Meetings Attended (C= Contribution, A= Attended Meeting) Organization Represented. Ken Macdonald (Chair) Roger Britt (Editor) Ron Magnuson

Transcription

1 TR Telecommunications Telephone Terminal Equipment Transmission Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline Telephones SP RV2 (to become ANSI/TIA-810-B) Draft 4 (January 12, 2005) Editor: Roger Britt rbritt@nortelnetworks.com Changes: (Ken and Roger) 1. Accepted the Draft 3 changes that were agreed to at the Teleconference meeting Jan 12, Added text for receive response and RLR low and high leak requirements 3. Added Annex A text to have LR definitions. 4. Added Work group attendance / contribution table. 5. Added note regarding ear simulator reference tabled until after SG12 meeting. 6. Added text discussed at December Teleconference meeting. 7. Editorial added Annex Heading style for cross references. 8. I tried to implement, or comment, proposals from Glenn s contribution for the handset section.. 9. Changed Send and Receive response text to not reference P Added Receive Volume control requirements from C. 11. Added latest RVC text proposed by Roger NOTICE Nortel Networks grants a free, irrevocable license to the Telecommunications Industry Association (TIA) to incorporate text or other copyrightable material contained in this contribution and any modifications thereof in the creation of a TIA Publication; to copyright and sell in TIA's name any TIA Publication even though it may include all or portions of this contribution; and at TIA's sole discretion to permit others to reproduce in whole or in part such contribution or the resulting TIA Publication. Nortel Networks will also be willing to grant licenses under such copyrights to third parties on reasonable, non-discriminatory terms and conditions for purpose of practicing a TIA Publication which incorporates this contribution. This document has been prepared by Nortel Networks to assist the TIA Engineering Committee. It is proposed to the Committee as a basis for discussion and is not to be construed as a binding proposal on Nortel Networks. Nortel Networks specifically reserves the right to amend or modify the material contained herein and nothing herein shall be construed as conferring or offering licenses or rights with respect to any intellectual property of Nortel Networks other than provided in the copyright statement above. Nortel Networks agrees that if it has changed or added text to the required language in the TIA contribution template as contained in the TIA Engineering Manual, any such change or addition which is inconsistent with such contribution template, is of no force or effect.

2 TR Working Group Meeting Participants (This list will be replaced in the Standard with a list of Contributors & Participants) Working Group Member Organization Represented Meetings Attended (C= Contribution, A= Attended Meeting) Feb, 2005 Las Vegas Jan, 2005 Teleconference Dec, 2004 Teleconference Nov, 2004 San Antonio Ken Macdonald (Chair) Roger Britt (Editor) Ron Magnuson David Stenner Microtronix Systems Ltd. C C C C Nortel Networks Acoustics for Communications Advent Instruments, Inc. A Roger Hunt Atlinks USA A Kirit Patel Cisco Systems A A Ruchir Dave Cisco Systems A C C C C A A A John Bareham Consultant in Electoacoustic A Bob Young IEEE STIT A A Glenn Hess MWM Acoustics C A A Miguel Nortel Networks DeAraujo A A Allen Woo Plantronics A A A Steve Graham Plantronics C A Tailey Tung Siemens Communications Inc. A Amar N. Ray Sprint Technologies A A Joachim Tenovis GmbH Pomy A Tom Harley Texas Instruments C Bogdan Texas Instruments Kosanovic A Benito Pēna UTSA Student A Trone Bishop Verizon A Steve Kropp Steve Whitesell VTech Communications VTech Communications A A

3 TR Al Baum Uniden A Jim Smith Wyle Labs A

4

5 TABLE OF CONTENTS SP RV2 (to become ANSI/TIA-810-B) 1. INTRODUCTION SCOPE LIMITS OF APPLICABILITY CATEGORIES OF CRITERIA FCC PART ENVIRONMENTAL SAFETY NORMATIVE REFERENCES DEFINITIONS, ABBREVIATIONS AND ACRONYMS CODEC EAR REFERENCE POINT (ERP) HATS POSITION MOUTH REFERENCE POINT (MRP) QUIET AND FULL SCALE CODE REFERENCE CODEC DIRECT DIGITAL PROCESSING SOUND PRESSURE LEVELS ELECTRIC POWER AND NOISE LEVELS ABBREVIATIONS AND ACRONYMS HANDSET TECHNICAL REQUIREMENTS HANDSET FREQUENCY RESPONSE Handset Send Frequency Response Handset Receive Frequency Response HANDSET LOUDNESS RATINGS AND RECEIVE VOLUME CONTROL Handset Send Loudness Rating (SLR) Handset Receive Loudness Rating Handset Receive Volume Control Handset Talker Sidetone HANDSET NOISE Handset Send Noise Handset Send Single Frequency Interference Handset Receive Noise Handset Receive Single Frequency Interference HANDSET RECEIVE COMFORT NOISE (ADVISORY) General Measurement Method Requirement HANDSET DISTORTION AND NOISE Handset Send Distortion and Noise Handset Receive Distortion and Noise i

