TR41.3.3-08-08-004-L
Telecommunications
Telephone Terminal Equipment
Transmission Requirements for Wideband Digital
Telephones
PN-3-4705-RV1 (to become ANSI/TIA-920-A)
Draft 11 (Aug 4, 2008)
Editor: Tom Harley
tharley@ti.com
Changes Note: DISABLE CHANGE TRACKING BEFORE SAVING.
Draft 11 Changes made during Portland OR meeting
Draft 10 Changes made during New Orleans meeting
Draft 9 Changes made during Albuquerque NM meeting
Draft 7 & 8 Changes made during Ottawa, ON meeting
Draft 5 & 6 (Tom & Allen Wo & Roger ):
1. Changes made during St. Petersburg, FL meeting
2. Changes agreed to at FL but deferred, including duplicating handset updates into headset section.
3. Acknowledgements page updated.
Draft 4 (Tom):
1. Changed handsfree to speakerphone.
2. Scattered G711 references deleted.
3. Changed speakerphone RLR=16 dB target to 2 dB by adding 14 dB correction.
4. Changed handset frequency responses for send and receive, as agreed.
5. Updated references.
6. TCLw handset 52 dB.
7. Specified 10N for high leak position of handset, added “artificial ear/mouth vs ear/mouth
simulator” and “preferred ear simulator” sections to mirror 810B updates.
8. Reordered Section 4 and 5 to be more like 810B.
9. Rewrote handset Rx frequency response requirement.
Draft 3 (Roger):
1. In 2.1 Scope adds ITU-T reference, deletes “comparison of different products”. Note added that
when comparing different products, following identical procedures is important.
2. In 3. Normative References adds IETF RFC 1890 RTP reference for L16-256.
3. In 4.1 Codec L16-256 definition refined.
4. In 6.1 Vocoder Mandatory Requirements, G711 reference deleted.
Draft 2 (Roger):
1. Tweaked first sentence Scope.
2. In 6.1 deleted “mandatory G.711 codec for narrowband operation” from 1 st sentence and deleted
2nd paragraph about also meeting TIA-810-A requirements.
Draft 1 (Roger):
3. Imported the 810-B changes into the handset section, clause 7 and some of Annex A.
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
TABLE OF CONTENTS
1.
INTRODUCTION ..................................................................................................................... 1
2.
OVERVIEW .............................................................................................................................. 2
2.1.
SCOPE .................................................................................................................................. 2
2.2.
2.3.
2.4.
LIMITS OF APPLICABILITY ................................................................................................ 2
CATEGORIES OF CRITERIA ................................................................................................ 2
FCC PART 68 ....................................................................................................................... 3
2.5.
2.6.
ENVIRONMENTAL .............................................................................................................. 3
SAFETY ................................................................................................................................ 3
3.
NORMATIVE REFERENCES ................................................................................................ 4
4.
DEFINITIONS, ABBREVIATIONS AND ACRONYMS ..................................................... 6
4.1.
4.2.
4.3.
4.4.
4.5.
4.6.
4.7.
4.8.
CODEC ................................................................................................................................. 6
EAR REFERENCE POINT (ERP) .......................................................................................... 6
ARTIFICIAL EAR/MOUTH VS. EAR/MOUTH SIMULATOR............................................... 6
HATS POSITION .................................................................................................................. 6
NOMINAL VOLUME CONTROL SETTING ......................................................................... 6
REFERENCE VOLUME CONTROL SETTING ...................................................................... 6
PREFERRED EAR SIMULATOR ........................................................................................... 6
STANDARD TEST POSITION .............................................................................................. 6
4.9.
4.10.
4.11.
4.12.
4.13.
4.14.
RECOMMENDED TEST POSITIONS (RTP) ......................................................................... 7
MOUTH REFERENCE POINT (MRP) .................................................................................. 7
SESSION DESCRIPTION PROTOCOL (SDP) ....................................................................... 7
REFERENCE CODEC ........................................................................................................... 8
DIRECT DIGITAL PROCESSING ......................................................................................... 8
SOUND PRESSURE LEVELS................................................................................................ 9
4.15.
4.16.
4.17.
ELECTRIC POWER AND NOISE LEVELS ........................................................................... 9
50TP ..................................................................................................................................... 9
ABBREVIATIONS AND ACRONYMS .................................................................................. 9
4.18. TEST SIGNALS................................................................................................................... 10
1.1.1. Choice of Test Signal .................................................................................................. 10
1.1.2. Frequency Tolerance of Test Signals and Analysis .................................................... 10
4.19. TESTING MODE ................................................................................................................. 11
4.20. PRECAUTIONS ................................................................................................................... 11
5.
GENERAL TECHNICAL REQUIREMENTS .................................................................... 12
5.1.
VOICE CODING MANDATORY REQUIREMENTS ........................................................... 12
1.1.3. Transmission Format of L16-256 Codec .................................................................... 12
1.1.4. Overload Point ............................................................................................................ 13
1.1.5. Quiet Code and Full Scale Code ................................................................................. 13
1.1.6. 0 dBm0 (Digital Milliwatt) ......................................................................................... 13
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6.
HANDSET TECHNICAL REQUIREMENTS ..................................................................... 14
6.1.
HANDSET FREQUENCY RESPONSE ................................................................................. 14
1.1.7. Handset Send Frequency Response ............................................................................. 14
1.1.8. Handset Receive Frequency Response ........................................................................ 16
6.2.
HANDSET WIDEBAND LOUDNESS RATINGS AND RECEIVE VOLUME CONTROL ..... 18
1.1.9. Handset Wideband Send Loudness Rating (SLR) ...................................................... 18
1.1.10. Handset Wideband Receive Loudness Rating (RLR) ................................................ 18
1.1.11. Handset Receive Volume Control Range ................................................................... 19
1.1.12. Magnetic Field for Hearing Aid Coupling ................................................................. 19
1.1.13. Handset Talker Sidetone (STMR) .............................................................................. 20
1.1.14. Handset Sidetone Delay ............................................................................................. 20
6.3.
HANDSET NOISE ............................................................................................................... 20
1.1.15. Handset Send Noise .................................................................................................... 20
1.1.16. Handset Send Single Frequency Interference............................................................. 21
1.1.17. Handset Receive Noise ............................................................................................... 21
1.1.18. Handset Receive Single Frequency Interference ........................................................ 21
6.4.
HANDSET RECEIVE COMFORT NOISE (ADVISORY) ...................................................... 22
1.1.19. General ....................................................................................................................... 22
1.1.20. Measurement Method ................................................................................................. 22
1.1.21. Requirement ............................................................................................................... 22
6.5.
HANDSET DISTORTION AND NOISE ................................................................................ 23
1.1.22. Handset Send Distortion and Noise............................................................................ 23
1.1.23. Handset Receive Distortion and Noise ....................................................................... 24
6.6.
WEIGHTED TERMINAL COUPLING LOSS (TCLW) ......................................................... 25
1.1.24. Measurement Method ................................................................................................. 25
1.1.25. Requirements .............................................................................................................. 26
6.7.
STABILITY LOSS ............................................................................................................... 27
1.1.26. Measurement Method ................................................................................................. 28
1.1.27. Requirement ............................................................................................................... 28
6.8.
LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE INPUT) .......... 28
1.1.28. General ....................................................................................................................... 28
1.1.29. Measurement Method ................................................................................................. 28
1.1.30. Requirements .............................................................................................................. 28
6.9.
SHORT DURATION MAXIMUM ACOUSTIC PRESSURE (PEAK) ..................................... 29
1.1.31. General ....................................................................................................................... 29
1.1.32. Measurement Method ................................................................................................. 29
1.1.33. Requirements .............................................................................................................. 29
6.10. VOIP TELEPHONE DELAY ................................................................................................ 29
1.1.34. Requirement ............................................................................................................... 29
1.1.35. Handset Send Delay ................................................................................................... 29
1.1.36. Handset Receive Delay............................................................................................... 30
7.
HEADSET TECHNICAL REQUIREMENTS...................................................................... 31
7.1.
HEADSET FREQUENCY RESPONSE.................................................................................. 31
1.1.37. Headset Send Frequency Response ............................................................................ 31
1.1.38. Headset Receive Frequency Response ....................................................................... 34
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7.2.
HEADSET WIDEBAND LOUDNESS RATINGS ................................................................. 36
1.1.39. Headset Wideband Send Loudness Rating (SLR) ..................................................... 36
1.1.40. Headset Wideband Receive Loudness Rating (RLR) ................................................ 36
1.1.41. Headset Talker Sidetone ............................................................................................ 37
7.2.2. HEADSET SIDETONE DELAY ........................................................................................... 37
7.3.
HEADSET NOISE ............................................................................................................... 37
1.1.42. Headset Send Noise ................................................................................................... 37
1.1.43. 7.3.2 Headset Send Single Frequency Interference ................................................ 38
1.1.44. ........................................................................................................................................ 38
1.1.45. 7.3.3 Headset Receive Noise .................................................................................. 38
1.1.46. Headset Receive Single Frequency Interference ....................................................... 39
7.4.
HEADSET DISTORTION AND NOISE................................................................................ 39
1.1.47. Headset Send Distortion and Noise ........................................................................... 39
1.1.48. Headset Receive Distortion and Noise ...................................................................... 40
7.5.
WEIGHTED TERMINAL COUPLING LOSS (TCLW) ........................................................ 41
1.1.49. Measurement Method ................................................................................................ 41
1.1.50. Requirements ............................................................................................................. 42
7.6.
HEADSET LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE) ... 44
1.1.51. Requirements ............................................................. Error! Bookmark not defined.
7.7.
SHORT DURATION MAXIMUM ACOUSTIC PRESSURE (PEAK) .................................... 45
1.1.52. Requirements ............................................................................................................. 45
8.
SPEAKERPHONE TECHNICAL REQUIREMENTS (ADVISORY) .............................. 46
8.1.
SPEAKERPHONE FREQUENCY RESPONSE ..................................................................... 46
1.1.53. Speakerphone Send Frequency Response .................................................................. 46
1.1.54. Speakerphone Receive Frequency Response ............................................................. 48
8.2.
SPEAKERPHONE WIDEBAND LOUDNESS RATINGS AND RECEIVE VOLUME
CONTROL ........................................................................................................................................ 49
1.1.55. Speakerphone Wideband Send Loudness Rating (SLR) ........................................... 49
1.1.56. Speakerphone Wideband Receive Loudness Rating (RLR) ...................................... 50
1.1.57. Speakerphone Receive Volume Control .................................................................... 50
8.3.
SPEAKERPHONE NOISE.................................................................................................... 50
1.1.58. Speakerphone Send Noise.......................................................................................... 50
1.1.59. Speakerphone Send Single Frequency Interference................................................... 50
1.1.60. Speakerphone Receive Noise ..................................................................................... 51
1.1.61. Speakerphone Receive Single Frequency Interference .............................................. 51
8.4.
