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Telecommunications

Telephone Terminal Equipment


Transmission Requirements for Wideband
Digital Wireline Telephones with Headset

TIA-920.130-A February 2011


(Revision of TIA-920) 
 
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TIA-920.130-A

Table of Contents
1. SCOPE........................................................................................................................................ 1
1.1. LIMITS OF APPLICABILITY .................................................................................................... 1
1.2. CATEGORIES OF CRITERIA .................................................................................................... 1
1.3. FCC PART 68........................................................................................................................ 2
1.4. ENVIRONMENTAL ................................................................................................................. 2
1.5. SAFETY ................................................................................................................................. 2
2. NORMATIVE REFERENCES ................................................................................................ 3
3. DEFINITIONS, ABBREVIATIONS AND ACRONYMS ..................................................... 4
3.1. DEFINITIONS ......................................................................................................................... 4
3.1.1. Artificial Ear/Mouth .and Ear/Mouth Simulators ......................................................... 4
3.1.2. Codec ............................................................................................................................ 4
3.1.3. Direct Digital Signal Processing ................................................................................... 4
3.1.4. Ear Reference Point (ERP) ........................................................................................... 4
3.1.5. Electric Power and Noise Levels .................................................................................. 4
3.1.6. Mouth Reference Point (MRP) ..................................................................................... 4
3.1.7. Nominal Volume Control Setting ................................................................................. 4
3.1.8. Preferred Ear Simulator ................................................................................................ 4
3.1.9. Recommended Test Position (RTP).............................................................................. 4
3.1.10. Reference Codec .......................................................................................................... 4
3.1.11. Sound Pressure Level................................................................................................... 5
3.2. ABBREVIATIONS AND ACRONYMS ....................................................................................... 5
4. GENERAL TECHNICAL REQUIREMENTS ...................................................................... 6
4.1. REFERENCE CODECS ............................................................................................................ 6
4.1.1. L16-256 Reference Codec ............................................................................................ 6
4.1.2. Other Reference Codecs................................................................................................ 7
4.2. TEST SIGNALS....................................................................................................................... 7
4.2.1. Choice of Test Signal.................................................................................................... 7
4.2.2. Analysis......................................................................................................................... 7
4.2.3. Analysis Limitation Using ITU-T P.50......................................................................... 7
4.3. PRECAUTIONS ....................................................................................................................... 7
4.4. TEST ARRANGEMENTS ......................................................................................................... 8
4.4.1. Reference Codec Arrangement ..................................................................................... 8
4.4.2. Direct Digital Signal Processing Arrangement ............................................................. 8
4.4.3. Send General Test Setup Arrangement ......................................................................... 8
4.4.4. Receive General Test Setup Arrangement .................................................................... 9
5. HEADSET TECHNICAL REQUIREMENTS ..................................................................... 10
5.1. SEND ................................................................................................................................... 10
5.1.1. Headset Send Frequency Response............................................................................. 10
5.1.2. Headset Wideband Send Loudness Rating (SLR)....................................................... 12
5.1.3. Headset Send Noise .................................................................................................... 12
5.1.4. Headset Send Single Frequency Interference ............................................................. 12
5.1.5. Headset Send Distortion and Noise............................................................................. 13
5.1.6. Headset Send Delay .................................................................................................... 14
5.2. RECEIVE.............................................................................................................................. 16

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TIA-920.130-A

5.2.1. Headset Receive Frequency Response ........................................................................16


5.2.2. Headset Wideband Receive Loudness Rating (RLR)..................................................18
5.2.3. Headset Receive Noise ................................................................................................18
5.2.4. Headset Receive Single Frequency Interference .........................................................18
5.2.5. Headset Receive Distortion and Noise ........................................................................19
5.2.6. Headset Receive Delay................................................................................................21
5.3. HEADSET TALKER SIDETONE ..............................................................................................22
5.3.1. Headset Sidetone Masking Rating (STMR) ................................................................22
5.3.2. Headset Sidetone Delay...............................................................................................22
5.4. ECHO RETURN LOSS AND STABILITY .................................................................................23
5.4.1. Weighted Terminal Coupling Loss (TCLw)................................................................23
5.4.2. Stability Loss ...............................................................................................................24
ANNEX A (NORMATIVE) – CALCULATION OF LOUDNESS RATINGS AND
SIDETONE MASKING RATING ....................................................................................................26
A.1 WIDEBAND SEND LOUDNESS RATING (HANDSET AND HEADSET): .......................................26
A.2 WIDEBAND RECEIVE LOUDNESS RATING (HANDSET AND HEADSET): ..................................26
A.3 SIDETONE MASKING RATING (HANDSET AND HEADSET):.....................................................27
ANNEX B (NORMATIVE) – DRP TO ERP TRANSFER FUNCTION.................................28
ANNEX C (INFORMATIVE) – MEASUREMENT AND LEVEL CONVERSIONS ...........29
ANNEX D (INFORMATIVE) – R40 PREFERRED FREQUENCIES ...................................30
ANNEX E (INFORMATIVE) – RECEIVE ACOUSTIC LIMITING ....................................32
ANNEX F (INFORMATIVE) – TRANSMISSION REQUIREMENTS FOR WIDEBAND
DIGITAL WIRELINE TELEPHONES DOCUMENT(S)..............................................................33
ANNEX G (INFORMATIVE) – BIBLIOGRAPHY..................................................................35

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TIA-920.130-A

Table of Figures
Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec ......................................... 8
Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Signaling............................. 8
Figure 3 – Headset Send Test Setup Arrangement................................................................................. 9
Figure 4 – Headset Receive Test Setup Arrangement ............................................................................ 9
Figure 5 – Headset Send Frequency Response Mask........................................................................... 11
Figure 6 – Headset Receive Frequency Response Mask at ERP.......................................................... 17
Figure 7 – Terminal Coupling Loss Measurement Method ................................................................. 24
Figure 8 – Reference Corner ................................................................................................................ 25

Table of Tables
Table 1 – Co-ordinates of Headset Send Response Limits .................................................................. 11
Table 2 – Limits for Send SDNR vs Frequency................................................................................... 13
Table 3 – Limits for Send SDNR vs Level........................................................................................... 13
Table 4 – Send SDNR Stimulus Center Frequencies and Band Limits................................................ 14
Table 5 – Send SDNR Analysis Notch Frequency Band ..................................................................... 14
Table 6 – Headset Send Delay Requirements ...................................................................................... 15
Table 7 – Co-ordinates of Headset Receive Response Limits at ERP ................................................. 17
Table 8 – Limits for Receive SDNR vs. Frequency ............................................................................. 19
Table 9 – Limits for Receive SDNR vs Level...................................................................................... 19
Table 10 – Receive SDNR Stimulus Center Frequencies and Band Limits......................................... 20
Table 11 – Receive SDNR Analysis Notch Frequency Band .............................................................. 20
Table 12 – Headset Receive Delay Requirements................................................................................ 21
Table A.1– (Ref. ITU-T P.79 Annex A, Table A.2 and Table B.2) Weighting factors for calculating
wideband loudness ratings and STMR ......................................................................................... 27
Table B.1 – DRP to ERP Correction Factors ....................................................................................... 28
Table D.1 – ISO R40 and R10 Preferred Frequencies, 1/12th and 1/3rd Octave Frequencies .............. 30

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TIA-920.130-A

FOREWORD
(This foreword is not part of this standard.)

This document is a TIA Telecommunications standard produced by Working Group TR-41.3.3 of


Committee TR-41. This standard was developed in accordance with TIA 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-920.

There are six annexes in this Standard. Annex A and Annex B are normative and are considered part
of this Standard. Annex C, Annex D Annex E and Annex G are informative and are not considered
part of this Standard.

The leadership of the TR-41.3.3 VoIP and PCM Digital Transmission Performance Working Group
(Chair: Al Baum, Uniden and Allen Woo, Plantronics) acknowledge the written contributions
provided by the following individuals in the development of this standard.

