
(en-bloc sending) <600 ms
Through-connection delay <250 ms
Incoming call indication sending delay (overlap receiving) <400 ms
(en-bloc receiving) <600 ms
Connection release delay <250 ms
A.4 INTEGRATION OF SIGNALLING where N is the number of transit nodes in the connection
TO GIVE A SPECIFIC DELAY MODEL and TtxnTOT is the total delay arising from the transmission
systems, expressed as
Refinement of the model to produce Y = N+1

a specific delay model
∑ TtxnY
Y=1
A.4.1 To be of practical value, it is necessary to refine
the general model described above by:
where TtxnY is the delay between a pair of CCs, calculated
a) mapping the signalling for a particular telephone as (10 + 0.01 × G) ms, and G is the geographical distance
facility to deliver a specific delay model; in kilometres.
b) identifying the specific call-processing tasks to be Note.— A model can also be constructed for the
performed at each stage of signalling in the specific instantaneous access facility. However, since the required
delay model; and signalling is similar, it does not differ significantly from the
model for direct access.
c) attaching a delay element for each identified task.
A.4.2 By mapping appropriate signalling exchanges to Allocation of values

the delay model, planning values can be calculated for each
specific ground-user telephone facility. This can be done for A.4.5 Allocating the previously given values to the
a particular network topology, thus allowing network plan- derived formulae shows that there is ample scope for
ners to optimize a network design to avoid excessive delay. network planners to specify connection topologies that fall
well within the performance requirements of both the direct
access and instantaneous access facilities. Taking the case
Example delay model for the of a direct access connection involving a single transit node
direct access facility with distances of 1 500 km and 4 100 km between the
adjacent nodes would give an allowance as follows:
A.4.3 Figure A-4 shows a typical signalling model for
the direct access facility. The three-digit numbers represent 100 + 300 + (1 × 300) + (10 + 0.01 × 1 500)
specific processing tasks (and hence, delay) to be + (10 + 0.01 × 4 100) + 300 + 100 = 1 176 ms
performed by CCAs and CCs at each stage of signalling.
These delays are defined in Table A-2.
This value is well within the two-second performance
required for direct access.
Note.— Since a CCA and its serving CC will almost
always be co-located (i.e. usually in the same building), the
delay between them can effectively be ignored. A.4.6 At first sight it appears that an instantaneous
access connection involving a single transit node could not
A.4.4 Total call set-up delay, the parameter under- meet the performance requirement (one second). However,
pinning the performance requirement for direct access as previously stated, available modern equipment generally
connections can be obtained by summing delays as follows: has no difficulty in exceeding the allowances by a wide
margin. Moreover, using relatively old fashioned micro-
Call set-up delay = processors, actual values for TccORIG, etc., are typically
nearer to 150 ms than to 300 ms, and values for TccaA, etc.,
TccaA T T T T should be nearer to 20–30 ms than to 100 ms. Thus, the
+ ccORIG + (N*TccTRANSIT) + txnTOT + ccTERM + ccaB
000 000 300 400 allowance of 1 176 ms drops to 726 ms or less.
Appendix A. Method for assigning signalling delay to network elements A-5
MSC Direct_Access
CC “A” ORIG CC Transit CC TERM CC CCA “B”

[000] DA_request
[100] SET UP
CALL_PROCEEDING [200]
SET UP
CALL_PROCEEDING [300]
ALERTING incoming_DA_indication
[202] [400]
ALERTING
alerting_indication DA_answer [403]

[002]
CONNECT [303]
[203]
CONNECT
[103]
DA_answered
[003]
Figure A-4. Signalling model for the direct access facility

