Tia Eia 810 A Final

ANSI/TIA/EIA-810-A-2000
Approved: December 19, 2000
TIA/EIA
STANDARD
TIA/EIA-810-A
Telecommunications
Telephone Terminal Equipment
Transmission Requirements for
Narrowband Voice over IP and
Voice over PCM Digital Wireline
Telephones
TIA/EIA-810-A
(Upgrade and Revision of TIA/EIA/IS-810)
DECEMBER 2000
TELECOMMUNICATIONS INDUSTRY ASSOCIATION
The Telecommunications Industry Association

Represents the Communications Sector of
NOTICE
TIA/EIA Engineering Standards and Publications are designed to serve the public interest
through eliminating misunderstandings between manufacturers and purchasers, facilitating
interchangeability and improvement of products, and assisting the purchaser in selecting and
obtaining with minimum delay the proper product for his particular need. Existence of such
Standards and Publications shall not in any respect preclude any member or nonmember of
TIA/EIA from manufacturing or selling products not conforming to such Standards and
Publications, nor shall the existence of such Standards and Publications preclude their voluntary
use by those other than TIA/EIA members, whether the standard is to be used either domestically
or internationally.
Standards and Publications are adopted by TIA/EIA in accordance with the American National
Standards Institute (ANSI) patent policy. By such action, TIA/EIA does not assume any liability
to any patent owner, nor does it assume any obligation whatever to parties adopting the Standard
or Publication.
This Standard does not purport to address all safety problems associated with its use or all
applicable regulatory requirements. It is the responsibility of the user of this Standard to
establish appropriate safety and health practices and to determine the applicability of regulatory
limitations before its use.
(From Standards Proposal No. 4352-URV, formulated under the cognizance of the TIA TR-41.3
Subcommittee on Analog and Digital Wireline Terminals.)
Published by
TELECOMMUNICATIONS INDUSTRY ASSOCIATION 2000
Standards and Technology Department
2500 Wilson Boulevard
Arlington, VA 22201
PRICE: Please refer to current Catalog of

EIA ELECTRONIC INDUSTRIES ALLIANCE STANDARDS and ENGINEERING
PUBLICATIONS or call Global Engineering Documents, USA and Canada
(1-800-854-7179) International (303-397-7956)
All rights reserved

Printed in U.S.A.
PLEASE!
DON'T VIOLATE
THE
LAW!
This document is copyrighted by the TIA and may not be reproduced without
permission.
Organizations may obtain permission to reproduce a limited number of copies

through entering into a license agreement. For information, contact:
Global Engineering Documents

15 Inverness Way East
Englewood, CO 80112-5704 or call
U.S.A. and Canada 1-800-854-7179, International (303) 397-7956
TIA/EIA-810-A
TABLE OF CONTENTS
1. INTRODUCTION ..................................................................................................................... 1
2. SCOPE........................................................................................................................................ 2
2.1. LIMITS OF APPLICABILITY ................................................................................................ 2
2.2. CATEGORIES OF CRITERIA ................................................................................................ 2
2.3. FCC PART 68....................................................................................................................... 2
2.4. ENVIRONMENTAL .............................................................................................................. 2
2.5. SAFETY ................................................................................................................................ 2
3. NORMATIVE REFERENCES................................................................................................ 3
4. DEFINITIONS, ABBREVIATIONS AND ACRONYMS..................................................... 5
4.1. CODEC ................................................................................................................................. 5
4.2. EAR REFERENCE POINT (ERP).......................................................................................... 5
4.3. HATS POSITION.................................................................................................................. 5
4.4. MOUTH REFERENCE POINT (MRP) .................................................................................. 5
4.5. QUIET AND FULL SCALE CODE ........................................................................................ 5
4.6. REFERENCE CODEC ........................................................................................................... 5
4.7. DIRECT DIGITAL PROCESSING ......................................................................................... 6
4.8. SOUND PRESSURE LEVELS................................................................................................ 7
4.9. ELECTRIC POWER AND NOISE LEVELS ........................................................................... 7
4.10. ABBREVIATIONS AND ACRONYMS .................................................................................. 7
5. HANDSET TECHNICAL REQUIREMENTS....................................................................... 9
5.1. HANDSET FREQUENCY RESPONSE................................................................................... 9
5.1.1. Handset Send Frequency Response .............................................................................. 9
5.1.2. Handset Receive Frequency Response ....................................................................... 11
5.2. HANDSET LOUDNESS RATINGS AND RECEIVE VOLUME CONTROL .......................... 13
5.2.1. Handset Send Loudness Rating (SLR) ....................................................................... 13
5.2.2. Handset Receive Loudness Rating.............................................................................. 13
5.2.3. Handset Receive Volume Control .............................................................................. 14
5.2.4. Handset Talker Sidetone............................................................................................. 14
5.3. HANDSET NOISE ............................................................................................................... 14
5.3.1. Handset Send Noise .................................................................................................... 14
5.3.2. Handset Send Single Frequency Interference ............................................................. 14
5.3.3. Handset Receive Noise ............................................................................................... 15
5.3.4. Handset Receive Single Frequency Interference ........................................................ 15
5.4. HANDSET RECEIVE COMFORT NOISE (ADVISORY) ..................................................... 16
5.4.1. General........................................................................................................................ 16
5.4.2. Measurement Method ................................................................................................. 16
5.4.3. Requirement ................................................................................................................ 16
5.5. HANDSET DISTORTION AND NOISE ............................................................................... 16
5.5.1. Handset Send Distortion and Noise ............................................................................ 17
i
TIA/EIA-810-A
5.5.2. Handset Receive Distortion and Noise........................................................................17
5.6. WEIGHTED TERMINAL COUPLING LOSS (TCLW) .........................................................18
5.6.1. Measurement Method..................................................................................................18
5.6.2. Requirements...............................................................................................................19
5.7. STABILITY LOSS ...............................................................................................................20
5.7.2. Requirement ................................................................................................................20
5.8. LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE INPUT)..........21
5.8.1. General ........................................................................................................................21
5.8.3. Requirements...............................................................................................................22
5.9. SHORT DURATION (PEAK) ACOUSTIC PRESSURE .........................................................22
5.9.1. General ........................................................................................................................22
5.9.3. Requirements...............................................................................................................23
5.10. PACKET VOICE LATENCY (ADVISORY) .........................................................................23
5.10.1. Handset Send Latency ................................................................................................24
5.10.2. Handset Receive Latency ...........................................................................................24
6. HEADSET TECHNICAL REQUIREMENTS......................................................................25
6.1. HEADSET FREQUENCY RESPONSE..................................................................................25
6.1.1. Headset Send Frequency Response .............................................................................25
6.1.2. Headset Receive Frequency Response ........................................................................26
6.2. HEADSET LOUDNESS RATINGS .......................................................................................27
6.2.1. Headset Send Loudness Rating ...................................................................................27
6.2.2. Headset Receive Loudness Rating ..............................................................................27
6.2.3. Headset Talker Sidetone..............................................................................................28
6.3. HEADSET NOISE ................................................................................................................29
6.3.1. Headset Send Noise.....................................................................................................29
6.3.2. Headset Send Single Frequency Interference..............................................................29
6.3.3. Headset Receive Noise ................................................................................................30
6.3.4. Headset Receive Single Frequency Interference.........................................................30
6.4. HEADSET DISTORTION AND NOISE ................................................................................30
6.4.1. Headset Send Distortion and Noise.............................................................................31
6.4.2. Headset Receive Distortion and Noise........................................................................31
6.5. WEIGHTED TERMINAL COUPLING LOSS (TCLW) .........................................................32
6.5.2. Requirements...............................................................................................................32
6.6. LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE INPUT)..........33
6.6.1. General ........................................................................................................................33
6.6.3. Requirements...............................................................................................................34
6.7. SHORT DURATION (PEAK) ACOUSTIC PRESSURE .........................................................34
6.7.1. General ........................................................................................................................34
6.7.3. Requirements...............................................................................................................35
ii
TIA/EIA-810-A
7. HANDSFREE TECHNICAL REQUIREMENTS (ADVISORY) ...................................... 36
7.1. HANDSFREE FREQUENCY RESPONSE ............................................................................ 36
7.1.1. Handsfree Send Frequency Response......................................................................... 36
7.1.2. Handsfree Receive Frequency Response .................................................................... 38
7.2. HANDSFREE LOUDNESS RATINGS AND RECEIVE VOLUME CONTROL ..................... 40
7.2.1. Handsfree Send Loudness Rating ............................................................................... 40
7.2.2. Handsfree Receive Loudness Rating .......................................................................... 40
7.2.3. Handsfree Receive Volume Control ........................................................................... 40
7.3. HANDSFREE NOISE .......................................................................................................... 40
7.3.1. Handsfree Send Noise................................................................................................. 40
7.3.2. Handsfree Send Single Frequency Interference.......................................................... 41
7.3.3. Handsfree Receive Noise............................................................................................ 41
7.3.4. Handsfree Receive Single Frequency Interference..................................................... 42
7.4. HANDSFREE DISTORTION AND NOISE ........................................................................... 42
7.4.1. Handsfree Send Distortion and Noise......................................................................... 42
7.4.2. Handsfree Receive Distortion and Noise.................................................................... 43
7.5. WEIGHTED TERMINAL COUPLING LOSS (TCLW) ........................................................ 43
7.5.2. Requirements .............................................................................................................. 43
7.6. STABILITY LOSS ............................................................................................................... 44
7.6.2. Requirement ................................................................................................................ 44
8. QUALITY OF SERVICE (ADVISORY) .............................................................................. 45
8.1. EQUIPMENT IMPAIRMENT FACTOR, IE .......................................................................... 45
8.2. PACKET LOSS.................................................................................................................... 45
8.3. QOS .................................................................................................................................... 47
ANNEX A (INFORMATIVE) – CALCULATION OF LOUDNESS RATINGS......................... 50
ANNEX B (INFORMATIVE) – MEASUREMENT AND LEVEL CONVERSIONS................. 53
ANNEX C (INFORMATIVE) – PREFERRED 1/12 OCTAVE FREQUENCIES....................... 55
iii
TIA/EIA-810-A
FOREWORD
(This foreword is not part of this standard.)
This document is a TIA/EIA Telecommunications standard produced by Working Group

TR-41.3.3 of Committee TR-41. This standard was developed in accordance with TIA/EIA
procedural guidelines, and represents the consensus position of the Working Group and its parent
Subcommittee TR-41.3, which served as the formulating group. This standard is based on
TIA/EIA/IS-810.
The TR-41.3.3 VoIP/PCM Transmission Performance Working Group acknowledges the

contribution made by the following individuals in the development of this standard.
Name Representing
Ron Magnuson Siemens Chair
Roger Britt Nortel Networks Editor
John Bareham Consultant in Electroacoustics
Kevin Cross Malden Electronics Ltd.
Steve Graham Nortel Networks
Phil Holland Circa Communications Ltd.
Michael Knappe Cisco Systems
Ken Simpson Simon Fraser University Engineering Student
Stephen Whitesell Lucent Technologies TR-41.3 Chair
Allen Woo Plantronics
Bob Young Bob Young Associates
Copyrighted parts of ITU-T Appendix I to Recommendation G.113 and Recommendation P.79 are
used with permission of the ITU. The ITU owns the copyright for the ITU Recommendations.
Copyrighted parts of ISO 3 are used with permission of the ISO. The ISO owns the copyright for the
ISO Standards.
The three annexes in this Standard are informative and are not considered part of this Standard.
Suggestions for improvement of this standard are welcome. They should be sent to:
Telecommunications Industry Association

Engineering Department
Suite 300
250 Wilson Boulevard
Arlington, VA 22201
iv
TIA/EIA-810-A
1. Introduction
This revision of TIA/EIA/IS-810 establishes handset, headset and handsfree telephone audio
performance requirements for digital wireline telephones regardless protocol or digital format. A
number of improvements and corrections have been made, particularly related to single frequency
interference and acoustic pressure.
This standard only addresses conventional narrowband performance, where narrowband is defined as
the frequency range between 300 and 3400 Hz. Wideband telephony, in the frequency range between
150 and 6800 Hz, is an enhancement that is likely to offered by VoIP telephones. The performance
requirements of wideband telephony will be addressed in a future TIA/EIA standard.
1
TIA/EIA-810-A
2. Scope
This standard establishes voice performance requirements for narrowband digital wireline telephones
with codecs that conform to the ITU-T G-Series Recommendations and where transmission is in
digital format. A telephone is defined as a device that terminates networks and provides telephony
voice service. Transmission may be over Local Area Networks, Firewire/IEEE1394, Universal Serial
Bus (USB), public ISDN or digital over twisted pair wire. Applications include Voice over Internet
Protocol (VoIP) and PCM-based telephones, whether connected through modems, gateways, or PBXs
and personal computer-based telephones that may or may not have handsets.
Technical requirements are set for handset, headset and handsfree (speakerphone) modes of
operation. Quality of Service is also addressed in Section 8. These requirements apply regardless of
the technology used to couple the handset or headset to the telephone. Coupling may be by a cord, a
short range air interface such as, but not limited to, a radio interface, an electric field interface, a
magnetic field interface or an infra-red interface.
2.1. Limits of Applicability