6 5.6. WEIGHTED TERMINAL COUPLING LOSS (TCLW) Measurement Method Requirements STABILITY LOSS Measurement Method Requirement LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE INPUT) General Measurement Method Requirements SHORT DURATION (PEAK) ACOUSTIC PRESSURE General Measurement Method Requirements PACKET VOICE LATENCY (ADVISORY) Handset Send Latency Handset Receive Latency HEADSET TECHNICAL REQUIREMENTS HEADSET FREQUENCY RESPONSE Headset Send Frequency Response Headset Receive Frequency Response HEADSET LOUDNESS RATINGS Headset Send Loudness Rating Headset Receive Loudness Rating Headset Talker Sidetone HEADSET NOISE Headset Send Noise Headset Send Single Frequency Interference Headset Receive Noise Headset Receive Single Frequency Interference HEADSET DISTORTION AND NOISE Headset Send Distortion and Noise Headset Receive Distortion and Noise WEIGHTED TERMINAL COUPLING LOSS (TCLW) Measurement Method Requirements LONG DURATION MAXIMUM ACOUSTIC PRESSURE (STEADY STATE INPUT) General Measurement Method Requirements SHORT DURATION (PEAK) ACOUSTIC PRESSURE General Measurement Method Requirements HANDSFREE TECHNICAL REQUIREMENTS (ADVISORY)...41 ii

7 7.1. HANDSFREE FREQUENCY RESPONSE Handsfree Send Frequency Response Handsfree Receive Frequency Response HANDSFREE LOUDNESS RATINGS AND RECEIVE VOLUME CONTROL Handsfree Send Loudness Rating Handsfree Receive Loudness Rating Handsfree Receive Volume Control HANDSFREE NOISE Handsfree Send Noise Handsfree Send Single Frequency Interference Handsfree Receive Noise Handsfree Receive Single Frequency Interference HANDSFREE DISTORTION AND NOISE Handsfree Send Distortion and Noise Handsfree Receive Distortion and Noise WEIGHTED TERMINAL COUPLING LOSS (TCLW) Measurement Method Requirements STABILITY LOSS Measurement Method Requirement ANNEX A (INFORMATIVE) CALCULATION OF LOUDNESS RATINGS ANNEX B (INFORMATIVE) MEASUREMENT AND LEVEL CONVERSIONS ANNEX C (INFORMATIVE) PREFERRED 1/12 OCTAVE FREQUENCIES iii

8 FOREWORD (This foreword is not part of this standard.) This document is a TIA/EIA Telecommunications standard produced by Working Group TR of Committee TR-41. This standard was developed in accordance with TIA/EIA procedural guidelines, and represents the consensus position of the Working Group and its parent Subcommittee TR-41.3, which served as the formulating group. This standard is based on TIA/EIA/IS-810. The TR VoIP/PCM Transmission Performance Working Group acknowledges the contribution made by the following individuals in the development of this standard. Name Representing Ken Macdonald Microtronix Systems Ltd. Chair Roger Britt Nortel Networks Editor Copyrighted parts of ITU-T Appendix I to Recommendation G.113 and Recommendation P.79 are used with permission of the ITU. The ITU owns the copyright for the ITU Recommendations. Copyrighted parts of ISO 3 are used with permission of the ISO. The ISO owns the copyright for the ISO Standards. The three annexes in this Standard are informative and are not considered part of this Standard. Suggestions for improvement of this standard are welcome. They should be sent to: Telecommunications Industry Association Engineering Department Suite Wilson Boulevard Arlington, VA iv

9 1. Introduction This revision of TIA/EIA/IS-810 establishes handset, headset and handsfree telephone audio performance requirements for digital wireline telephones regardless protocol or digital format. A number of improvements and corrections have been made, particularly related to single frequency interference and acoustic pressure. This standard only addresses conventional narrowband performance, where narrowband is defined as the frequency range between 300 and 3400 Hz. Wideband telephony, in the frequency range between 150 and 6800 Hz, is an enhancement that is likely to offered by VoIP telephones. The performance requirements of wideband telephony will be addressed in a future TIA/EIA standard. 1

10 2. Scope This standard establishes voice performance requirements for narrowband digital wireline telephones with codecs that conform to the ITU-T G-Series Recommendations and where transmission is in digital format. A telephone is defined as a device that terminates networks and provides telephony voice service. Transmission may be over Local Area Networks, Firewire/IEEE1394, Universal Serial Bus (USB), public ISDN or digital over twisted pair wire. Applications include Voice over Internet Protocol (VoIP) and PCM-based telephones, whether connected through modems, gateways, or PBXs and personal computer-based telephones that may or may not have handsets. Technical requirements are set for handset, headset and handsfree (speakerphone) modes of operation.. These requirements apply regardless of the technology used to couple the handset or headset to the telephone. Coupling may be by a cord, a short range air interface such as, but not limited to, a radio interface, an electric field interface, a magnetic field interface or an infra-red interface. The test measurement methods in this standard reference procedures in IEEE Std 269 where applicable. Several performance measurement procedures are established today, each of which yields standardized measurement data that may be used for the comparison of different products. This document may reference only a single procedure or specific equipment, however the intent is not to be all-inclusive. Any measurement procedure and equipment that can result in an identical measurement is considered valid. While the procedures may call out specific test points within the requirements, the full range of the requirements take precedent Limits of Applicability These requirements are not intended to describe specific requirements for the following types of digital voice terminal equipment: telephones with carbon transmitters, ISDN terminal adapters and cellular voice terminals Categories of Criteria Mandatory requirements are designated by the word "shall". Advisory requirements are designated by the word "should," or "may," or "desirable" which are used interchangeably in this standard. Advisory criteria represent product goals or are included in an effort to ensure universal product compatibility. Where both a mandatory and an advisory level are specified for the same criterion, the advisory level represents a goal currently identifiable as having distinct compatibility or performance advantages toward which future designs should strive 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 Environmental The telephone will also be subject to the applicable environmental conditions specified in EIA/TIA

