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8/16/2002 1 Digital Transmission Key Learning Points • Fundamentals of Voice Digitization • Pulse Code Modulation • Quantification Noise • Multiplexed Digital Lines

8/16/20021 Digital Transmission Key Learning Points Fundamentals of Voice Digitization Pulse Code Modulation Quantification Noise Multiplexed Digital Lines

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8/16/2002

1

Digital Transmission

Key Learning Points

• Fundamentals of Voice Digitization

• Pulse Code Modulation

• Quantification Noise

• Multiplexed Digital Lines

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2.5.2 Digital Leased Circuits (from public carriers)

- supports high level of inter-site traffic, generally more expensive than modem based service

- provides direct digital connection between DTE’s- basis of most private data & voice networks

goal: understand organization & capacity of digital

networks

Digital switching & transmission for voice & data used in most public carrier networks

• Eliminates Need for Modem – Voice data must be ‘digitized’

• ISDN: Network that allows Transmission of voice & data

• Public Carriers leased digital circuit rates from kbps..100’sMbps - digital circuits must co-exist with other circuits for inter-

change traffic

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Voice Digitization• Voice signals are inherently analog • Spectral Content of Voice 4000Hz (except ) • Requires Analog To Digital Signal Conversion (ADC)

Nyquist Sampling Theorem: - Must Sample Twice Highest Frequency Component- Sample Rate for Analog Voice Signal = 8000Hz - Sampling Interval for Voice Signal = 0.125 ms

0 T 2T 3T 4T 5T

Amplitude

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(1) Pulse Amplitude Modulation (PAM):

• Analog Voice Signal Sampled converted to pulse stream

• Pulse Amplitude: discrete analog signal, amplitude = continuous analog signal

(2) Pulse Code Modulation (PCM): Quantize each Pulse into Binary Form

- 8 bits used to quantize pulse range = 0..256 levels

- R = 8 bits * 8000Hz = 64kbps per voice channel

- minimum unit of capacity available for lease

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8 bits

125 us

sample clock

PAM signal

PCM signal

Digital Voice Signal

sampling circuit

Quantizer & Compander

sampleclock

PCM Signal

PAMsignal

Analog Voice Signal

10001010 10111010

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q = quantization interval: signal (voltage) difference between adjacent discrete signal levels

- accuracy determined by number of bits in signal: n bits 2n levels- signals within a level are represented by same binary codeword- one bit may be used for signal polarity (+ or - )

e.g. n = 3 8 levelsrange 8volts (16 volts)interval q = 16/8 = 2 volts

0 110 100 010 001 001 011 101 11

binary codeword

analogvoltage range

6..8 volts4..6 volts2..4 volts0..2 volts0..-2volts-2..-4 volts-4..-6 volts-6..-8volts

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Each codeword corresponds to nominal input voltage centered at q • actual input may differ by q/2

• quantization error, , actual signal amplitude – quantized signal amplitude

• quantization noise: random quantization error variance between samples

0 110 100 010 001 001 011 101 11

q

V

-V

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actual amplitude = quantization error

+q/2

0

-q/2

-

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analog voltage

discrete voltage

% maximum error

binary codeword

6..8 7 1 14% 011

4..6 5 1 20% 010

2..4 3 1 33% 001

0..2 1 1 100% 000

0..-2 -1 1 100% 100

-2..-4 -3 1 33% 101

-4..-6 -5 1 20% 110

-6..-8 -7 1 14% 111

e.g. analog voltage range = 8 volts, n = 3• 8 signal levels q = 2 volts• smaller amplitudes more sensitive to • ear is sensitive to noise on quiet, low amplitude speech signals

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Practically non-linear PCM used to overcome quantization noise

• 2 level digitization: segment level and quantization level

• range of input signal amplitudes associated with each

quantization interval

• input signal amplitude increases corresponding code words represent larger signal range

vovi compressor expanderA/D D/ANetwork

•continuous analog signal passed into compressor then into A/D•expander reverses the operation performed at output of D/A

Compression and Expansion

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At transmitter: analog voice non-linearly encoded into binary data

1. compressor stage: analog input signal compressed - encoded value depends on segment level

2.ADC stage: compressed analog signal is digitized &linearly quantized

At receiver codewords converted to analog voice signal

1. DAC stage: compressed digital signal is linearly converted to analog signal

2. expander stage: analog output passed through expander – reverses compressor operation