These requirements are not intended to describe specific requirements for the following types of
digital voice terminal equipment: telephones with carbon transmitters, ISDN terminal adapters and
cellular voice terminals.
The loudness ratings in this standard intentionally differ from loss plan published in the PBX
standard ANSI/TIA/EIA-464-B. At the time of approval of this standard, Project, PN-3673, was
active to revise the ANSI/TIA/EIA-464-B loss plan to agree with this standard and extend its
applicability to IP Gateways. Mixing digital telephone loudness ratings and PBX/Gateway loss plans
may not provide optimum performance.
2.2. Categories of Criteria

Mandatory requirements are designated by the word "shall". Advisory requirements are designated
by the word "should," or "may," or "desirable" which are used interchangeably in this standard.
Advisory criteria represent product goals or are included in an effort to ensure universal product
compatibility. Where both a mandatory and an advisory level are specified for the same criterion, the
advisory level represents a goal currently identifiable as having distinct compatibility or performance
advantages toward which future designs should strive.
2.3. FCC Part 68

This standard is intended to be in conformity with Part 68 of the Federal Communications
Commission (FCC) Rules and Regulations, but is not limited to the scope of those rules. In the event
that Part 68 requirements are more stringent than those contained in this standard, the provisions of
Part 68 apply.
2.4. Environmental
The telephone will also be subject to the applicable environmental conditions specified in EIA/TIA-
571.
2.5. Safety
This standard does not contain safety requirements. Compliance with the applicable UL and CSA
safety standards may be required in certain locations.
2
TIA/EIA-810-A
3. Normative References
The following standards contain provisions, which, through reference in this text, constitute
provisions of this Standard. At the time of publication, the editions indicated were valid. All
standards are subject to revision, and parties to agreements based on this Standard are encouraged to
investigate the possibility of applying the most recent editions of the standards indicated below, or
their successors. ANSI and TIA maintain registers of currently valid national standards published by
them.
[1] ANSI/TIA/EIA-464-B-1996, Requirements for Private Branch Exchange (PBX) Switching

Equipment.
[2] ANSI/TIA/EIA-504-A-1997, Telecommunications – Telephone Terminal Equipment –

Magnetic Field and Acoustic Gain Requirements for Handset Telephones Intended for use by
the Hard of Hearing.
[3] ANSI/EIA/TIA-571-1991, Environmental Considerations for Telephone Terminals.
[4] ANSI/TIA/EIA-579-A-1998, Telecommunications – Telephone Terminal Equipment –

Transmission Requirements for Digital Wireline Telephones.
[5] ANSI/IEEE Standard 269-1992, Standard Methods for Measuring Transmission

Performance of Analog and Digital Telephone Sets.
[6] ANSI/IEEE Standard 661-1979 (Reaff 1998), Standard Method for Determining Objective
Loudness Ratings of Telephone Connections.
[7] ANSI/IEEE Standard 1206-1994, Standard Methods for Measuring Transmission

Performance of Telephone Handsets and Headsets.
[8] ANSI/IEEE Standard 1329-1999, Standard Method for Measuring Transmission

Performance of Hands-Free Telephone Sets.
[9] ANSI S1.4-1990, Sound Level Meters.
[10] 47 CFR Part 68, Connection of Terminal Equipment to the Telephone Network.
[11] ITU-T Recommendation G.107 (1998), The E-Model, A Computational Model for use in
Transmission Planning.
[12] ITU-T Recommendation G.109 (1999), Definition of categories of speech transmission

quality.
[13] ITU-T Recommendation G.113 (1996), Transmission impairments.
[14] ITU-T Appendix I to Recommendation G.113 (1998), Transmission impairments – Appendix

I: Provisional planning values for the equipment impairment factor Ie.
[15] ITU-T Recommendation G.114 (1996), One-way transmission time.
[16] ITU-T Recommendation G.122 (1993), Loudness ratings (LRs) of national systems.
[17] ITU-T Recommendation G.131 (1993), Control of talker echo.
3
TIA/EIA-810-A
[18] ITU-T Recommendation G.175 (1997), Transmission planning for private/public network
interconnection of voice traffic.
[19] ITU-T Recommendation G.711 (1988), Pulse code Modulation (PCM) of voice frequencies.
[20] ITU-T Recommendation G.712 (1996), Transmission performance characteristics of pulse

code modulation.
[21] ITU-T Recommendation G.723.1 (1996), Dual rate speech coder for multimedia
communications transmitting at 5.3 and 6.3 kbit/s.
[22] ITU-T Recommendation G.729 (1996), Coding of speech at 8 kbit/s using conjugate-
structure algebraic-code-excited linear-prediction (CS-ACELP).
[23] ITU-T Recommendation O.41 (1994), Psophometer for use on telephone-type circuits.
[24] ITU-T Recommendation O.131 (1988), Quantizing distortion measuring equipment using a
pseudo-random noise test signal.
[25] ITU-T Recommendation P.51 (1996), Artificial mouth.
[26] ITU-T Recommendation P.56 (1993), Objective measurement of active speech level.
[27] ITU-T Recommendation P.57 (1996), Artificial ears.
[28] ITU-T Recommendation P.58 (1996), Head and torso simulator for telephonometry.
[29] ITU-T Recommendation P.64 (1999), Determination of sensitivity/frequency characteristics

of local telephone systems.
[30] ITU-T Recommendation P.79 (1999), Calculation of loudness ratings for telephone sets.
[31] ITU-T Recommendation P.310 (1996), Transmission characteristics for telephone band (300
- 3400 Hz) digital telephones.
[32] ITU-T Recommendation P.360 (1998), Efficiency of devices for preventing the occurrence of
excessive acoustic pressure by telephone receivers.
[33] ITU-T Recommendation P.501 (1996), Test signals for use in telephonometry.
[34] ISO 3: 1973 Preferred numbers - Series of preferred numbers.
[35] ETSI EG 201 377-1 (1999), Specification and measurement of speech transmission quality;
Part 1: Introduction to objective comparison measurement methods for one-way speech
quality across networks.
4
TIA/EIA-810-A
4. Definitions, Abbreviations and Acronyms

For the purposes of this Standard, the following definitions apply.
4.1. Codec
A codec is a combination of an analog-to-digital encoder and a digital-to-analog decoder operating in
opposite directions of transmission in the same equipment.
4.2. Ear Reference Point (ERP)

A virtual point for geometric reference located at the entrance to the listener's ear, traditionally used
for calculating telephonometric loudness ratings.
4.3. HATS Position

The HATS (head and torso simulator) position (ITU-T P.64 Annex D and Annex E) is the correct
artificial head handset position for measuring sensitivity and frequency response characteristics. The
HATS position has been shown to be essentially identical to the LRGP (loudness rating guard-ring
position) position, except for the mouth simulator direction, which has been corrected with a 19
degrees downwards rotation to more closely match real talkers. For handsets with omnidirectional
microphones, measurements on the two heads may differ slightly, typically less than 1dB. For
handsets with directional or noise-canceling microphones, the differences will be larger, and the
HATS position will give the more realistic results. Some equipment may use the term “LRGP-H” for
the HATS position.
4.4. Mouth Reference Point (MRP)

The mouth reference point is located on axis and 25 mm in front of the lip ring of a mouth simulator.
4.5. Quiet and Full Scale Code
Table 1 – PCM Codes for Zero (Quiet Code) and Full Scale
Mu-Law A-Law
Level
Sign Bit Chord Bit Step Bits Sign Bit Chord Bits Step Bits
+ Full Scale 1 000 0000 1 010 1010
+ Zero 1 111 1111 1 101 0101
- Zero 0 111 1111 0 101 0101
- Full Scale 0 000 0000 0 010 1010
4.6. Reference Codec

A reference codec is used for testing digital telephone terminals with analog test equipment. Figure 1
shows the basic test setup using a reference codec. A codec that approaches an ideal codec and has
superior, well-defined, characteristics qualifies as a reference codec.
When a 0.775 volt rms analog signal is applied to the coder input, a 0 dBm0 digital code is present at
the digital reference. When a 0 dBm0 digital code is applied to the decoder, a 0.775 volt rms analog
signal appears at the decoder output. At the digital reference point 0 dBm0 is 3.14 (A-law) or 3.17
(mu-law) dB below digital full scale.
5
TIA/EIA-810-A
This implementation of a reference codec eliminates the 600 ohm source and load resistors specified
by other standards. The coder input impedance is high relative to the generator and the decoder
output impedance is low relative to the measuring voltmeter.
The interface block, shown in Figures 1 and 2, passes the voice channel digital bit stream to the
terminal without modification. There is no gain or loss in the send or receive direction due to the
interface. If the interface does change the digital voice stream then the terminal and interface shall be
considered jointly as the terminal. An example of this is a receive volume control implemented in a
PBX or gateway.
4.7. Direct Digital Processing

Direct digital generation of the receive signal and analysis of the send signal may be used in place of
the reference codec as shown in Figure 2. This method is preferred when possible.
Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec
Digital Reference
Point
(Junction j)
Send
vS E N D
pM Decoder v
Mouth Sound Pressure
at MRP Digital
Interface
Set
pE Coder GEN
Ear Sound Pressure
at ERP
vRCV
Receive Reference Codec
Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Generation and
Analysis
Digital Reference
Point
(Junction j)
Send
pM Digital
Analysis
Mouth Sound Pressure
at MRP Digital
Interface
Set
pE Digital
Generation
Ear Sound Pressure
at ERP
Receive
6
TIA/EIA-810-A
4.8. Sound Pressure Levels

Sound pressure level is a value expressed as a ratio of the pressure of a sound to a reference pressure.
The following sound level units are used in this standard:
dBPa: The sound pressure level, in decibels of a sound is 20 times the logarithm to the base 10 of
the ratio of the pressure of this sound to the reference pressure of
1 Pascal (Pa). Note: 1 Pa = 1 N/m2.
dBSPL: The sound pressure level, in decibels of a sound is 20 times the logarithm to the base 10 of
the ratio of the pressure of this sound to the reference pressure of
2 X 10-5 N/m2 (0 dBPa = 94 dBSPL).
dBA: The A-weighted sound level is the sound pressure level in dBSPL, weighted by use of
metering characteristics and A-weighting specified in ANSI S1.4.
4.9. Electric Power and Noise Levels

The following electric power and noise level units are used in this standard:
dBm0: The absolute power level at a digital reference point of the same signal that would be
measured as the absolute power level, in dBm, if the reference point was analog. The
absolute power in dBm is defined as 10 log (power in mW / 1 mW). When the impedance
is 600 ohm resistive, dBm can be referred to a voltage of 0.775 volts, which results in a
reference active power of 1 mW. Note that 0 dBm0 is not the maximum digital code. For
Mu Law codecs 0 dBm0 is 3.17 dB below digital full scale. For A Law codecs 0 dBm0 is
3.14 dB below digital full scale.
dBm0p: The noise level, measured by a psophometer with a special noise weighting filter as
described in ITU-T Recommendations O.41 and P.53. The small letter “p” comes from the
French word “ponderé”. The equivalent English word is “weighted”, but the “p” refers
specifically to psophometric weighting.
4.10. Abbreviations and Acronyms

Abbreviations and acronyms, other than in common usage, which appear in this standard, are defined
below.
AGC Automatic Gain Control

DRP Drum Reference Point
EFR Enhanced Full Rate
ERP Ear Reference Point
FFT Fast Fourier Transform
GoB Good or Better
HATS Head and Torso Simulator
Ie Equipment Impairment factor
ISDN Integrated Services Digital Network
LRGP Loudness Rating Guard-ring Position
MOS Mean Opinion Score
MRP Mouth Reference Point
OLR Objective Loudness Rating
PBX Private Branch Exchange
PCM Pulse Code Modulation
7
TIA/EIA-810-A
PL Packet Loss
PLC Packet Loss Concealment
PoW Poor or Worse
QoS Quality of Service
RLR Receive Loudness Rating
RTP Recommended Test Position
SLR Send Loudness Rating
STMR Sidetone Masking Rating
TCLt Temporally weighted Terminal Coupling Loss
TCLw Weighted Terminal Coupling Loss
VAD Voice Activity Detector
VoIP Voice over Internet Protocol
8
TIA/EIA-810-A
5. Handset Technical Requirements

All telephones shall support G.711 A-law and mu-law. The handset technical requirements apply
only to mu-law and A-law G.711 codecs. If the telephone uses other G-Series low-bit rate vocoders,
the manufacturer must ensure that their implementation passes the standard test vectors associated
with that codec. For bit exact vocoders, such as G.729, it is important to ensure that vector testing has
been performed and found to be compliant with the associated ITU requirement.
Unless specified otherwise:
• Encoding and decoding is assuming to be G.711 in mu-law. In particular, this applies to the
requirements in the Distortion and Noise sections.
Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test results or
require test signals other than sinewaves. ITU-T Recommendation P.64 allows several types of test
signals. The test signal used should be stated. The test signal levels specified in this standard shall be
used. Test signal levels that differ from those specified in this standard may also be required.
Packet voice latency may introduce significant delay that must be accounted for by the test
equipment.
Equipment using nonlinear voice signal processing may require subjective testing.
Suitable artificial ears for tests involving the handset receiver are documented in ITU-T
Recommendation P.57. The correct artificial ear is selected by the size, ear coupling method,
impedance and bandwidth characteristics of the device under test. All tests involving the handset
receiver shall be done with the same artificial ear. All test reports shall document the model of
artificial ear used in the tests.
5.1. Handset Frequency Response