11 2.5. Safety SP RV2 (to become ANSI/TIA-810-B) This standard does not contain safety requirements. Compliance with the applicable UL and CSA safety standards may be required in certain locations. 3

12 3. Normative References The following standards contain provisions, which, through reference in this text, constitute provisions of this Standard. At the time of publication, the editions indicated were valid. All standards are subject to revision, and parties to agreements based on this Standard are encouraged to investigate the possibility of applying the most recent editions of the standards indicated below, or their successors. ANSI and TIA maintain registers of currently valid national standards published by them. [1] ANSI/TIA/EIA-464-B-1996, Requirements for Private Branch Exchange (PBX) Switching Equipment. [2] ANSI/TIA/EIA-504-A-1997, Telecommunications Telephone Terminal Equipment Magnetic Field and Acoustic Gain Requirements for Handset Telephones Intended for use by the Hard of Hearing. [3] ANSI/TIA/EIA-571-A-1999, Environmental Considerations. [4] ANSI/TIA/EIA-579-A-1998, Telecommunications Telephone Terminal Equipment Transmission Requirements for Digital Wireline Telephones. [5] ANSI/IEEE Standard , Standard Methods for Measuring Transmission Performance of Analog and Digital Telephone Sets. [6] ANSI/IEEE Standard (Reaff 1998), Standard Method for Determining Objective Loudness Ratings of Telephone Connections. [7] ANSI/IEEE Standard , Standard Methods for Measuring Transmission Performance of Telephone Handsets and Headsets. [8] ANSI/IEEE Standard , Standard Method for Measuring Transmission Performance of Hands-Free Telephone Sets. [9] ANSI S , Sound Level Meters. [10] 47 CFR Part 68, Connection of Terminal Equipment to the Telephone Network. [11] ITU-T Recommendation G.107 (1998), The E-Model, A Computational Model for use in Transmission Planning. [12] ITU-T Recommendation G.109 (1999), Definition of categories of speech transmission quality. [13] ITU-T Recommendation G.113 (1996), Transmission impairments. [14] ITU-T Appendix I to Recommendation G.113 (1998), Transmission impairments Appendix I: Provisional planning values for the equipment impairment factor Ie. [15] ITU-T Recommendation G.114 (1996), One-way transmission time. [16] ITU-T Recommendation G.122 (1993), Loudness ratings (LRs) of national systems. [17] ITU-T Recommendation G.131 (1993), Control of talker echo. 4

13 [18] ITU-T Recommendation G.175 (1997), Transmission planning for private/public network interconnection of voice traffic. [19] ITU-T Recommendation G.711 (1988), Pulse code Modulation (PCM) of voice frequencies. [20] ITU-T Recommendation G.712 (1996), Transmission performance characteristics of pulse code modulation. [21] ITU-T Recommendation G (1996), Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. [22] ITU-T Recommendation G.729 (1996), Coding of speech at 8 kbit/s using conjugatestructure algebraic-code-excited linear-prediction (CS-ACELP). [23] ITU-T Recommendation O.41 (1994), Psophometer for use on telephone-type circuits. [24] ITU-T Recommendation O.131 (1988), Quantizing distortion measuring equipment using a pseudo-random noise test signal. [25] ITU-T Recommendation P.51 (1996), Artificial mouth. [26] ITU-T Recommendation P.56 (1993), Objective measurement of active speech level. [27] ITU-T Recommendation P.57 (1996), Artificial ears. [28] ITU-T Recommendation P.58 (1996), Head and torso simulator for telephonometry. [29] ITU-T Recommendation P.64 (1999), Determination of sensitivity/frequency characteristics of local telephone systems. [30] ITU-T Recommendation P.79 (1999), Calculation of loudness ratings for telephone sets. [31] ITU-T Recommendation P.310 (1996), Transmission characteristics for telephone band ( Hz) digital telephones. [32] ITU-T Recommendation P.360 (1998), Efficiency of devices for preventing the occurrence of excessive acoustic pressure by telephone receivers. [33] ITU-T Recommendation P.501 (1996), Test signals for use in telephonometry. [34] ISO 3: 1973 Preferred numbers - Series of preferred numbers. [35] ETSI EG (1999), Specification and measurement of speech transmission quality; Part 1: Introduction to objective comparison measurement methods for one-way speech quality across networks. 5