Table A-2. Call-processing tasks and delay elements for the direct access facility
Delay element
Call-processing task definition (Note 1)
Between A-party requesting a service and CCA “A” sending a DA_request to the originating CC. TccaA
000
Between CCA “A” receiving an indication that the B-party is alerting (alerting_indication) and the TccaA
A-party being given an indication of that. 002
Between CCA “A” receiving an indication that the B-party has answered (DA_answered) and the TccaA
A-party being given an indication of that. 003
Between an originating CC receiving a DA_request from CCA “A” and sending SET UP to the next CC. TccORIG
100
(Note 2)
Between an originating CC receiving an indication that the B-party is alerting (ALERTING) and TccORIG
passing that indication on to CCA “A” (alerting_indication). 102
Between an originating CC receiving an indication that the B-party has answered (CONNECT) and TccORIG
passing that indication on to CCA “A” (DA_answered). 103
Between a transit CC receiving SET UP from the preceding CC and sending SET UP to the subsequent CC. TccTRANSIT
200
(Note 2)
Between a transit CC receiving an indication from the preceding CC that the B-party is alerting TccTRANSIT
(ALERTING) and passing that indication on to the subsequent CC. 202
Between a transit CC receiving an indication from the preceding CC that the B-party has answered TccTRANSIT
(CONNECT) and passing that indication on to the subsequent CC. 203
Between a terminating CC receiving SET UP and sending an incoming_DA_ indication to CCA “B”. TccTERM
300
(Note 3)
Between a terminating CC receiving an indication (DA_answer) from CCA “B” that the B-party has TccTERM
answered and sending a CONNECT message to the next CC. 303
Between CCA “B” receiving an indication of an incoming call (incoming_DA_ indication) and the TccaB
B-party being given an indication of that call. 400
Between the B-party answering and CCA “B” giving an indication (DA_answer) of that to the TccaB
terminating CC. 403
Note 1.— The signalling transfer delay imposed by a node (equivalent to the signalling transfer delay parameter
shown in Table A-1) is the average of all the delays for that node.
Note 2.— Equivalent to the call set-up delay parameter shown in Table A-1.
Note 3.— Equivalent to the incoming call indication sending delay parameter shown in Table A-1.
Appendix B
REFERENCED PUBLICATIONS AND
INFORMATION SOURCES
B.1 GENERAL 6. ETR 250 (1996): “Transmission and Multiplexing

(TM); Speech communication quality from mouth to
The manual incorporates, by reference, provisions from ear for 3,1 kHz handset telephony across networks”.
other publications. These references are cited at the
appropriate places in the text and the publications are listed 7. EUROCONTROL ASM.ET1.ST18.1000-REP-01.00
in Section B.2, together with other publications that may be (1998): “Guidelines for the Application of the ECAC
useful for gaining a fuller understanding of the subject Radar Separation Minima”, Edition 2.0.
matter of this manual. Section B.3 provides a table of
equivalence for PSS1 specifications generated by different 8. ICAO Annex 10, Volume III, Part II, Chapter 4,
standardization bodies. Web site addresses of stan- “Aeronautical Speech Circuits”.
dardization bodies are provided in Section B.4.
9. ICAO Annex 11, Chapter 6, “Air Traffic Services
Requirements for Communications”.
B.2 REFERENCES
10. ICAO State letter T8/3.1, dated 13 May 1998, on the
For dated references, subsequent amendments to any of “Asia/Pacific Regional Interface Control Document
these publications apply to this manual only when (ICD) for Signalling System for Digital ATS Speed
incorporated by amendment. For undated references, the Circuit Network”.
latest edition of the publication referred to applies.
11. ISO/IEC 11571 (1998): “Information technology,
1. ECMA-312 (2000): “Private Integrated Services
Telecommunications and information exchange
Network (PISN) — Profile Standard for the Use of
between systems — Private Integrated Services
PSS1 (QSIG) in Air Traffic Services Networks” (also
Networks — Addressing”.
published by ETSI as EN 301 846).
2. ECMA TR/76 (1999): “Private Integrated Services 12. ISO/IEC 11572: (2000) “Information technology —
Network — Architecture and Scenarios for Private Telecommunications and information exchange
Integrated Services Networking”. between systems — Private Integrated Services
Network (PISN) — Circuit mode bearer services —
Note.— The content of ECM TR/76 is fully aligned Inter-exchange signalling procedures and protocol”.
with ISO/IEC TR 14475, Second Edition.
13. ISO/IEC 11573 (1994): “Information technology —
3. EG 201 050 (1999): “Speech Processing, Transmission Telecommunications and information exchange
and Quality Aspects (STQ); Overall Transmission Plan between systems — Synchronization methods and
Aspects for Telephony in a Private Network”, V1.2.2. technical requirements for Private Integrated Services
Networks”.
4. EN 301 846 (2001): “Private Integrated Services
Network (PISN) — Profile Standard for the Use of
PSS1 (QSIG) in Air Traffic Services Networks”, (also 14. ISO/IEC 11582 (1995): “Information technology —
published by ECMA as ECMA-312). Telecommunications and information exchange
5. ES 201 168 (1998): “Corporate Networks (CN); Network — Generic functional protocol for the
Transmission characteristics of digital Private Branch support of supplementary services — Inter-exchange
eXchanges (PBXs)”. signalling procedures and protocol”.
B-1
B-2 Ground-Ground Voice Switching and Signalling
15. ISO/IEC 13868 (1995): “Information technology — 24. ITU-T Recommendation E.180/Q.35 (1998): “Techni-
Telecommunications and information exchange cal characteristics of tones for the telephone service”.
Network — Inter-exchange signalling protocol — 25. ITU-T Recommendation G.107 (2000): “The E-Model,
Name identification supplementary services”. a computational model for use in transmission
planning”.
Telecommunications and information exchange 26. ITU-T Recommendation G.108 (1999): “Application
between systems — Private Integrated Services of the E-model: A planning guide”.
Network — Functional requirements for static
circuit-mode inter-PINX connections”. 27. ITU-T Recommendation G.109 (1999): “Definition of
categories of speech transmission quality”.
Telecommunications and information exchange 28. ITU-T Recommendation G.113 (1996): “Transmission
between systems — Private Integrated Services impairments”.
Network — Inter-exchange signalling protocol — Call
intrusion supplementary service”. 29. ITU-T Recommendation G.114 (1996): “Transmission
systems and media — General characteristics of inter-
18. ISO/IEC 15056 (1997): “Information technology — national telephone connections and international
Telecommunications and information exchange telephone circuits — One-way transmission time”.
Network — Inter-exchange signalling protocol — 30. ITU-T Recommendation G.702 (1988): “Digital
Transit counter additional network feature”. hierarchy bit rates”.
19. ISO/IEC 15992 (1998): “Information technology — 31. ITU-T Recommendation G.703 (1998): “General
Telecommunications and information exchange aspects of digital transmission systems — Terminal
between systems — Private Integrated Services equipments physical/electrical characteristics of
Network — Inter-exchange signalling protocol — Call hierarchical digital interfaces.”
priority interruption and call priority interruption
protection supplementary services”. 32. ITU-T Recommendation G.711 (1988): “Pulse code
modulation (PCM) of voice frequencies”.
Telecommunications and information exchange 33. ITU-T Recommendation G.728 (1992): “Coding of
between systems — Private Integrated Services speech at 16 kbit/s using low-delay code excited linear
Network — Mapping functions for the employment of prediction”.
64 kbit/s circuit mode connections with 16 kbit/s
sub-multiplexing”. 34. ITU-T Recommendation G.729 (1996): “Coding of
speech at 8 kbit/s using conjugate-structure algebraic-
21. ISO/IEC 17311 (2000): “Information technology — code-excited linear-prediction (CS-ACELP)”.
Telecommunications and information exchange
between systems — Private Integrated Services 35. ITU-T Recommendation G.729 — Annex A (1996):
Network — Mapping functions for the employment of “C source code and test vectors for implementation
64 kbit/s circuit mode connections with 8 kbit/s verification of the G.729 reduced complexity 8 kbit/s
sub-multiplexing”. CS-ACELP speech coder”.
22. ISO/IEC TR 14475 (2001): “Information technology 36. ITU-T Recommendation Q.140 (1988): “Specifi-
— Telecommunications and information exchange cations of signalling system No 5 — Definition and
between systems — Private Integrated Services function of signals”.
Network — Architecture and Scenarios for Private
Integrated Services Networking”, Second Edition. 37. ITU-T Recommendation Q.141 (1988): “Specifi-
cations of signalling system No 5 — Signal code for
Note.— The content of ISO/IEC TR 14475, Second line signalling”.
Edition, is also published by ECM as ECMA TR/76.
38. ITU-T Recommendation Q.151 (1988): “Specifi-
23. ITU-T Recommendation E.164 (1997): “The inter- cations of signalling system No 5 — Signal code for
national public telecommunication numbering plan”. register signalling”.
Appendix B. Referenced publications and information sources B-3
39. ITU-T Recommendation Q.152 (1988): “Specifi- 48. EUROCONTROL document COM-GUI-01-00: Guide-
cations of signalling system No 5 — End-of-pulsing lines for the Implementation of the Automatic Voice
conditions — register arrangements concerning ST Communication Network, Edition 2.0.
(end-of-pulsing) signal”.
40. ITU-T Recommendation Q.400 (1988): “Specifi- B.3 EQUIVALENCES FOR