SPEAKERPHONE DISTORTION AND NOISE .................................................................... 51
1.1.62. Speakerphone Send Distortion and Noise.................................................................. 52
1.1.63. Speakerphone Receive Distortion and Noise ............................................................. 52
8.5.
WEIGHTED TERMINAL COUPLING LOSS (TCLW) ........................................................ 54
1.1.64. Measurement Method ................................................................................................ 54
1.1.65. Requirements ............................................................................................................. 54
8.6.
STABILITY LOSS ............................................................................................................... 54
1.1.66. Measurement Method ................................................................................................ 54
1.1.67. Requirement ............................................................................................................... 54
ANNEX A (NORMATIVE) – CALCULATION OF LOUDNESS RATINGS ............................. 55
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
ANNEX B (INFORMATIVE) – MEASUREMENT AND LEVEL CONVERSIONS ................. 58
ANNEX C (INFORMATIVE) – R40 PREFERRED FREQUENCIES ......................................... 60
ANNEX D (NORMATIVE) – DRP TO ERP TRANSFER FUNCTION ....................................... 61
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-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.
The TR-41.3.3 VoIP and PCM Transmission Performance Working Group acknowledge the
contribution made by the following individuals in the development of this standard.
Name:
Representing:
Al Baum
Tom Harley
Roger Britt
Uniden
Texas Instruments
Nortel
Miguel De Araujo
Nortel
Michael Chen
VTech
Juan Corona
Steve Whitesell
Joachim Pomy
James Bress
John Bareham
Glenn Hess
Amar Ray
Steve Graham
Allen Woo
Kirit Patel
Bob Young
Ron Magnuson
VTech
VTech
Avaya/ETSI
AST Technology Labs
Consultant
MWM Acoustics
Embarq
Plantronics
Plantronics
Cisco Systems
Consultant
Consultant
Chair
Editor
Emeritus Chair
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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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
1. Introduction
This revision of TIA-920 establishes handset, headset and speakerphone telephone audio
performance requirements for wideband digital telephones regardless of protocol or digital format. A
number of improvements and corrections have been made, particularly related to the use of improved
ear simulators.
This standard addresses wideband performance, where wideband is 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.
Editor’s note; upper limit frequency on all lower masks recently went from 6800 to 6500 Hz.
Stability and TCLw test up to 6700 Hz. Should 6800 Hz reference above be lowered?
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
2. Overview
2.1. Scope
This standard establishes voice performance requirements for devices that function as a wideband
digital telephone. Transmission may be over any digital interface including wired, or wireless, Local
or Wide Area Networks, Firewire/IEEE1394, 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, PBXs, or
personal computer-based telephones. Examples include, but are not limited to: ISDN telephones,
digital proprietary telephones, VoIP telephones (corded and cordless), softphones (such as a laptop
computers), IEEE 802.11 telephones, USB telephones, USB devices, DECT telephones, Bluetooth®
Telephones and Bluetooth devices.
For telephone systems that incorporate a Universal Serial Bus (USB) type interface or a Bluetooth
type interface to a host (such as a laptop computer), it may be desirable for the USB or Bluetooth
device to meet the requirements of relevant clauses of this standard, where the host device is assumed
to have a 0 dB loss plan, in its default state. It may be desirable for the device to provide gain
adjustment for both the send and receive channels. When connected to a host device, the full system
shall then meet all of the associated requirements of this standard. A USB or Bluetooth device may
have a handset, headset or speakerphone configuration.
Technical requirements are specified for handset, headset and speakerphone modes of operation
regardless of the technology used to couple the handset or headset to the telephone.
The test measurement methods in this standard reference procedures in IEEE Standards 269, 269a
and 1329 where applicable, as well as the appropriate 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 allinclusive. 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.
2.2. 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 (cell phones), and group audio terminals.
2.3. 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
1
Bluetooth is a registered trademark of the Bluetooth SIF. This standard and TIA do not endorse
Bluetooth products or services.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
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.
2.4. 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.
2.5. Environmental
This standard does not contain environmental requirements. Environmental requirements can be
found in ANSI/TIA/EIA-571-B.
2.6. Safety
This standard does not contain safety requirements. Compliance with the applicable UL and CSA
safety standards may be required in certain locations.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-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/EIA/TIA-571-B-1999, Environmental Considerations.
[2]
ANSI/EIA/TIA-810-B-2007, Transmission Requirements for Narrowband Voice over IP and
Voice over PCM Digital Wireline Telephones. ( http://www.tiaonline.org/standards/ip/ )
[3]
ANSI S1.4-1990, Sound Level Meters.
[4]
ASTM D 2240-2002, Standard Test Method for Rubber Property – Durometer Hardness
[5]
IEEE Standard 269-2002 & 269a-2006, Standard Methods for Measuring Transmission
Performance of Analog and Digital Telephone Sets, Handsets, and Headsets.
[6]
IEEE Standard 1329-1999, Standard Method for Measuring Transmission Performance of
HandsFree Telephone Sets.
[7]
IETF RFC 1890 AVT Profiles.
[8]
IETF RFC 1890 RTP Profile for Audio and Video Conferences with minimal control.
[9]
ISO 3: 1973 Preferred numbers - Series of preferred numbers.
[10]
ITU-T Recommendation G.122 (1993), Influence of national systems on stability and talker
echo in international connections.
[11]
ITU-T Recommendation G.131 (2003), Talker Echo and Its Control.
[12]
ITU-T Recommendation G.711 (1988), Pulse code Modulation (PCM) of voice frequencies.
[13]
ITU-T Recommendation O.41 (1994), Psophometer for use on telephone-type circuits.
[14]
ITU-T Recommendation P.10 (1998), Vocabulary of terms of telephone transmission quality
and telephone sets.
[15]
ITU-T Recommendation P.51 (1996), Artificial mouth.
[16]
ITU-T Recommendation P.57 (2005), Artificial ears.
[17]
ITU-T Recommendation P.58 (1996), Head and torso simulator for telephonometry.
[18]
ITU-T Recommendation P.64 (1999), Determination of sensitivity/frequency characteristics
of local telephone systems.
[19]
ITU-T Recommendation P.79 (1999), Calculation of loudness ratings for telephone sets.
[20]
ITU-T Recommendation P.79, Annex G (2001), Wideband loudness rating algorithm
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
[21]
ITU-T Recommendation P.311 (1998), Transmission characteristics for wideband (150-7000
Hz) digital handset telephones.
[22]
ITU-T Recommendation P.341 (1998), Transmission characteristics for wideband (150-7000
Hz) digital handsfree telephony terminals.
[23]
ITU-T Recommendation P.360 (1998), Efficiency of devices for preventing the occurrence of
excessive acoustic pressure by telephone receivers.
[24]
ITU-T Recommendation P.501 (2000), Test signals for use in telephonometry.
[25]
ITU-T Recommendation P.1010 (2004), Fundamental voice transmission objectives for VoIP
terminals and gateways
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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. 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 voice signal coder and encoder. For the
purposes of this standard L16-256 is the linear 16 bit codec as defined by RFC 1890 at 256 kilobits
per second (16 bits per sample and 16,000 samples per second).
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. Artificial Ear/Mouth vs. Ear/Mouth Simulator
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 and 269a.
4.4. HATS Position
The HATS (head and torso simulator) position (ITU-T P.64 Annex D and Annex E) is the correct
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 degree
downwards rotation to more closely match real talkers. For handsets with omnidirectional
microphones, measurements on the two heads may differ slightly, typically less than 1 dB. 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.
4.5. 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. All tests shall be performed with the receive volume control set to
the nominal volume control setting, unless otherwise specified.
For handsets the RLR is measured with the receiver in the high leak position.
4.6. Reference Volume Control Setting
The reference volume control setting is the quietest volume control setting that complies with the
mandatory low leak RLR requirement in 6.2.3.
4.7. Preferred Ear Simulator
The preferred ear simulator is the Type 3.3. For alternative ear simulators, see relevant sections of
IEEE Std 269.
4.8. Standard Test Position
The Standard Test Position consists of a high leak position and a low leak position.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
The high leak position is defined as the Type 3.3 artificial ear with the receiver contacting the
pinna with a force of 10 N.
The low leak position is defined as the Type 3.3 artificial ear with the receiver contacting the
pinna with a force of 18 N.
4.9. Recommended Test Positions (RTP)
RTP for a handset may be defined (using coordinates as defined in ITU-T P.64, Annex E) by
following these steps:
1. Find the Ear-Cap Reference Point (ECRP) on the handset. Unless otherwise specified by the
manufacturer, the ECRP is the intersection of the external ear-cap reference plane with a
normal axis through the effective acoustic center of the sound outlet ports. Generally, the
acoustic center of the sound outlet ports is at the center of their distribution. For some
handsets, the ear cap reference plane has to be estimated. For example, a tangent to a curved
surface at the effective acoustic center.
2. Line up the handset ECRP at the Ear simulator ERP on the positioning device. The ear cap
reference plane shall be identical to the reference plane of the positioning device. When the
positioning device is set to ERP, then the ear cap reference plane, the reference plane of the
positioning device and the ERP plane are identical.
3. Translation: Move the handset ECRP in ear cap reference plane relative to ERP along the y e
and/or ze axis. If no coordinates are given, leave the ECRP centered on ERP, equivalent to
(0, 0) coordinates. The ye -axis is defined along the length of the handset with positive ye
being in a direction towards the microphone from ECRP. Positive ze axis is in a downward
direction towards the floor.
4. Rotation: Adjust the angle(s) of the handset positioner about the ERP of the coordinate
system.
5. If no coordinates are given, use angles consistent with the HATS position.
6. Application of force: Adjust pressure or distance along the axis of motion of the positioning
device. This axis is defined by a line that passes through the ERP of the left and right ears. It
is parallel to the Ym axis. If no force or position is given, use 6N.
7. RTP can then be defined as the combination of the above translation, rotation and force
specifications. The manufacturer of the device under test is responsible for providing this
data.
4.10.
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.
4.11.
Session Description Protocol (SDP)
Session Description Protocol is a standard way of defining dynamically an RTP media payload
(media format). RFC stands for Request for Comment and is used by the Internet Engineering Task
Force (IETF) to define IP protocol standards.
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4.12.
Reference Codec
A reference codec is used for testing digital telephone terminals with analog test equipment. Error!
Reference source not found. 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 3.17 dB below digital
full scale.
Although power levels are referenced to 600 ohms, the reference codec does not require physical 600
ohm source and termination resistors. 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 Error! Reference source not found. and Error! Reference source
not found., 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.
Figure 1 – Digital Telephone Set Test Arrangement with Reference Codec
Digital Reference
Point
(Junction j)
Send
pM
Mouth Sound Pressure
at MRP
Digital
Set
Decoder
v
Coder
GEN
Interface
pE
Ear Sound Pressure
at ERP
vRCV
Receive
4.13.
vSEND
Reference Codec
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 Error! Reference source not found.. Although this method is
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
preferred, the test methodology usually refers to Error! Reference source not found., the reference
codec method, for the sake of clarity.