5.1.1. Handset Send Frequency Response
Send frequency response is the ratio of the voltage output of the reference codec to the sound
pressure at the Mouth Reference Point (MRP) for each frequency or frequency band (Fi) as shown in
the equation below:
SMJ = 20 log (VSEND / PM) dB rel 1 V / Pa [1]
Where
SMJ Send Sensitivity, Mouth to Junction, at Fi.
PM Sound pressure at the MRP at Fi.
VSEND RMS output voltage of the reference codec at Fi.
5.1.1.1. Measurement Method

The handset shall be mounted in the HATS position. Measurements should be done in ISO 1/12
octave intervals or smaller, over a minimum range of 100 Hz through 4000 Hz. For loudness
calculations ISO 1/3 octave data is required. See Annex C.
The send frequency response is measured according to ITU-T Recommendation P.64 using the
measurement set-up shown in Figure 3. The test signal level shall be -4.7 dBPa at the MRP.
9
TIA/EIA-810-A
Figure 3 – Handset Send Frequency Response Measurement Method
vS E N D
Send
pM Decoder v
Measuring
GEN
Digital
Interface Amplifier
Set
Mouth Simulator Coder
Quiet Room
Reference Codec
5.1.1.2. Requirement
The send frequency response shall be below the upper limit and above the lower limit defined in
Table 2 and shown in Figure 4. The limit curves shall be determined by straight lines joining
successive co-ordinates given in the table, when frequency response is plotted on a linear scale
against frequency on a logarithmic scale. Note: The frequency response mask is a floating or “best
fit” mask.
Table 2 – Co-ordinates of Handset Send Response
Limit Curve Frequency Send Response Limit

(Hz) (dB) [arbitrary level]
upper limit 100 -7
150 0
1000 0
2000 +5
3400 +5
4000 0
lower limit 300 - infinity
300 -8
500 -6
3000 -6
3400 -9
3400 -infinity
10
TIA/EIA-810-A
Figure 4 – Handset Send Frequency Response Mask
10
Arbitrary Level (dB)
-10
-20
100 1000 10000
Frequency (Hz)
5.1.2. Handset Receive Frequency Response

Receive frequency response is the ratio of the sound pressure measured in the ear simulator to the
voltage input to the reference codec for each frequency or frequency band (Fi) as shown in the
equation below:
SJE = 20 log (PE / VRCV) dB rel 1 Pa / V [2]
Where
SJE Receive Sensitivity, Junction to Ear, at Fi.
PE ERP Sound pressure measured by ear simulator at Fi. Measurements collected
at other points, e.g., DRP and free field, must be corrected back to ERP.
VRCV RMS Input voltage to the reference codec at Fi.

Mount the handset with the receiver coupled to the artificial ear. The receive frequency response is
measured according to ITU-T Recommendation P.64 using the measurement set-up shown in Figure
5. Measurements should be done in ISO 1/12 octave intervals or smaller, over a minimum range of
100 Hz through 8000 Hz. For loudness calculations ISO 1/3 octave data is required. See Annex C.
The test signal level shall be -18.2 dBV (-16 dBm0). Telephone sets with adjustable receive levels
shall be adjusted so that their RLR is as close as possible to the nominal value of Section 5.2.2.2 for
this test.
11
TIA/EIA-810-A
Figure 5 – Handset Receive Frequency Response Measurement Method
Receive pE
Decoder
Ear
Simulator
Digital
Interface
Set
Sound Pressure
Measuring
Coder GEN
Amplifier
vRCV
Quiet Room Reference Codec
The receive frequency response shall be below the upper limit and above the lower limit defined in
fit” mask.
Table 3 – Co-ordinates of Handset Receive Response Limits
Limit Curve Frequency Receive Response Limit

upper limit 100 -7
170 +2
300 +2
1000 0
3000 +2
4000 +2
8000 -18
300 -7
500 -5
3000 -5
3400 -8
3400 -infinity
12
TIA/EIA-810-A
Figure 6 – Handset Receive Frequency Response Mask
10
-10
-20
100 1000 10000
Frequency (Hz)
5.2. Handset Loudness Ratings and Receive Volume Control

5.2.1. Handset Send Loudness Rating (SLR)
The SLR for a digital telephone set is the conversion ratio of a defined acoustic signal at the mouth
reference point to the transmit signal at the digital reference point. Refer to Annex A and ITU-T
Recommendation P.79.

The SLR shall be calculated using the 1/3 octave sensitivity data collected from the send frequency
response measurement. Use equation [A1] of Annex A and bands 4 to 17, Table 18.
The terminal should be designed to have a nominal SLR value of 8 dB, with a tolerance of ±4.0 dB.
5.2.2. Handset Receive Loudness Rating

The RLR for a digital telephone set is the conversion ratio of a defined acoustic signal at the receive
digital reference point to an acoustic output signal from the receiver. Refer to Annex A and ITU-T
Recommendation P.79.

The RLR shall be calculated from the 1/3 octave sensitivity data collected from the receive frequency
13
TIA/EIA-810-A
The terminal should be designed to have a nominal RLR value of 2 dB, with a tolerance of ±4.0 dB.
5.2.3. Handset Receive Volume Control

The test method and acoustic output requirements for receive-amplified handset telephones are
documented in ANSI/TIA/EIA-504-A. This standard includes the magnetic output requirements for
handset telephones intended for use by the hard of hearing.
5.2.4. Handset Talker Sidetone

The sidetone masking rating (STMR) for a digital telephone set is the ratio of a defined input
acoustic signal at the mouth reference point to the resulting acoustic output signal from the receiver.

Mount the handset in the HATS position with the receiver coupled to the artificial ear. The test signal
level at the MRP shall be -4.7 dBPa. For each frequency given in Table 18, bands 1 to 20, the sound
pressure in the artificial ear shall be measured. The STMR shall be calculated using equation [A3] of
Annex A.
Telephone sets with adjustable receive levels shall be tested at the minimum, nominal and maximum
settings. For the nominal setting, adjust the level so that the RLR is as close as possible to the
nominal RLR value.
The value of STMR shall be within the range of 18 dB ± 6 dB, for any adjustable receive level.
5.3. Handset Noise

The noise levels are related to the SLR and RLR requirements.
5.3.1. Handset Send Noise

5.3.1.1. General
The send noise of a digital telephone is the 5 second average background noise at the digital transmit
output with the telephone transmitter isolated from sound input and mechanical disturbances.

With the handset in the HATS position with the receiver coupled to the artificial ear in a quiet
environment (ambient noise less than 30 dBA), measure the noise level at the digital interface output
or the reference codec decoder output with apparatus that includes psophometric weighting,
according to ITU-T Recommendation 0.41.
The send noise shall be less than -68 dBm0p.
5.3.2. Handset Send Single Frequency Interference

5.3.2.1. General
Narrow-Band noise, including single frequency interference, is an impairment that can be perceived
as a tone depending on its level relative to the overall weighted noise level. This test measures the
14
TIA/EIA-810-A
weighted noise level characteristics in narrow bands of not more than 31 Hz, which can then be
compared to the overall weighted background noise level.

With the handset in the HATS position and the receiver coupled to the ear simulator in a quiet
environment (ambient noise less than 30 dBA), measure the psophometrically-weighted noise level at
VSEND with a selective voltmeter or spectrum analyzer with an effective bandwidth of not more than
31 Hz, over the frequency range of 100 to 3500 Hz. If FFT analysis is used, then “Flat Top”
windowing shall be employed.
The send single frequency interference shall be less than -78 dBm0p.
5.3.3. Handset Receive Noise

5.3.3.1. General
The receive noise of a digital telephone is the short-term average background noise level measured at
the output of the telephone receiver with the digital telephone receiving the digital quiet code.

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. A signal
corresponding to a decoder value quiet code is applied at the digital interface. The A-weighted noise
level is measured in the artificial ear. The ambient noise for this measurement shall not exceed
30 dBA.
Telephone sets with adjustable receive levels shall be adjusted so that the RLR is as close as possible
to the nominal RLR value when driven by quiet code.
The receive noise shall be less than 38 dBA.
5.3.4. Handset Receive Single Frequency Interference

5.3.4.1. General
compared to the overall weighted background noise level. Narrow-band noise is measured at the
output of the telephone receiver with the digital telephone receiving the digital quiet code.

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. A signal
corresponding to a decoder quiet code is applied at the digital interface. The A-weighted noise level
is measured in the artificial ear with a selective voltmeter or spectrum analyzer, with an effective
bandwidth of not more then 31 Hz, over the frequency range of 100 to 8100 Hz. If FFT analysis is
used, then “Flat Top” windowing shall be employed. The ambient noise for this measurement shall
not exceed 30 dBA.
15
TIA/EIA-810-A
The receive single frequency interference shall be 10 dB quieter than the A-weighted broadband
noise floor.
5.4. Handset Receive Comfort Noise (Advisory)

If comfort noise is introduced to replace actual background noise the level should roughly match the
loudness of the original background noise. There is more likely to be annoyance if the comfort noise
is greater than the original noise than if it is less.
5.4.1. General
The receive comfort noise of a digital telephone is the short-term average background noise level
measured at the output of the telephone receiver with the digital telephone receiving either a silence
indication packet from the transmitting telephone or no packets from the transmitting telephone for
some non-transient period of time.
5.4.2. Measurement Method

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. The
receive volume control is adjusted as close as possible to the nominal RLR value. The digital
interface is sent the quiet code  the code that represents silence for the coder format.
With both VAD disabled at the transmitting source and comfort noise generation on the receiving
unit under test turned off, a white noise test signal should be sent from the transmitting end such that
the receive noise level measured at the receiving telephone is 48 dBA. This test signal at this level
will be assigned the level of ‘N dB’ as a calibrated point for the purpose of the comfort noise test,
since it may be generated either as an acoustic signal at a ‘golden’ transmitting telephone (and
measured in dBA) or injected digitally (and measured in dBm0p).
The following test sequence must be followed for all calibrated test noise levels of ‘M dB’, which
will range from N-10 to N+10 dB.
1. The echo canceller at both ends should be disabled.
2. 10 seconds of silence (or idle code) is inserted at the transmitting point.
3. 300-3400 Hz band-limited white noise of level M dB is inserted at the transmitting point for 130
seconds.
4. During the final 10 seconds of level M noise insertion, the acoustic noise level at the receive will
be measured.
5. Steps 2-4 are repeated for varying M in 1 dB gradations.
5.4.3. Requirement
For all input noise levels M in the range of N-10 to N+10, with N calibrated to give 48 dBA receive
noise levels at the receiver, acoustic noise levels measured at the receiver must be within +0.5/-3.0
dB of the expected acoustic receive noise level for that input. This expected receive noise level for
any given M and N would be 48 dBA – (N-M).
5.5. Handset Distortion and Noise

The distortion and noise requirements only apply to G.711 codecs in mu-law.
16
TIA/EIA-810-A
5.5.1. Handset Send Distortion and Noise

5.5.1.1. Method of Measurement
The distortion and noise is measured according to IEEE 1329, 9.3.6. Apply a sinewave signal at the
MRP, with the levels given in Table 4 and the following frequencies: 315, 502, 803 and 1004 Hz.
The ratio of the signal-to-total distortion and noise power of the digitally encoded signal output is
measured. The test frequency tolerance is 3%, but even submultiples of the sampling frequency must
not be used.
Note: In cases where the sound pressure exceeds +6 dBPa, the linearity of the artificial mouth should
be checked, as it exceeds the limits of ITU-T Recommendation P.51.
The ratio of signal-to-total distortion and noise power of the digitally encoded signal output shall be
above the limits given in Table 4. Limits for intermediate levels are found by drawing straight lines
between the breaking points in the table on a linear (dB signal level) – linear (dB ratio) scale.
Table 4 – Handset Send Signal-to-Total Distortion and Noise Ratio Limits
Send level at the MRP Send Ratio

(dBPa) (dB)
-30 20
-24 25
-17 31
-10 33
0 33
+4 33
+8 24
+10 20
5.5.2. Handset Receive Distortion and Noise

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. The
distortion and noise is measured according to IEEE 1329, 9.4.6. Apply a digitally simulated
sinewave, with the levels given in Table 5 and the following frequencies: 315, 502, 803 and 1004 Hz.
The ratio of signal-to-total distortion and noise power is measured. The test frequency tolerance is
3%, but even submultiples of the sampling frequency must not be used.
to the nominal RLR value.
The ratio of signal-to-total distortion and noise power measured in the artificial ear, with A-
weighting applied, shall be above the limits given in Table 5, unless the signal in the artificial ear
exceeds +10 dBPa or is less than –50 dBPa.
17
TIA/EIA-810-A
Table 5 – Handset Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the digital Receive Ratio Receive Ratio

interface @ 315 Hz @ 502, 803 and 1004 Hz
(dBm0) (dB) (dB)
-40 19 20
-34 24 25
-27 30 31
-20 32 33
-10 32 33
-6 32 33
-3 28 29
0 23 24
5.6. Weighted Terminal Coupling Loss (TCLw)

The weighted terminal coupling loss (TCLw) provides a measure of the echo performance under
normal conversation, i.e., single far-end talker conditions. It is possible that echo control devices
such as echo suppressors or echo cancellation with non-linear processing may be used on handset
connections to provide sufficient echo return loss to mitigate increased echo notice-ability associated
with longer network delays.
The use of echo control devices on the handset can affect the measurement of TCLw. The result
would likely be different under cases of either single far-end talker or double-talk. The TCLw
measurement is intended to represent a single far-end talker. This may provide idealized and
unrealistic performance measurements when non-linear processing on the transmit side is used as
part of the echo control algorithm. It may be more appropriate to measure TCLw either with non-
linear processing disabled or with a near-end signal present that is a) capable of enabling echo
control’s double-talk detector with the subsequent removal of non-linear processing and b) can be
filtered out from the final return signal so as not to affect the accuracy of the TCLw measurement.
The latter may be the only method that can used consistently across products in a black-box testing
setup. A suitable signal may be a pulsed sine wave, but will depend on the temporal characteristics of
the double-talk detector.
The ‘proper’ measurement of TCLw then becomes specific to the echo control implementation.
These issues are still under study and are not addressed in these requirements. For further
information see IEEE 1329, Clause 11.

