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IP/VoIP Analysis & Simulation

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Protocol Analyzer

Products
IP/VoIP Analysis & Simulation
Voice Codecs

Voice Codecs

GL Communications products support a variety of signaling and audio processing applications in both VoIP and TDM. Using these
tools, one can emulate, analyze, and troubleshoot audio signaling over both VoIP and TDM. Each of these tools support the
following narrow-band, and wideband (HD audio) codec standards:

VAD Packetisation
Codec Data Rate Sampling Rate
Supported Time (Ptime)
G.711 (PCM µ-law/A-law) 64 kbps 8000 No Multiples of 10 ms
G.711 App II (PCM µ-law/A-law with
64 kbps 8000 Yes Multiples of 10 ms
VAD)
G.722 64 kbps 16000 No Multiples of 10 ms
24 kbps
G.722.1 (Wideband) 16000 No Multiples of 10 ms
32 kbps
G.729 8 kbps 8000 No Multiples of 10 ms
G.729B 8 kbps 8000 Yes Multiples of 10 ms
GSM 6.10 FR 13.2 kbps 8000 No Multiples of 20 ms
Fixed at 20 ms. Multiple
GSM EFR 12.2 kbps 8000 Yes
Ptime Not Supported
GSM HR 5.6 kbps 8000 Yes Multiples of 20 ms
5 bit 40 kbps
4 bit 32 kbps
G.726 8000 No Multiples of 10 ms
3 bit 24 kbps
2 bit 16 kbps
5 bit 40 kbps
4 bit 32 kbps
G.726 (with VAD) 8000 Yes Multiples of 10 ms
3 bit 24 kbps
2 bit 16 kbps
4.75 kbps
5.15 kbps
5.9 kbps
AMR 6.7 kbps
8000 Yes Multiples of 20 ms
(requires additional license) 7.4 kbps
7.95 kbps
10.2 kbps
12.2 kbps
6.60 kbps
8.85 kbps
12.65 kbps
14.25 kbps
AMR WB (Wideband)
15.85 kbps 16000 Yes Multiples of 20 ms
(requires additional license)
18.25 kbps
19.85 kbps
23.05 kbps
23.85 kbps
EVRC, EVRC0 EVRC Rates -
8000 No Multiples of 20 ms
(requires additional license) 1/8, 1/2 and 1
EVRCB Rates
EVRCB , EVRCB0
- 1/8, 1/4, 1/2 8000 Yes Multiples of 20 ms
(requires additional license)
and 1
EVRC_C (Wideband)
16000 Yes Multiples of 20 ms
(requires additional license)
SMV Modes - 0, 1,
8000 No Multiples of 20 ms
×
(requires additional license) 2 and 3
15.2 kbps Multiples of 20 ms
iLBC 8000 No
13.33 kbps Multiples of 30 ms
Fixed at 20 ms. Multiple
SPEEX (Narrow Band) 8 kbps 8000 Yes
Ptime Not Supported
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SPEEX (Wideband) 11.2 kbps 16000 Yes Fixed at 20 ms. Multiple
Ptime Not Supported
EVS (Narrow Band) 5.9 kbps 8000 Yes 20 ms to 100ms in multiples
(requires additional license) 7.2 kbps of 20msec
8 kbps
9.6 kbps
13.2 kbps
16.4 kbps
24.4 kbps
EVS (Wideband) 5.9 kbps 16000 Yes 20 ms to 100ms in multiples
(requires additional license) 7.2 kbps of 20msec
8 kbps
9.6 kbps
13.2 kbps
16.4 kbps
24.4 kbps
32 kbps
48 kbps
64 kbps
96 kbps
128 kbps
EVS (Super Wideband) 9.6 kbps 32000 Yes 20 ms to 100ms in multiples
(requires additional license) 13.2 kbps of 20msec
16.4 kbps
24.4 kbps
32 kbps
48 kbps
64 kbps
96 kbps
128 kbps
EVS (Full Band) 16.4 kbps 48000 Yes 20 ms to 100ms in multiples
(requires additional license) 24.4 kbps of 20msec
32 kbps
48 kbps
64 kbps
96 kbps
128 kbps
OPUS (Narrow Band) 6 to 128 kbps 8000 Yes Fixed at 20 ms
(requires additional license)
OPUS (Middle Band) 6 to 128 kbps 12000 Yes Fixed at 20 ms
(requires additional license)
OPUS (Wideband) 6 to 128 kbps 16000 Yes Fixed at 20 ms
(requires additional license)
OPUS (Super Wideband) 6 to 128 kbps 24000 Yes Fixed at 20 ms
(requires additional license)
OPUS (Full Band) 6 to 128 kbps 48000 Yes Fixed at 20 ms
(requires additional license)