Figure 2 – Digital Telephone Set Test Arrangement using Direct Digital Generation and
Analysis
Digital Reference
Point
(Junction j)
Send
Digital
Analysis
pM
Mouth Sound Pressure
at MRP
Digital
Set
Interface
Digital
Generation
pE
Ear Sound Pressure
at ERP
Receive
4.14.
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 corresponds to 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.
4.15.
Electric Power and Noise Levels
The following electric power and noise level units are used in this standard:
dBm0:
4.16.
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 and L16-256 wideband codecs, 0 dBm0 is 3.17 dB below digital full scale.
50TP
The acoustic test point 50 cm from the front center of the speakerphone telephone and 30 cm above
the test table.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
4.17.
Abbreviations and Acronyms
Abbreviations and acronyms, other than in common usage, which appear in this standard, are defined
below.
AGC
CSS
DRP
EFR
ERP
FFT
HATS
ISDN
LRGP
L16-256
MRP
OLR
PBX
PCM
RLR
RFC
RTP
RTP
SDP
SLR
STMR
TCLT
TCLw
VAD
VoIP
4.18.
Automatic Gain Control
Composite Source Signal
Drum Reference Point
Enhanced Full Rate
Ear Reference Point
Fast Fourier Transform
Head and Torso Simulator
Integrated Services Digital Network
Loudness Rating Guard-ring Position
Linear, Sixteen Bit, 256 kbit/s Codec
Mouth Reference Point
Overall Loudness Rating
Private Branch Exchange
Pulse Code Modulation
Receive Loudness Rating
Request for Comment (used by the IETF to define IP Protocol)
Real Time Protocol
Recommended Test Position
Session Description Protocol
Send Loudness Rating
Sidetone Masking Rating
Temporally weighted Terminal Coupling Loss
Weighted Terminal Coupling Loss
Voice Activity Detector
Voice over Internet Protocol
Test Signals
4.18.1. Choice of Test Signal
IEEE Std. 269 & 269a, IEEE Std. 1329 and ITU-T Recommendation P.501 have recommendations
on which test signals are appropriate. In particular, see Annex F, “Test Signals” and Annex G,
“Analysis Methods” of IEEE Std. 269.
The bandwidth of the test signal used shall nominally cover 100 Hz to 8000 Hz unless otherwise
specified.
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. Algorithmic
processes, such as Echo Control, VAD and AGC, may influence the test results or require test signals
other than sinusoidal.
Speakerphone modes of operation almost always employ algorithms that are sensitive to the temporal
and spectral characteristics of the signals. For these devices the use of sinusoidal test signals may not
be appropriate and should be used with caution. Unless explicitly stated otherwise in this document,
for testing to determine TIA-920A compliance tones shall not be used to measure the following.
Send, Receive, and Sidetone Spectral Response, Send, Receive, and Sidetone Loudness, TCLw and
Stability.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
4.18.2. Frequency Tolerance of Test Signals and Analysis
Test signals shall have a frequency within 3% of the specified value. The range shall be within 3% of
the specified test range. This is to accommodate different generation and analysis methods.
Analysis using sine waves may be done at R40 preferred frequencies or at 1/12 th octave band center
frequencies. Analysis using other test signals shall be done in 1/12th octave bands unless otherwise
specified.
R40 analysis shall be done from 100 Hz to 8000 Hz. One-twelfth octave analysis shall be done from
92 Hz to 7286 Hz.
Wherever a frequency range of 100 to 8000 Hz is specified, a range of 92 Hz to 7286 Hz may be
used.
4.19.
Testing mode
Technical requirements apply to the telephone operating in the PCM 256 kbit/s mode. The telephone
must be able to be put into the L16-256 codec mode regardless of which other codecs are supported.
4.20.
Precautions
Coding, decoding, packetization and other signal processing may introduce significant delays that
must be accounted for by the measurement system. For example, when measuring the frequency
response of a device using a stepped sine signal it is possible for the generator and measuring device
to track incorrectly. The measuring device might be measuring the level of delayed frequency, fn,
while the generator has progressed to fn+1. Refer to IEEE Std. 269 & 269a 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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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
5. General Technical Requirements
5.1. Voice Coding Mandatory Requirements
A telephone supporting wideband operation shall support the L16-256 codec. Note: Existing
wideband conferencing systems commonly use the G.722 and G.722.1 codecs. Interoperability with
existing systems requires additional codecs either within the telephone itself or in other system
equipment such as voice gateways.
If the telephone uses additional vocoders, the manufacturer must ensure that their implementation
passes the standard test vectors associated with that codec. For bit exact vocoders it is important to
ensure that vector testing has been performed and found to be compliant with the associated ITU
requirement.
Technical requirements of this standard apply only to linear PCM L16-256 codec at 256 kbit/s.
Unless specified otherwise:


Test methods are given in IEEE Std 269.
When sine wave stimulus is used the frequency tolerance is 3%, and even submultiples of the
sampling frequency (normally 16000 Hz) must not be used.
Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test results or
require test signals other than sine waves. IEEE Standards 269, 269 allows several types of test
signals. The test signal used should be stated. The test signal levels shall be equivalent to the test
signal levels specified in this standard. Test signal levels that differ from those specified in this
standard may also be required. Note that the use of inappropriate test signals may result in erroneous
test results.
Packet voice latency may introduce significant delay that must be accounted for by the test
equipment.
Telephones using nonlinear voice signal processing may require subjective testing.
NOTE: For telephones where 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.
5.1.1. Transmission Format of L16-256 Codec
L16-256 denotes uncompressed 16-bit linear PCM coding of wideband speech sampled at 16 kHz
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 given in
Table 1. The value will be represented in two’s complement notation and transmitted in network byte
order with the most significant byte first.
Table 1 – Codec PCM Codes
Reference
Digital representation
0 dBm0
See sub clause 6.1.4
12
Level
Relative to
0 dBm0
(dB)
0
RMS
Analog
Voltage
(V)
0.775
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Full Scale
 32767
+Full Scale
32767 is x7FFF hex
-Full Scale
-32768 is x8000 hex.
Quiet Code
x0000 hex
Note: All values except the Quiet Code are sinusoidal.
3.17
1.116






5.1.2. Overload Point
Digital full scale is +3.17 dB above the 0 dBm level.
5.1.3. Quiet Code and Full Scale Code
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.
5.1.4. 0 dBm0 (Digital Milliwatt)
The 1 kHz 0 dBm0 Sine wave is represented by the following values:
Sample [ 0]
Sample [ 1]
Sample [ 2]
Sample [ 3]
Sample [ 4]
Sample [ 5]
Sample [ 6]
Sample [ 7]
Sample [ 8]
Sample [ 9]
Sample [10]
Sample [11]
Sample [12]
Sample [13]
Sample [14]
Sample [15]
= 0
= 8705
= 16085
= 21016
= 22748
= 21016
= 16085
= 8705
= 0
= -8705
= -16085
= -21016
= -22748
= -21016
= -16085
= -8705
The values are calculated for a 0 dBm0 level which is 3.17 dB below the digitally encoded peak of
+ 32767. When converting from A-Law, Mu-Law or other encoding formats the precise code
representation may differ somewhat due to processing errors. Care should be taken with conversion
algorithms to minimize distortion.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
6. Handset Technical Requirements
All tests shall be performed with a Type 3.3 ear simulator with the handset in the HATS position.
The Type 3.3 ear simulator shall comply with the specifications given in ITU-T Recommendation
P.57. The Type 3.3 shall have a hardness of 35 ±6 degrees Shore-OO, as measured according to
ASTM 2240. (ITU-T Recommendation P.57-2002 originally specified a hardness of 55 ±10 degrees
Shore-OO for Type 3.3.)
Unless otherwise specified, all tests shall be performed in the Standard Test Position with the
receiver located at the high leak position. A manufacturer may specify Recommended Test Positions
for the high leak and the low leak positions. The RTP may specify the handset position with respect
to ERP, or other aspects of the test position intended to simulate actual use. If the CPE is tested at
the RTP, the RTP shall be documented and used for all handset tests.
All tests shall be preformed with the receive volume control set to the nominal volume control
setting, unless otherwise specified.
6.1. Handset Frequency Response
6.1.1. Handset 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 the equation below:
SMJ = 20 log (VSEND / PM) dB rel 1 V / Pa
Equation [1]
Where
SMJ
PM
VSEND
Send Sensitivity, Mouth to Junction, at Fi.
Sound pressure at the MRP at Fi.
RMS output voltage of the reference codec at Fi.
6.1.1.1.
Measurement Method
Measurements should be done in ISO 1/12th octave bands or R40 intervals or smaller, over a
minimum range of 100 Hz through 8000 Hz using the measurement set-up shown in Figure 3. Direct
digital processing may be employed as explained in clause 4.13. The test signal level shall be 4.7
dBPa at the MRP.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Figure 3 – Handset Send Frequency Response Measurement Method
vSEND
HATS
Decoder
Measuring
Amplifier
Digital
Interface
GEN
Set
Send pM
v
Coder
Mouth Simulator
Quiet Room
Reference Codec
6.1.1.2.
Requirement
The send frequency response shall fall between the upper and lower limits given in Table 2 and
shown in Figure 4. The limit curves shall be determined by straight lines joining successive coordinates given in the table, where frequency response is plotted on a linear dB scale against
frequency on a logarithmic scale. Note: The frequency response mask is a floating or “best fit” mask.
Figure 4 includes the nominal send frequency response characteristic to illustrate the design intent of
the limits.
Table 2 – Co-ordinates of Handset Send Response Limits
Limit Curve
Frequency
(Hz)
Send Response Limit
(dB) [arbitrary level]
Upper Limit
100
140
1000
2000
5000
8000
-1
+3
+3
+8
+8
+3
Lower Limit
200
200
250
1000
3000
5000
6500
6500
- infinity
-6
-3
-3
-1
-1
-6
- infinity
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Figure 4 – Handset Send Frequency Response Mask
10
Arbitrary Level (dB)
5
0
-5
-10
-15
-20
100
1000
Frequency (Hz)
10000
6.1.2. 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, or digital bit stream equivalent, for each frequency or frequency
band (Fi) as shown in the equation below:
SJE
= 20 log (PE / VRCV) dB rel 1 Pa / V
Equation [2]
Where
SJE
PE
VRCV
Receive Sensitivity, Junction to Ear, at Fi.
ERP Sound pressure measured by ear simulator at Fi.
Measurement data are converted from the Drum Reference Point, DRP, to the ERP.
RMS Input voltage to the reference codec, or digital bit stream equivalent at Fi.
6.1.2.1.
Measurement Method
The receive frequency response shall be measured with the receiver at the high leak position. The
receive frequency response is measured using the measurement set-up shown in Figure 5. Direct
digital processing may be employed as explained in clause 4.13. Measurements should be done in
1/12th octave bands, or R40 intervals over a range of 100 Hz through 3350 Hz. Measurements should
be done in 1/3rd octave bands for the R40 frequencies of 4000, 5000, 6300, 8000 Hz. The test signal
level shall be -18.2 dBV (-16 dBm0), or digital bit stream equivalent. The frequency response
measured with the ear simulator must be transformed to the ear reference point (ERP).