Organization Representative Previous Version TIA-920.130-A


Plantronics Allen Woo 9 9 Editor
Texas Instruments Warren Karapetian - 9 Editor
AST Technology Labs Don McKinnon – 9 Editor
Nortel Networks Roger Britt 9 Editor 9
Siemens Ron Magnuson 9 Editor –
Alcatel Francois Pinier 9 –
AST Technology Labs James Bress - 9
Bob Young & Associates Bob Young 9 9
Cisco Systems Ram Jagadeesan 9 –
Cisco Systems Michael Knappe 9 –
Cisco Systems Kirit Patel 9 –
Consultant in Electroacoustics John Bareham 9 9
CenturyLink Amar Ray – 9
IEEE STIT John Bareham 9 9
IEEE STIT Rodolfo Ceruti 9 –
IEEE STIT Glenn Hess 9 –
IEEE STIT Bob Young 9 –
Intel Corporation Mike Dunn 9 –
Microsoft Hong Sodoma – 9
MWM Acoustics Glenn Hess 9 9
Nortel Networks Steve Graham 9 –
PictureTel Corporation Dave Lindbergh 9 –
Plantronics Steve Graham – 9
Plantronics Bill Rains - 9
Polycom Peter Baker 9 –
Texas Instruments Tom Harley – 9
Uniden Al Baum – 9
Vtech Steve Whitesell – 9

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TIA-920.130-A

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.

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
( http://www.tiaonline.org )

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TIA-920.130-A

Introduction
This revision of TIA-920 establishes headset telephone audio performance requirements for wideband
digital wireline telephones regardless of protocol or digital format. Significant improvements and
corrections have been made, particularly related to the use of improved ear simulators.

This standard addresses wideband performance, where wideband is nominally defined as the
frequency range between 150 and 6800 Hz. Requirements for conventional narrowband telephony, in
the frequency range between 300 and 3400 Hz are defined in ANSI/TIA-810-B.

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TIA-920.130-A

Telephone Terminal Equipment – Transmission Requirements for Wideband Digital


Wireline Telephones with Headsets
1. Scope
This standard establishes transmission performance requirements for headset devices that function as a
wideband digital wireline telephone. Transmission may be over any digital interface including Local or
Wide Area Networks, Firewire/IEEE Std 1394, Universal Serial Bus (USB), public ISDN or digital over
twisted pair wire. This includes TDM-based and packet-based (e.g. VoIP) telephones. These telephones
may be connected through modems, voice gateways, wireless access points, or PBXs, or they may be
personal computer-based telephones. Examples include, but are not limited to: ISDN telephones, digital
proprietary telephones, VoIP telephones (corded and cordless), softphones running on personal
computers, IEEE Std 802.11 telephones, USB telephony devices, DECT (CAT-iq) telephones, Bluetooth®
telephony devices, and HD (High Definition) voice telephony devices.

Some telephone systems consist of a host (such as a laptop computer) with an interface for a Universal
Serial Bus (USB) or Bluetooth device. If the host device is assumed to have a 0 dB loss plan in its default
state, then the relevant clauses of this standard are directly applicable to the USB or Bluetooth device. If
the host system provides gain or loss in the send path, receive path, or both, then the relevant clauses of
this standard apply to the composite system.

The test measurement methods in this standard reference procedures in IEEE Std 269 and ITU-T
Recommendations. Several performance measurement procedures are established, each of which yields
standardized measurement data that may be used for the determination of compliance with this standard.
Although this document may reference specific procedures or test equipment, the intent is not to be all-
inclusive. Any measurement procedure and equipment that can result in an identical measurement is
considered valid.
NOTE: If the main purpose for testing to this standard is comparison testing of different
products, rather than compliance testing, then it is important that identical test
procedures and equipment be used when testing the different products.

While the procedures may call out specific test points within the requirements, the full range of the
requirements take precedence.
1.1. Limits of Applicability
This standard is not intended to describe specific requirements for the following types of digital voice
terminal equipment: telephones with carbon transmitters, ISDN terminal adapters, cellular voice terminals
(e.g. cell phones).
1.2. 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.

1Bluetooth is a registered trademark of the Bluetooth SIG. This standard and TIA do not endorse
Bluetooth products or services.

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TIA-920.130-A

1.3. 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.
1.4. Environmental
This standard does not contain environmental requirements. Environmental requirements can be found in
ANSI/TIA-571-B.
1.5. 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-920.130-A

2. 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. ANSI and TIA maintain registers of
currently valid national standards published by them.
1. ANSI S1.4-1983 (R2006)/ANSI S1.4a-1985 (R2006), Sound Level Meters.
2. ASTM D 2240-05(2010), Standard Test Method for Rubber Property – Durometer Hardness
3. IEEE Standard 269-2010, Standard Methods for Measuring Transmission Performance of
Analog and Digital Telephone Sets, Handsets, and Headsets.
4. ITU-T Recommendation G.122 (1993-03), Influence of national systems on stability and talker
echo in international connections.
5. ITU-T Recommendation P.57 (2009-04), Artificial ears.
6. ITU-T Recommendation P.58 (1996-08) and P.58 Note (2008), Head and torso simulator for
telephonometry.
7. ITU-T Recommendation P.79 (2007-11), Calculation of loudness ratings for telephone sets.
8. ITU-T Recommendation P.501 (2009-12), Test signals for use in telephonometry.
9. ITU-T G.100.1 (2001-11), The use of decibel and of relative levels in speechband
communications
10. ITU-T Recommendation G.722 (1988-11), 7 kHz audio-coding within 64 kbit/s.

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TIA-920.130-A

3. Definitions, Abbreviations and Acronyms


For the purposes of this standard, the following definitions apply.
3.1. Definitions
3.1.1. Artificial Ear/Mouth .and Ear/Mouth Simulators
This standard uses the terms “ear simulator” and “mouth simulator” synonymously with the terms
“artificial ear” (ITU-T P.57) and “artificial mouth” (ITU-T P.51), respectively, to harmonize with IEEE
Std 269.
3.1.2. 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. An example of such a codec is the reference
codec, which is described later. In this document if the context does not indicate the reference codec,
then the term codec refers to the digital-to-digital voice signal coder and encoder.
3.1.3. Direct Digital Signal Processing
Direct digital signal processing utilized digital means of signal generation and analysis and avoids the use
of an analog network interface.
3.1.4. 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.
3.1.5. Electric Power and Noise Levels
The following electric power and noise level units are used in this standard:

The rms level of a sine wave with a peak value which is a specified number of dB below digital overload
is defined to be 0 dBm0. This shall be considered equivalent to 0 dBm (0.775 Vrms for 600 ohms) in an
analog system. Each different codec will have its own specified 0 dBm0 level relative to digital overload.
See ITU-T G.100.1.
NOTE: 0 dBm0 is generally not the maximum digital code.

3.1.6. Mouth Reference Point (MRP)


The mouth reference point is located on axis and 25 mm in front of the lip plane of a mouth simulator.
3.1.7. Nominal Volume Control Setting
The nominal volume control setting is the receive volume control setting that results in the RLR closest to
the nominal RLR value.
3.1.8. Preferred Ear Simulator
The preferred ear simulator is the Type 3.3. For alternative ear simulators, see relevant sections of IEEE
Std 269.
3.1.9. Recommended Test Position (RTP)
A Recommended Test Position (RTP) may be defined by the manufacturer for use in testing a headset
instead of the Standard Test Position. See IEEE Std 269 for the procedure to use in defining an RTP.
3.1.10. Reference Codec
A codec that approaches an ideal codec2 and has superior, well-defined, characteristics. Although power
levels are referenced to 600 ohms, the reference codec does not require a physical 600 ohm source
resistor.

2
Ideal codec characteristics, such as attenuation versus frequency distortion, idle channel noise, and
quantizing distortion are specified in ITU-T Recommendation G.712.

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TIA-920.130-A

3.1.11. Sound Pressure Level


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
Where: 0 dBSPL = 20 micro-Pascals and 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.