µ-Law, A-law (G.711)

PCM has been the standard for digital voice transmission in telephony since 1972. It captures speech in a range of 300 to 3.4
kHz, samples at 8000 samples/second with 8 bits per sample resulting in 64 kbps with an encoding frame length of 10 ms.
The two algorithms defined in the standard are µ-Law (North America & Japan) and A-law (used in Europe and the rest of the
world). Both are logarithmic, but A-law was specifically designed to be simpler for a computer to process.

G.722, G.722.1
G.722[1] is a ITU-T 16 kHz (with 14 bits per sample) wideband speech codec standard operating at 48, 56 and 64 kbps with
an encoding frame length of 10 ms. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). G.722 sample
audio data at a rate of 16 kHz (using 14 bits) with an encoding frame length of 10 ms, double that of traditional telephony
interfaces, which results in superior audio quality and clarity.

G.729, G.729B ×
G.729 operates at a bit rate of 8 kbps with an encoding frame length of 10 ms and 5 ms look ahead, but there are extensions,
commonly designated as G.729a and G.729b. Annex A and Annex B Voice encoding using CS-ACELP (Conjugate-Structure
Algebraic Code Excited Linear Prediction) 8 kbps, is the lowest bit rate ITU-T standard with toll quality. Annex A is a low-
complexity version of the G.729 standard. Annex B defines VAD/CNG/DTX (Voice Activity Detection/Comfort Noise
Generator/Discontinuous Transmission) for G.729 and G.729A.
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GSM-FR
GSM-FR is a Full Rate speech coder standardized by the European Telecommunications Standards Institute (ETSI) for
compressing toll quality speech (8000 samples / second) and was the first digital speech coding standard used in GSM digital
mobile phone systems. The coder has a bit rate of 13 kbps with an encoding frame length of 20 ms.

This coder uses the principle of Regular Pulse Excitation-Long Term Prediction-Linear Predictive coding. The coder works on
a frame of 160 speech samples with an encoding frame length of 20 ms, and no look ahead is required.

GSM EFR
GSM-EFR (6.60) is an improved and hence the Extended version of GSM-FR (6.10) codec. With sampling frequency of 8000
samples/sec and frame size of 31 bytes it achieves the bit rate of 12.2kbps with an encoding fixed frame length of 20 ms.
Codec supports Voice Activity Detection (VAD) to allow saving of bandwidth.

GSM HR
GSM HR 6.20 operates with sampling frequency of 8000 samples/sec. This codec outputs the frames of size 14 Bytes, which
puts the bit rate of encoder at 5.6kbps with an encoding frame length of 20 ms. Codec supports Voice Activity Detection (VAD)
to allow saving of bandwidth.

G.726 (ADPCM)
This is an ADPCM (Adaptive Differential Pulse Code Modulation). Originally, a half-rate alternative to ITU-T G.711 and
includes both the G.721 and G.723 standards. G.726 compresses by converting between linear, A-law (used in Europe) or µ-
Law (used in the U.S and Japan) PCM and 40, 32, 24 or 16 kbps with an encoding frame length of 10 ms.