Note: 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.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Figure 5 – Handset Receive Frequency Response Measurement Method
Receive p
E Ear
Simulator
Decoder
Digital
Sound Pressure
Measuring
Amplifier
Interface
Set
Coder
GEN
vRCV
Quiet Room
Reference Codec
6.1.2.2.
Requirement
The receive frequency response requirements between 100 Hz and 8000 Hz (referenced to the ERP)
are as follows:
1. With the receiver at the high leak position, the receive frequency response:
Shall fall within the mandatory limits in
a. Table 3 (shown in Figure 6).
Should fall within the desired limits in
b. Table 3 (shown in Figure 6).
The limit curves shall be determined by straight lines joining successive co-ordinates given in the
table, where frequency response is plotted on a linear dB scale against frequency on a logarithmic
scale. The frequency response mask is a floating or “best fit” mask.
See Error! Reference source not found. for the details on the derivation of the nominal frequency
response characteristics.
Table 3 – Co-ordinates of Handset Receive Response Limits
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Limit Curve
Frequency
(Hz)
Desirable Receive Response
Limit (dB) [arbitrary level]
Mandatory Receive Response
Limit (dB) [arbitrary level]
Upper Limit
100
130
700
1000
1400
2000
4000
8000
200
200
800
4000
6500
6500
+1
+4
+1
+4
+4
Lower Limit
+4
+9
+9
+9
+8
- infinity
-10
-4
-4
-10
-infinity
+9
+8
- infinity
-10
-4
-4
-10
-infinity
Figure 6 – Handset Receive Frequency Response Mask
15
Arbitrary Level (dB)
10
5
0
-5
-10
-15
100
1000
Frequency (Hz)
10000
Note: The lower mask above has no relative maximum around 3 kHz. The TIA-920A working
group’s new intent is to consistently remove the 3 kHz bump from the narrow band in the future TIA810C standard.
6.2. Handset Wideband Loudness Ratings and Receive Volume Control
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
18
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
sound to a user. An important characteristic of P.79 loudness ratings is that the louder the telephone,
the more negative the loudness rating.
6.2.1. Handset Wideband Send Loudness Rating (SLR)
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.
6.2.1.1.
Measurement Method
The SLR shall be calculated from the send frequency response measurement (clause 6.1.1) using
equations Error! Reference source not found. and Error! Reference source not found. in Error!
Reference source not found. and frequency bands 1 to 20, Table 14.
6.2.1.2.
Requirement
The terminal shall be designed to have an SLR value of 8 dB, with a tolerance of ±4.0 dB.
6.2.2. Handset Wideband Receive Loudness Rating (RLR)
The RLR is defined in Annex A.
6.2.2.1.
Measurement Method
The RLR shall be calculated from the receive frequency response measurement (clause Error!
Reference source not found.) using equations Error! Reference source not found. and Error!
Reference source not found. in Error! Reference source not found. and frequency bands 1 to 20,
in Table 14.
6.2.2.2.
Requirement
The RLR values measured with the receiver at the high leak position shall have an RLR value of 2
dB, with a tolerance of -4.0/+8.0 dB and should have a nominal RLR value of 2 dB, with a tolerance
of ±4.0 dB.
6.2.3. Handset Receive Volume Control Range
The current regulatory volume control requirements are specified in 47 CFR Part 68.317.
NOTE: The wideband RLR measurements in this document use the HATS (Head and Torso
Simulator) while the current 47 CFR Part 68.317 references narrowband ROLR measurements in
ANSI/EIA/TIA-579-1991, which specifies the Type 1 ear (ITU-T Recommendation P.57).
6.2.3.1.
Measurement Method
The RLR shall be calculated from the receive frequency response measured with the receiver at the
low leak position. The measurement shall be done with the volume control at either the Reference
Volume Control Setting, or the manufacturer’s defined reference volume setting, and the maximum
setting. Use equations Error! Reference source not found. and Error! Reference source not
found. in Error! Reference source not found. and frequency bands 1 to 20, Table 14. Measure the
receive distortion and noise (see clause Error! Reference source not found.) with a 1004 Hz sine
wave at -16 dBm0 input at the maximum volume control setting.
6.2.3.2.
Requirement
The RLR values measured with the receiver at the low leak position shall have a nominal RLR value
of 2 dB, with a tolerance of ±4.0 dB at the Reference Volume Control Setting or the manufacturer’s
defined reference volume setting.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
With the receiver at the low leak position the RLR at the maximum volume control setting shall be at
least 12 dB louder than the RLR at the Reference Volume Control Setting, or the manufacturer
defined reference volume setting (also measured at the low leak position). If the RLR at the
maximum volume control setting is more than 18 dB louder than the RLR at the Reference Volume
Control Setting or the manufacturer’s defined reference volume setting, then the CPE shall
automatically reset to either the Nominal Volume Control Setting, Reference Volume Control
Setting, or the manufacturer’s defined reference volume setting, after ending the call. To ensure that
there is no significant clipping, the receive signal to total distortion and noise ratio at the maximum
volume control setting shall be greater than 20 dB with a 1004 Hz, -16 dBm0 input. (See clause 6.5.2
for test method.)
NOTE: Some special purpose CPE provide high receive gain for hearing impaired users. These CPE
are intended to provide the highest gain for below normal input signal levels, and they might fail the
distortion requirement at the maximum volume control setting when measured with the specified test
signal level.
6.2.4. Magnetic Field for Hearing Aid Coupling
This standard does not contain Magnetic Field for Hearing Aid Coupling requirements. The current
regulatory hearing aid compatibility magnetic output requirements are specified in 47 CFR Part
68.316.
NOTE: Part 68.316 does not provide suitable references for testing Digital Telephones. Suitable test
procedures are currently in TSB-31-C-1, Part 68 Rationale and Measurement Guidelines."
6.2.5. Handset Talker Sidetone (STMR)
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.
It’s desirable for the STMR to be constant over the receive volume control range.
6.2.5.1.
Measurement Method
The test signal level at the MRP shall be -4.7 dBPa. For each frequency given in Table 14, bands 1 to
20, the sound pressure in the artificial ear shall be measured. The frequency response measured with
the ear simulator must be transformed to the ear reference point (ERP). Refer to Annex D. The
STMR shall be calculated using equation Error! Reference source not found. of Error! Reference
source not found..
Telephone sets with adjustable receive levels shall be tested at the minimum, nominal and maximum
settings.
6.2.5.2.
Requirement
The value of STMR shall be within the range of 18 dB ± 6 dB, for any adjustable receive level.
6.2.6. Handset Sidetone Delay
In a digital telephone, sidetone echo occurs when significant delay is introduced into the speech path
between the handset microphone and the handset receiver by the sidetone feedback algorithm.
Ideally, the sidetone signal should be a real-time signal. Sidetone delay less than 5 ms is generally
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
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. Since the sidetone level could be as loud, or
louder than a talker echo signal, sidetone delay greater than 5 ms is undesirable.
6.2.6.1.
Measurement Method
See the method described in IEEE Standards 269, 269.
6.2.6.2.
Requirement
Sidetone delay shall be less than 5 ms. Sidetone delay should be less than 1 ms.
6.3. Handset Noise
6.3.1. Handset Send Noise
6.3.1.1.
General
The send noise of a digital telephone is the 5 second average noise level at the digital transmit output
with the telephone transmitter isolated from sound input and mechanical disturbances.
6.3.1.2.
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.
6.3.1.3.
Requirement
The overall send noise shall be less than -68 dBm0, A-weighted.
6.3.2. Handset Send Single Frequency Interference
6.3.2.1.
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.
6.3.2.2.
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 a selective voltmeter or spectrum analyzer 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.
6.3.2.3.
Requirement
The A-weighted send single frequency interference shall be less than -78 dBm0 in each band.
6.3.3. Handset Receive Noise
6.3.3.1.
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.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Receive noise measurement results must be transformed from the DRP of the ear simulator to the
ERP. If a single wideband measurement is made the transfer function must be realized using a
minimum phase, parametric filter (or equivalent). Refer to IEEE Std. 269.
6.3.3.2.
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 artificial ear over the frequency range of 100 to 8500 Hz. The ambient
noise for this measurement shall not exceed 30 dBA.
6.3.3.3.
Requirement
The receive noise shall be less than 40 dBA.
6.3.4. Handset Receive Single Frequency Interference
6.3.4.1.
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 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.
6.3.4.2.
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 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 8500 Hz. If FFT
analysis is used, then “Flat Top” windowing shall be employed. The ambient noise for this
measurement shall not exceed 30 dBA.
6.3.4.3.
Requirement
The receive single frequency interference shall be 10 dB quieter than the A-weighted receive noise,
and shall be below 30 dBA.
6.4. 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 than the original noise.
6.4.1. 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 terminal receiving either a silence
indication packet or no packets for some non-transient period of time.
6.4.2. Measurement Method
The digital interface is sent a quiet code  the code that represents silence for the coder format.
Disable comfort noise generation and any echo canceller on the terminal. Apply a white noise test
signal to the terminal and adjust the signal amplitude such that the receive noise level measured at
the terminal is 48 dBA. The test signal level is assigned the level of ‘N dB’. Enable comfort noise
generation on the terminal.
The following test sequence must be followed for all test noise levels of ‘M dB’, which will range
from N-10 to N+10 dB.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
1. The digital interface is sent the quiet code for 10 seconds.
2. Apply 300-3400 Hz band-limited white noise of level M dB to the terminal for 60 seconds.
3. Stop sending data to the terminal; this should cause the comfort noise generator to trigger and
apply comfort noise to the terminal receiver. Wait 10 seconds.
4. During the next 10 seconds the acoustic noise level at the receiver is measured.
5. Steps 1-4 are repeated for varying M in 5 dB increments.
6.4.3. Requirement
For a test signal level of M = N, verify that the measured receive noise level is 48 dBA +0.5/-3.0.
For all input noise levels M in the range of N-10 to N+10 the receive noise level measured 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).
Note; Some committee members feel this comfort noise section should be removed, or suggestions
made for “black box” testing. Commercial phones often have no means to disable CNG and AEC.
6.5. Handset Distortion and Noise
The distortion and noise requirements only apply to linear 16 bit PCM at 256 kbit/s.
6.5.1. Handset Send Distortion and Noise
6.5.1.1.
Method of Measurement
The highest signal levels employed for this test may exceed the published specifications for the
mouth simulator. At high test levels, short duty cycles may be required to prevent overheating of the
mouth simulator.
Prior to testing the telephone, the output level and distortion of the mouth simulator should be
verified at the maximum sound pressure level used for each frequency. This need not be verified
before each test. The distortion of the mouth simulator should be at least 10 dB less than the
maximum allowable telephone distortion for each frequency and level. The mouth verification should
be done at the MRP.