14 4. Definitions, Abbreviations and Acronyms For the purposes of this Standard, the following definitions apply 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 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 HATS Position The HATS (head and torso simulator) position (ITU-T P.64 Annex D and Annex E) is the correct artificial head handset position for measuring sensitivity and frequency response characteristics. The HATS position has been shown to be essentially identical to the LRGP (loudness rating guard-ring position) position, except for the mouth simulator direction, which has been corrected with a 19 degrees downwards rotation to more closely match real talkers. For handsets with omnidirectional microphones, measurements on the two heads may differ slightly, typically less than 1dB. For handsets with directional or noise-canceling microphones, the differences will be larger, and the HATS position will give the more realistic results. Some equipment may use the term LRGP-H for the HATS position Standard Test Position The Standard Test Position consists of a high leak position and a low leak position. The high leak position as defined below: 1. For the Type 3.3.? artificial ear the receiver shall be located at the ERP. 2. For the Type 3.4 artificial ear the receiver shall be applied with 6 N force which corresponds to the ERP position. The low leak position as defined below: 1. For the Type 3.3.? artificial ear the receiver contacting the pinna with a force of 18 N. 2. For the Type 3.4 artificial ear the receiver contacting the pinna with a force of 15 N. Although the Type 3.3.? and 3.4 artificial ears are similar, they are not identical. Therefore, the same type of artificial ear shall be used for all tests and the ear type shall be documented Recommended Test Positions (RTP) RTP may be defined by following these steps: 1. Find Ear-Cap Reference Point (ECRP) on the phone. Unless specified by the manufacturer, the ECRP is the intersection of the external ear-cap reference plane with a normal axis (Xaxis as defined by ITU-T P.64, Annex E) 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 many handsets, the ear-cap reference plane is parallel to the reference plane of the positioning device. 2. Line up ECRP at the Ear simulator ERP on the positioning device. The ear-cap reference plane shall be identical to the plane of the ERP as defined by the positioning device. 3. If desired, move ECRP in ear-cap reference plane relative to ERP. This can be defined as (Y, Z) coordinates (ITU-T P.64, Annex E) in the ear-cap reference plane. If none given, 6

15 leave the ECRP centered on ERP, equivalent to (0, 0) coordinates. The Y-axis is defined along the length of the phone with positive Y being in a direction towards the microphone from ECRP (moving the phone down on the positioner). The Z-axis intersects at ECRP, and is perpendicular to the Y axis, with positive Z being towards the right as the phone is observed from the front (moving the phone to the right on positioner) (defined only for right ear). 4. If desired, adjust the three angles (A, B, C) as defined for device under test. If none given, use angles consistent with the HATS position, which shall be defined by the handset positioner equipment manufacturer. 5. For the Type ear adjust the force to 6 N for the high leak position and 18 N for the low leak position. For the Type 3.4 ear adjust the force to 6 N for the high leak position and 15 N for the low leak position. 6. RTP can then be defined as the combination (Y, Z) coordinates, three angles and force or X coordinate. The manufacturer of the device under test is responsible for providing this data Mouth Reference Point (MRP) The mouth reference point is located on axis and 25 mm in front of the lip ring of a mouth simulator Quiet and Full Scale Code Level Table 1 PCM Codes for Zero (Quiet Code) and Full Scale Mu-Law A-Law Sign Bit Chord Bit Step Bits Sign Bit Chord Bits Step Bits + Full Scale Zero Zero Full Scale Reference Codec A reference codec is used for testing digital telephone terminals with analog test equipment. Figure 1 shows the basic test setup using a reference codec. A codec that approaches an ideal codec and has superior, well-defined, characteristics qualifies as a reference codec. When a 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 volt rms analog signal appears at the decoder output. At the digital reference point 0 dbm0 is 3.14 (A-law) or 3.17 (mu-law) db below digital full scale. This implementation of a reference codec eliminates the 600 ohm source and load resistors specified by other standards. The coder input impedance is high relative to the generator and the decoder output impedance is low relative to the measuring voltmeter. The interface block, shown in Figures 1 and 2, passes the voice channel digital bit stream to the terminal without modification. There is no gain or loss in the send or receive direction due to the interface. If the interface does change the digital voice stream then the terminal and interface shall be considered jointly as the terminal. An example of this is a receive volume control implemented in a PBX or gateway. 7

16 4.9. Direct Digital Processing Direct digital generation of the receive signal and analysis of the send signal may be used in place of the reference codec as shown in Figure 2. This method is preferred when possible. Figure 1 Digital Telephone Set Test Arrangement with Reference Codec Send Digital Reference Point (Junction j) v SEND p M Mouth Sound Pressure at MRP Digital Set Interface Decoder v p E Ear Sound Pressure at ERP Receive Coder GEN vrcv Reference Codec Figure 2 Digital Telephone Set Test Arrangement using Direct Digital Generation and Analysis Send Digital Reference Point (Junction j) p M Mouth Sound Pressure at MRP p E Ear Sound Pressure at ERP Digital Set Interface Digital Analysis Digital Generation Receive 8