G.726 with Voice Activity Detection (ADPCM)

This is an ITU-T Adaptive differential pulse code modulation (ADPCM) voice codec, which transmits at bit rates of 16, 24, 32,
and 40 kbps with an encoding frame length of 10 ms. It supports Voice Activity detection and generates SID packets during
Silence Period. ADPCM provides the following functionality:
Voice mail recording and playback, which is a requirement for Internet voice mail.
Voice transport for cellular, wireless, and cable markets.
High voice quality voice transport at 32 kbps.

AMR NB and WB (requires additional license)

AMR is the 3GPP mandatory standard codec for narrowband speech and multimedia messaging services over GSM and
evolved GSM (WCDMA, GPRS and EDGE) networks. Designed to provide transcoder free connectivity between GSM, US-
TDMA and Personal Digital Cellular networks (Japan).

AMR operates at eight bit rates in the range of 4.75 to 12.2 kbps with an encoding frame length of 20 ms and was specifically
designed to improve link robustness.

AMR-WB provides improved speech quality because of a wider speech bandwidth that is of 50–7000 Hz compared to
narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz.

Enhanced Variable Rate Codec (EVRC), EVRC-B, EVRC-C

For EVRC codec type three rates are provided (1/8, ½ and 1 with an encoding frame length of 20 ms). Default 1/8 and 1 are
selected as the minimum rate & maximum rate. Minimum rate should be less than or equal to maximum rate. There is option
to select RTP packet format between Header Free Format and Bundled Format. By default, Bundled Format is set.

EVRC-B is an enhancement to EVRC. EVRCB codec type compresses each 20 milliseconds of 8000 Hz, 16-bit sampled
speech input into output frames of one of the four rates:(1/8- 16 bits, ¼- 40 bits, ½- 80 bits, and 1- 171 bits with an encoding
frame length of 20 ms). By default, 1/8 and 1 are selected as the minimum rate & maximum rate. There is option to select RTP
packet format between Header Free Format and Bundled Format. By default, Bundled Format is set.

Important enhancement in EVRC-B is the use of 1/4 rate frames that were not used in EVRC. This provides lower average
data rates (ADRs) compared to EVRC, for a given voice quality.

EVRC-C adds the feature of encoding wideband signals sampled at 16 kHz with signal bandwidth up to 7 kHz.

Enhanced Voice Services Codec (EVS)

EVS provides vastly improved voice quality, network capacity and advanced features for voice services over LTE and other
radio access technologies standardized by 3GPP. It is the first 3GPP conversational codec providing up to 20 kHz audio
bandwidth, offering speech quality that of highest standard.

EVS codec includes a multi-rate audio codec, a source controlled variable bit-rate (SC-VBR) scheme, a VAD, a comfort noise×
generation (CNG) system, and an error concealment (EC) mechanism to offset the effects of transmission errors resulting in
lost packets. Its channel-aware mode feature further improves frame/packet error resilience.

OPUS
The Opus codec scales from 6 kbit/s narrowband mono speech to 510 kbit/s fullbandLive
stereo music. Supports both constant
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 bitrate (CBR) and variable bitrate (VBR). Provides audio bandwidth such as Narrow Band (8 kHz), Middle Band (12 kHz),
Wideband (16 kHz), Super Wideband (24 kHz), and Full Band (48 kHz).

SMV
The Selectable Mode Vocoder (SMV) [2] compresses each 20 milliseconds of 8000 Hz, 16-bit sampled speech input into
output frames of one of the four different sizes: Rate 1 (171 bits), Rate 1/2 (80 bits), Rate 1/4 (40 bits), or Rate 1/8 (16 bits)
with an encoding frame length of 20 ms. SMV is the preferred speech codec standard for CDMA2000, and will be deployed in
third generation handsets.