Apply a sine wave signal at the MRP, at the levels given in Table 4 at the following frequencies: 160,
315, 502, 803, 1004, 2008 and 3150 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.
6.5.1.2.
Requirement
The ratio of signal-to-total distortion and noise (SDN) of the digitally encoded signal output shall be
above the limits given in Table 4, with A-weighting applied to the measured distortion and noise
output. Limits for intermediate levels are found by drawing straight lines between the successive
coordinates in the table on a linear (dB signal level) – linear (dB ratio) scale.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 4 – Handset Send Signal-to-Total Distortion and Noise Ratio Limits
-30
Send Ratio
160 to 803 Hz
(dB)
26
Send Ratio
1004 Hz
(dB)
26
Send Ratio
2008 to 3150 Hz
(dB)
NA
-25
26
26
26
-20
31
31
31
-10
33
33
33
0
33
33
33
+5
33
33
33
+10
26
26
26
Send Level at MRP
(dB Pa)
+15
NA
20
NA
Note: 20 dB = 10%, 26 = 5%, 31 = 2.8%, 33 = 2.2%. NA = Not Applicable.
Note; Fewer tests are warranted. If one column is emphasized, 400 Hz more appropriate because this
is more prominent during speech. Move lower frequency boundary to 203 Hz (unless lower Send
Mask frequency response revised to go down to 160 Hz).
6.5.2. Handset Receive Distortion and Noise
6.5.2.1.
Method of Measurement
Apply a digitally simulated sine wave, with the signal levels given in Table 5 and the following
frequencies: 160, 315, 502, 1004, 2008 and 3150 Hz. The ratio of the signal-to-total distortion and
noise power is measured with the artificial ear. The test frequency tolerance is 3%, and even
submultiples of the sampling frequency must not be used.
6.5.2.2.
Requirement
The ratio of signal-to-total distortion and noise (SDN) measured with the artificial ear shall be above
the limits given in Table 5, with A-weighting applied to the measured distortion and noise output,
unless the signal in the artificial ear exceeds +10 dBPa or is less than -50 dBPa.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 5 – Handset Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the
digital interface
(dBm0)
-40
Receive Ratio
@ 160 Hz
(dB)
20
Receive Ratio
@ 315 Hz
(dB)
24
Receive Ratio
@ 502 to 3150 Hz
(dB)
24
-34
24
24
24
-27
28
30
30
-20
28
32
32
-10
28
32
32
-6
28
32
32
-3
28
28
28
0
24
24
28
Note: 20 dB = 10%, 24= 6.3%, 28 = 4%, 30=3.2%, 32 = 2.5%.
Note; Fewer tests are warranted. If one column is emphasized, 400 Hz more appropriate because this
is more prominent during speech. Move lower frequency boundary to 203 Hz (unless lower Receive
Mask frequency response revised to go down to 160 Hz). Tests that vary level only and frequency
only might be listed separately.
6.6. 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 cancellers with non-linear processing may be used on handset
connections to provide sufficient echo return loss to mitigate increased echo 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 nonlinear 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 be used consistently across products in a black-box
testing setup. The ‘proper’ measurement of TCLw is addressed in IEEE Std. 269, Annex O.
6.6.1. Measurement Method
TCLw shall be measured with the handset receiver at the high leak Standard Test Position on the
HATS. The TCLw measurement shall be made at an input signal level of -16 and -10 dBm0. The test
should be performed in a quiet environment (the ambient noise level shall be less than 30 dBA.)
The TCLw measurement shall not be performed using 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. The test signal shall be band-limited to 100 through 8000 Hz. The calibration shall be
determined during the ON portions of the signal. 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.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
The attenuation from digital input (Receive) to digital output (Send) is measured in 1/12 th octave
bands, using the measurement arrangement shown in Figure 7. See Error! Reference source not
found..
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 rather than 300 to 3400.
Figure 7 – Terminal Coupling Loss Measurement Method
vSEND (Echo Return)
Ear
Simulator
Decoder
v
Coder
GEN
Digital
Set
Interface
vRCV
Anechoic Chamber
Reference Codec
6.6.2. Requirements
The normalized value of TCLw at the high leak position 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 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 = TCLw measured + (8 - SLR) dB + (2 - RLR) dB = 48 dB
+ (8 - 11) dB + (2 - 0) dB = 47 dB.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
NOTES
1. The requirement of 52 dB for IP sets is a function of the -16 dBm0 test signal level and
the -68 dBm0A send noise requirement. Measuring TCLw > 52 dB can be difficult.
2. 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.
3. 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.
4. 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.
Figure 8 – Reference Corner
25 cm
50 cm
6.7. 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 frequency.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
6.7.1. 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 and represented by
the L16-256 codec. The recommended pattern is 250 msec of noise (on), 150 msec 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 handset and transmission circuit fully active, measure the
attenuation from the digital input to the digital output using Method 1 and Method 2.
6.7.1.1.
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.
6.7.1.2.
Method 2
Place the handset with the earcap and mouthpiece facing a hard, smooth surface free of any other
object for 50 cm. The telephone set shall be fully active.
6.7.2. Requirement
Stability loss using both method 1 and method 2, (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.
6.8. Long Duration Maximum Acoustic Pressure (Steady State Input)
6.8.1. General
The long duration maximum acoustic pressure is the steady state (longer than 500 ms) sound pressure
disturbance 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.
6.8.2. Measurement Method
The steady-state A-weighted sound pressure level shall be measured using the digital terminals test
procedure in IEEE Standards 269, 269.
6.8.3. Requirements
The measured maximum rms level shall be less than 125 dB(A) at ERP, required in UL/CSA 609502003.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
6.9. Short Duration Maximum Acoustic Pressure (Peak)
6.9.1. General
The short duration maximum acoustic pressure is the sound pressure impulse (less than 500 ms)
emitted from a handset receiver.
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
auxiliary leads of the digital telephone.
6.9.2. Measurement Method
The peak acoustic pressure level shall be measured using the digital terminals test procedure in IEEE
Standards 269, 269.
6.9.3. Requirements
The maximum peak acoustic pressure shall be less than 136 dBSPL at ERP, as required in UL/CSA
60950-2003.
6.10.
VoIP Telephone Delay
Delay is a complex end-to-end issue. Certain aspects of delay can be optimized in VoIP telephones,
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 farend 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 in VoIP telephones. Therefore, this standard now
specifies delay in terms of categories for network planning purposes, similar to ITU-T Rec. P.1010.
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.
6.10.1. Requirement
The wired terminals shall be configurable so that requirements of at least Category B are met.
Wireless terminals should be configurable so that the requirements of at least Category C are met.
6.10.2. Handset Send Delay
6.10.2.1.
General
The send delay is defined here as the time from when an acoustic signal leaves an artificial mouth
playing into a VoIP telephone’s handset to the time its digitized, packetized representation arrives at
that telephone’s packet network interface.
6.10.2.2.
Measurement Method
A digital audio measuring device capable of measuring the delay between an injected signal (to the
mouth simulator) 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 L16-256 packets with a speech frame
rate of 20 ms and with one speech frame per packet.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
An acoustic signal of -4.7 dBPa shall 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 packet network interface shall
be measured. The send delay shall be used to determine the corresponding category.




Category A: Ts ≤ 25 ms
Category B: Ts ≤ 35 ms
Category C: Ts ≤ 50 ms
Category D: Ts > 50 ms
6.10.3. Handset Receive Delay
6.10.3.1.
General
The receive delay is defined here as the time from when a digitized, packetized representation of a
signal arrives at that VoIP telephone’s packet network interface to the time its analog reproduction is
received at an artificial ear sealed to that telephone’s handset.
The handset receive delay requirements include the depacketization, hardware/firmware processing
and de-jitter delays, plus any delay associated with the radio link for wireless products.
6.10.3.2.
Measurement Method
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 packet 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 L16-256 packets with a speech frame rate of 20
ms and with one speech frame per packet.
A pulsed digital signal of -16 dBm0 shall 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 shall be measured. The receive delay shall be used to determine the
corresponding category.




Category A: Tr ≤ 30 ms
Category B: Tr ≤ 65 ms
Category C: Tr ≤ 100 ms
Category D: Tr > 100 ms
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7. 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 with a Type 3.3 ear simulator. The Type 3.3 ear simulator shall comply
with the specifications given in ITU-T Recommendation P.57. The Type 3.3 shall have a hardness of
35 ±6 degrees Shore-OO, as measured according to ASTM 2240. All tests involving the headset
receiver shall be done with the same ear and mouth simulator. All test reports shall document the
model of ear and mouth simulator used.
All tests shall be preformed with the receive volume control set to the nominal volume control
setting, unless otherwise specified.
The headset test method is given in IEEE Std. 269 and 269a.
7.1. Headset Frequency Response
7.1.1. 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 the equation below:
SMJ = 20 log (VSEND / PM) dB rel 1 V / Pa
[3]
Where
SMJ
PM
VSEND
Send Sensitivity, Mouth to Junction, at Fi.
Sound pressure at the MRP at Fi.
RMS output voltage of the reference codec, or digital bit stream equivalent at Fi.
7.1.1.1.
Measurement Method
The test setup is shown in Figure 3, clause 6.1.1.1, except the handset is replaced by the headset.
Measurements should be done in ISO 1/12th octave bands or R40 intervals or smaller, over a
minimum range of 100 Hz through 8000 Hz using the measurement set-up shown in Figure 3 except
the handset is replaced by the headset. Direct digital processing may be employed as explained in
clause 4.13. The test signal level shall be 4.7 dBPa at the MRP.
.
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 and 269a.
7.1.1.2.
Requirement
The headset send frequency response shall fall between the upper limit and the lower limit given in
Error! Reference source not found. and shown in Error! Reference source not found.. 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 dB scale against frequency on a logarithmic scale. Note that
the frequency response mask is a floating or “best fit” mask.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 6 – Co-ordinates of Headset Send Response Limits
Limit Curve
Frequency
(Hz)
Send Response Limit
(dB) [arbitrary level]
upper limit
100
120
1000
2000
6000
8000
-1
+4
+4
+9
+9
+8
lower limit
200
200
250
1000
3000
6500
6500
- infinity
-6
-3
-3
-1
-10
- infinity
Figure 9 – Headset Send Frequency Response Mask
15
Arbitrary Level (dB)
10
5
0
-5
-10
-15
100
1000
Frequency (Hz)
32
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
(Editor; Accepted for now. However, in future handset and headset Tx responses should be
harmonized. Needs further investigation. Variability of headset boom mic position is a factor
discussed, and reduced noise for boom mics, and some claim less signal in speech justifies high
frequency roll off to be more aggressive than previous draft. Also SNR improves if no high
frequency component of speech exists, some claim, due to aggressive high frequency rolloff.)