3.2. Abbreviations and Acronyms


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

AGC Automatic Gain Control


CPE Customer Premises equipment
CSS Composite Source Signal
DRP earDrum Reference Point
ERP Ear Reference Point
FFT Fast Fourier Transform
HATS Head and Torso Simulator
ISDN Integrated Services Digital Network
LRGP Loudness Rating Guard-ring Position
L16-256 Linear, Sixteen Bit, 256 kbit/s Codec
MRP Mouth Reference Point
OLR Overall Loudness Rating
PBX Private Branch Exchange
PCM Pulse Code Modulation
RLR Receive Loudness Rating
RFC Request for Comment (used by the IETF to define IP Protocol)
RTP Recommended Test Position
SDNR Signal to Distortion and Noise Ratio
SLR Send Loudness Rating
STMR Sidetone Masking Rating
TCLT Temporally weighted Terminal Coupling Loss
TCLw Weighted Terminal Coupling Loss
TDM Time Division Multiplexing
VAD Voice Activity Detector
VoIP Voice over Internet Protocol

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TIA-920.130-A

4. General Technical Requirements


4.1. Reference Codecs

This standard describes the use of an L16-256 reference codec for testing purposes. If the telephone
supports the L16-256 codec, then the L16-256 reference codec shall be used when testing for compliance
with this standard. If the L16-256 codec is not supported, then a telephone is considered compliant to this
standard if all requirements of this standard are met when tested with at least one of the telephone's
supported codecs.
NOTES:
1. For telephones where single or tandem codecs other than L16-256 are used (e.g.
cordless interface to a telephone set), the codec may affect the test results and some voice
transmission technologies may be unable to meet specified noise and/or distortion
requirements. Such devices need further investigation.
2. Telephones using nonlinear voice signal processing may require subjective testing.

4.1.1. L16-256 Reference Codec


The L16-256 codec is specified as follows:

• L16-256 uses uncompressed 16-bit linear PCM coding sampled at 16 kHz and having a bit rate of
256 kbit/s. The L16-256 coder shall use 16-bit signed representation with 65535 equally divided
steps between minimum and maximum signal level from -32768 to 32767.
• Each L16-256 sample value will be represented in two’s complement notation and transmitted in
network byte order with the most significant byte first.
• The L16-256 0 dBm0 level is 3.17 dB below the digital overload point. This means that a L16-
256 coded sine wave that has an rms level of 0 dBm0 has peak values which are 3.17 dB below
digital full scale.
• The L16-256 quiet code is the digital code representing the smallest encoded analog level. Full
scale code is the digital code representing the largest encoded analog level.
• A 1 kHz 0 dBm0 sine wave can be represented by the following values:
Sample [ 0] = 0
Sample [ 1] = 8705
Sample [ 2] = 16085
Sample [ 3] = 21016
Sample [ 4] = 22748
Sample [ 5] = 21016
Sample [ 6] = 16085
Sample [ 7] = 8705
Sample [ 8] = 0
Sample [ 9] = -8705
Sample [10] = -16085
Sample [11] = -21016
Sample [12] = -22748
Sample [13] = -21016
Sample [14] = -16085
Sample [15] = -8705

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TIA-920.130-A

4.1.2. Other Reference Codecs


The performance requirements in this standard were developed based on testing with an L16-256
reference codec. The performance requirements and testing methods specified in this standard may also
be applicable for devices using codecs other than L16-256. If a codec other than L16-256 is used, the 0
dBm0 level of the codec, and a means to interface to and from this codec should be provided by the
device manufacturer or a relevant standard. (Example: For a G.722 codec, 0 dBm0 = 9 dB below full
scale.) If the 0 dBm0 level is not provided, then 3.17 dB below full scale may be assumed. (Example: In
most pc-based applications in which no other definition supersedes, use 3.17 dB.)
NOTE: 0 dBm0 level has a significant effect on measurements. It is important to state the
level being used for the test.

4.2. Test Signals


4.2.1. Choice of Test Signal
Unless stated otherwise, the test stimulus signal shall be ITU-T P.50 artificial voice or an IEEE Std 269
published real speech test signal. In the event of different test results when using the P.50 and the IEEE
Std 269 real speech signal, the results with the IEEE Std 269 real speech shall take precedence. The test
signal used should be stated.

The bandwidth of the test signal used shall nominally cover 100 Hz to 8000 Hz unless otherwise
specified. The test signal levels specified in this standard shall be used.
NOTE: Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test
results.

Unless explicitly stated otherwise in this document sinusoidal tones shall not be used to measure the
following: Send, Receive, and Sidetone Spectral Response, Send, Receive, and Sidetone Loudness,
TCLw and Stability.

4.2.2. Analysis
Analysis shall be done in 1/12th octave bands unless otherwise specified.

4.2.3. Analysis Limitation Using ITU-T P.50


Wherever ITU-T P. 50 is used as a stimulus signal a range of 92 Hz to 7286 Hz shall be used:

1. Frequency response measurements for send, receive and sidetone shall have the 7286 Hz 1/12th
octave band as the highest frequency band measured.

2. To calculate SLR, RLR, and STMR, the 1/12th octave 7286 Hz band frequency response value
shall be used for the 8 kHz 1/3rd octave band data point.

4.3. Precautions
Coding, decoding, packetization and other signal processing may introduce significant delays that must be
accounted for by the measurement system. Refer to IEEE Std 269 for additional precautions regarding
test signal usage.

Telephones using nonlinear voice signal processing may require subjective testing to validate or
supplement objective measurement.

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4.4. Test Arrangements


4.4.1. Reference Codec Arrangement
A reference codec, as shown in Figure 1, may be used for testing digital telephone terminals with analog
test equipment. The interface block passes the voice channel digital bit stream to the terminal without
modification. There is no gain or loss in the 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.
Digital Reference
Point
(Junction j)
Send
vSEND

pM Decoder v
Mouth Sound Pressure
at MRP Digital
Interface
Set
pE Coder GEN
Ear Sound Pressure
at ERP
vRCV
Receive Reference Codec

Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec

4.4.2. Direct Digital Signal Processing Arrangement


Direct digital signal generation of the receive signal and analysis of the send signal may be used as shown
in Figure 2. This arrangement may be used in place of the reference codec. The interface block passes
the voice channel digital bit stream to the terminal without modification. There is no gain or loss in the
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.
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

Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Signaling
4.4.3. Send General Test Setup Arrangement
The send general test setup is shown below in Figure 3.

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Headset
VSEND

Decoder
GEN V
Digital Interface
Set Measuring
Amplifier
Coder

HATS
Quiet Room Reference Codec

Figure 3 – Headset Send Test Setup Arrangement

4.4.4. Receive General Test Setup Arrangement


The receive general test setup is shown below in Figure 4.

Quiet Room
Headset

Decoder
Digital Interface VRCV
Set

Sound Coder
Pressure GEN
Measuring
Amplifier HATS
Reference Codec

Figure 4 – Headset Receive Test Setup Arrangement

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5. Headset Technical Requirements


The requirements of this section apply to the headset and terminal together. It is not intended to be a
specification for a headset as a component separate from the terminal.

All tests shall be performed using a head and torso simulator (HATS) complying with ITU-T
Recommendation P.58 with a Type 3.3 ear simulator complying with ITU-T Recommendation P.57. The
pinna for the Type 3.3 ear simulator shall have a hardness rating of 35 ±6 Shore-OO, as measured
according to ASTM D 2240. All tests involving the headset receiver shall be done with the same HATS.

Measurements should be done at the Recommended Test Position (RTP) from the manufacturer. If the
RTP is not available, then the headset should be positioned using the guidelines outlined in IEEE Std 269.
However the CPE is tested, the test position shall be documented and used for all headset tests.

Unless otherwise specified, the headset test method is given in IEEE Std 269.

5.1. Send
The send performance of a telephone is determined by its acoustical-to-electrical transfer characteristics.
To determine the send performance, requirements for frequency response, Send Loudness Rating (SLR),
distortion, noise and delay are specified.

All send tests shall be performed with the receive volume control set to the nominal volume control
setting, unless otherwise specified.

5.1.1. Headset Send Frequency Response


The send frequency response is the overall response of the transducer, send amplifier, and the codec send
filter. Send sensitivity is the ratio of the voltage output of the reference codec, or digital bit stream
equivalent, to the sound pressure at the Mouth Reference Point (MRP) for each frequency or frequency
band (Fi) as shown in Equation 1 below:

Equation 1:
SMJ = 20 log (VSEND / PM) dB rel 1 V / Pa
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, or digital bit stream equivalent, at Fi.