TCLw is measured in free-air in such a way that the inherent mechanical coupling of the handset is
not affected. The TCLw measurement shall be made at an input signal level of -16 dBm0. The test
shall be performed with the handset suspended in a noose around the earcap with the handset cord
hanging freely below the handset.
For devices that incorporate non-linear processes, additional measurements using signal levels of
-26 dBm0 and -10 dBm0 may be performed.
Noise and reflections in the test space must not influence the measurement. The test should be
performed in an anechoic chamber with the handset positioned at least 50 cm away from the nearest
part of the test chamber. The ambient noise level shall be less than 30 dBA.
18
TIA/EIA-810-A
The test signal is white noise, band limited to 100 through 4000 Hz, and modulated at a rate of 250
ms ON and 150 ms OFF. The measurement and calibration shall be determined during the ON
portions of the signal. Sine wave signals may be used with G.711 codecs.
The attenuation from digital input to digital output is measured at 1/12 octave frequencies as given
by the R.40-series of preferred numbers in ISO 3 for frequencies from 290 to 3255 Hz, using the
measurement arrangement shown in Figure 7. See Annex C.
The weighted terminal coupling loss is calculated according to ITU-T Recommendation G.122
(1993) Annex B, Section B.4 (trapezoidal rule).
Telephone sets with adjustable receive levels shall be tested at the nominal setting. For the nominal
setting, adjust the level so that the RLR is as close as possible to the nominal RLR value.
Figure 7 – Terminal Coupling Loss Measurement Method
Handset Suspended
vS E N D ( E c h o R e t u r n )
Decoder v
Digital
Interface
Set
Coder GEN
vRCV
Anechoic Chamber Reference Codec
5.6.2. Requirements
The normalized value of TCLw loss shall be greater than 52 dB for IP sets and 45 dB for PCM sets
when measured under free field conditions and with SLR normalized to 8 dB and RLR normalized to
2 dB. It is desirable that the normalized value of TCLw for IP sets to be greater than 55 dB and that
the normalized value of TCLw for PCM sets be greater than 50 dB to meet ITU-T Recommendation
G.131 talker echo objective requirements.
For example, if the measured TCLw is 48 dB, the measured SLR is 11 dB and the measured RLR is
0 dB, then the normalized value of TCLw = 48 dB + (8 - 11) dB + (2 - 0) dB = 47 dB.
NOTE 1: If equipped with adjustable receive level, the TCLw will decrease in proportion with the
increased gain relative to the nominal RLR in most cases. For example, if the measured
TCLw is 45 dB at nominal RLR and the adjustable receive level adds 12 dB of gain, then
TCLw (maximum receive level) = 45 dB - 12 dB = 33 dB.
NOTE 2: The echo impairment perceived by the person at the opposite end of the connection from a
telephone set is a function of the magnitude of the talker echo signal as well as the talker
echo path delay. The echo signal becomes more disturbing as the talker echo path delay
increases. Thus, a telephone set with adequate TCLw performance on low delay
19
TIA/EIA-810-A
connections may provide satisfactory performance while the same may not be true for
connections that have a long delay.
NOTE 3: Temporally weighted terminal coupling loss (TCLt) is an alternate method for echo
measurement, which may be more subjectively relevant, especially in devices with echo
suppression or cancellation features. (See IEEE Standard 1329-1999.) The performance
requirements may need to be changed when using this method. This issue is currently
under study.
5.7. Stability Loss

The stability loss is a measure of the contribution of the telephone set to the overall network stability
requirements. Stability loss is defined as the minimum loss from the digital input (receive) to the
digital output (send), at any test frequency.

The stability measurement shall be made at an input signal level of -16 dBm0, at 1/12 octave bands
centered at 205 Hz to 3868 Hz. The test signal is white noise, band limited to 100 through 4000 Hz,
and modulated at a rate of 250 ms ON and 150 ms OFF. The measurement and calibration shall be
determined during the ON portions of the signal. Sine wave signals may be used with G.711 codecs.
With the handset and transmission circuit fully active, measure the attenuation from the digital input
to the digital output using Method 1 and Method 2. See Annex C.
For devices that incorporated non-linear processes, additional measurements using signal levels of
–26 dBm0 and –10 dBm0 may be performed.
5.7.1.1. Method 1
Place the handset in the reference corner, as shown in Figure 8, with the earcap and mouthpiece
facing a hard, smooth surface. The handset shall be placed along the diagonal from the apex of the
reference corner to the outside corner, with the earcap end of the handset 250 mm from the apex. The
telephone set shall be fully active.
The reference corner consists of three perpendicular plane, smooth, hard surfaces extending 0.5 m
from the apex of the corner.
5.7.1.2. Method 2
Place the handset with the earcap and mouthpiece facing a hard, smooth surface free of any other
object for 0.5 m. The telephone set shall be fully active.
5.7.2. Requirement
The stability loss, i.e., minimum loss, at any frequency shall be greater than 6 dB. It is desirable that
this loss be greater than 10 dB.
Telephone sets with adjustable receive level should maintain stability over the entire range of
adjustable receive levels.
20
TIA/EIA-810-A
Figure 8 – Reference Corner
25 cm
50 cm
5.8. Long Duration Maximum Acoustic Pressure (Steady State Input)

5.8.1. General
The long duration acoustic pressure is the maximum, steady state sound pressure disturbance, greater
than 500 ms, emitted from a telephone receiver, caused by the maximum excursions of the receive
digital signal.
Additional consideration should be given to the acoustic pressure caused by tones, other audio
signals or long duration, high amplitude electrical signals applied to power, network, handset or
auxiliary leads of the digital telephone.

The steady-state A-weighted sound pressure level shall be measured using the digital terminals test
procedure in ITU-T Recommendation P.360, with the following modifications.
Apply a digital square wave to the receive input, switched between the maximum positive and the
maximum negative values (see Table 1), as defined in ITU-T Recommendation G.711. The switching
rate shall range from 1 Hz to 4000 Hz over a sweep time of not less than 30 seconds. The
measurement shall be made with a RMS detector set to 1 second effective averaging time (RMS
Slow).
Telephone sets with adjustable receive levels shall be adjusted to the maximum setting.
Maximum acoustic pressure limits can be referenced to the ERP, DRP, free field (0 degrees elevation
and azimuth) or to a diffuse field, as required by the appropriate safety standard. This may require
that measurements made at one reference point be translated to the required reference point.
21
TIA/EIA-810-A
For types 2, 3.1, 3.3 and 3.4 ear simulators, DRP to ERP transformation shall be performed by adding
the values in table 2b/P.57 of ITU-T Recommendation P.57 to the data measured at the DRP. For
type 3.2 ear simulators, DRP to ERP transformation shall be performed by using the transfer function
supplied by the manufacturer of the ear simulator.
Transformation to free field or diffuse field shall be made using the transfer function supplied by the
manufacturer of the ear simulator, if available. Alternatively, the transfer functions specified in ITU-
T P.58 may be used. Transfer functions with resolution of at least 1/12 octave or R40 format shall be
used if available.
Maximum acoustic pressure measurements shall be made on the same ear simulator and with the
same positioning and force as used for receive frequency response measurements. For handsets
measured on a type 3.3 or type 3.4 ear simulator, one additional measurement shall be made using a
force of 15 N. If both measurements fall below the applicable performance limit, the measurement
that most closely approaches the limit shall be taken as the final result. If one or both measurements
exceed the limit, the one that most exceeds the limit shall be taken as the final result.
5.8.3. Requirements
The requirements are currently under study.
5.9. Short Duration (Peak) Acoustic Pressure

5.9.1. General
The short duration acoustic pressure is the maximum short duration, sound pressure impulse, less
than 500 ms, from a telephone receiver, caused by the maximum excursions of the receive digital
signal.
This short duration test stresses nonlinear processes, like AGC, and doesn’t directly replace a short
duration surge. Additional consideration should be given to the peak acoustic pressure caused by
tones or short duration, high amplitude electrical pulses applied to power, network, handset or

The peak acoustic pressure level shall be measured using the digital terminals test procedure in ITU-
T Recommendation P.360, with the following modifications instead of applying electrical impulses
to the send and receive pairs. The short duration acoustic pressure shall be determined by applying
digital codes to the receive input. The codes shall be switched between the maximum positive and the
maximum negative values, defined in ITU-T Recommendation G.711. The switching rate shall range
from 2 Hz to 4000 Hz. The duration of the ON codes shall be a number of complete cycles
approximating but not exceeding 500 ms. The ON codes must be followed by a quiet interval of at
least 500 ms before repeating the codes, as shown in Figure 9.
Peak acoustic pressure limits can be referenced to the ERP, DRP, free field (0 degrees elevation and
azimuth) or to a diffuse field, as required by the appropriate safety standard. This may require that
measurements made at one reference point be translated to the required reference point. The
translations shall be implemented using a minimum phase, parametric filter (or equivalent) as peak
measurements must be made in real time. Both the magnitude and phase of the transfer function is
necessary to best preserve the waveshape for a proper measurement of its peak value.
The filter parameters for transformation from DRP to ERP shall be based on the transfer function
specified in Section 5.8.2. The filter parameters for transformation to free field or diffuse field shall
22
TIA/EIA-810-A
be made using the transfer function supplied by the manufacturer of the ear simulator, if available.
Alternatively, the transfer function specified in ITU-T P.58 can be used. Transfer functions with
resolution of at least 1/12 octave or R40 format shall be used if available.
The magnitude of the filter response shall follow the transfer function within a tolerance of ±1dB.
In practice, most devices will pass peak acoustic limits without the reduction provided by the
translation filter. In these cases, the filter may not be required.
Figure 9 – Short Duration Acoustic Pressure Test Signal
≤ 500 ms ≥ 500 ms
maximum positive
digital word
maximum negative
digital word
5.9.3. Requirements
5.10. Packet Voice Latency (Advisory)

Telephone sets that employ packet voice transmission may add significant delay to the voice signal
due to packetization, compression, de-jitter and queuing mechanisms. ITU-T G.114 recommends that
the maximum end-to-end latency should be less than 150 ms, to minimize the effects on the dynamics
of conversations. This section gives some guidelines for the delay engineering of the telephone set.
The requirements are advisory, but the intent is to ensure that any combination of different telephone
vendor send and receive latencies (which may not be symmetric) using G.711 will remain under 100
ms for the overall telephone component. This allows for an additional 50 ms of delay in the packet
network while still meeting G.114 recommendations. Note that this 100 ms requirement may not be
sufficient to achieve good overall latency in congested, voice compressed and/or multi-hop packet
networks and vendors are encouraged to minimize telephone set latency as much as possible.
23
TIA/EIA-810-A
5.10.1. Handset Send Latency

5.10.1.1. General
The send latency is defined here as the time measured from when an acoustic signal leaves an
artificial mouth playing into a telephone set’s handset to the time its digitized, packetized
representation arrives at that telephone’s packet network interface.

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. A digital
audio measuring device capable of measuring the delay between an injected signal (to the artificial
mouth) and a digitally transmitted signal should be connected to the artificial mouth and directly to
the network output of the telephone. All delays inherent in the measurement system itself must be
calibrated out. The telephone should be set to transmit G.711 packets.
An acoustic signal of –4.7 dBPa should be generated at the artificial mouth. The delay between the
time the pulse left the mouth to the time it was received at the telephone’s network interface should
be measured.
The send latency should be 35 ms or less with a maximum speech frame rate of 20 ms and with one
speech frame per packet.
5.10.2. Handset Receive Latency

5.10.2.1. General
The receive latency is defined here as the time measured from when a digitized, packetized
representation of a signal arrives at that telephone’s packet network interface to when its analog
reproduction is received at an artificial ear sealed to that telephone’s handset.