SPEEX NB and WB
SPEEX NB is based on CELP Narrowband (8 kHz with an encoding fixed frame length of 20 ms), open source codec
specifically used for VoIP and file-based applications

SPEEX WB Codec has a sampling rate of 16000 samples/sec with an encoding fixed frame length of 20 ms, which makes it a
wide band codec. This codec supports different codec options such as Sampling Rate, Variable Bit Rate, Voice Activity
Detection and Perceptional Enhancement.

iLBC Codec
iLBC (internet Low Bitrate Codec) is a narrow band speech codec that operates at either 13.33 kbps with an encoding frame
length of 30 ms or 15.20 kbps with an encoding length of 20 ms. Companies that are using iLBC in their commercial products
include:
Applications/Soft phones: Skype, Nortel, Webex, Hotsip, Marratech, Gatelinx, K-Phone, XTen;
IP Phones: WorldGate, Grandstream, Pingtel;
Chip: Audiocodes, TI Telogy, LeadTek, Mindspeed.

The 13.33 kbps rate 30ms frame encodes packets of 399 bits, (50 bytes) and is designated in RTP Toolbox as iLBC_13_33.

The 15.2 kbps 20 ms frame creates packets of 303 bits, (38 bytes). This is labeled iLBC in RTP Toolbox. The basic quality is
higher than G.729A.

Some Definitions:

Number of bits per second which needs to be transmitted to deliver a voice call. (codec bit rate =
codec sample size / codec sample interval).
Codec Bit Rate (Kbps)
We can calculate bit-rate as follows: For G.711 – 64 kbps = (160 bytes * 8 bits) * (1/20 ms)
For G.729 – 8 kbps = (20 bytes * 8 bits) * (1/20 ms)
Voice Payload in Bytes The voice payload size represents the number of bytes (or bits) that are filled into a packet.
PPS represents the number of packets that need to be transmitted every second in order to deliver
PPS the codec bit rate. To retrieve the PPS you can just do 1/(voice payload in ms). For Example, 50
PPS=(1/20 ms), 33 PPS = (1/33 ms)
Quality score based on various end point and network parameters. Includes codecs, packet loss, and
R-Factor
delay.
Conversational The voice quality metric that measures voice quality based on transmission delay, burst packet loss,
R-Factor and burst loss recency.
Listening R-Factor The voice quality metric based only on burst packet loss and codec selection.
Mean Opinion Score based on listening quality. Does not consider recency or delay. ITU-T P.862
MOS-LQ
Listening Quality implementations.
MOS-CQ Mean Opinion Score based on conversational quality. Includes recency and delay effects.
MOS-PQ ITU-T P.862 normalized raw quality score.
MOS-Nom Nominal quality or maximum score for the codec selected. Similar to the G.107 E-model defaults
A time factor used to weight scores based on the time from a burst packet loss to the end of the call
Recency
or next packet loss event.

Estimating Speech Quality

The mean opinion score (MOS) is a commonly used method to determine the quality of speech. Every codec has a certain speech
quality characteristics. With MOS, the quality of a speech is rated on a scale of 1 (bad) to 5 (excellent).

Recently. the speech quality estimates are based on the ITU G.107 E Model. These models considered the entire Ear-Mouth path ×
and all relevant conditions such as end-to-end level, echo, side tone, and frequency characteristics of the various path segments.

The E Model uses a computational method that includes factors such as noise, signal level, loudness ratings, impairments, delay,
codec type, and even network type to derive a quality score. This transmission quality rating is called as the ‘R’ factor. Over time
and based on experience with subjective and objective measurements, the E Model's R-Factor score was mapped to an
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equivalent Mean Opinion Score (Excellent to Bad) to predict the quality of the “mouth to ear” (M2E) speech path. Scoring
includes consideration for the type of subjective test used for scoring. Passive/listening or active/conversational tests produce
slightly different scores.

For IP networks, the score assumes ideal conditions outside the IP cloud and bases the scores on the relevant IP impairments such
as packet loss, latency, jitter, and even when these impairments occur over the duration of the call.