17 4.10. 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/m 2. 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/m 2 (0 dbpa = 94 dbspl). dba: The A-weighted sound level is the sound pressure level in dbspl, weighted by use of metering characteristics and A-weighting specified in ANSI S Electric Power and Noise Levels The following electric power and noise level units are used in this standard: dbm0: The absolute power level at a digital reference point of the same signal that would be measured as the absolute power level, in dbm, if the reference point was analog. The absolute power in dbm is defined as 10 log (power in mw / 1 mw). When the impedance is 600 ohm resistive, dbm can be referred to a voltage of volts, which results in a reference active power of 1 mw. Note that 0 dbm0 is not the maximum digital code. For Mu Law codecs 0 dbm0 is 3.17 db below digital full scale. For A Law codecs 0 dbm0 is 3.14 db below digital full scale. dbm0p: The noise level, measured by a psophometer with a special noise weighting filter as described in ITU-T Recommendations O.41 and P.53. The small letter p comes from the French word ponderé. The equivalent English word is weighted, but the p refers specifically to psophometric weighting Abbreviations and Acronyms Abbreviations and acronyms, other than in common usage, which appear in this standard, are defined below. AGC DRP EFR ERP FFT GoB HATS Ie ISDN LRGP MOS MRP OLR PBX PCM Automatic Gain Control Drum Reference Point Enhanced Full Rate Ear Reference Point Fast Fourier Transform Good or Better Head and Torso Simulator Equipment Impairment factor Integrated Services Digital Network Loudness Rating Guard-ring Position Mean Opinion Score Mouth Reference Point Objective Loudness Rating Private Branch Exchange Pulse Code Modulation 9

18 PL PLC PoW QoS RLR RTP SLR STMR TCLt TCLw VAD VoIP Packet Loss Packet Loss Concealment Poor or Worse Quality of Service Receive Loudness Rating Recommended Test Position Send Loudness Rating Sidetone Masking Rating Temporally weighted Terminal Coupling Loss Weighted Terminal Coupling Loss Voice Activity Detector Voice over Internet Protocol 10

19 5. Handset Technical Requirements All telephones shall support G.711 A-law and mu-law. The handset technical requirements apply only to mu-law and A-law G.711 codecs. If the telephone uses other G-Series low-bit rate vocoders, the manufacturer must ensure that their implementation passes the standard test vectors associated with that codec. For bit exact vocoders, such as G.729, it is important to ensure that vector testing has been performed and found to be compliant with the associated ITU requirement. Unless specified otherwise: Encoding and decoding is assuming to be G.711 in mu-law. In particular, this applies to the requirements in the Distortion and Noise sections. Algorithmic processes, such as Echo Control, VAD and AGC, may influence the test results or require test signals other than sinewaves. ITU-T Recommendation P.64 allows several types of test signals. The test signal used should be stated. The test signal levels specified in this standard shall be used. Test signal levels that differ from those specified in this standard may also be required. Packet voice latency may introduce significant delay that must be accounted for by the test equipment. Equipment using nonlinear voice signal processing may require subjective testing. All tests shall be performed with a Type 3.3.? or a Type 3.4 ear simulator (ITU-T Recommendation P.57) with the handset in the HATS position. 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, to be used for all handset tests. 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 Handset Frequency Response Handset Send Frequency Response Send frequency response is the ratio of the voltage output of the reference codec to the sound pressure at the Mouth Reference Point (MRP) for each frequency or frequency band (Fi) as shown in the equation below: S MJ = 20 log (V SEND / P M ) db rel 1 V / Pa [1] Where S MJ P M V SEND Send Sensitivity, Mouth to Junction, at Fi. Sound pressure at the MRP at Fi. RMS output voltage of the reference codec at Fi Measurement Method Measurements should be done in ISO 1/12 octave intervals or smaller, over a minimum range of 100 Hz through 4000 Hz using the measurement set-up shown in Figure 3. The test signal level shall be dbpa at the MRP. 11

20 Figure 3 Handset Send Frequency Response Measurement Method v SEND HATS Decoder v GEN Send p M Digital Set Interface Coder Measuring Amplifier Mouth Simulator Quiet Room Reference Codec v SEND Send p M Decoder v GEN Digital Set Interface Measuring Amplifier Mouth Simulator Coder Quiet Room Reference Codec Requirement The send frequency response shall be below the upper limit and above the lower limit defined in Table 2 and shown in Figure 4. The limit curves shall be determined by straight lines joining successive co-ordinates given in the table, when frequency response is plotted on a linear scale against frequency on a logarithmic scale. Note: The frequency response mask is a floating or best fit mask. 12

21 Table 2 Co-ordinates of Handset Send Response Limit Curve Frequency (Hz) SP RV2 (to become ANSI/TIA-810-B) Send Response Limit (db) [arbitrary level] upper limit lower limit infinity infinity Figure 4 Handset Send Frequency Response Mask 10 Arbitrary Level (db) Frequency (Hz) Handset Receive Frequency Response Receive frequency response is the ratio of the sound pressure measured in the ear simulator to the voltage input to the reference codec for each frequency or frequency band (Fi) as shown in the equation below: 13