The handset is mounted in the HATS position and the receiver coupled to the artificial ear. A digital
audio measuring device capable of measuring the delay between an injected digital packet signal and
the output of an artificial ear should be connected to the network input of the telephone and to the
artificial ear. All delays inherent in the measurement system itself must be calibrated out. The
telephone should be set to receive G.711 packets.
A pulsed digital signal of –16 dBm0 should be injected as packets to the telephone’s network
interface. The delay between the time the packet was injected at the telephone network interface to
the time it was received at the artificial ear should be measured.
The receive latency should be 65 ms or less with a maximum speech frame rate of 20 ms and with
one speech frame per packet.
24
TIA/EIA-810-A
6. Headset Technical Requirements

All telephones shall support G.711 A-law and mu-law. The headset technical requirements apply
equipment.
Equipment using nonlinear voice signal processing may require subjective testing.
Suitable artificial ears for tests involving the headset receiver are documented in ITU-T
Recommendation P.57. The correct artificial ear is selected by the size, ear coupling method,
impedance and bandwidth characteristics of the device under test. All tests involving the handset
receiver shall be done with the same artificial ear. All test reports shall document the model of
artificial ear used in the tests.
The headset test method is given in IEEE 1206.
6.1. Headset Frequency Response

6.1.1. Headset Send Frequency Response
The send frequency response is the overall response of the transducer, send amplifier, and the codec
send filter. The send sensitivity is expressed in terms of dBV/Pa.

The send frequency response is measured according to IEEE 1206 using the measurement set-up
shown in Figure 3, substituting the handset with the headset. Measurements should be done in ISO
1/12 octave intervals or smaller, over a minimum range of 100 Hz through 4000 Hz. The test signal
level shall be -4.7 dBPa at the MRP.
The send frequency response shall be below the upper limit and above the lower limit defined in
fit” mask.
25
TIA/EIA-810-A
Table 6 – Co-ordinates of Headset Send Response Limits
Limit Curve Frequency Send Response Limit

upper limit 100 -7
150 0
1000 0
2000 +5
3400 +5
4000 0
300 -12
1000 -6
3000 -6
3400 -9
3400 -infinity
Figure 10 – Headset Send Frequency Response Mask
10
-10
-20
100 1000 10000
Frequency (Hz)
6.1.2. Headset Receive Frequency Response

The receive frequency response is the overall response of the codec receive filter, receive amplifier
and transducer. The receive sensitivity is expressed in terms of dBPa/V.
26
TIA/EIA-810-A

The receive frequency response is measured according to IEEE 1206 using the measurement set-up
shown in Figure 5, substituting the handset with the headset. Measurements should be done in ISO
1/12 octave intervals or smaller, over a minimum range of 100 Hz through 8000 Hz. Couple the
receiver to the appropriate artificial ear. The test signal level shall be -18.2 dBV (-16 dBm0).
Telephone sets with adjustable receive levels shall be adjusted so that their RLR is as close as
possible to the nominal value of Section 5.2.2.2 for this test.
The receive frequency response shall be below the upper limit and above the lower limit defined in
fit” mask.
6.2. Headset Loudness Ratings

6.2.1. Headset Send Loudness Rating
The SLR for a digital telephone set is the conversion ratio of a defined acoustic signal at the mouth
reference point to the transmit signal at the interface.

The terminal should be designed to have a nominal SLR value of 8 dB, with a tolerance of ±5.0 dB.
6.2.2. Headset Receive Loudness Rating

The RLR for a digital telephone set is the conversion ratio of a defined acoustic signal at the receive
input interface to an acoustic output signal from the headset receiver.

The headset is mounted as specified by IEEE 1206 and the receiver is coupled to the appropriate
artificial ear. The RLR shall be calculated from the 1/3 octave sensitivity data collected from the
receive frequency response measurement. Use equation [A2] of Annex A and bands 4 to 17, Table
18.
The monaural terminal should have a nominal RLR value of 0 dB, with a tolerance of ±4.0 dB. The
binaural terminal should have a nominal RLR value of 6 dB, with a tolerance of ±4.0 dB, for each of
the receivers measured separately.
Note 1: Headset RLRs are louder than handset RLRs to compensate for lack of noise occlusion.
Note 2: Either the terminal or the headset should have a receive volume control that is capable of
amplification and attenuation.
27
TIA/EIA-810-A
Table 7 – Co-ordinates of Headset Receive Response Limits
Limit Curve Frequency Receive Response Limit

upper limit 100 -7
150 0
1000 0
2000 5
3400 5
8000 -18
300 -11
600 -8
3000 -8
3400 -11
3400 -infinity
Figure 11 – Headset Receive Frequency Response Mask
10
-10
-20
100 1000 10000
Frequency (Hz)
6.2.3. Headset Talker Sidetone

The sidetone masking rating (STMR) for a digital telephone set is the ratio of a defined input
acoustic signal at the mouth reference point to the resulting acoustic output signal from the receiver.
It’s desirable for the STMR to be constant over the receive volume control range.
28
TIA/EIA-810-A

artificial ear. The test signal level at the MRP shall be -4.7 dBPa. For each frequency given in Table
18, bands 1 to 20, the sound pressure in the artificial ear shall be measured. The STMR shall be
calculated using equation [A3] of Annex A.
Telephone sets with adjustable receive levels shall be tested at the minimum, nominal and maximum
settings. For the nominal setting, adjust the level so that the RLR is as close as possible to the
nominal RLR value.
For any adjustable receive level, the value of STMR shall be within the range of 21 dB ± 6 dB for
supraural, 18 dB ± 6 dB for insert, 18 dB ± 6 dB for interconchial. The value of STMR for binaural
terminals should be 6 dB quieter, for each of the receivers measured separately.
6.3. Headset Noise

The noise levels are related to the SLR and RLR requirements.
6.3.1. Headset Send Noise

The following requirements apply to headset and 64 kbit/s, PCM digital telephones only.
6.3.1.1. General
The send noise of a digital telephone is the 5 second average background noise at the digital transmit
output with the headset transmitter isolated from sound input and mechanical disturbances.

With the headset mounted as specified by IEEE 1206 and the receiver coupled to the appropriate
artificial ear in a quiet environment (ambient noise less than 30 dBA), free of mechanical
disturbances, measure the noise level at the digital interface output or the reference codec decoder
output with apparatus that includes psophometric weighting, according to ITU-T Recommendation
0.41.
The send noise shall be no greater than -64 dBm0p.
6.3.2. Headset Send Single Frequency Interference

6.3.2.1. General

With the headset mounted as specified by IEEE 1206 and the receiver coupled to the appropriate
artificial ear in a quiet environment (ambient noise less than 30 dBA), free of mechanical
disturbances, measure the psophometrically-weighted noise level at VSEND with a selective voltmeter
or spectrum analyzer with an effective bandwidth of not more than 31 Hz, over the frequency range
of 100 to 3500 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed.
29
TIA/EIA-810-A
The send single frequency interference shall be no greater than -74 dBm0p.
6.3.3. Headset Receive Noise

6.3.3.1. General
The receive noise of a digital telephone is the short-term average background noise level measured at
the output of the headset receiver with the digital telephone receiving the digital quiet code.

artificial ear. A signal corresponding to a decoder quiet code is applied at the digital interface. The
A-weighted noise level is measured in the artificial ear. The ambient noise for this measurement shall
not exceed 30 dBA.
The receive noise shall be less than 40 dBA for a monaural headset. The receive noise for binaural
headsets should be less than 32 dBA, for each of the receivers measured separately.
6.3.4. Headset Receive Single Frequency Interference

6.3.4.1. General

artificial ear. A signal corresponding to a decoder quiet code is applied at the digital interface. The
A-weighted noise level is measured in the artificial ear with a selective voltmeter or spectrum
analyzer, with an effective bandwidth of not more then 31 Hz, over the frequency range of 100 to
8100 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed. The ambient noise
for this measurement shall not exceed 30 dBA.
The receive single frequency interference shall be 10 dB quieter than the A-weighted broadband
noise floor.
6.4. Headset Distortion and Noise

30
TIA/EIA-810-A
6.4.1. Headset Send Distortion and Noise

The distortion and noise is measured according to IEEE 1329, 9.3.6. Apply a sinewave signal at the
MRP, with the levels given in Table 4 and the following frequencies: 315, 502, 803 and 1004 Hz.
The ratio of the signal-to-total distortion and noise power of the digitally encoded signal output is
measured. The test frequency tolerance is 3%, but even submultiples of the sampling frequency must
not be used.
Note: In cases where the sound pressure exceeds +6 dBPa, the linearity of the artificial mouth should
be checked, as it exceeds the limits of ITU-T Recommendation P.51.
Table 8 – Headset Send Signal-to-Total Distortion and Noise Ratio Limits

(dBPa) (dB)
-30 20
-24 25
-17 31
-10 33
0 33
+4 33
+8 24
+10 20
The ratio of signal-to-total distortion and noise power of the digitally encoded signal output shall be
above the limits given in Table 8. Limits for intermediate levels are found by drawing straight lines
between the breaking points in the table on a linear (dB signal level) – linear (dB ratio) scale.
6.4.2. Headset Receive Distortion and Noise

artificial ear. The distortion and noise is measured according to IEEE 1329, 9.4.6. Apply a digitally
simulated sinewave, with the levels given in Table 5 and the following frequencies: 315, 502, 803
and 1004 Hz. The ratio of signal-to-total distortion and noise power is measured in the artificial ear.
The test frequency tolerance is 3%, but even submultiples of the sampling frequency must not be
used.
The ratio of signal-to-total distortion and noise power measured in the artificial ear, with A-
weighting applied, shall be above the limits given in Table 9, unless the signal in the artificial ear
exceeds +10 dBPa or is less than –50 dBPa.
31
TIA/EIA-810-A
Table 9 – Headset Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the digital Receive Ratio Receive Ratio

interface @ 315 Hz @ 502, 803 and 1004 Hz
(dBm0) (dB) (dB)
-40 19 20
-34 24 25
-27 30 31
-20 32 33
-10 32 33
-6 32 33
-3 28 29
0 23 24

normal conversation.

TCLw is measured in free-air in such a way that the inherent mechanical coupling of the headset is
not affected. The TCLw measurement shall be made at an input signal level of
-16 dBm0. The test shall be performed with the headset suspended in a noose around the earcap with
the headset cord hanging freely below the headset.
For devices that incorporate non-linear processes, additional measurements using signal levels of
-26 dBm0 and -10 dBm0 may be performed.
Noise and reflections in the test space must not influence the measurement. The test should be
performed in an anechoic chamber with the headset positioned at least 50 cm away from the nearest
part of the test chamber. The ambient noise level shall be less than 30 dBA.
The test signal is white noise, band limited to 100 through 4000 Hz, and modulated at a rate of 250
ms ON and 150 ms OFF. The measurement and calibration shall be determined during the ON
portions of the signal. Sine wave signals may be used with G.711 codecs.
The attenuation from digital input to digital output is measured at 1/12 octave frequencies as given
by the R.40-series of preferred numbers in ISO 3 for frequencies from 290 to 3255 Hz, using the
measurement arrangement shown in Figure 12. See Annex C.
The weighted terminal coupling loss is calculated according to ITU-T Recommendation G.122
(1993) Annex B, Section B.4 (trapezoidal rule).
Telephone sets with adjustable receive levels shall be tested at the nominal setting. For the nominal
setting, adjust the level so that the RLR is as close as possible to the nominal RLR value.
6.5.2. Requirements
The normalized value of TCLw loss shall be greater than 52 dB for IP sets and 45 dB for PCM sets
when measured under free field conditions and with SLR normalized to 8 dB and RLR normalized to
32
TIA/EIA-810-A
0 dB. It is desirable that the normalized value of TCLw for IP sets to be greater than 55 dB and that
the normalized value of TCLw for PCM sets be greater than 50 dB.
For example, if the measured TCLw is 48 dB, the measured SLR is 10 dB and the measured RLR is
2 dB, then the normalized value of TCLw = 48 dB + (8 - 10) dB + (0 - 2) dB = 44 dB.
NOTE 1: If equipped with adjustable receive level, the TCLw will decrease in proportion with the
increased gain relative to the nominal RLR in most cases. For example, if the measured
TCLw is 45 dB at nominal RLR and the adjustable receive level adds 12 dB of gain, then
TCLw (maximum receive level) = 45 dB - 12 dB = 33 dB.
NOTE 2: The echo impairment perceived by the person at the opposite end of the connection from a
telephone set is a function of the magnitude of the talker echo signal as well as the talker
echo path delay. The echo signal becomes more disturbing as the talker echo path delay
increases. Thus, a telephone set with adequate TCLw performance on low delay
connections may provide satisfactory performance while the same may not be true for
connections that have a long delay.
under study.
Figure 12 – Terminal Coupling Loss Measurement Method
Headset Suspended
vS E N D ( E c h o R e t u r n )
Decoder v
Digital
Interface
Set
Coder GEN
Anechoic Chamber vRCV