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7.1.2. Headset Receive Frequency Response
The receive frequency response is the overall response of the codec receive filter, receive amplifier
and transducer. The receive 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 the equation below:
SJE
= 20 log (PE / VRCV) dB rel 1 Pa / V
[4]
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.
7.1.2.1.
Measurement Method
The receive frequency response is measured according to IEEE Std. 269a using the HATS position
and using the measurement set-up shown in Figure 5 of the handset section except the handset is
replaced by the headset. Direct digital processing may be employed as explained in clause 4.13.
Measurements should be done in 1/12th octave bands, or R40 intervals over a range of 100 Hz
through 8000 Hz. Measurements should be done in 1/3rd octave bands for the R40 frequencies of
4000, 5000, 6300, 8000 Hz. The test signal level shall be -18.2 dBV (-16 dBm0), or digital bit stream
equivalent. The frequency response measured with the ear simulator must be transformed to the ERP,
refer to Annex D. The measurement is made at the reference volume control setting.
Note: 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.
7.1.2.2.
Requirement
The receive frequency response shall be within the upper limit and lower limits given in
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 7 and shown in Figure 10. 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 dB scale
against frequency on a logarithmic scale. Note: The frequency response mask is a floating or “best
fit” mask.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 7 – Co-ordinates of Headset Receive Response Limits
Limit Curve
Frequency
(Hz)
Receive Response Limit
(dB) [arbitrary level]
Upper Limit
100
130
1000
2000
4000
8000
1
4
4
9
9
8
Lower Limit
200
200
800
4000
6500
6500
-infinity
-10
-4
-4
-10
-infinity
Figure 10 – Headset Receive Frequency Response Mask
15
Arbitrary Level (dB)
10
5
0
-5
-10
-15
100
1000
Frequency (Hz)
36
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7.2. Headset Wideband Loudness Ratings
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.
7.2.1. Headset Wideband Send Loudness Rating (SLR)
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.
7.2.1.1.
Measurement Method
The SLR shall be calculated using the 1/3rd octave sensitivity data collected from the send frequency
response measurement referred to in 7.1.1. Use equation [A1] of Annex A and bands 4 to 17 of
Table 14.
Note; In Ottawa it was resolved to use NB LR computation for all 920A LR requirements, but
corresponding new LR targets were not yet resolved, and the reference to equation [A1] of Annex A
has not yet been modified consistently. Regarding targets, options were keeping current LR targets,
making SLR 2 dB quieter, making SLR & RLR 2 dB quieter. This global change will obsolete the
above reference to bands 1 through 20 here and elsewhere, since bands 1 to 3 (<200 Hz) and bands
18 to 20 (>4000 Hz) are not included in the NB LR computation.
7.2.1.2.
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 because the mic boom position my vary
relative to the MRP.
7.2.2. Headset Wideband Receive Loudness Rating (RLR)
The RLR is the loudness loss in the receiving direction from the digital reference point to the ear
reference point. Refer to Annex A and ITU-T Recommendation P.79.
7.2.2.1.
Measurement Method
The RLR shall be calculated using the 1/3rd octave sensitivity data collected from the receive
frequency response measurement referred to in 7.1.2. Use equation [A2] of Annex A and bands 4 to
17 of Table 14.
The reference volume control setting shall be used.
7.2.2.2.
Requirement
The monaural terminal shall have an RLR value of 0 dB, with a tolerance of -4.0/+8.0 dB. The
binaural terminal should have an RLR value of 6 dB, with a tolerance of -4.0/+8.0 dB, for each of the
left and right receivers measured separately.
Editor’s note; The old 920 RLR requirement was 2 dB +/-4 dB, but 810-B is 0 dB +/-4 dB. The
proposed “shift” to a 0 dB target to harmonize with 810-B seems a bit artificial, given that a 0 dB
+4/-8 dB requirement is essentially identical to a 2 dB +/-6 dB requirement.
NOTE: Either the terminal or the headset should have a receive volume control that is capable of
amplification and attenuation.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7.2.3. Headset Talker Sidetone
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.
It’s desirable for the STMR to be constant over the receive volume control range.
7.2.3.1.
Measurement Method
The test signal level at the MRP shall be -4.7 dBPa. For each frequency given in Table 14, 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 D. 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
volume control settings.
7.2.3.2.
Requirement
For any adjustable receive level, the value of STMR shall be within the range of 21 dB ±6 dB for
supra-aural, 18 dB ±6 dB for insert, 20 dB ±6 for intra-conch, e.g. earbud.. The value of STMR for
binaural terminals should be 6 dB quieter, for each of the receivers measured separately.
NOTE - In practice, sidetone measurements in the high leak position are limited to a value of
approximately 24 dB by the influence of the test setup (HATS).
7.2.4. Headset Sidetone Delay
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. Since the sidetone level could be as loud, or
louder than a talker echo signal, sidetone delay greater than 5 ms is undesirable.
7.2.4.1.
Measurement Method
See the method described in IEEE Standards 269, 269 and 269a.
7.2.4.2.
Requirement
Sidetone delay shall be less than 5 ms. Sidetone delay should be less than 1 ms.
7.3. Headset Noise
7.3.1. Headset Send Noise
7.3.1.1.
General
The send noise of a digital telephone is the 5 second average background noise level at the digital
transmit output with the telephone headset transmitter isolated from sound input and mechanical
disturbances.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
7.3.1.2.
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.
7.3.1.3.
Requirement
The overall send noise shall be less than or equal to -64 dBm0, A-weighted. (This agrees with the
810B headset requirement, and is 4 dB louder than the 810B handset requirement, ignoring the
discrepancy in spectral weighting. Typically the headset mic boom may not be as close to the MRP
as the handset.)
7.3.2. Headset Send Single Frequency Interference
7.3.2.1.
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.
7.3.2.2.
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 a selective voltmeter or spectrum analyzer having 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.
7.3.2.3.
Requirement
The A-weighted send single frequency interference shall be no greater than -74 dBm0.
7.3.3.
Headset Receive Noise
7.3.3.1.
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 the
ERP. If a single wideband measurement is made, the transfer function must be realized using a
minimum phase, parametric filter (or equivalent). Refer to IEEE Std. 269a.
7.3.3.2.
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.
7.3.3.3.
Requirement
The receive noise shall be less than 40 dBA for a monaural headset. The receive noise for binaural
headsets should be less than 34 dBA, for each of the receivers measured separately.
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7.3.4. Headset Receive Single Frequency Interference
7.3.4.1.
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 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.
7.3.4.2.
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 then 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.
7.3.4.3.
Requirement
The receive A-weighted single frequency interference shall be 10 dB quieter than the overall Aweighted receive noise and shall be below 30 dBA.
Editors Note; A very quite phone could in principle have , for example, 10 dBA overall noise but 1
dBA noise within 180 +/-15.5 Hz and thus fail this receive noise requirement. The committee felt
that room and measurement mic noise are present at higher levels both overall and from 100 to 8500
Hz, so this type of theoretical unmerited failure does not occur in practice..
7.4. Headset Distortion and Noise
The distortion and noise requirements only apply to linear 16 bit PCM at 256 kbit/s.
7.4.1. Headset Send Distortion and Noise
7.4.1.1.
Method of Measurement
The highest signal levels employed for this test may exceed the published specifications for the
mouth simulator. At high test levels, short duty cycles may be required to prevent overheating of the
mouth simulator.
Prior to testing the telephone, the output level and distortion of the mouth simulator should be
verified at the maximum sound pressure level used for each frequency. This need not be verified
before each test. The distortion of the mouth simulator should be at least 10 dB less than the
maximum allowable telephone distortion for each frequency and level used for test. The mouth
verification should be done at the MRP.
Apply a sine wave signal at the MRP, at the levels given in Table 8 and at the following frequencies:
160, 315, 502,803, 1004, 2008 and 3150 Hz. The ratio of the signal-to-total-distortion and noise
power of the digital 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.
7.4.1.2.
Requirement
The ratio of the signal-to-total distortion and noise (SDN) of the digitally encoded signal output shall
be above the limits given in Table 8, when the A-weighting is applied to the measured distortion and
noise output. Limits for intermediate levels are found by drawing straight lines between the
successive coordinates in the table on a linear (dB signal level) – linear (dB ratio) scale.
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Table 8 – Headset Send Signal-to-Total Distortion and Noise Ratio Limits
-30
Send Ratio
160 to 803 Hz
(dB)
26
Send Ratio
1004 Hz
(dB)
26
Send Ratio
2008 to 3150 Hz
(dB)
NA
-25
26
26
26
-20
31
31
31
-10
33
33
33
0
33
33
33
+5
33
33
33
+10
26
26
26
Send Level at MRP
(dB Pa)
+15
NA
20
NA
Note: 20 dB = 10%, 26 = 5%, 31 = 2.8%, 33 = 2.2%. NA = Not Applicable.
Note: Most sound source equipment will generate significant distortion for acoustic signals above 5
dBPa. Editor’s Note: Note 160 Hz is below the lower mask. Work is in progress on a major revision
of the SDN measurement procedure, using gated 1/3 octave band limited white noise stimulus.
7.4.2. Headset Receive Distortion and Noise
7.4.2.1.
Method of Measurement
Apply a digitally simulated sine wave, with the signal levels given in Table 9 and the following
frequencies: 160, 315, 502, 1004, 2008 and 3150 Hz. The ratio of the signal-to-total distortion and
noise power is measured in the ear simulator. The test frequency tolerance is 3%, and even
submultiples of the sampling frequency must not be used.
7.4.2.2.
Requirement
The ratio of the signal-to-total distortion and noise (SDN) measured in the ear simulator, shall be
above the limits given in Table 9, with A-weighting applied to measured distortion and noise output,
unless the signal in the ear simulator exceeds +10 dBPa or is less than -50 dBPa.
Table 9 – Headset Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the
digital interface
(dBm0)
-40
Receive Ratio
@ 160 Hz
(dB)
20
Receive Ratio
@ 315 Hz
(dB)
24
Receive Ratio
@ 502 to 3150 Hz
(dB)
24
-34
24
24
24
-27
28
30
30
-20
28
32
32
-10
28
32
32
-6
28
32
32
-3
28
28
28
0
24
24
Note: 20 dB = 10%, 24 = 6.3%, 28 = 4%, 30 = 3.2%, 32 = 2.5%.
41
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Editor’s Note: We must remove 160 Hz during the future major rework of this requirement, because the Rx
lower mask only goes to 200 Hz, and high pass filtering is encouraged below this frequency.
7.5. 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
world 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
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 be used consistently across products in a black-box
testing setup. The ‘proper’ measurement of echo is addressed in IEEE std. 269, Annex O.
7.5.1. 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.
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 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.
The attenuation from digital input (receive) to digital output (send) is measured at 1/12th octave
bands, using the measurement arrangement shown in Figure 11.
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.
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Figure 11 – Terminal Coupling Loss Measurement Method
7.5.2. 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 10 dB and RLR normalized
to 0 dB. Note that the normalized value of TCLw for IP sets to be greater than 50 dB to meet ITU-T
recommendation G.131 talker echo objective requirements.