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e.g. Let Signal Range ± 30 volts and n = 5 bits

32 total levels divided into• 1 polarity level• 2 segment levels• 2 quantum levels

signal polarity segmentcode

segment size

quantamcode

q

7 < S 15

1 11 8 V 00-11 2

3 < S 7 1 10 4 V 00-11 11 < S 3 1 01 2 V 00-11 0.50 S 1 1 00 1 V 00-11 0.25

% 1 12.5

0.5 12.50.25 12.50.125 12.5

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polaritysegment

levelquantam

levelinput range

output range

1 11

11 14.0-16.0 1510 12.0-14.0 1301 10.0-12.0 1100 8.00-10.00 9

1 10

11 6.50-8.00 710 5.50-6.50 601 4.50-5.50 500 3.50-4.50 4

1 01

11 2.75-3.50 3.010 2.25-2.75 2.501 1.75-2.25 2.000 1.25-1.75 1.5

1 00

11 0.875-1.25 1.010 0.625-0.875 0.7501 0.375-0.625 0.5000 0.00-0.375 0.25

• ‘+’ signal encoding• similar for ‘-’signal

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0

1

11

10

01

00

00

01

10

11

polaritybit

segment levels

quantum levels

V-16 -8 -4 -2

2 4 8 16-V

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PCM codecs (coder/decoder)• older codecs operated as above• newer codecs use 2 digital compression/expansion techniques

u-law: (N. America, Japan)A-law: (ITU-T)

- similar in principal to companding-expansion- conversion needed when using leased & switched circuits that span continents

- necessary only for voice

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Multiplexing (MUX)

Link Exchange Circuits: T1, T3, E1…- carry multiple calls concurrently- TDM Used: multiple digital signals assigned time slices

voice data: 8 bit sample @ 125us = 64kbps/ voice channel

control overhead: (i) start of frame (frame synchronization)(ii) call set-up (signaling)

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DS1 or T1 Links:

• 24 voice channels grouped 1.536Mbps (North America)

• (1 frame/125us 24 slots) = 192 bits/125 us

• 192 bits + 1 framing bit = 193 bits/125 us 1.544Mbps

Signaling Info: carried in 1st bit of time slots 6-12 - leaves 7 bits for data

Frame synchronization: bit (framing bit) at start of ‘frame 1’ - toggles from 1,0 for consecutive frames

•slots 6,12: 1 signal bit, 7 data bits 56 kbps•slots 7-11, 13-24: 8 data bits 64 kbps

slot 23 slot 23 … slot 1 slot 0 frame bit

125 us

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clock=8 KHz

64Kbps links digital links

01..

23

23 …. 1,0

193 bits

23 …. 1,0

synch bits

slots

DS1 or T1 Link

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E1 Link: (ITU-T)

• 30 voice channels at 64Kbps 1.920 Mbps

• two additional slots for signaling and control

32 (8/125us) = 2.048 Mbps

• Signaling info: carried in time slot 16• Frame synchronization: time slot 0

- used for frame alignment

- allows receiver to interpret time slots in each frame on aligned boundaries

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Fractional T1, E1: Lower Bit Rates on T1, E1 systems

Higher Aggregate Link Rates: MUX several groups (DSxx, Ey)• Higher order mux circuits: known as

- plesiochronous (nearly synchronous) - asynchronous

• PDH: (plesiochronous digital hierarchy) results in higher-order mux rates• higher bit rate links require additional bits for framing & control

Link Type 64Kbps channels

payload rate total rate including control data

E3 = 16 E1 480 30.72Mbps 34.368MbpsT3 = 28 T1 672 43.008Mbps 44.736 Mbps

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clock=8 KHz

64Kbps links

01..

23

T1 = 2464Kbps links + control

T1 link 0

01..

23

T1 link 1

01..

23

T1 link 27 T3 = 28T1 links + control

clock=8 KHzT3 link

justification bit

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Leased Line Interconnection

channel service unit (CSU): •electrical barrier •keep alive signal•loopback test

data service unit (DSU) translate data format between entities•T1 uses TDM DSX frames for Data•LAN serial data frame format (e.g. ethernet)•physical connector to LAN

Public CarrierNetworkCSU/DSU CSU/DSU

CSU/DSUrouter

hubCSU/DSU router

hubT3 lines