cations of signalling system R2 — Definition and PSS1 (QSIG) SPECIFICATIONS
function of signals — Forward line signals”.
Specifications for PSS1 (QSIG) are mainly developed by
41. ITU-T Recommendation Q.411 (1988): “Specifi- ECMA, an international Europe-based industry association
cations of signalling system R2 — Line signalling, dedicated to the standardization of information and com-
analogue version — Line signalling code”. munications technology systems. They are then submitted
to ISO/IEC for adoption as international standards. Once an
42. ITU-T Recommendation Q.440 (1988): “Specifi- international standard is published, the specification is also
cations of signalling system R2 — Interregister submitted to the ETSI for publication as an EN. Table B-1
signalling — General”. lists the equivalent publications from each of the three
organizations, ECMA, ISO/IEC and ETSI. Generally, the
43. ITU-T Recommendation Q.441 (1988): “Specifications publications of the three organizations are technically
of signalling system R2 — Interregister signalling — aligned with one another. However, in some instances there
Signalling code”. are some technical differences between the publications of
ISO/IEC and those of the other two organizations that
44. ITU-T Recommendation Q.543 (1993): “Digital reflect specific requirements of the European marketplace.
Exchanges — Digital exchange performance design
objectives”.
B.4 WEB SITES OF
45. ITU-T Recommendation Q.921 (1997): “ISDN user- STANDARDIZATION BODIES
network interface — Data link layer specification”.
ECMA: www.ecma.ch
46. ITU-T Recommendation Q.921 — Amendment 1
ETSI: www.etsi.org
(2000): “Digital Subscriber Signalling System No.1 —
Data link layer”. ICAO: www.icao.int
ISO: www.iso.ch
47. TIA/EIA-TSB-32-A (1998): “Overall Transmission
Plan Aspects for Telephony in a Private Network”. ITU: www.itu.int
B-4 Ground-Ground Voice Switching and Signalling
Table B-1. Equivalent publications for PSS1 (QSIG)
ECMA International European