22 S JE = 20 log (P E / V RCV ) db rel 1 Pa / V [2] Where S JE P E V RCV Receive Sensitivity, Junction to Ear, at Fi. ERP Sound pressure measured by ear simulator at Fi. Measurements collected at other points, e.g., DRP and free field, must be corrected back to ERP. RMS Input voltage to the reference codec at Fi Measurement Method The receive frequency response shall be measured with the receiver at the high leak position and also with the receiver at the low leak position. The receive frequency response is measured using the measurement set-up shown in Figure 5. Measurements should be done in ISO 1/12 octave intervals or smaller, over a minimum range of 100 Hz through 8000 Hz. The test signal level shall be dbv (-16 dbm0). The frequency response measured with the ear simulator must be transformed to the ear reference point (ERP). Telephone sets with adjustable receive levels shall be adjusted so that their RLR is as close as possible to the nominal value of Section for this test. Figure 5 Handset Receive Frequency Response Measurement Method Receive p E Ear Simulator Decoder Sound Pressure Measuring Amplifier Digital Set Interface Coder GEN Quiet Room vrcv Reference Codec Receive p E Ear Simulator Sound Pressure Measuring Amplifier Digital Set Interface Decoder Coder GEN Quiet Room vrcv Reference Codec Requirement The receive frequency response shall be below the upper limit and above the lower limit defined in Table 3 and shown in Figure 6. The receive frequency response requirements between 100 Hz and 8 khz (referenced to the ERP) are as follows: 14

23 1. With the receiver at the high leak position, the receive frequency response should fall within the desired limits in Table 3 (shown in Figure 6). 2. With the receiver at the low leak position, the receive frequency response: a. Shall fall within the mandatory limits in Table 3 (shown in Figure 6). b. Should fall within the desired limits in Table 3 (shown in Figure 6). The limit curves shall be determined by straight lines joining successive co-ordinates given in the table, when frequency response is plotted on a linear scale against frequency on a logarithmic scale. Note: The frequency response mask is a floating or best fit mask. Table 3 Co-ordinates of Handset Receive Response Limits Limit Curve Frequency (Hz) Receive Response Limit (db) [arbitrary level] upper limit lower limit infinity infinity 15

24 Figure 6 Handset Receive Frequency Response Mask 10 Arbitrary Level (db) Frequency (Hz) 5.2. Handset Loudness Ratings and Receive Volume Control Handset Send Loudness Rating (SLR) The SLR is defined in Annex A Measurement Method The SLR shall be calculated using the 1/3 octave sensitivity data collected from the send frequency response measurement. Use equation [A3] of Annex A and bands 4 to 17, Table A Requirement The terminal shall be designed to have a nominal SLR value of 8 db, with a tolerance of ±4.0 db Handset Receive Loudness Rating The RLR for a digital telephone set is the conversion ratio of a defined voltage input to the reference codec to an acoustic output signal from the receiver. Refer to Annex A and ITU-T Recommendation P Measurement Method The RLR shall be calculated from the 1/3 octave sensitivity data collected from the receive frequency response measurement. Use equation [A4] of Annex A and bands 4 to 17, Table A.1. 16

25 Requirement SP RV2 (to become ANSI/TIA-810-B) The RLR values measured with the receiver at the high leak position shall have a nominal RLR value of 2 db, with a tolerance of y.y/+x.x db and should have a nominal RLR value of 2 db, with a tolerance of ±4.0 db. The RLR values measured with the receiver at the low leak position shall The terminal shall be designed to have a nominal RLR value of 2 db, with a tolerance of ±4.0 db. The volume control position that is closest to the nominal RLR value, with the receiver in the low leak position, is defined to be the nominal volume control position Handset Receive Volume Control Range Roger s Jan 26 proposed text: The current regulatory volume control requirements are specified in 47 CFR Part NOTE: The RLR measurements in this document use the HATS (Head and Torso Simulator) while the current 47 CFR Part references ROLR measurements in ANSI/EIA/TIA , which specifies the Type 1 ear (ITU-T Recommendation P.57). Roger s old proposed Text: The RLR shall be calculated for the Minimum, Reference and Maximum volume control settings. The Reference volume control setting is the lowest setting where the RLR is equal to or louder than the nominal RLR value specified in The RLR at the Maximum volume control setting shall be at least 12 db louder than the Reference volume control setting RLR. No Minimum volume control setting is specified for handset telephones, but it is desirable that the RLR at the Minimum volume control setting be at least 6 db quieter than the Reference volume control setting The following is from C as agreed at the December meeting, but is still being discussed. The receiver volume control performance requirements are below. Reference 47 CFR Part for regulatory volume control requirements. NOTE: The RLR measurements in this document use the HATS (Head and Torso Simulator) while the current 47 CFR Part references ROLR measurements in ANSI/EIA 470 A 1987 which use a Type 1 ear (ITU-T P.57). The test method and acoustic output requirements for receive-amplified handset telephones are documented in ANSI/TIA/EIA-504-A. This standard includes the magnetic output requirements for handset telephones intended for use by the hard of hearing Measurement Method Roger s Jan 26 proposed text: The RLR shall be calculated from the 1/3 octave sensitivity data collected from the receive frequency response measurement with the receiver at the low leak position. The measurement shall be done with the volume control at both the minimum and maximum settings. Use equation [A2] of Annex A and bands 4 to 17, Table A.1. Old method: 17