Reference Codec
6.6. Long Duration Maximum Acoustic Pressure (Steady State Input)

6.6.1. General
The long duration acoustic pressure is the maximum, steady state sound pressure disturbance, greater
than 500 ms, emitted from a headset receiver, caused by the maximum excursions of the receive
digital signal.
33
TIA/EIA-810-A
Additional consideration should be given to the acoustic pressure caused by tones, other audio
signals or long duration, high amplitude electrical signals applied to power, network, handset or

The steady-state A-weighted sound pressure level shall be measured using the digital terminals test
procedure in ITU-T Recommendation P.360, with the following modifications.
Apply a digital square wave to the receive input, switched between the maximum positive and the
maximum negative values (see Table 1), as defined in ITU-T Recommendation G.711. The switching
rate shall range from 1 Hz to 4000 Hz over a sweep time of not less than 30 seconds. The
measurement shall be made with a RMS detector set to 1 second effective averaging time (RMS
Slow).
Maximum acoustic pressure limits can be referenced to the ERP, DRP, free field (0 degrees elevation
and azimuth) or to a diffuse field, as required by the appropriate safety standard. This may require
that measurements made at one reference point be translated to the required reference point.
For types 2, 3.1, 3.3 and 3.4 ear simulators, DRP to ERP transformation shall be performed by adding
the values in table 2b/P.57 of ITU-T Recommendation P.57 to the data measured at the DRP. For
type 3.2 ear simulators, DRP to ERP transformation shall be performed by using the transfer function
supplied by the manufacturer of the ear simulator.
Transformation to free field or diffuse field shall be made using the transfer function supplied by the
manufacturer of the ear simulator, if available. Alternatively, the transfer functions specified in ITU-
T P.58 may be used. Transfer functions with resolution of at least 1/12 octave or R40 format shall be
used if available.
6.6.3. Requirements
6.7. Short Duration (Peak) Acoustic Pressure

6.7.1. General
The short duration acoustic pressure is the maximum short duration, sound pressure impulse, less
than 500 ms, from a headset receiver, caused by the maximum excursions of the receive digital
signal.
This short duration test stresses nonlinear processes, like AGC, and doesn’t directly replace a short
duration surge. Additional consideration should be given to the peak acoustic pressure caused by
tones or short duration, high amplitude electrical pulses applied to power, network, handset or
34
TIA/EIA-810-A

The peak acoustic pressure level shall be measured using the digital terminals test procedure in ITU-
T Recommendation P.360, with the following modifications instead of applying electrical impulses
to the send and receive pairs. The short duration acoustic pressure shall be determined by applying
digital codes to the receive input. The codes shall be switched between the maximum positive and the
maximum negative values, defined in ITU-T Recommendation G.711. The switching rate shall range
from 2 Hz to 4000 Hz. The duration of the ON codes shall be a number of complete cycles
approximating but not exceeding 500 ms. The ON codes must be followed by a quiet interval of at
least 500 ms before repeating the codes, as shown in Figure 9.
Peak acoustic pressure limits can be referenced to the ERP, DRP, free field (0 degrees elevation and
azimuth) or to a diffuse field, as required by the appropriate safety standard. This may require that
measurements made at one reference point be translated to the required reference point. The
translations shall be implemented using a minimum phase, parametric filter (or equivalent) as peak
measurements must be made in real time. Both the magnitude and phase of the transfer function is
necessary to best preserve the waveshape for a proper measurement of its peak value.
The filter parameters for transformation from DRP to ERP shall be based on the transfer function
specified in Section 6.6.2. The filter parameters for transformation to free field or diffuse field shall
be made using the transfer function supplied by the manufacturer of the ear simulator, if available.
Alternatively, the transfer function specified in ITU-T P.58 can be used. Transfer functions with
resolution of at least 1/12 octave or R40 format shall be used if available.
The magnitude of the filter response shall follow the transfer function within a tolerance of ±1dB.
In practice, most devices will pass peak acoustic limits without the reduction provided by the
translation filter. In these cases, the filter may not be required.
6.7.3. Requirements
35
TIA/EIA-810-A
7. Handsfree Technical Requirements (Advisory)

All handsfree requirements are advisory.
All telephones shall support G.711 A-law and mu-law. The handset technical requirements apply
equipment.
Equipment using nonlinear voice signal processing may subjective testing.
The handsfree test method is given in IEEE 1329.
Handsfree telephones designed for other than traditional tabletop or desktop positioning should be
tested with the appropriate user positioning in mind. This position shall be defined as the
“recommended test position” (RTP). The RTP should be obtained from the manufacturer, and should
be based upon the product’s intended use. For testing purposes, this will dictate the distance and
position geometry relationship between the handsfree and the mouth simulator and microphone.
Measurements performed at other distances or positions shall be noted, and in the absence of a RTP,
the 50 cm test position as defined in IEEE 1329, is recommended.
The volume control setting resulting in nominal RLR is the reference volume control (RVC) setting
defined in IEEE 1329.
7.1. Handsfree Frequency Response

7.1.1. Handsfree Send Frequency Response
The send frequency response is the overall response of the transducer, send amplifier, and the codec
send filter. The send sensitivity is expressed in terms of dBV/Pa.

The send frequency response is measured in or converted to 1/3 octave bands, according to IEEE
1329, over a minimum range of 100 Hz through 4000 Hz. For desktop handsfree use the
measurement set-up shown in Figure 13. For other handsfree devices use the RTP. The nominal test
signal level shall be -4.7 dBPa at the MRP.
36
TIA/EIA-810-A
Figure 13 – Handsfree Send Frequency Response Measurement Method
vS E N D
Mouth Simulator
GEN
Send
pM
Decoder v
Measuring
50 cm Interface Amplifier
30 cm
Coder
40 cm Digital
Set
Anechoic Chamber (On Table)
Reference Codec
The send frequency response should be below the 1/3 octave band upper limit and above the 1/3
octave band lower limit defined in Table 10 and shown in Figure 14. Note: The frequency response
mask is a floating or “best fit” mask.
Table 10 – Co-ordinates of Handsfree Send Response Limits
Limit Curve 1/3 Octave Send Response Limit

Band (Hz) (dB) [arbitrary level]
upper limit 100 -7
125 -3
160 to 1000 0
1250 +2
1600 +4
2000 to 3150 +5
4000 0
315 -13
400 -12
500 -11
630 -10
800 -9
1000 to 2500 -7
3150 -10
4000 -infinity
37
TIA/EIA-810-A
Figure 14 – Handsfree Send Frequency Response Mask
Arbitrary Level (dB) 10
-10
-20
100 1000 10000
Frequency (Hz)
7.1.2. Handsfree Receive Frequency Response

The receive frequency response is the overall response of the codec receive filter, receive amplifier
and transducer. The receive sensitivity is expressed in terms of dBPa/V.

The receive frequency response is measured in or converted to 1/3 octave bands, according to IEEE
1329, over a minimum range of 100 Hz through 8000 Hz. For desktop handsfree use the
measurement set-up shown in Figure 15. For other handsfree devices use the RTP. The test signal
level shall be -25 dBV. Telephone sets with adjustable receive levels shall be adjusted so that their
RLR is as close as possible to the nominal value of Section 7.2.2.2 for this test.
Figure 15 – Handsfree Receive Frequency Response Measurement Method
To Sound Pressure
Measuring Amplifier
Free Field Microphone Decoder

R e c e i v e pE
50 cm Interface
30 cm
Coder GEN
40 cm
Digital
Set
(On Table)
Anechoic Chamber vRCV
Reference Codec
38
TIA/EIA-810-A
The receive frequency response should be below the 1/3 octave band upper limit and above the 1/3
octave band lower limit defined in Table 11 and shown in Figure 16. Note: The frequency response
mask is a floating or “best fit” mask.
Table 11 – Co-ordinates of Handsfree Receive Response Limit Curves
Limit Curve 1/3 Octave Receive Response Limit

Band (Hz) (dB) [arbitrary level]
upper limit 100 -9
125 -6
160 -3
200 to 4000 0
5000 -7
6300 -14
8000 -20
315 -12
400 -11
500 to 2500 -10
3150 -13
4000 -infinity
Figure 16 – Handsfree Receive Frequency Response Mask
10
-10
-20
100 1000 10000
Frequency (Hz)
39
TIA/EIA-810-A
7.2. Handsfree Loudness Ratings and Receive Volume Control

Correlation factors relating handsfree loudness ratings to handset loudness ratings are under
investigation and are not used in this standard. The currently accepted correlation factors for
personal, wireline telephone applications are shown below. These correlation factors may not be
appropriate for other handsfree applications such as conference, hand-held, in-car applications or any
applications where the relationship between the talker and the handsfree varies from the 50 cm
position, or the reverberation characteristics or the background noise levels vary from typical office
environments.
The handsfree SLR should be 5 dB quieter than the handset SLR due to:
• a 3 dB increase in the average talking level when using a handsfree,

• a 1-2 dB decrease in the actual handset talking levels compared to those measured at the MRP,
• other small differences related to different frequency responses, etc.
Subjective evaluations have determined that the handsfree RLR should be 14 dB quieter than the
handset RLR.
7.2.1. Handsfree Send Loudness Rating

The terminal should be designed to have a nominal handsfree SLR = 13 dB, with a tolerance of ±4.0
dB.
7.2.2. Handsfree Receive Loudness Rating

The RLR shall be calculated from the 1/3 octave sensitivity data collected from the receive frequency
The terminal should be designed to have a nominal handsfree RLR = 16 dB, with a tolerance of ±4.0
dB.
7.2.3. Handsfree Receive Volume Control

The handsfree receive volume control shall provide greater than or equal to 8 dB of gain relative to
the nominal volume control setting. The volume control should provide at least 16 dB of attenuation
relative to the nominal volume control setting. The volume control step size shall be less than 6 dB.
7.3. Handsfree Noise
7.3.1. Handsfree Send Noise

7.3.1.1. General
The send noise of a digital telephone is the short-term average background noise at the digital
transmit output with the microphone isolated from sound input and mechanical disturbances.
40
TIA/EIA-810-A

Handsfree send noise is measured according to IEEE 1329, 9.3.4, Noise. With the handsfree in a
quiet environment (ambient noise less than 30 dBA), the noise level at the digital output is measured
with apparatus including psophometric weighting according to ITU-T Recommendation 0.41.
The handsfree send noise should be no greater than -64 dBm0p.
7.3.2. Handsfree Send Single Frequency Interference

7.3.2.1. General

Handsfree send noise is measured according to IEEE 1329, 9.3.4, Noise. In a quiet environment
(ambient noise less than 30 dBA), measure the psophometrically-weighted noise level at VSEND with a
selective voltmeter or spectrum analyzer with an effective bandwidth of not more than 31 Hz, over
the frequency range of 100 to 3500 Hz. If FFT analysis is used, then “Flat Top” windowing shall be
employed.
The handsfree send single frequency interference should be less than -74 dBm0p.
7.3.3. Handsfree Receive Noise

7.3.3.1. General
The receive noise of a digital handsfree telephone is the short-term average background noise level
measured at the speaker output with the digital telephone receiving the digital quiet code.

The handsfree receive A-weighted noise level is measured according to IEEE 1329, 9.4.4, Noise. A
signal corresponding to a decoder value number 0 (µ-law) is applied at the digital interface. The
ambient noise for this measurement shall not exceed 30 dBA.
The handsfree receive noise should be less than 40 dBA at the maximum volume control setting and
less than 35 dBA with the volume control at the nominal RLR value, with the comfort noise turned
off.
41
TIA/EIA-810-A
7.3.4. Handsfree Receive Single Frequency Interference

7.3.4.1. General

Handsfree receive noise is measured according to IEEE 1329, 9.4.4, Noise. A signal corresponding to
a decoder quiet code is applied at the digital interface. The A-weighted noise level is measured with a
selective voltmeter or spectrum analyzer, with an effective bandwidth of not more then 31 Hz, over
the frequency range of 100 to 8100 Hz. If FFT analysis is used, then “Flat Top” windowing shall be
employed. The ambient noise for this measurement shall not exceed 30 dBA.
to the nominal RLR value when driven quiet code.
The handsfree receive single frequency interference should be 10 dB quieter than the A-weighted
broadband noise floor.
7.4. Handsfree Distortion and Noise

7.4.1. Handsfree Send Distortion and Noise

Handsfree send distortion and noise is measured according to IEEE 1329, 9.3.6, Distortion. Apply a
test signal at the MRP, with the levels given in Table 12 and the following frequencies: 502, 803 and
1004 Hz. The ratio of the signal-to-total distortion and noise power of the digitally encoded signal
output is measured.
Table 12 – Handsfree Send Signal-to-Total Distortion and Noise Ratio Limits

(dBPa) (dB)
-12 18
-8 30
-4 33
0 33
+4 33
+8 31
The ratio of signal-to-total distortion and noise power of the digitally encoded signal output should
be above the limits given in Table 12.
42
TIA/EIA-810-A
Limits for intermediate levels are found by drawing straight lines between the breaking points in the
table on a linear (dB signal level) – linear (dB ratio) scale.
7.4.2. Handsfree Receive Distortion and Noise

Handsfree receive distortion and noise is measured according to IEEE 1329, 9.4.6, Distortion. Apply
a digitally simulated test signal, with the levels given in Table 13 and the following frequencies: 502,
803 and 1004 Hz. The ratio of signal-to-total distortion and noise power is measured.
The ratio of signal-to-total distortion and noise power measured with A-weighting applied, should be
above the limits given in Table 13, unless the measured sound pressure is less than –50 dBPa. The
measurement microphone may be placed at 25 cm for this measurement if the measured signal levels
are too low.
Table 13 – Handsfree Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the Receive Ratio

digital interface
(dBm0) (dB)
-34 25
-27 28
-20 31
-10 31
-6 31
-3 23

normal conversation.