Editor’s Note: This needs further investigation. G.131 references P.310 (45 dB) and P.311 (35 dB)
and ETSI (55 dB?). Perhaps this sentence should be removed for handset and headset entirely
because the 50 dB reference cannot be found.
For 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 = 48 dB + (10 - 11) dB + (0 – (-2)) dB = 49 dB.
See clause 7.6 for additional notes regarding TCLw.
NOTES
1. The requirement of 52 dB for IP sets is a function of the -16 dBm0 test signal level and
the -68 dBm0A send noise requirement. Measuring TCLw > 52 dB can be difficult.
2. 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.
3. 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
connections may provide satisfactory performance while the same may not be true for
connections that have a long delay.
4. Temporally weighted terminal coupling loss (TCLt) is an alternate method for echo
measurement, which may be more subjectively relevant, especially in devices with echo
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
suppression or cancellation features. (See IEEE Std 1329.) The performance requirements
may need to be changed when using this method.
7.6. 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.
7.6.1. 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 and represented by
the L16-256 codec. The recommended pattern is 250 msec of noise (on), 150 msec 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.
7.6.1.1.
Method 1
Place the headset in the reference corner, as shown in Figure 12. 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 250
mm 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.
7.6.1.2.
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.
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Figure 12 – Reference Corner
25 cm
50 cm
Editor’s note: needs to change Figure 12 to show headset.
7.6.2. Requirement
Stability loss using both method 1 and method 2, (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.
7.7. Headset Long Duration Maximum Acoustic Pressure (Steady State)
7.7.1. General
The long duration maximum acoustic pressure is the steady state (longer than 500 ms) sound pressure
disturbance 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, headset or
auxiliary leads of the digital telephone.
7.7.2. Measurement Method
The steady-state A-weighted sound pressure level shall be measured using the digital terminals test
procedure in IEEE 269.
7.7.3. Requirements
The measured maximum rms level shall be less than 118 dB(A) at ERP, required in UL/CSA 609501, 2007.
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7.8. Short Duration Maximum Acoustic Pressure (Peak)
7.8.1. General
The short duration maximum acoustic pressure is the sound pressure impulse (less than 500 ms)
emitted from a headset receiver.
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, headset or
auxiliary leads of the digital telephone.
7.8.2. Measurement Method
The peak acoustic pressure level shall be measured using the digital terminals test procedure in IEEE
269..
7.8.3. Requirements
The maximum peak acoustic pressure shall be less than 136 dBSPL, as required in UL/CSA 60950-1,
2007.
Editor’s Note: Some committee members felt UL should be referenced and no safety requirements
should be in this document. Al Baum will check out the TIA legal policy regarding liability on this
issue.
Editor’s note: The Clause 6.20 of VoIP about telephone delay does not belong to handset. It should
be a standalone clause including both handset and headset. In general handset and headset do not
affect the overall VoIP telephone delay very much, and handsfree may have only a few msecs more
delay for noise filtering, etc.)
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8. Speakerphone Technical Requirements (Advisory)
All speakerphone requirements are advisory. Speakerphone test methods are given in IEEE Std.
1329.
Speakerphone modes of operation almost always employ algorithms that are sensitive to the temporal
and spectral characteristics of the signals. For these devices the use of sinusoidal test signals is not
recommended.
Speakerphone 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 speakerphone telephone and the mouth simulator and
microphone. Measurements performed at other distances or positions shall be noted, and in the
absence of an RTP, the 50 cm test position as defined in IEEE Std. 1329 is recommended.
8.1. Speakerphone Frequency Response
8.1.1. Speakerphone 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.
8.1.1.1.
Measurement Method
The send frequency response is measured directly in 1/3rd octave bands or converted to 1/3rd octave
bands according to IEEE Std. 1329, clause 9.3.8. The measurement is made over a minimum range of
100 Hz through 8000 Hz. For desktop speakerphones, use the measurement set-up shown in Figure
13. For other speakerphone devices, use the RTP. The test signal level shall be -4.7 dBPa at the
MRP.
Figure 13 – Speakerphone Send Frequency Response Measurement Method
vSEND
Mouth Simulator
Send
GEN
Decoder
pM
Interface
50 cm
30 cm
Coder
40 cm
Anechoic Chamber
Digital
Set
(On Table)
Reference Codec
47
v
Measuring
Amplifier
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
8.1.1.2.
Requirement
The speakerphone send response should be within the limits given in Table 10 and shown in Figure
14. The limits break at the crossover frequencies of the ISO R10 series of preferred frequencies.
The crossover frequencies may be adjusted up to 3% to accommodate non-R10 1/3rd octave
measurement data. The frequency response mask is a floating or “best fit” mask.
Table 10 – Co-ordinates of Speakerphone Send Response Limits
Limit Curve
Frequency
Bands
Send Response Limit
(dB) [arbitrary level]
Upper Limit
100 to 1120
1120 to 1780
1780 to 2820
2820 to 4470
4470 to 7080
7080 to 8910
0
+1
+2
+3
+4
-5
Lower Limit
224 to 282
282 to 355
355 to 4470
4470 to 5620
-12
-10
-8
-12
Figure 14 – Speakerphone Send Frequency Response Mask
10
Arbitrary Level (dB)
5
0
-5
-10
-15
-20
100
1000
Frequency (Hz)
48
10000
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8.1.2. Speakerphone 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.
8.1.2.1.
Measurement Method
The receive frequency response is measured directly in 1/3rd octave bands or converted to 1/3rd
octave bands according to IEEE Std. 1329 clause 9.3.8. The measurement is made over a minimum
range of 100 Hz through 8000 Hz. For desktop speakerphones, use the measurement set-up shown in
Figure 15. For other speakerphone devices, use the RTP. The test signal level shall be -25 dBV.
The speakerphone reference volume control setting shall be used. A free field microphone is used
for the measurement.
Figure 15 – Speakerphone Receive Frequency Response Measurement Method
To Sound Pressure
Measuring Amplifier
Decoder
Free Field Microphone
Receive pE
Interface
50 cm
30 cm
40 cm
Anechoic Chamber
GEN
Coder
Digital
Set
(On Table)
vRCV
Reference Codec
8.1.2.2.
Requirement
The speakerphone receive frequency response should be below the 1/3rd octave band upper limit and
above the 1/3rd octave band lower limit given in Table 11 and shown in Figure 16. The limits break at
the crossover frequencies of the ISO R10 series of preferred frequencies. The crossover frequencies
may be adjusted up to 3% to accommodate non-R10 1/3rd octave measurement data.
Table 11 – Co-ordinates of Speakerphone Receive Response Limit Curves
Limit Curve
Frequency
Receive Response Limit
(dB) [arbitrary level]
upper limit
100 to 7080
7080 to 8910
0
-5
lower limit
224 to 282
282 to 4470
4470 to 5620
-14
-12
-14
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SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Figure 16 – Speakerphone Receive Frequency Response Mask
10
Arbitrary Level (dB)
5
0
-5
-10
-15
-20
100
1000
Frequency (Hz)
10000
8.2. Speakerphone Wideband Loudness Ratings and Receive Volume Control
Correlation factors relating speakerphone 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 described in sub clause 8.2.2.2. These correlation
factors may not be appropriate for other speakerphone applications such as conference, hand-held, incar applications or any applications where the relationship between the talker and the speakerphone
varies from the 50 cm position, or the reverberation characteristics or the background noise levels
vary from typical office environments.
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.
8.2.1. Speakerphone Wideband Send Loudness Rating (SLR)
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.
8.2.1.1.
Measurement Method
The SLR shall be calculated using the 1/3rd octave sensitivity data collected from the send frequency
response measurement. Use equation [A4] of Annex A and bands 1 to 20, Table 14.
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8.2.1.2.
Requirement
The terminal should be designed to have a speakerphone SLR value of 13 dB, with a tolerance of
±4.0 dB.
8.2.2. Speakerphone Wideband Receive Loudness Rating (RLR)
8.2.2.1.
Measurement Method
The RLR shall be calculated from the 1/3rd octave sensitivity data collected from the receive
frequency response measurement. Use equation [A5] of Annex A and bands 1 to 20, Table 14.
8.2.2.2.
Requirement
At the reference volume control setting the terminal should be designed to have a speakerphone RLR
of 2 dB, with a tolerance of ±4.0 dB.
Note that in most standards, such as IEEE Std. 1329 and ITU-T Recommendation P.340, there is a
correction factor of 14 dB subtracted from calculated RLR (RLR = RLR -14). In order to be
consistent with TIA-810-B, this standard does not use the correction factor. In order to compare
RLR as measured by other standards the correction factor may be used. For example, an RLR of 16
dB measured per this standard would be: 16 dB -14 dB = +2 dB. If a correction factor is used, it
must be stated with the data.
8.2.3. Speakerphone Receive Volume Control
The speakerphone receive volume control shall provide greater than or equal to 8 dB of gain relative
to the reference volume control setting. The volume control should provide at least 16 dB of
attenuation relative to the reference volume control setting. The volume control step size shall be less
than 6 dB.
8.3. Speakerphone Noise
8.3.1. Speakerphone Send Noise
8.3.1.1.
General
The send noise of a digital telephone is the 5 second average noise level at the digital transmit output
with the microphone isolated from sound input and mechanical disturbances.
8.3.1.2.
Measurement Method
Speakerphone send noise is measured according to IEEE Std. 1329, clause 9.3.4. With the
speakerphone telephone in a quiet environment (ambient noise less than 30 dBA), the A-weighted
noise level at the digital output is measured.
8.3.1.3.
Requirement
The speakerphone send noise should be no greater than -63 dBm0, A-weighted.
8.3.2. Speakerphone Send Single Frequency Interference
8.3.2.1.
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
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.
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8.3.2.2.
Measurement Method
Speakerphone send noise is measured according to IEEE Std. 1329, clause 9.3.4. In a quiet
environment (ambient noise less than 30 dBA), measure the noise level at VSEND with a selective
voltmeter or spectrum analyzer having 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.
8.3.2.3.
Requirement
The speakerphone send single frequency interference should be less than -70 dBm0.
8.3.3. Speakerphone Receive Noise
8.3.3.1.
General
The receive noise of a digital speakerphone telephone is the 5 second average noise level measured at
the speaker output with the digital telephone receiving the digital quiet code.
8.3.3.2.
Measurement Method
The speakerphone receive A-weighted noise level is measured according to IEEE Std. 1329 clause
9.4.4. A signal corresponding to the quiet code is applied at the digital interface. The ambient noise
for this measurement shall not exceed 30 dBA.
8.3.3.3.
Requirement
The speakerphone receive noise should be less than 40 dBA at the maximum volume control setting
and less than 35 dBA at the reference volume control setting, with the comfort noise turned off.
8.3.4. Speakerphone Receive Single Frequency Interference
8.3.4.1.
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 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.
8.3.4.2.