26 The receive frequency response shall be measured with the receiver at the low leak position. The measurement is performed according to with the volume control at the maximum volume control setting. The RLR shall be calculated from these responses as described in Annex A Requirement Roger s Jan 26 proposed text: At the maximum volume control setting, the RLR shall be less than 10 db without significant clipping of the test signal; if the RLR is less than 16 db then the terminal shall automatically reset to the nominal volume control setting after going On-Hook to end the call. At the minimum volume control setting, the RLR should be greater than +8 db. NOTE: Objective distortion requirements are specified in Old Requirement: 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 (measured in at the low leak position). If the maximum volume control setting RLR is more than 18 db louder than the RLR at the Reference Volume Control Setting, then the CPE shall automatically reset to the Reference Volume Control Setting after going On-Hook to end the call Handset Receive Volume Control Settings The volume control shall have at least two unique settings between the nominal and maximum volume control settings Magnetic Field for Hearing Aid Coupling Roger s Jan 26 proposed text: The current regulatory hearing aid compatibility magnetic output requirements are specified in 47 CFR Part NOTE: Part is old and does not provide suitable references for testing Digital Telephones. See ANSI/TIA/EIA-504-1, ISDN or special use telephones, for the appropriate test level. Old Note: NOTE - See 47 CFR Part for these requirements Handset Talker Sidetone The sidetone masking rating (STMR) for a digital telephone set is the ratio of a defined input acoustic signal at the mouth reference point to the resulting acoustic output signal from the receiver Measurement Method The test signal level at the MRP shall be -4.7 dbpa. For each frequency given in Table A.1, bands 1 to 20, the sound pressure in the artificial ear shall be measured. The STMR shall be calculated using equation [A5] of Annex A. 18

27 Telephone sets with adjustable receive levels shall be tested at the minimum, nominal and maximum settings. For the nominal setting, adjust the level so that the RLR is as close as possible to the nominal RLR value Requirement The value of STMR shall be within the range of 18 db ± 6 db, for any adjustable receive level Handset Noise The noise levels are related to the SLR and RLR requirements Handset Send Noise General The send noise of a digital telephone is the 5 second average background noise at the digital transmit output with the telephone transmitter isolated from sound input and mechanical disturbances Measurement Method In a quiet environment (ambient noise less than 30 dba), measure the noise level at the digital interface output or the reference codec decoder output over the frequency range of 100 to 3400 (or 4000) Hz with apparatus that includes psophometric weighting, according to ITU-T Recommendation Requirement The send noise shall be less than -68 dbm0p Handset Send Single Frequency Interference 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 Measurement Method In a quiet environment (ambient noise less than 30 dba), measure the psophometrically-weighted noise level at V SEND with a selective voltmeter or spectrum analyzer with an effective bandwidth of not more than 31 Hz, over the frequency range of 100 to 3500 (Glenn proposes changing to 3400 to be consistent with 269) Hz. If FFT analysis is used, then Flat Top windowing shall be employed Requirement The send single frequency interference shall be less than -78 dbm0p Handset Receive Noise General The receive noise of a digital telephone is the short-term average background noise level measured at the output of the telephone receiver with the digital telephone receiving the digital quiet code. 19

28 Measurement Method A signal corresponding to a decoder value quiet code is applied at the digital interface. The A- weighted noise level is measured in the artificial ear over the frequency range of 100 to 8500 Hz. The ambient noise for this measurement shall not exceed 30 dba. Telephone sets with adjustable receive levels shall be adjusted so that the RLR is as close as possible to the nominal RLR value when driven by quiet code Requirement The receive noise shall be less than 38 dba Handset Receive Single Frequency Interference 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 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 8100 (Glenn proposes changing to 8500 to be consistent with 269) Hz. If FFT analysis is used, then Flat Top windowing shall be employed. The ambient noise for this measurement shall not exceed 30 dba. Telephone sets with adjustable receive levels shall be adjusted so that the RLR is as close as possible to the nominal RLR value when driven by quiet code Requirement The receive single frequency interference shall be 10 db quieter than the A-weighted broadband noise floor. (Glenn proposes changing to an absolute value of 28 dba to be consistent with the send SFI requirement) 5.4. Handset Receive Comfort Noise (Advisory) If comfort noise is introduced to replace actual background noise the level should roughly match the loudness of the original background noise. There is more likely to be annoyance if the comfort noise is greater than the original noise than if it is less General The receive comfort noise of a digital telephone is the short-term average background noise level measured at the output of the telephone receiver with the digital telephone receiving either a silence indication packet from the transmitting telephone or no packets from the transmitting telephone for some non-transient period of time Measurement Method The receive volume control is adjusted as close as possible to the nominal RLR value. The digital interface is sent the quiet code the code that represents silence for the coder format. 20