5.1.1.1. Requirement
The send frequency response shall fall between the upper limit and the lower limit given in Table 1 and
shown in Figure 5. The limit curves shall be determined by straight lines joining successive co-ordinates
given in the table. The frequency response is plotted using the 1/12th octave band center frequency
measurement points on a linear dB scale against frequency on a logarithmic scale.
NOTE: The frequency response mask is a floating or “best fit” mask.

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Table 1 – Co-ordinates of Headset Send Response Limits

Limit Curve Frequency Send Response Limit


(Hz) (dB) [arbitrary level]
upper limit 100 -1
120 +4
1000 +4
2000 +9
6000 +9
8000 +8
lower limit 200 - infinity
200 -6
250 -3
1000 -3
3000 -1
6500 -10
6500 - infinity

15

10
Arbitrary Level (dB)

-5

-10

-15
100 1000 10000
Frequency (Hz)

Figure 5 – Headset Send Frequency Response Mask


5.1.1.2. Measurement Method
Measurements should be done in 1/12th octave bands or smaller, over a range of 100 Hz through 8000 Hz
using the measurement setup shown in Figure 3. Direct digital signal processing may be employed as
explained in clause 3.1.3. The test signal level shall be −4.7 dBPa at the MRP.

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5.1.2. Headset Wideband Send Loudness Rating (SLR)


The loudness rating algorithm is defined in ITU-T Recommendation P.79 and summarized in Annex A.
Loudness ratings are calculated from a telephone’s send, receive and sidetone frequency response
measurement data. They provide single number metrics, which describe how loud the telephone will
sound to a user. An important characteristic of P.79 loudness ratings is that the louder the telephone, the
more negative the loudness rating.

The SLR is the loudness loss in the send direction from the acoustic signal at the mouth reference point to
the send signal at the digital reference point. Refer to Annex A and ITU-T Recommendation P.79.
5.1.2.1. Requirement
The terminal shall be designed to have an SLR value of 10 dB, with a tolerance of ±5.0 dB.
NOTE: This SLR target is 2 dB quieter relative to a handset, because people typically speak
louder into a headset relative to a handset. The tolerance is wider than a handset
because the microphone boom position may vary relative to the MRP.
5.1.2.2. Measurement Method
Use band averaging as described in IEEE Std 269 to convert the 1/12th octave sensitivity data collected
from the send frequency response measurement in 5.1.1 to 1/3rd octave data. Then calculate SLR using
Equation A-1 of Annex A and bands 1 to 20 of Table A.1.
NOTE: When using ITU-T P.50 see 4.2.3.

5.1.3. Headset Send Noise


5.1.3.1. General
The send noise of a digital telephone is the 5 second average noise level at the digital transmit output with
the telephone headset transmitter isolated from sound input and mechanical disturbances.
5.1.3.2. Requirement
The overall send noise shall be less than or equal to -64 dBm0, A-weighted.
NOTE: This agrees with the TIA-810-B headset requirement, and is 4 dB louder than the TIA-
810-B handset requirement, ignoring the discrepancy in spectral weighting. Typically
the headset microphone boom may not be as close to the MRP as the handset.
5.1.3.3. Measurement Method
In a quiet environment (ambient noise less than 30 dBA), free of mechanical disturbances, measure the A-
weighted, 5 second average, noise level at the digital interface output or the reference codec decoder
output over the frequency range of 100 to 8000 Hz.
NOTE: A-weighting is specified for this measurement because traditional Psophometric
weighting for electrical measurements assumes a narrowband receiver at the far end
phone.

5.1.4. Headset Send Single Frequency Interference


5.1.4.1. 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.
5.1.4.2. Requirement
The A-weighted send single frequency interference shall be less than -74 dBm0.

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5.1.4.3. Measurement Method


In a quiet environment (ambient noise less than 30 dBA), free of mechanical disturbances, measure the A-
weighted noise level at VSEND with an effective bandwidth of not more than 31 Hz, over the frequency
range of 100 to 8000 Hz. If FFT analysis is used, then “Flat Top” windowing shall be employed.

5.1.5. Headset Send Distortion and Noise


5.1.5.1. Requirement
1. The ratio of the signal power to the total distortion and noise power of the signal output shall be
above the limits given in Table 2 and Table 3.
2. It is recommended that the send distortion and noise performance also be confirmed using subjective
listening tests.
Table 2 – Limits for Send SDNR vs Frequency
Center Frequency SDNR (dB)
(Hz) (+5 dBPa Send Level)
400 26
800 26
1600 26
2500 26
Table 3 – Limits for Send SDNR vs Level
Send Level at the MRP SDNR (dB)
(dBPa) (800 Hz)
-20 26
-15 26
-10 26
-5 26
0 26
NOTE: 26 dB = 5%

5.1.5.2. Measurement Method


1. Send distortion and noise is measured according to IEEE Std 269 using the setup shown in Figure 3.
2. The test stimulus is white noise band limited to 1/3rd octave and pulsed 250 ms ON (active period),
150 ms OFF (conditioning period). This sequence should be repeated ten times for a total stimulus of
four seconds. This test signal should place the telephone in a well-defined, reproducible state for the
period of the measurement.
If the headset does not maintain the reproducible state for the entire measurement period, then the 150
ms silent conditioning period may be replaced with an alternative conditioning signal (e.g. real speech
signal, longer “OFF” period, etc.).
3. Apply the test signal at the MRP, at the levels given in Table 2 using the center frequencies and band
limits specified in Table 4.

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Table 4 – Send SDNR Stimulus Center Frequencies and Band Limits


Center Frequency Band Edges
(Hz) (Hz)
400 320 480
800 675 925
1600 1375 1815
2500 2205 2855

4. Process the last 200 ms of each measured noise burst using Hanning windowing with 5 Hz resolution
FFT from 100 Hz to 8000 Hz. After time averaging, remove the data within the notch frequency
band specified in Table 5.

Table 5 – Send SDNR Analysis Notch Frequency Band


Center Frequency Notch Band Edges
(Hz) (Hz)
400 315 485
800 670 930
1600 1370 1820
2500 2200 2860

5. Calculate the ratio of the signal power to the total A-weighted distortion and noise power of the signal
output.

5.1.6. Headset Send Delay


5.1.6.1. General
Delay is a complex end-to-end issue. Certain aspects of delay can be optimized, such as the internal
hardware/firmware delay and the optimization of the jitter buffer operation, which must trade-off the
impairment of packet loss against the expected delay variation of the far-end telephone and/or the
network. Other aspects, such as packetization and depacketization are also important sources of delay but
they are a function of the selected codec and the number of speech frames per packet, so they cannot be
optimized.

When reporting compliance with this standard, only the category with the largest measured delay shall be
reported if the send and receive categories are different. If codecs or speech frame rates other than those
specified in the measurement methods are used, then they must be clearly identified when reporting
compliance.

The send delay is defined here as the time from when an acoustic signal leaves the mouth simulator
playing into a headset to the time its digitized representation arrives at the digital network interface.

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5.1.6.2. Requirement
1. Corded terminals shall be configurable so that requirements of at least Category B are met.
2. Cordless terminals should be configurable so that the requirements of at least Category C are
met.
3. The compliance category of the product shall be the same as the longest measured latency
category for send or receive. Products that do not meet category C cannot be categorized.
Table 6 – Headset Send Delay Requirements

Category Delay Time (ms)

A ≤ 35
B ≤ 55
C ≤ 70
NOTE: This standard specifies delay in terms of categories for network planning purposes,
similar to ITU-T Recommendation P.1010.

5.1.6.3. Method of Measurement


A digital audio measuring device capable of measuring the delay between an injected signal (to the mouth
simulator) and a digitally transmitted signal is connected to the mouth simulator and directly to the
network output of the telephone. All delays inherent in the measurement system itself must be subtracted
from the overall delay measurement to obtain the exact Send delay.