Measurement Method
Speakerphone receive noise is measured according to IEEE Std. 1329, clause 9.4.4. 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 having an effective bandwidth of not
more then 31 Hz, over the frequency range of 100 to 8000 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 reference volume control setting shall be used.
8.3.4.3.
Requirement
The receive A-weighted single frequency interference shall be 10 dB quieter than the overall Aweighted receive noise. In no case shall the single frequency interference be greater than 28 dBA in
each measured band.
8.4. Speakerphone Distortion and Noise
The distortion and noise requirements only apply to linear PCM at 256 kbits/s.
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8.4.1. Speakerphone Send Distortion and Noise
8.4.1.1.
Method of Measurement
Speakerphone send distortion and noise is measured according to IEEE Std. 1329, clause 9.3.6.
Apply a test signal at the MRP, at the levels given in Table 12 and at the following frequencies: 315,
502, 803, 1004, 2008 and 3150 Hz. The ratio of the signal power –to the total A-weighted distortion
and noise power of the digitally encoded signal output is measured.
Table 12 – Speakerphone Send Signal-to-Total Distortion and Noise Ratio Limits
Send level at the MRP
(dBPa)
Send Ratio (dB)
@ 315 Hz
Send Ratio (dB)
@ 502 to 3150 Hz
-10
-5
0
+5
+10
26
30
30
30
30
26
30
30
30
30
Note: 26 dB = 5%, 30 dB = 3.2%
8.4.1.2.
Requirement
The ratio of the signal power to the total distortion and noise power of the digitally encoded signal
output should be above the limits given in Table 12.
Limits for intermediate levels are found by drawing straight lines between the successive coordinates
in the table on a linear (dB signal level) – linear (dB ratio) scale.
8.4.2. Speakerphone Receive Distortion and Noise
8.4.2.1.
Method of Measurement
Speakerphone receive distortion and noise is measured according to IEEE Std. 1329, 9.4.6. Apply a
test signal, at the levels given in Table 13 and at the following frequencies: 315, 502, 1004, 2008 and
3150 Hz. The ratio of the signal power to the total A-weighted distortion and noise power is
measured.
The reference volume control setting shall be used.
8.4.2.2.
Requirement
The ratio of the signal power to the total A-weighted distortion and noise power 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.
53
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 13 – Speakerphone Receive Signal-to-Total Distortion and Noise Ratio Limits
Receive level at the
digital interface
(dBm0)
Receive Ratio (dB)
@ 315 & 502 Hz
Receive Ratio (dB)
@ 803 to 2008 Hz
-30
-20
-10
-6
-3
28
28
28
28
24
28
30
30
30
24
Note: 24 dB = 6.3%, 28 dB = 4%, 30 dB =3.2%
54
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
8.5. 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 speakerphone
connections to provide sufficient echo return loss to mitigate increased echo associated with longer
network delays.
The use of echo control devices on the speakerphone 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. The ‘proper’ measurement of TCLw is
addressed in IEEE Std. 269, Annex O.
Editors note; check against handset
8.5.1. Measurement Method
Refer to IEEE Std. 1329, clause 11, for the test method. The test range is 300 to 6700 Hz. The TCLw
measurement shall not be performed using a sinusoidal test signal for the receive path input.
8.5.2. Requirements
The normalized value of speakerphone 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 2 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 speakerphone TCLw is 27 dB, the measured SLR is 15 dB and the
measured RLR is -1 dB, then the normalized value of TCLw = 27 dB + (2-(-1)) dB + (13 - 15) dB
= 28 dB.
8.6. 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 frequency.
8.6.1. Measurement Method
Place the speakerphone 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, with no
horizontal dimension of the table less than 0.8 m (see IEEE Std. 1329, clause 7.3.4).
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 1/12th octave bands, or R40 intervals centered at 300 Hz to 6700 Hz. The
TCLw 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 and
represented by the L16-256 codec. The recommended pattern is 250 msec of noise (on), 150 msec
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 speakerphone and transmission circuit fully
active, measure the attenuation from the digital input to the digital output.
8.6.2. Requirement
The speakerphone 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 a minimum 6 dB stability loss over the
entire range of adjustable receive levels.
55
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Annex A (normative) – Calculation of Loudness Ratings
This Annex details the loudness rating calculations and weighting factors relevant to this document.
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 G has the weighting factors used for wideband SLR and RLR
calculation. The STMR weighting factors are unchanged. For convenience, the P.79 weighting
information is included in this document in Table 14.
Wideband Send Loudness Rating (Handset and Headset):
Band 20
SLR = - 57.1 log10

10
(0.1 * 0.175 * (SMJ – Wsi ))
[A1]
i = Band 1
Where:
i
SMJ
Wsi
Frequency bands from Table 14, bands 1-20.
Send frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured
per this standard.
Send weighting factor from Table 14.
Wideband Receive Loudness Rating (Handset and Headset):
Band 20
RLR = - 57.1 log10

10(0.1 * 0.175 * (SJE – Wri ))
[A2]
i = Band 1
Where:
i
SJE
Wri
Frequency bands from Table 14, bands 1-20.
Receive frequency response data (Sensitivity, Junction-to-Ear Reference Point) in
dBPa/V measured per this standard. See Annex D for DRP to ERP information.
Receive weighting factor from Table 14.
Note: There is no Leakage Correction, LE, when using the Type 2, 3.2, 3.3 or Type 3.4 ear simulator.
56
SP-3-4705-RV1 (to be published as ANSI/TIA-920-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:
i
SmeST
WMSi
Frequency bands from Table 14, bands 1-20.
Sidetone frequency response data (Sensitivity, Mouth-to-Ear) in dB Pa/Pa measured
per this standard.
Sidetone weighting factor from Table 14.
Wideband Send Loudness Rating (Speakerphone):
Band 20
SLR = - 57.1 log10

10
(0.1 * 0.175 * (SMJ – Wsi ))
[A4]
i = Band 1
Where:
i
SMJ
Wsi
Frequency bands from Table 14, bands 1-20.
Send frequency response data (Sensitivity, Mouth-to-Junction) in dBV/Pa measured
per this standard.
Send weighting factor from Table 14.
Wideband Receive Loudness Rating (Speakerphone):
Band 20
RLR = - 57.1 log10

10(0.1 * 0.175 * (SJE – Wri )) - CorrRFF
[A5]
i = Band 1
Where:
i
SJE
Frequency bands from Table 14, bands 1-20.
Receive frequency response data (Sensitivity, Junction-to-Ear) in dBPa/V measured
per this standard.
Wri
Receive weighting factor from Table 14.
Correction of 14 dB for receive measured in the free field as recommended
CorrRFF
in ITU-T P.340 (05/2000)
Note that in most standards, such as IEEE Std. 1329 and ITU-T Recommendation P.340, there is a
correction factor of 14 dB subtracted from calculated RLR (RLR = RLR-14). In order to be
consistent with TIA-810-A, this standard does not require the correction factor. In order to compare
RLR as measured by other standards the correction factor may be used. For example, an RLR of 16
calculated using equation [A5] would be 8 dB-14 dB = 2 dB. If a correction factor is used, it must
be stated with the data.
57
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Table 14 – ITU-T P.79 Annex G
Weighting factors for calculating wideband loudness ratings
Band No.
Midfrequency
(Hz)
Send
Wsi
Receive
Wri
Sidetone
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
8000
181.7
166.7
52.8
20
Note:
The send and receive weighting values for bands 4-17 are different
from the values used for narrowband (200 to 4000 Hz) calculation and
therefore should not be used for narrowband loudness calculation. ITU-T P.79
advises that the sealed weighting be used for STMR computations, even though the
HATs is not sealed.
58
SP-3-4705-RV1 (to be published as ANSI/TIA-920-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 1 mW, typically using a circuit impedance of 600  or 900 .
0 dBm = 10 log 1(mW)
dBV = 10 log V2
= 20 log V
or,
V = 10 dBV/20
P = V2/R, where for dBm reference, R = 600 
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  (use -2.2 dB)
P = V2/R, where for dBm reference, R = 900 
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  (use -0.5 dB)
This means that if we substitute 600  for 900  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 " to "Z2 = 900 ", 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 , -20 dBm signal to dBV, subtract 2.2
to get -22.2 dBV. Another example is if -20 dBm is measured across 600 , then when measuring
across 900 , add a correction of -1.8 dB to get -21.8 dBm (since less power is dissipated by the
higher resistance).
59
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Acoustic Sound Pressure Conventions
dB Pa (dB Pascal)
dBSPL (dB Sound Pressure Level)
Where,
0 dB Pa = 94 dBSPL, 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”.
60
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Annex C (informative) – R40 Preferred Frequencies
The ISO 3, R40 basic series preferred frequencies are listed in Table 15. 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 15 – R40 Preferred Frequencies
#
0
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
Preferred Frequencies
(Hz)
#
100
106
112
118
125
132
140
150
160
170
180
190
200
212
224
236
250
265
280
300
315
335
355
375
400
425
450
475
500
530
560
600
630
670
710
750
800
850
900
950
1000
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
61
Preferred Frequencies
(Hz)
1000
1060
1120
1180
1250
1320
1400
1500
1600
1700
1800
1900
2000
2120
2240
2360
2500
2650
2800
3000
3150
3350
3550
3750
4000
4250
4500
4750
5000
5300
5600
6000
6300
6700
7100
7500
8000
8500
9000
9500
10000
SP-3-4705-RV1 (to be published as ANSI/TIA-920-A)
Annex D (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 16 to the measured data.
For complete information refer to ITU-T Recommendation P.58 and IEEE Standard 269 and 269a.
Table 16 – DRP to ERP Correction Factors
Frequency
92
97
103
109
115
122
130
137
145
154
163
173
183
193
205
218
230
244
259
274
SDE
(dB)
0.1
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.1
0.1
0.0
0.1
0.0
0.1
0.2
0.3
0.3
Frequency
(Hz)
290
307
325
345
365
387
410
434
460
487
516
546
579
613
649
688
729
772
818
866
SDE
(dB)
0.3
0.2
0.2
0.2
0.4
0.5
0.4
0.6
0.3
0.7
0.6
0.6
0.6
0.6
0.8
0.8
1.0
1.1
1.1
1.2
Frequency
(Hz)
917
972
1029
1090
1155
1223
1296
1372
1454
1540
1631
1728
1830
1939
2053
2175
2304
2441
2585
2738
62
SDE
(dB)
1.3
1.4
1.8
2.0
2.3
2.4
2.6
3.1
3.3
3.9
4.4
4.8
5.3
6.0
6.9
7.5
8.1
9.1
9.5
10.4
Frequency
(Hz)
2901
3073
3255
3447
3652
3868
4097
4340
4597
4870
5158
5464
5788
6131
6494
6879
7286
7718
8175
8659
SDE
(dB)
11.0
10.5
10.2
9.1
8.0
6.9
5.8
5.0
4.2
3.3
2.7
2.4
2.4
2.5
3.3
4.5
.9
9.0
14.2
20.7