Refer to IEEE 1329, Clause 11, for the test method.
7.5.2. Requirements
The normalized value of handsfree TCLw loss should be greater than 45 dB when measured under
free field conditions and with SLR normalized to 13 dB and RLR normalized to 16 dB. It is desirable
that the normalized value of TCLw be greater than 50 dB to meet ITU-T Recommendation G.131
talker echo objective requirements.
For example, if the measured handsfree TCLw is 27 dB, the measured SLR is 16 dB and the
measured RLR is 15 dB, then the normalized value of TCLw = 27 dB + (13 - 16) dB + (16 - 15) dB
= 25 dB.
43
TIA/EIA-810-A
under study.
7.6. Stability Loss

The stability loss is a measure of the telephone set's contribution to the overall network stability
requirements. Stability loss is defined as the minimum loss from the digital input (receive) to the
digital output (send), at any test frequency.

The stability measurement is made at an input signal level greater than or equal to -10 dBm0 and less
than or equal to 0 dBm0, at 1/12 octave bands centered at 205 Hz to 3868 Hz. With the handsfree and
transmission circuit fully active, measure the attenuation from the digital input to the digital output
using Method 1 and Method 2.
Place the handsfree telephone in the middle of a hard, smooth surface free of any other object for
0.5 m. The telephone set shall be fully active. The surface must be at least 1 square meter.
7.6.2. Requirement
The handsfree stability loss, i.e., minimum loss, at any frequency should be greater than 6 dB. It is
desirable that this loss be greater than 10 dB.
Telephone sets with adjustable receive level should maintain stability over the entire range of
adjustable receive levels.
44
TIA/EIA-810-A
8. Quality of Service (Advisory)

The term “Quality of Service” (QoS) refers to the customer’s level of satisfaction. In this case it is
related to voice quality. A number of tools are available to evaluate QoS. The tools can be divided
into three categories: subjective evaluations, objective measurements and objective analysis. Some of
the impairments introduced by IP networks, such as delay, echo loss and double talk do not lend
themselves to traditional one-way subjective evaluations or objective measurements. A number of
new objective measurement algorithms have recently been introduced, which use speech samples to
evaluate speech quality. Various commercial products are available on the marketplace. A
qualification process is underway in ITU-T Study Group 12, Question 13, and a new objective
speech quality standard is likely to be available by the year 2000. Further guidance on the
mechanisms of these algorithms is provided in ETSI EG 201 377 - 1.
One weakness of these objective measurement algorithms is that they do not account for the
conversational nature of speech. ITU-T Recommendation G.107, the E-Model, is useful in
transmission planning for evaluating end-to-end transmission performance of both PSTN and IP
networks. The main advantages of G.107 are that it is standardized, it can handle all IP QoS
impairments associated with modern signal processing devices and it models the conversational
nature of speech.
8.1. Equipment Impairment Factor, Ie

The input parameters of the E-Model reflect the various characteristics of voice quality, such as
loudness level, noise level and delay, etc. Of particular relevance to this standard is the equipment
impairment factor, Ie. This parameter quantifies the impairment introduced by low bit rate codecs
and digital channel impairments, such as corrupted data and lost/late packets. Ie is determined by a
combination of subjective tests and experience. It is independent of the other parameters.
Table 14 shows the Ie for a number of codecs in error-free conditions. This information was taken
from in ITU-T Recommendation G.113, APPENDIX I, Table I.1. Refer to the latest issue of G.113,
APPENDIX I for the currently accepted values of Ie.
8.2. Packet Loss

The impairment associated with different rates of packet loss can be determined for a given codec by
performing subjective tests, and an Ie value can be derived from the subjective result. Table 15
shows the provisional values of Ie for randomly distributed packet loss for the G.723.1A + voice
activity detection (VAD) (at 6.3 kbit/s) and the G.729A + VAD codecs. This information was taken
from ITU-T Recommendation G.113, APPENDIX I, Table I.2. Table 16 shows the provisional values
of Ie for randomly distributed and bursty packet loss for the G.711 codec with and without packet
loss concealment (PLC). This information was taken from ITU-T Recommendation G.113,
APPENDIX I, Table I.3.
These Ie values are provisional because they were determined in isolated experiments. Refer to the
latest issue of G.113, APPENDIX I for the currently accepted values of Ie.
45
TIA/EIA-810-A
Table 14 – Planning Values for the Equipment Impairment Factor Ie
Codec Type Reference Operating Rate Ie

kbit/s Value
PCM G.711 64 0
ADPCM G.726, G.727 40 2
G.721, G.726, G.727 32 7
G.726, G.727 24 25
G.726, G.727 16 50
LD-CELP G.728 16 7
12.8 20
CS-ACELP G.729 8 10
G.729-A + VAD 8 11
VSELP IS-54 8 20
ACELP IS-641 7.4 10
QCELP IS-96a 8 21
RCELP IS-127 8 6
VSELP Japanese PDC 6.7 24
RPE-LTP GSM 06.10, Full-rate 13 20
VSELP GSM 06.20, Half-rate 5.6 23
ACELP GSM 06.60, EFR 12.2 5
ACELP G.723.1 5.3 19
MP-MLQ G.723.1 6.3 15
Table 15 – Provisional Planning Values for Ie under Conditions of Random Packet Loss, for
Codecs G.729A + VAD, G.723.1A + VAD and GSM EFR
% Packet Loss G.729A + VAD G.723.1A + VAD GSM EFR

6.3 kbit/s
0 11 15 5
0.5 13 17 —
1 15 19 16
1.5 17 22 —
2 19 24 21
3 23 27 26
4 26 32 —
5 — — 33
8 36 41 —
16 49 55 —
NOTE – Number of frames per packet:
• G.729A + VAD: 2;
• G.723.1A + VAD: 1.
• GSM EFR: 1
46
TIA/EIA-810-A
Table 16 – Provisional Planning Values for the Equipment Impairment Factor Ie under
Conditions of Packet Loss, for Codec G.711 without and with Packet Loss Concealment (PLC)
G.711 w/ PLC
Packet Loss G.711 w/o PLC Random Packet Bursty Packet Loss
% Loss
0 0 0 0
1 25 5 5
2 35 7 7
3 45 10 10
5 55 15 30
7 — 20 35
10 — 25 40
15 — 35 45
20 — 45 50
Note:
Speech packet length: 10 ms
8.3. QoS
QoS can be expressed in many ways. Table 17 is based on information in ITU-T Recommendations
G.175, Table 3/G.175 and G.109, Table 1/G.109. It shows the correlation between the E-Model’s
objective outputs (R-value, MOS, GoB% and PoW%) and the corresponding categories of speech
transmission quality and user satisfaction.
Table 17 – The Relationship between R-value and MOS, GoB% and PoW%
and the Definition of Speech Transmission Quality Categories
and User Satisfaction Levels
R-value MOS %
GoB% %
PoW% Speech User satisfaction
lower Transmission
limit Quality Category
90 4.34 97 ~0 Best Very satisfied
80 4.03 89 ~0 High Satisfied
70 3.60 73 6 Medium Some users dissatisfied
60 3.10 50 17 Low Many users dissatisfied
50 2.58 27 38 Poor Nearly all users dissatisfied
Note 1: Connections with R-values below 50 are not recommended.
Note 2: Although the trend in transmission planning is to use R-values, equations to convert R-values
into other metrics e.g. MOS, %GoB, %PoW can be found in Annex B of G.107.
It is convenient to use the R-value to categorize the user’s subjective satisfaction, because the R-
value is linear and the categories separated by decades. Using the R-value also avoids mistakes
related to comparing subjective MOS results, obtained under certain conditions, with E-Model
objective predictions. Because the E-Model is a relative tool rather than an absolute tool, all E-Model
results should always be compared to a reference condition. That is, all E-Model results should
always be compared to a reference condition. The best reference condition is often the E-Model
default setting, which represents two ideal ISDN sets connected together with zero delay.
47
TIA/EIA-810-A
The four main impairments in VoIP telephony are delay, echo, voice compression and packet loss.
Graphs of the R-value vs network delay, with the Ie of a particular codec as the parametric variable
and all other E-Model parameters at their default setting, shows the best case QoS. Figure 17 shows
the R-value vs network delay for the reference case of the G.711 codec on an ISDN connection with
no delay (and obviously no packet loss), with the G.723.1A + VAD codec (6.3 kbit/s) at various
amounts of non-bursty packet loss. The actual distribution of packet loss on IP is likely to be bursty,
but it can be modeled by short-term variation in randomly distributed rates. In some cases, the short-
term packet loss may be much greater than shown in these figures. The fixed G.723.1A + VAD delay
was removed to better show the range of one-way network delay. This means that the G.723.1A +
VAD family of curves has been shifted to the left by 98 ms. Figure 18 shows a similar graph for the
G.729A + VAD codec.
The following is a list of the assumptions that were made for this analysis:
1. all input parameters are at the Version 19 E-Model default values, except Ie and delay T
2. absolute delay Ta = mean one-way delay T
3. round-trip delay Tr = 2 x mean one-way delay T
4. echo cancellation is ideal
5. G.711 ISDN reference connection delay is zero
6. G.723.1A + VAD delay is 3 packets X number of frames/packet X frame size (ms) + look ahead
(ms) = 3 X 1 X 30 ms + 7.5 ms = 97.5 ms ≅ 98 ms, where the 3 packets are a rule-of-thumb
related to one packet being packetized, one in transient and one being depacketized.
7. G.729A +VAD delay is 3 packets X number of frames/packet X frame size (ms) + look ahead
(ms) = 3 X 2 X 10 ms + 5 ms = 65 ms.
Figure 17 shows that G.711 is in the “very satisfied” category up to about 150 ms of network delay.
There are two slopes to the curve, with the break point at about 150 ms. The slope increases after 150
ms due to additional impairment related to the difficulty of two people conversing under long delay
conditions. For example, it becomes increasingly more difficult for one person to break into the
conversation and to decide whose turn it is to speak as the one-way delay increases.
The G.723.1A + VAD codec starts out in the middle of the “some users dissatisfied” category, with
no packet loss, and the family of G.723.1A + VAD curves shows the degradation in user satisfaction
with increasing packet loss and increasing network delay. It can be seen that relatively little network
delay, in IP terms, degrades the QoS significantly.
Figure 18 shows that G.729A +VAD codec provides somewhat better user satisfaction than the
G.723.1A + VAD codec, because of its better Ie value and shorter packetization delay. However, the
QoS of both codecs is significantly degraded relative the ISDN reference condition. Since this is the
best possible performance for these codecs, it is clear that neither can be considered as a viable
alternative to the existing ISDN QoS.
48
TIA/EIA-810-A
Figure 17 – G.711 and G.723.1A + VAD QoS
G.723.1A+VAD IP QoS for Several Values of Packet Loss Assumptions:

Compared to G.711 ISDN Reference QoS
G.711 ISDN Delay = 0 ms
User Satisfaction
100 G.723.1A IP Delay
= 3 X frames/packet X
Very satisfied frame size + look ahead
= 3 X 1 X 30 + 8
90 = 98 ms
with one speech frame/packet
Satisfied
G.711, Ie = 0, PL = 0%
80 G.723.1A+VAD, Ie = 15, PL = 0%
Some users G.723.1A+VAD, Ie = 19, PL = 1%
dissatisfied G.723.1A+VAD, Ie = 24, PL = 2%
G.723.1A+VAD, Ie = 27, PL = 3%
70 G.723.1A+VAD, Ie = 32, PL = 4%
Many users
dissatisfied
60 Ie = Equipment Impairment Factor
PL = Packet Loss
Nearly all users
dissatisfied
50
0 100 200 300 400 500
One-way Network Delay (ms).
Figure 18 – G.711 and G.729A + VAD QoS
G.729A+VAD IP QoS for Several Values of Packet Loss Assumptions:

Compared to G.711 ISDN Reference QoS
G.711 ISDN Delay = 0 ms
User Satisfaction
100 G.729A + VAD IP Delay
= 3 X frames/packet X
Very satisfied frame size + look ahead
= 3 X 2 X10 + 5
90 = 65 ms
with 2 speech frames/packet
Satisfied
G.711, Ie = 0, PL = 0%
80 G.729A+VAD, Ie = 11, PL = 0%
Some users G.729A+VAD, Ie = 15, PL = 1%
dissatisfied G.729A+VAD, Ie = 19, PL = 2%
G.729A+VAD, Ie = 23, PL = 3%
70 G.729A+VAD, Ie = 26, PL = 4%
Many users
dissatisfied
60 Ie = Equipment Impairment Factor
PL = Packet Loss
Nearly all users
dissatisfied
50
0 100 200 300 400 500
One-way Network Delay (ms).
49
TIA/EIA-810-A
Annex A (informative) – Calculation of Loudness Ratings