If the telephone uses packet based transmission, then the telephone should be set to transmit packets with
a speech frame rate of 20 ms and with one speech frame per packet.
NOTE: To minimize the delay measurement uncertainty, it is recommended to use the shortest
possible packet size for the delay measurement.

An acoustic signal of -4.7 dBPa shall be generated at the mouth simulator. The delay between the time
the pulse left the mouth to the time it was received at the telephone’s packet network interface shall be
measured. The send delay shall be used to determine the corresponding category.

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5.2. Receive
The receive performance of a telephone is determined by its electrical-to-acoustical transfer
characteristics. To determine the receive performance, requirements for frequency response, Receive
Loudness Rating (RLR), distortion, noise, volume control and delay are specified.

All receive tests shall be performed with the receive volume control set to the nominal volume control
setting as determined in clause 5.2.2, unless otherwise specified.

5.2.1. Headset Receive Frequency Response


The receive frequency response is the overall response of the codec receive filter, receive amplifier and
transducer. The receive frequency response is the ratio of the sound pressure measured in the ear
simulator to the voltage input to the reference codec, or digital bit stream equivalent, for each frequency
or frequency band (Fi) as shown in Equation 2 below:

Equation 2:
SJE = 20 log (PE / VRCV) dB rel 1 Pa / V
Where
SJE Receive Sensitivity, Junction to Ear, at Fi.
PE ERP Sound pressure measured by ear simulator at Fi. Measurement data are converted
from the Drum Reference Point, DRP to the ERP.
VRCV RMS Input voltage to the reference codec, or digital bit stream equivalent, at Fi.
5.2.1.1. Requirement
The receive frequency response shall be within the upper and lower limits given in Table 7 and shown in
Figure 6. The limit curves shall be determined by straight lines joining successive co-ordinates given in
the table. The frequency response is plotted using the 1/12th octave band center frequency measurement
points on a linear dB scale against frequency on a logarithmic scale. The frequency response mask is a
floating or “best fit” mask.
NOTE: The high frequency portion of the mask is made so that receive frequency response
that is relatively flat at free-field or diffuse-field will fit when translated to ERP.

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Table 7 – Co-ordinates of Headset Receive Response Limits at ERP

Limit Curve Frequency Receive Response Limit


(Hz) (dB) [arbitrary level]

Upper Limit 100 1


130 4
700 4
1400 9
4000 9
8000 8
Lower Limit 200 -infinity
200 -10
800 -4
4000 -4
6500 -10
6500 -infinity

15

10
Arbitrary Level (dB)

-5

-10

-15
100 1000 10000
Frequency (Hz)

Figure 6 – Headset Receive Frequency Response Mask at ERP


5.2.1.2. Measurement Method
The receive frequency response is measured according to IEEE Std 269 and using the measurement set-up
shown in Figure 4. Direct digital signal processing may be employed as explained in 3.1.3.
Measurements shall be done in 1/12th octave bands over a range of 100 Hz through 8000 Hz.
Measurements for higher frequency 1/12th octave bands should be converted to 1/3rd octave bands of
4000, 5000, 6300, 8000 Hz using band averaging per IEEE Std 269. The test signal level shall be -18.2
dBV (-16 dBm0), or digital bit stream equivalent. The frequency response measured with the ear
simulator shall be transformed to ERP, refer to Annex B. The measurement is made at the nominal
volume control setting.

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NOTES:
1. It is useful to look at 1/12th octave resolution all the way up to 8000 kHz, in order to
better understand variability in the receive side, low pass characteristics, etc.
2. When using ITU-T P.50 see 4.2.3.

5.2.2. Headset Wideband Receive Loudness Rating (RLR)


The RLR is defined in Annex A, clause A.2.
5.2.2.1. Requirement
1. Monaural headsets shall have a nominal RLR value of 0 dB, with a tolerance of -4.0/+8.0 dB.
2. Binaural and stereo headsets should have a nominal RLR value of 6 dB, with a tolerance of -4.0/+8.0
dB, for each of the left and right receivers measured separately.
NOTE: Either the terminal or the headset should have a receive volume control that is capable
of amplification and attenuation.

5.2.2.2. Measurement Method


Use band averaging as described in IEEE Std 269 to convert the 1/12th octave sensitivity data collected
from the receive frequency response measurement in 5.2.1 to 1/3rd octave sensitivity data for all required
frequency bands. Alternatively, re-measure the receive frequency response using 1/3rd octave bands.
Then calculate RLR using Equation A-2: of Annex A and bands 1 to 20 of Table A.1.

It may be necessary to repeat the receive frequency response measurement of 5.2.1.2 at various volume
control settings to determine the setting that results in the measured RLR value that is closest to the
nominal RLR value of 2 dB. This is the nominal volume control setting that is to be used for all
performance tests unless otherwise specified.
NOTE: When using ITU-T P.50 see 4.2.3.

5.2.3. Headset Receive Noise


5.2.3.1. General
The receive noise of a digital telephone is the 5 second average noise level measured at the output of the
telephone receiver with the digital telephone receiving the digital quiet code. Receive noise measurement
results must be transformed from the DRP of the ear simulator to ERP.
5.2.3.2. Requirement
1. Monaural headset receive noise shall be less than 40 dBA .
2. Binaural and stereo headsets receive noise should be less than 34 dBA, for each of the receivers
measured separately.
5.2.3.3. Measurement Method
A signal corresponding to a decoder quiet code is applied at the digital interface. The A-weighted noise
level is measured in the ear simulator over the frequency range of 100 to 8500 Hz. The ambient noise for
this measurement shall not exceed 30 dBA.

5.2.4. Headset Receive Single Frequency Interference


5.2.4.1. 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 weighted
noise level characteristics in narrow bands of not more than 31 Hz, which can then be compared to the

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TIA-920.130-A

overall weighted receive noise level. Narrow-band noise is measured at the output of the telephone
receiver with the digital telephone receiving the digital quiet code.
5.2.4.2. Requirement
The receive A-weighted single frequency interference shall be 10 dB quieter than the overall A-weighted
receive noise.
NOTE: For overall A-weighted receive noise levels lower than 25 dBA, single frequency
interference requirements and measurements are under study.

5.2.4.3. Measurement Method


A signal corresponding to a decoder quiet code is applied at the digital interface. The A-weighted noise
level is measured in the ear simulator with a selective voltmeter or spectrum analyzer, with an effective
bandwidth of not more than 31 Hz, over the frequency range of 100 to 8500 Hz. If FFT analysis is used,
then “Flat Top” windowing shall be employed. The ambient room noise for this measurement shall not
exceed 30 dBA.
5.2.5. Headset Receive Distortion and Noise
5.2.5.1. Requirement
The ratio of the signal power to the total A-weighted distortion and noise power should be greater
than or equal to the limits given in Table 8 and
1. Table 9.

2. It is recommended that the receive distortion and noise performance also be confirmed using
subjective listening tests.

Table 8 – Limits for Receive SDNR vs. Frequency


Center Frequency Receive Ratio (dB)
(Hz) (-10 dBm0 Test Level)
400 26
800 26
1600 26
2500 26

Table 9 – Limits for Receive SDNR vs Level


Receive Level Receive Ratio (dB)
(dBm0) (800 Hz)
-35 26
-30 26
-25 26
-20 26
-15 26
NOTE: 26 dB = 5%

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5.2.5.2. Measurement Method


1. Receive distortion and noise is measured according to IEEE Std. 269 using the test setup shown in
Figure 4 of this standard.

2. The test stimulus is white noise band limited to 1/3rd octave and pulsed 250 ms ON (active period),
150 ms OFF (conditioning period). This sequence should be repeated ten times for a total stimulus of
four seconds. This test signal should place the telephone in a well-defined, reproducible state for the
period of the measurement.

If the headset does not maintain the reproducible state for the entire measurement period, then the 150
ms silent conditioning period may be replaced with an alternative conditioning signal (e.g. real speech
signal, longer “OFF” period, etc.).