29 With both VAD disabled at the transmitting source and comfort noise generation on the receiving unit under test turned off, a white noise test signal should be sent from the transmitting end such that the receive noise level measured at the receiving telephone is 48 dba. This test signal at this level will be assigned the level of N db as a calibrated point for the purpose of the comfort noise test, since it may be generated either as an acoustic signal at a golden transmitting telephone (and measured in dba) or injected digitally (and measured in dbm0p). The following test sequence must be followed for all calibrated test noise levels of M db, which will range from N-10 to N+10 db. 1. The echo canceller at both ends should be disabled seconds of silence (or idle code) is inserted at the transmitting point Hz band-limited white noise of level M db is inserted at the transmitting point for 130 seconds. 4. During the final 10 seconds of level M noise insertion, the acoustic noise level at the receive will be measured. 5. Steps 2-4 are repeated for varying M in 1 db gradations Requirement For all input noise levels M in the range of N-10 to N+10, with N calibrated to give 48 dba receive noise levels at the receiver, acoustic noise levels measured at the receiver must be within +0.5/-3.0 db of the expected acoustic receive noise level for that input. This expected receive noise level for any given M and N would be 48 dba (N-M) Handset Distortion and Noise The distortion and noise requirements only apply to G.711 codecs in mu-law Handset Send Distortion and Noise Method of Measurement The distortion and noise is measured according to IEEE 1329, Apply a sinewave signal at the MRP, with the levels given in Table 4 and the following frequencies: 315, 502, 803 and 1004 Hz. The ratio of the signal-to-total distortion and noise power of the digitally encoded signal output is measured. The test frequency tolerance is 3%, but even submultiples of the sampling frequency must not be used. Note: In cases where the sound pressure exceeds +6 dbpa, the linearity of the artificial mouth should be checked, as it exceeds the limits of ITU-T Recommendation P Requirement The ratio of signal-to-total distortion and noise power of the digitally encoded signal output shall be above the limits given in Table 4. Limits for intermediate levels are found by drawing straight lines between the breaking points in the table on a linear (db signal level) linear (db ratio) scale. 21

30 Table 4 Handset Send Signal-to-Total Distortion and Noise Ratio Limits Send level at the MRP (dbpa) Send Ratio (db) (Glenn proposes changing the 33 db limits to 31 db) Handset Receive Distortion and Noise Method of Measurement The distortion and noise is measured according to IEEE 1329, Apply a digitally simulated sinewave, with the levels given in Table 5 and the following frequencies: 315, 502, 803 and 1004 Hz. The ratio of signal-to-total distortion and noise power is measured. The test frequency tolerance is 3%, but even submultiples of the sampling frequency must not be used. Telephone sets with adjustable receive levels shall be adjusted so that the RLR is as close as possible to the nominal RLR value Requirement The ratio of signal-to-total distortion and noise power measured in the artificial ear, with A- weighting applied, shall be above the limits given in Table 5, unless the signal in the artificial ear exceeds +10 dbpa or is less than 50 dbpa. Table 5 Handset Receive Signal-to-Total Distortion and Noise Ratio Limits Receive level at the digital interface (dbm0) Receive 315 Hz (db) Receive 502, 803 and 1004 Hz (db) (Glenn proposes changing the 32 and 33 db limits to 31 db) 22

31 5.6. Weighted Terminal Coupling Loss (TCLw) SP RV2 (to become ANSI/TIA-810-B) The weighted terminal coupling loss (TCLw) provides a measure of the echo performance under normal conversation, i.e., single far-end talker conditions. It is possible that echo control devices such as echo suppressors or echo cancellation with non-linear processing may be used on handset connections to provide sufficient echo return loss to mitigate increased echo notice-ability associated with longer network delays. The use of echo control devices on the handset can affect the measurement of TCLw. The result would likely be different under cases of either single far-end talker or double-talk. The TCLw measurement is intended to represent a single far-end talker. This may provide idealized and unrealistic performance measurements when non-linear processing on the transmit side is used as part of the echo control algorithm. It may be more appropriate to measure TCLw either with 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 used consistently across products in a black-box testing setup. A suitable signal may be a pulsed sine wave, but will depend on the temporal characteristics of the double-talk detector. The proper measurement of TCLw then becomes specific to the echo control implementation. These issues are still under study and are not addressed in these requirements. For further information see IEEE 1329, Clause Free-Air Measurement Method TCLw is measured in free-air in such a way that the inherent mechanical coupling of the handset is not affected. The TCLw measurement shall be made at an input signal level of -16 dbm0. The test shall be performed with the handset suspended in a noose around the earcap with the handset cord hanging freely below the handset. For devices that incorporate non-linear processes, additional measurements using signal levels of -26 dbm0 and -10 dbm0 may be performed. Noise and reflections in the test space must not influence the measurement. The test should be performed in an anechoic chamber with the handset positioned at least 50 cm away from the nearest part of the test chamber. The ambient noise level shall be less than 30 dba. The test signal is white noise, band limited to 100 through 4000 Hz, and modulated at a rate of 250 ms ON and 150 ms OFF(Glenn proposes changing the stimulus to be the same as TIA-920 at 6, -10 and 16 dbm0). The measurement and calibration shall be determined during the ON portions of the signal. Sine wave signals may be used with G.711 codecs. The attenuation from digital input to digital output is measured at 1/12 octave frequencies as given by the R.40-series of preferred numbers in ISO 3 for frequencies from 290 to 3255 Hz, using the measurement arrangement shown in Figure 7. See Annex C. The weighted terminal coupling loss is calculated according to ITU-T Recommendation G.122 (1993) Annex B, Section B.4 (trapezoidal rule). Telephone sets with adjustable receive levels shall be tested at the nominal setting. For the nominal setting, adjust the level so that the RLR is as close as possible to the nominal RLR value. 23