This Annex details the loudness rating calculations and weighting factors relevant to this document.
Loudness ratings are used to insure that the loudness of a connection from the Mouth Reference
Point (MRP) of the talker to the Ear Reference Point (ERP) of the far end listener is at a satisfactory
level. The loudness of the complete path is designated as the Overall objective Loudness Rating
(OLR). The MRP to electrical component is referred to as the Send Loudness Rating (SLR). The
electrical to ERP component is the Receive Loudness Rating (RLR). The loudness of the Sidetone
path of handsets and headsets is referred to as the Sidetone Masking Rating (STMR).
Loudness ratings are used rather than simple level measurements because of better subjective
correlation. Loudness ratings more closely account for the ear’s different sensitivity at different
frequencies and its nonlinear response to varying sound levels. The following calculations are based
on the 1993 and 1999 revisions of ITU-T Recommendation P.79. Older versions of P.79 should not
be used. ITU-T P.79 provides information on the derivation of the loudness rating algorithm.
Send Loudness Rating (Handset and Headset):
Band 17
SLR = - 57.1 log10 ∑ 10

(0.1 * 0.175 * (SMJ – Wsi )) [A1]
i = Band 4
Where:
i Frequency bands from Table 1 of ITU-T P.79-1999, bands 4-17.
SMJ Send Frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured
per this standard.
Wsi Send weighting factor from Table 1 of ITU-T P.79-1999, see Table 18.
Receive Loudness Rating (Handset and Headset):
Band 17
RLR = - 57.1 log10 ∑ 10(0.1 * 0.175 * (SJE – Wri - LE)) [A2]

i = Band 4
Where:
SJE Receive frequency response data (Sensitivity, Junction-to-Ear) in dBPa/V measured
per this standard.
Wri Receive weighting factor from Table 1 of ITU-T P.79-1999, see Table 18.
LE Leakage correction from Table 2 of ITU-T P.79-1999, see Table 18. Only used when
handset is sealed to IEC-318 ear. Not used with Handsfree, Headsets, or when using
ear simulators with controlled leakage.
50
TIA/EIA-810-A
Sidetone Masking Rating (Handset and Headset):
Band 20
STMR = - 44.4 log10 ∑ 10(0.1 * 0.225 * (SmeST – WMSi)) [A3]

i = Band 1
Where:
SmeST Sidetone frequency response data (Sensitivity, mouth-to-ear) in dB Pa/Pa measured
per this standard.
WMSi Sidetone weighting factor from Table 3 of ITU-T P.79-1999, see Table 18.
Send Loudness Rating (Handsfree):
Band 17
SLR = - 57.1 log10 ∑ 10

(0.1 * 0.175 * (SMJ – Wsi )) [A4]
i = Band 4
Where:
SMJ Send Frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured
per this standard.
Wsi Send weighting factor from Table 1 of ITU-T P.79-1999, see Table 18.
Note: There are no additional correction factors for handsfree send.
Receive Loudness Rating (Handsfree):
Band 17
RLR = - 57.1 log10 ∑ 10(0.1 * 0.175 * (SJE – Wri )) [A5]

i = Band 4
Where:
SJE Receive frequency response data (Sensitivity, Junction-to-Ear) in dBPa/V measured
per this standard.
Wri Receive weighting factor from Table 1 of ITU-T P.79-1999, see Table 18.
Note: There are no additional correction factors for handsfree receive.
51
TIA/EIA-810-A
Table 18 – ITU-T P.79-1999 Tables 1, 2 and 3

Weighting factors for calculating loudness ratings
Band No. Mid- Send Receive Receive Sidetone

frequency Wsi Wri LE WMSi
(Hz) (dB)
1 100 110.4
2 125 107.7
3 160 104.6
4 200 76.9 85.0 8.4 98.4
5 250 62.6 74.7 4.9 94.0
6 315 62.0 79.0 1.0 89.8
7 400 44.7 63.7 -0.7 84.8
8 500 53.1 73.5 -2.2 75.5
9 630 48.5 69.1 -2.6 66.0
10 800 47.6 68.0 -3.2 57.1
11 1000 50.1 68.7 -2.3 49.1
12 1250 59.1 75.1 -1.2 50.6
13 1600 56.7 70.4 -0.1 51.0
14 2000 72.2 81.4 3.6 51.9
15 2500 72.6 76.5 7.4 51.3
16 3150 89.2 93.3 6.7 50.6
17 4000 117.0 113.8 8.8 51.0
18 5000 49.7
19 6300 50.0
20 8000 52.8
52
TIA/EIA-810-A
Annex B (informative) – Measurement and Level Conversions

General
The following describes how to convert between various units of measurement used in telephone
testing.
Useful Conversions and Procedures
0 dBm (0 VU) is accepted as 1mW, typically using a circuit impedance of 600 Ω or 900 Ω.
0dBm = 10 log 1(mW)
dBV = 10 log V2 or, V = 10 dBV/20

= 20 log V
P = V2/R, where for dBm reference, R = 600 Ω
dBm = 10 log (V2/R * 1000)

= 10 log (V2/600 * 1000)
= 10 log (V2/0.600)
Therefore, for 0 dBm, V = 774.6 mV or 0 dBm = -2.218 dBV @ 600 Ω (use -2.2 dB)
P = V2/R, where for dBm reference, R = 900 Ω
dBm = 10 log (V2/R * 1000)

= 10 log (V2/900 * 1000)
= 10 log (V2/0.900)
Therefore, for 0 dBm, V = 948.7 mV or 0 dBm = -0.458 dBV @ 900 Ω (use -0.5 dB)
This means that if we substitute 600 Ω for 900 Ω or vice versa, and the voltage remains constant,
then we have:
Correction (dB) = -10 log 0.600/0.900 = 10 log 0.900/0.600 = 1.761 dB
To simplify,
Correction (dB) = 10 log( |Z1| / |Z2| ), that is, the log of the ratio of the magnitude of the impedances,
when converting from impedance Z1 to Z2.
If converting from "Z1 = 600 Ω" to "Z2 = 900 Ω", the correction factor is -1.76 dB (use -1.8 dB),
thus we subtract 1.8 dB from the measurement.
At this point, depending on the impedance being used, conversion factors can be applied dB for dB to
the measured or calculated result. As an example, to convert a 600 Ω -20dBm signal to dBV, simply
subtract 2.2 to get -22.2 dBV. Another example is if -20 dBm is measured across 600 Ω, then across
53
TIA/EIA-810-A
900 Ω, we add a correction of -1.8 dB to get -21.8 dBm (since less power is dissipated by the higher
resistance).
Loudness Rating Conversions
Conversions from the IEEE 661 methodology to the ITU-T P.79 methodology as specified by
TIA/EIA-579-A, Annex A, are as follows:
SLR (P.79) = TOLR (IEEE 661) + 57

RLR (P.79) = ROLR (IEEE 661) - 51
STMR (P.79) = SOLR (IEEE 661) + 9
The above conversions should be used as an approximation only. These conversions are based on the
nominal frequency response curves specified in TIA/EIA-579-A. Proper conversion may depend
upon actual measurements being made with each measurement standard, when frequency responses
deviate significantly from the specified nominals.
Acoustic Sound Pressure Conventions
dB Pa (dB Pascals)
dBSPL (dB Sound Pressure Level)
Where,
0 dB Pa = 94dBSPL, and 0 dBSPL = 20 microPascals, 1 Pa = 1 N/m2
An A weighted sound pressure level in dB (dBSPL, A weighted) is often abbreviated to “dBA”.
54
TIA/EIA-810-A
Annex C (informative) – Preferred 1/12 Octave Frequencies

The ISO 3, R40 basic series preferred numbers, in terms of 1/12 octave frequencies are listed in
Table 19. The frequencies highlighted in Italics are the R10, 1/3 octave frequencies.
Table 19 – Preferred 1/12 Octave Frequencies
# Preferred Frequencies # Preferred Frequencies

(Hz) (Hz)
0 100 40 1000
1 106 41 1060
2 112 42 1120
3 118 43 1180
4 125 44 1250
5 132 45 1320
6 140 46 1400
7 150 47 1500
8 160 48 1600
9 170 49 1700
10 180 50 1800
11 190 51 1900
12 200 52 2000
13 212 53 2120
14 224 54 2240
15 236 55 2360
16 250 56 2500
17 265 57 2650
18 280 58 2800
19 300 59 3000
20 315 60 3150
21 335 61 3350
22 355 62 3550
23 375 63 3750
24 400 64 4000
25 425 65 4250
26 450 66 4500
27 475 67 4750
28 500 68 5000
29 530 69 5300
30 560 70 5600
31 600 71 6000
32 630 72 6300
33 670 73 6700
34 710 74 7100
35 750 75 7500
36 800 76 8000
37 850 77 8500
38 900 78 9000
39 950 79 9500
40 1000 80 10000
55

Tia Eia 810 A Final

Transféré par

Informations du document

Titre original

Copyright

Formats disponibles

Partager ce document

Partager ou intégrer le document

Options de partage

Avez-vous trouvé ce document utile ?

Ce contenu est-il inapproprié ?

Droits d'auteur :

Formats disponibles

Tia Eia 810 A Final

Transféré par

Droits d'auteur :

Formats disponibles

ANSI/TIA/EIA-810-A-2000

Approved: December 19, 2000

TELECOMMUNICATIONS INDUSTRY ASSOCIATION

The Telecommunications Industry Association

PRICE: Please refer to current Catalog of

All rights reserved

Organizations may obtain permission to reproduce a limited number of copies

Global Engineering Documents

(This foreword is not part of this standard.)

This document is a TIA/EIA Telecommunications standard produced by Working Group

The TR-41.3.3 VoIP/PCM Transmission Performance Working Group acknowledges the

Telecommunications Industry Association

2.1. Limits of Applicability

2.2. Categories of Criteria

2.3. FCC Part 68

[1] ANSI/TIA/EIA-464-B-1996, Requirements for Private Branch Exchange (PBX) Switching

[2] ANSI/TIA/EIA-504-A-1997, Telecommunications – Telephone Terminal Equipment –

[3] ANSI/EIA/TIA-571-1991, Environmental Considerations for Telephone Terminals.

[4] ANSI/TIA/EIA-579-A-1998, Telecommunications – Telephone Terminal Equipment –

[5] ANSI/IEEE Standard 269-1992, Standard Methods for Measuring Transmission

[7] ANSI/IEEE Standard 1206-1994, Standard Methods for Measuring Transmission

[8] ANSI/IEEE Standard 1329-1999, Standard Method for Measuring Transmission

[9] ANSI S1.4-1990, Sound Level Meters.

[12] ITU-T Recommendation G.109 (1999), Definition of categories of speech transmission

[13] ITU-T Recommendation G.113 (1996), Transmission impairments.

[14] ITU-T Appendix I to Recommendation G.113 (1998), Transmission impairments – Appendix

[15] ITU-T Recommendation G.114 (1996), One-way transmission time.

[17] ITU-T Recommendation G.131 (1993), Control of talker echo.

[20] ITU-T Recommendation G.712 (1996), Transmission performance characteristics of pulse

[25] ITU-T Recommendation P.51 (1996), Artificial mouth.

[27] ITU-T Recommendation P.57 (1996), Artificial ears.

[29] ITU-T Recommendation P.64 (1999), Determination of sensitivity/frequency characteristics

[34] ISO 3: 1973 Preferred numbers - Series of preferred numbers.

4. Definitions, Abbreviations and Acronyms

4.2. Ear Reference Point (ERP)

4.3. HATS Position

4.4. Mouth Reference Point (MRP)

4.5. Quiet and Full Scale Code

4.6. Reference Codec

4.7. Direct Digital Processing

Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec

4.8. Sound Pressure Levels

4.9. Electric Power and Noise Levels

4.10. Abbreviations and Acronyms

AGC Automatic Gain Control

5. Handset Technical Requirements

Unless specified otherwise:

5.1. Handset Frequency Response

SMJ = 20 log (VSEND / PM) dB rel 1 V / Pa [1]

5.1.1.1. Measurement Method

Figure 3 – Handset Send Frequency Response Measurement Method

Table 2 – Co-ordinates of Handset Send Response

Limit Curve Frequency Send Response Limit

Figure 4 – Handset Send Frequency Response Mask

5.1.2. Handset Receive Frequency Response

SJE = 20 log (PE / VRCV) dB rel 1 Pa / V [2]

5.1.2.1. Measurement Method

Figure 5 – Handset Receive Frequency Response Measurement Method

Table 3 – Co-ordinates of Handset Receive Response Limits

Limit Curve Frequency Receive Response Limit

Figure 6 – Handset Receive Frequency Response Mask

5.2. Handset Loudness Ratings and Receive Volume Control

5.2.1.1. Measurement Method