3. Apply the test signal at the levels given in Table 8 using the center frequencies and Band Limits
specified in Table 10.
Table 10 – Receive SDNR Stimulus Center Frequencies and Band Limits
Center Frequency Band Edges
(Hz) (Hz)
400 320 480
800 675 925
1600 1375 1815
2500 2205 2855

4. Process the last 200 ms of each measured noise burst using Hanning window with 5 Hz resolution
FFT from 100 Hz to 8000 Hz. After time averaging, remove the data within the notch frequency
band specified in Table 11.
Table 11 – Receive SDNR Analysis Notch Frequency Band
Center Frequency Notch Band Edges
(Hz) (Hz)
400 315 485
800 670 930
1600 1370 1820
2500 2200 2860

5. Calculate the ratio of the signal power to the total A-weighted distortion and noise power of the signal
output.

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5.2.6. Headset Receive Delay


See 5.1.6 for more information on delay.
5.2.6.1. General
The receive delay is defined here as the time from when a digitized representation of a signal arrives at
that telephone’s network interface to the time its analog reproduction is received at the ear simulator
coupled to the headset.

The headset receive delay requirements include any depacketization, hardware/firmware processing and
de-jitter delays, plus any delay associated with the radio link for cordless products.
5.2.6.2. Requirement
1. Corded terminals shall be configurable so that requirements of at least Category B are met.
2. Cordless terminals should be configurable so that the requirements of at least Category C are met.
3. The compliance category of the product shall be the same as the longest measured latency category
for send or receive. Products that do not meet category C cannot be categorized.
Table 12 – Headset Receive Delay Requirements

Category Delay Time (ms)

A ≤ 40
B ≤ 85
C ≤ 100
NOTE: This standard specifies delay in terms of categories for network planning purposes,
similar to ITU-T Recommendation P.1010.

5.2.6.3. Measurement Method


A digital audio measuring device capable of measuring the delay between an injected digital signal and
the output of an ear simulator is connected to the network input of the telephone and to the ear simulator.
All delays inherent in the measurement system itself must be subtracted from the overall delay
measurement to obtain the exact receive delay.

If the telephone uses packet based transmission, then the telephone should be set to receive packets with a
speech frame rate of 20 ms and with one speech frame per packet.
NOTE: To minimize the delay measurement uncertainty, it is recommended to use the shortest
possible packet size for the delay measurement.

A pulsed digital signal of -16 dBm0 shall be injected to the telephone’s network interface without test
signal jitter. The delay between the time test signal was injected at the telephone network interface to the
time it was received at the ear simulator shall be measured. The receive delay shall be used to determine
the corresponding category.

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5.3. Headset Talker Sidetone


All sidetone tests shall be performed with the receive volume control set to the nominal volume control
setting as determined in 5.2.2, unless otherwise specified.
5.3.1. Headset Sidetone Masking Rating (STMR)
5.3.1.1. General
The sidetone masking rating (STMR) of a digital telephone set is the loudness of the path from the mouth
to the ear of the same headset. STMR is calculated from the ratio of the acoustic output signal from the
receiver at the ear reference point (ERP) to the acoustic input signal at the mouth reference point (MRP)
over the specified frequency band.

Sidetone is a desirable feature for telephony applications however some products, such as cordless
headsets, may not provide sidetone.

It’s desirable for the STMR to be constant over the receive volume control range.
5.3.1.2. Requirement
1. Monaural headset STMR should be higher (quieter) than 12 dB for any adjustable receive level.
2. Binaural and stereo headset STMR should be higher (quieter) than 18 dB for each of the receivers for
any adjustable receive level.
NOTE: It is desirable for the STMR to be constant over the receive volume control range.

5.3.1.3. Measurement Method


The test signal level at the MRP shall be -4.7 dBPa. For each frequency given in Table A.1, bands 1 to
20, the sound pressure in the ear simulator shall be measured. The frequency response measured with the
ear simulator must be transformed to the ear reference point (ERP). Refer to Annex B. The STMR shall
be calculated using Equation A-3: of Annex A.

Telephones with adjustable receive levels shall be tested at the minimum, nominal and maximum volume
control settings.
NOTE: When using ITU-T P.50 see 4.2.3.

5.3.2. Headset Sidetone Delay


5.3.2.1. General
In a digital telephone, sidetone echo occurs when significant delay is introduced into the speech path
between the headset microphone and the headset receiver by the sidetone feedback algorithm. Ideally, the
sidetone signal should be a real-time signal. Sidetone delay less than 5 ms is generally perceived as
normal sidetone. Sidetone delay between 5 and 10 ms is generally perceived as unnatural sidetone, with
an uncomfortable hollow characteristic. Sidetone delay greater than 10 ms is generally perceived as a
distinct talker echo signal.
5.3.2.2. Requirement
1. Sidetone delay shall be less than 5 ms.
2. It is desirable for Sidetone delay to be less than 1 ms.
5.3.2.3. Measurement Method
See the method described in IEEE Std 269.

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5.4. Echo Return Loss and Stability


5.4.1. 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 cancellers with non-linear processing may be used on headset connections to provide
sufficient echo return loss to mitigate increased echo associated with longer network delays.

The use of echo control devices on the headset 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 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 be used consistently across products
in a black-box testing setup. The ‘proper’ measurement of echo is addressed in IEEE Std 269, Annex O.
5.4.1.1. Requirements
The normalized value of TCLw loss shall be greater than 52 dB for packet-based headsets and 45 dB for
TDM-based headsets and with SLR normalized to 10 dB and RLR normalized to 0 dB. This requirement
shall be met for at least one of the two input signal levels specified in 5.4.1.2.
NOTES:
1. As a calculation example, if the measured TCLw is 48 dB, the measured SLR is 11 dB and the
measured RLR is -2 dB, then the normalized value of TCLw = TCLw measured + (10 - SLR)
dB + (0 - RLR) dB = 48 dB + (10 - 11) dB + (0 - (-2)) dB = 49 dB.
2. The requirement of 52 dB for packet-based headsets is a function of the -16 dBm0 test signal
level and the -64 dBm0A send noise requirement. Measuring TCLw > 52 dB can be difficult.
3. If equipped with adjustable receive level, the un-normalized 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 un-normalized TCLw (maximum receive level) = 45 dB - 12 dB = 33 dB.
4. The echo impairment perceived by the person at the far-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 may not provide satisfactory
echo loss for connections that have a long delay.
5. 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 Std 1329.) The performance requirements may
need to be changed when using this method.

5.4.1.2. Measurement Method


The TCLw measurement shall be made at an input signal level of -16 and -10 dBm0. The test shall be
performed in a quiet environment (the ambient noise level shall be less than 30 dBA).

The TCLw measurement shall not be performed using a sinusoidal test signal for the receive path input.
The test signal may be a composite source signal (CSS) as defined in ITU-T P.501 or bursted white noise
as defined in IEEE Std. 269. The test signal shall be band-limited to 100 through 8000 Hz. The
calibration shall be determined during the ON portions of the test signal, not the average of the on and off
times. The measurement shall be performed after system stability is reached (including convergence of
any echo algorithms): this shall be accomplished by invoking the test signal for at least 2 seconds before
the actual measurement occurs.

23
TIA-920.130-A

The attenuation from digital input (receive) to digital output (send) is measured at 1/12th octave bands,
using the measurement arrangement shown in Figure 7.

The weighted terminal coupling loss is calculated according to ITU-T G.122 Annex B.4 (trapezoidal rule)
using the frequency range of 300 to 6700 Hz rather than 300 to 3400 Hz.
NOTE: ITU-T P.311 uses 100 to 8 kHz, and G.122 does not explicitly list frequencies for an
extension to wideband. Some suggest 150 to 6800 Hz would be more appropriate.