Service standard standard standard
Basic call ECMA-143 ISO/IEC 11572 EN 300 172

GFP ECMA-165 ISO/IEC 11582 EN 300 239
Number identification (Note) (Note) (Note)
Name identification ECMA-164 ISO/IEC 13868 EN 300 238
Call intrusion ECMA-203 ISO/IEC 14846 EN 300 426
Transit counter ECMA-225 ISO/IEC 15056 EN 301 048
Call priority interruption and protection ECMA-264 ISO/IEC 15992 EN 301 656
Addressing ECMA-155 ISO/IEC 11571 EN 300 189
Mapping/static circuit mode ECMA-226 ISO/IEC 14474 EN 301 765
Mapping/16 ECMA-253 ISO/IEC 17310 EN 301 039
Mapping/8 ECMA-289 ISO/IEC 17311 EN 301 924
Note.— The protocol specification for the identification SSs is contained within the basic call protocol
standards (ECMA-143, ISO/IEC 11572 and EN 300 172).
— END —
© ICAO 2002
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Manual on Air Traffic

Approved by the Secretary General

First Edition — 2002

International Civil Aviation Organization

Manual on Air Traffic

Approved by the Secretary General

First Edition — 2002

International Civil Aviation Organization

RECORD OF AMENDMENTS AND CORRIGENDA

No. Date Entered by No. Date Entered by

The designations employed and the presentation of the material in this

Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . (vii) 2.2 QoS and network performance . . . . . . . . . . . . . 2-7

2.1 Network planning, implementation and

Page Page Page

Appendix B. Referenced publications and

ACSE Association control service element Ie Equipment impairment factor

1.1 CORRELATION TO THE operational requirements, including those specified in

1.2.2 Applicability to domestic Note.— Some administrations may prefer that it be

1.3.3.5 Under normal conditions the B-party can

Table 1-1. Performance requirements

Facility Establishment of communication should occur:

1.4.2 Indication of calling, 1.4.3.2 In networks where switching and signalling

1.4.2.3 When a call is alerting, the identity of the

b) allow the priority call to intrude in a call already

1.4.5 Conference capabilities 1.4.6 Automatic recording

2.1.3.1 To avoid having to fully interconnect each

a) copper cables; 2.1.3.2 The use of switching elements in a network

The use of alternative technologies is not precluded,

Use of sub-multiplexing and low

Core technology b) where the use of a satellite is required, e.g. on a link

Table 2-1. Characteristics of speech encoding algorithms

PCM (G.711) 64 0.75 0

Note 1.— All delay values are one-way.

2.1.6 Planning guidelines i) Networks should not be permitted to have a single

Table 2-2. Possible configurations based on considerations of

Interface characteristics Sub-multiplexing and Sub-multiplexing and

Match interface characteristics of the Directly connected Directly connected

Figure 2-1. Directly connected configuration

Figure 2-2. Indirectly connected configuration

Table 2-3. Example configurations

Bit rate of voice channels

– at point A 64 kbit/s 16 kbit/s 64 kbit/s 64 kbit/s

– at point A A/µ-law PCM 16 kbit/s LD-CELP A/µ-law PCM A/µ-law PCM

Interface at point B 64 kbit/s or E1/T1 64 kbit/s G.703 E1 or T1 64 kbit/s G.703

Bit rate of signalling

– at point A 64 kbit/s 16 kbit/s 64 kbit/s 64 kbit/s

a) rapid call set-up times (in order to meet the

b) transmission planning; 2.2.3.2 To ensure optimum speech quality, planners

Upper limit for total impairment value (Icpif) Speech quality

Calling party speech path

Calling party echo return path

Called party speech path

Figure 2-3. Echo return path

2.3.3.2 Worldwide, AGVNs operate collectively to

Figure 2-4. Format of numbers in the recommended numbering plan

2.4.1 General 2.4.2.3 The manner in which PSS1 (QSIG) should be

Table 2-5. PSS1 (QSIG) capabilities required to support

Primary user ground telephone facilities PSS1 (QSIG) capabilities required