Codec Name Nominal MOS Nominal R factor

G.711 µ-Law (64 kbps) 4.2 93
G.711 A-law (64 kbps) 4.2 93
G.722 (64 kbps) 3.91 96
G.722.1 (32kbps) 4.09 102
G.722.1 (24 kbps) 3.98 98
G.729A/G.729AB (8 kbps) 3.91 82
GSM-FR 3.57 73
GSM HR 3.53 72
GSM EFR 4.16 91
G.726-40k 4.16 91
G.726-32k 4.04 86
G.726-24k 3.35 68
G.726-16k 2.82 57
G.726-40k with VAD 4.16 91
G.726-32k with VAD 4.04 86
G.726-24k with VAD 3.35 68
G.726-16k with VAD 2.82 57
AMR NB Mode 0 (4.75k)
3.65 75
(requires additional license)
AMR NB Mode 1 (5.15) 3.73 77
AMR NB Mode 2 (5.9) 3.84 80
AMR NB Mode 3 (6.7) 3.95 83
AMR NB Mode 4 (7.4) 3.98 84
AMR NB Mode 5 (7.95) 4.04 86
AMR NB Mode 6 (10.2) 4.11 89
AMR NB Mode 7 (12.2) 4.16 91
AMR WB Mode 0 (6.6k)
3.39 81
(requires additional license)
AMR WB Mode 1 (8.85) 3.81 92
AMR WB Mode 2 (12.65) 4.04 100
AMR WB Mode 3 (14.25) 4.09 103
AMR WB Mode 4 (15.85) 4.11 104
AMR WB Mode 5 (18.25) 4.14 105
AMR WB Mode 6 (19.85) 4.18 107
AMR WB Mode 7 (23.05) 4.18 107
AMR WB Mode 8 (23.85) 4.18 107
EVRC
3.94 83
(requires additional license)
EVRCB
3.98 84
(requires additional license)
SMV 3.88 81
Speex WB (11.2 kbps) 4.16 106
Speex NB 4.16 91
iLBC 13.3k 3.88 81
iLBC 15.2k 3.95 83
×
GL’s Applications supporting Speech Codecs
Capture and Analysis Products Simulation Products
ISDN Triggered Call Capture & Analysis Application (over TDM) RTP ToolBox™
ISDN PRI Analyzer PacketGen™
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Multi-Channel Audio Bridge MAPS™ MGCP, MAPS™ MEGACO
Spectral Analysis and Oscilloscope Display MAPS™ SIP, MAPS™ SIP I
Voice Band Analyzer (VBA) MAPS™ SIGTRAN, MAPS™ ISDN SIGTRAN
PacketScan™ (over IP) MAPS™ BICC IP, MAPS™ CAMEL IP
VQT Analysis for NB, WB, and SWB speech VQuad™ SIP

Buyer's Guide
Item No. Item Description
PCD103 Optional Codec – AMR – Narrowband (requires additional license)
PCD104 Optional Codec - EVRC (requires additional license)
PCD105 Optional Codec – EVRC-B (requires additional license)
PCD106 Optional Codec – EVRC-C (requires additional license)
PCD107 Optional Codec – AMR - Wideband (requires additional license)
PCD108 Optional Codec - EVS (requires additional license)
PCD109 Optional Codec - Opus (requires additional license)
Related Software
PKV100 PacketScan™ (Online and Offline)
PKV120 PacketScan™ HD – includes PKV100 – Online (not Offline) for temporary audio codec support
PKB100 RTP ToolBox™ Application
PKS100 PacketGen™ with PacketScan™
PKS101 SIP Core (additional)
PKS102 RTP Soft Core for RTP Traffic Generation (additional)
PKS103 RTP IuUP Softcore
PKS107 RTP EUROCAE ED137
PKS106 RTP Video Traffic Generation
PKS108 RTP Voice Quality Measurements
PKS120 MAPS™ SIP Emulator
VQT010 VQuad™ Software (Stand Alone)
VQT013 VQuad™ with SIP (VoIP) Call Control

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