Headset Mounted on HATS


vSEND (Echo Return)

Decoder v
Digital
Interface
Set

Coder GEN

Anechoic Chamber v RCV


Reference Codec

Figure 7 – Terminal Coupling Loss Measurement Method

5.4.2. 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 frequency.
5.4.2.1. Requirement
1. Stability loss (i.e., minimum loss, at any frequency) using both measurement method 1 in 5.4.2.2.1
and measurement method 2 in 5.4.2.2.2 shall be greater than 6 dB.
2. It is desirable that Stability loss using both method 1 and method 2 be greater than 10 dB.
3. Telephone sets with adjustable receive level should maintain stability over the entire range of
adjustable receive levels.
5.4.2.2. Measurement Method
The stability measurement shall be made at input signal levels of -16 and -10 dBm0. The stability
measurement shall not be performed using a sinusoidal test signal for the receive path input. The test
signal may be CSS or bursted white noise, band-limited to 100 through 8000 Hz. The recommended
pattern is 250 ms of noise (on), 150 ms silence (off). The measurement and calibration shall be
determined during the ON portions of the signal, not the average of on and off times. With the headset
and transmission circuit fully active, measure the attenuation from the digital input to the digital output
using Method 1 and Method 2.

24
TIA-920.130-A

5.4.2.2.1. Method 1
Place the headset in the reference corner, as shown in Figure 8. Test stability while placing headset on
both sides (face down and face up) with mic boom closest to corner. If there is a retractable mic boom,
extend mic to normal use position for HATS. The headset shall be placed along the diagonal from the
apex of the reference corner to the outside corner, with the receiver end of the headset 25 cm from the
apex. The telephone set shall be fully active.

The reference corner consists of three perpendicular planes, smooth, hard surfaces extending 0.5 m from
the apex of the corner.
Side B

cm
25

Side A
Base

Figure 8 – Reference Corner


5.4.2.2.2. Method 2
Place the headset with the receiver and mouthpiece facing a hard, smooth surface free of any other object
for 50 cm. The telephone set shall be fully active.

25
TIA-920.130-A

Annex A (Normative) – Calculation of Loudness Ratings and Sidetone Masking


Rating
This Annex details the loudness rating calculations and weighting factors relevant to the TIA-920 series
of documents. Loudness ratings are a measure of loudness loss and are used in network planning 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 wideband Overall Loudness Rating (OLR). The wideband Send Loudness Rating
(SLR) is the loudness loss from the MRP to the electrical output. The wideband Receive Loudness
Rating (RLR) is the loudness loss from the electrical input to the ERP. The Sidetone Masking Rating
(STMR) is the loudness loss from the MRP to the ERP via the electric sidetone path.

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 ITU-T
Recommendation P.79. ITU-T Recommendation P.79 provides information on the derivation of the
loudness rating algorithm.

Loudness ratings determined in accordance with P.79 are analogous to loss, resulting in the characteristic
that the louder the telephone, the more negative the loudness rating.

ITU-T Recommendation P.79 Annex A, Table A.2 has the weighting factors used for wideband SLR and
RLR calculation. ITU-T Recommendation P.79 Table B.2 has the weighting factors used for STMR
calculation. For convenience, the P.79 weighting information is included in this document in Table A.1.
A.1 Wideband Send Loudness Rating (Handset and Headset):
Equation A-1:
Band 20

SLR = - 57.1 log10 ∑ 10


(0.1 * 0.175 * (SMJ – Wsi ))
i = Band 1
Where:
i Frequency bands from Table A.1, bands 1-20.
SMJ Send frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured per
this standard.
Wsi Send weighting factor from Table A.1.

A.2 Wideband Receive Loudness Rating (Handset and Headset):


Equation A-2:
Band 20

RLR = - 57.1 log10 ∑ 10(0.1 * 0.175 * (SJE


– Wri ))
i = Band 1
Where:
i Frequency bands from Table A.1, bands 1-20.
SJE Receive frequency response data (Sensitivity, Junction-to-Ear Reference Point) in
dBPa/V measured per this standard. See Annex B for DRP to ERP information.
Wri Receive weighting factor from Table A.1.

NOTE: There is no Leakage Correction, LE, when using the Type 2, 3.2, 3.3 or Type 3.4 ear
simulator.

26
TIA-920.130-A

A.3 Sidetone Masking Rating (Handset and Headset):


Equation A-3:
Band 20

STMR = - 44.4 log10 ∑ 10(0.1 * 0.225 * (SmeST – WMSi))


i = Band 1
Where:
i Frequency bands from Table A.1, bands 1-20.
SmeST Sidetone frequency response data (Sensitivity, Mouth-to-Ear) in dB Pa/Pa measured per
this standard.
WMSi Sidetone weighting factor from Table A.1.

Table A.1– (Ref. ITU-T P.79 Annex A, Table A.2 and Table B.2)
Weighting factors for calculating wideband loudness ratings and STMR

Band No. Mid-frequency Send Receive Sidetone


(Hz) Wsi Wri WMSi
1 100 154.5 152.8 110.4
2 125 115.4 116.2 107.7
3 160 89.0 91.3 104.6
4 200 77.2 85.3 98.4
5 250 62.9 75.0 94.0
6 315 62.3 79.3 89.8
7 400 45.0 64.0 84.8
8 500 53.4 73.8 75.5
9 630 48.8 69.4 66.0
10 800 47.9 68.3 57.1
11 1000 50.4 69.0 49.1
12 1250 59.4 75.4 50.6
13 1600 57.0 70.7 51.0
14 2000 72.5 81.7 51.9
15 2500 72.9 76.8 51.3
16 3150 89.5 93.6 50.6
17 4000 117.3 114.1 51.0
18 5000 157.3 144.6 49.7
19 6300 172.2 165.8 50.0
20 8000 181.7 166.7 52.8

NOTE: ITU-T P.79 advises that the sealed weighting be used for STMR computations, even
though the HATS is not sealed.

27
TIA-920.130-A

Annex B (Normative) – DRP to ERP Transfer Function


Frequency response measurements using the Type 3.3 or 3.4 ear are made at the ear Drum Reference
Point (DRP). Data collected shall be converted to the Ear Reference Point (ERP) before comparing
against the tolerance limits or calculating loudness ratings. The conversion is accomplished by adding the
correction factor, SDE, given in Table B.1to the measured data.

For complete information refer to ITU-T Recommendation P.58 and IEEE Std 269.
Table B.1 – DRP to ERP Correction Factors
SDE Frequency SDE Frequency SDE Frequency SDE
Frequency
(dB) (Hz) (dB) (Hz) (dB) (Hz) (dB)
92 0.1 290 −0 3 917 −1 3 2901 −11 0
97 0.0 307 −0.2 972 −1.4 3073 −10.5
103 0.0 325 −0.2 1029 −1.8 3255 −10.2
109 0.0 345 −0.2 1090 −2.0 3447 −9.1
115 0.0 365 −0.4 1155 −2.3 3652 −8.0
122 0.0 387 −0.5 1223 −2.4 3868 −6.9
130 0.0 410 −0.4 1296 −2.6 4097 −5.8
137 0.0 434 −0.6 1372 −3.1 4340 −5.0
145 0.0 460 −0.3 1454 −3.3 4597 −4.2
154 0.0 487 −0.7 1540 −3.9 4870 −3.3
163 0.0 516 −0.6 1631 −4.4 5158 −2.7
173 −0.1 546 −0.6 1728 −4.8 5464 −2.4
183 −0.1 579 −0.6 1830 −5.3 5788 −2.4
193 0.0 613 −0.6 1939 −6.0 6131 −2.5
205 0.1 649 −0.8 2053 −6.9 6494 −3.3
218 0.0 688 −0.8 2175 −7.5 6879 −4.5
230 −0.1 729 −1.0 2304 −8.1 7286 −5.9
244 −0.2 772 −1.1 2441 −9.1 7718 −9.0
259 −0.3 818 −1.1 2585 −9.5 8175 −14.2
274 −0.3 866 −1.2 2738 −10.4 8659 −20.7

28
TIA-920.130-A

Annex C (Informative) – Measurement and Level Conversions

The following describes how to convert between various units of measurement used in telephone testing.

0 dBm (0 VU) is accepted as 1 mW, typically using a circuit impedance of 600 Ω or 900 Ω.

0 dBm = 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),
therefore subtract 1.8 dB from the measurement.

At this point, depending on the impedance, conversion factors can be applied dB for dB to the measured
or calculated result. For example, to convert a 600 Ω, -20 dBm signal to dBV, subtract 2.2 to get -22.2
dBV. Another example is if -20 dBm is measured across 600 Ω, then when measuring across 900 Ω, add
a correction of -1.8 dB to get -21.8 dBm (since less power is dissipated by the higher resistance).

29
TIA-920.130-A

Annex D (Informative) – R40 Preferred Frequencies


The ISO 3, R40 basic series preferred frequencies are listed in Table D.1 . The frequencies highlighted in
Italics are the R10 series of preferred frequencies. The R40 series of preferred frequencies is based on
1/12th octave frequencies, but the numbers are rounded in a convenient pattern. The R10 series is based
on the 1/3rd octave frequencies.
Table D.1 – ISO R40 and R10 Preferred Frequencies, 1/12th and 1/3rd Octave Frequencies
R40 Preferred 1/12 Oct. Band 1/3 Oct. Band R10 Preferred
Frequencies, Hz. Center Freq, Hz. Center Freq, Hz. Frequencies, Hz.
90
91.73
95
97.16
100 100.00 100
102.92
106
109.02
112
115.48
118
122.32
125 125.89 125
129.57
132
137.25
140
145.38
150
153.99 158.49 160
160
163.12
170
172.78
180
183.02
190
193.87
200 199.53 200
205.35
212
217.52
224
230.41
236
244.06
250 251.19 250
258.52
265
273.84
280
290.07
300
307.26
315 316.23 315
325.46
335
344.75
355
365.17
375
386.81
400 398.11 400
409.73
425
434.01
450
459.73
475
486.97
500 501.19 500
515.82
530
546.39
560
578.76
600
613.06
630 630.96 630
649.38
670
687.86
710
728.62
750
771.79
800 794.33 800
817.52
850
865.96
900
917.28 1000.00 1000
950
971 63
30
TIA-920.130-A

1000
1029.20
1060
1090.18
1120
1154.78
1180
1223.21
1250
1295.69
1258.93 1250
1320
1372.46
1400
1453.78
1500
1539.93
1600 1584.89 1600
1631.17
1700
1727.83
1800
1830.21
1900
1938.65
2000 1995.26 2000
2053.53
2120
2175.20
2240
2304.09
2360
2440.62
2500 2511.89 2500
2585.23
2650
2738.42
2800
2900.68
3000
3072.56
3150 3162.28 3150
3254.62
3350
3447.47
3550
3651.74
3750
3868.12
4000 3981.07 4000
4097.32
4250
4340.10
4500
4597.27
4750
4869.68
5000 5011.87 5000
5158.22
5300
5463.87
5600
5787.62
6000
6130.56
6300 6309.57 6300
6493.82
6700
6878.60
7100
7286.18
7500
7717.92
8000 7943.28 8000
8175.23
8500
8659.64
9000
9172.76
9500
9716.28
10000 10000.00 10000
10292.01
10600
11200 10901.84

31
TIA-920.130-A

Annex E (Informative) – Receive Acoustic Limiting

Previous versions of this standard have included Receive Acoustic Limiting requirements. These
requirements are now covered in UL/CSA: 60950-1.

32
TIA-920.130-A

Annex F (Informative) – Transmission Requirements for Wideband Digital


Wireline Telephones Document(s)

dBm: (sometimes also called dBmW) is an abbreviation for the power ratio in decibels (dB) of the
measured power referenced to one milliwatt (mW). The logarithmic expression for power, referenced to
1 milliwatt is dBm = 10 log (power / 1 mW). For electrical power level measurements, 0 dBm can also
be referred to as the power of 0.775 volts RMS Sinusoidal signal applied across a defined resistive load
(often 600 Ohms in telecommunications), which results in a reference active power level of 1 mW. Note:
specifying dBm without specifying the impedance may be misleading. For digital networks, since there is
no impedance used, it should also be defined by the corresponding Sinusoidal peak code value (dBFS, or
dBov) for the given codec.

dBov/dBFS: dBov stands for dB overload. It is the rms signal level of a digital signal relative to the
overload or maximum peak level of the digital format. This is also commonly referred to as dBFS (Full
Scale). For example, a rectangular function with only the positive or negative maximum number is 0
dBov; a single frequency tone with peak at maximum level is -3.01 dBov. Note: 0 dBm0 equals (-3.17-
3.01) = -6.18 dBFS.

TLP-Max: Maximum Transmission Level Point is defined as the maximum dBm power measured across
a defined impedance (e.g. 600 Ohm resistive) by applying a full scale (0 dBFS or dBov) Sinusoidal
signal (just below its clipping point). Consequently, Max-TLP for A-law and μ-law codecs in a system,
are defined as +3.14 dBm and +3.17 dBm, respectively.

0-TLP: Zero Transmission Level Point is an arbitrarily established point in a system relative to which
transmission levels at all other points in the system are measured. 0-TLP for A-law and μ-law codecs in a
system is defined as -3.14 dB and -3.17 dB, below TLP-Max, respectively. In digital transmissions
networks, the 0-TLP is often referred to as 0 dBm0, which is also equivalent to 0 dBm).

dBm0: Is defined as the power level in dBm relative to a reference point called the Zero Transmission
Level Point, or 0-TLP. This level for A-law and μ-law codecs is defined as -3.14 dB and -3.17 dB below
the full scale (or TLP-Max.), respectively. Consequently, for an A-law and μ-law codecs, the 0 dBm0
(which is also referred to as 0 digital reference point or digital milliwatt) and 0-TLP are all the same, and
may be used interchangeably. Also note that 0-TLP, sometimes is referred to as 0 dBr (or dB relative to 0
dBm) in some earlier telecom literature.

For the purposes of this standard, 0 dBm0 (or 0-TLP) is the RMS level of a Sinusoidal wave with -3.17
dB below L16-256 TLP-Max level, and -9 dB below G.722 codecs TLP Max level. Consequently, the 0
dBm0 digital reference point may be defined for 0 dBm Sinusoidal test signal considering the following
TLP-Max designations:

Codec Type Defined TLP-Max in dBm 0 dBm0 reference point


at 600 Ohm is defined as 0 dBm at
A-law +3.14 (TLP-Max) – 3.14
μ-law +3.17 (TLP-Max) – 3.17
L16-256 +3.17 (TLP-Max) – 3.17
G.722 +9.00 (TLP-Max) – 9.0

33
TIA-920.130-A

NOTE: It is the responsibility of the transcoder to correctly implement the digital full scale to
dBm0 level translation for each codec when digital signals are translated from one
codec format to another, and to ensure that signal levels at one port of the transcoder
are correctly translated to the appropriate signal levels at the other port of the
transcoder.

This simply means that a continuity test tone (1004 Hz) entering one end of the network at 0 dBm level
(terminated with 600 Ohm impedance), should produce a 0 dBm output signal across 600 Ohm load on
the other end of the network independent of any codec types or transcoding used in between the
two points.

34
TIA-920.130-A

Annex G (Informative) – Bibliography


The following standards are identified for informational purposes only and do not constitute provisions of
this Standard. At the time of publication, the editions indicated were valid. All standards are subject to
revision, and users this Standard may want to consider the most recent editions of the standards indicated
below when looking for further information. ANSI and TIA maintain registers of currently valid national
standards published by them.

1. ANSI/TIA-571-B-2007, Telecommunications Telephone Terminal Equipment Electrical, Thermal,


Mechanical Environmental Performance Requirements.
2. ANSI/TIA-810-B-2006, Telecommunications Telephone Terminal Equipment Transmission
Requirements for Narrowband Digital Telephones.
3. IETF RFC3551 (2003) (L16-256 codec) RTP Profile for Audio and Video Conferences with
Minimal Control.
4. ITU-T Recommendation P.51 (1996-08), Artificial mouth.
5. ITU-T Recommendation P.311 (2005-06), Transmission characteristics for wideband (150-7000
Hz) digital handset telephones.
6. ITU-T Recommendation P.360 (2006-07), Efficiency of devices for preventing the occurrence of
excessive acoustic pressure by telephone receivers.
7. ITU-T Recommendation P.1010 (2004-07), Fundamental voice transmission objectives for VoIP
terminals and gateways.
8. ITU-T Recommendation G.712 (2001-11), Transmission performance characteristics of pulse code
modulation channels.

35