[1] fundamental of telecommunic

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Fundamental of Telecommunication

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Switch mode

Circuit switch

Packet switch

Telephone communication

Connection-oriented

Data communication

Computer net communication

Newfangled IP Phone

Message switch

Store-and-forward switch

Telegram, FAX, Mails

X.25

Frame Relay

ATM

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SourceA C DB

Destination

Call Request Signal

Call Accept Signal

Talking

Acknowledgement Signal

Time

Circuit Switching

Time delay

Signaling process

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Telephone communication is Connection-oriented network,

and it needed Signaling system used to control the

connection and release of calls.

Before the talk, needed connection voice path signaling

process, and after the talk, needed release circuit signaling

process.

The advantage of the circuit switching, is the real time, and

the short transmission time delay, which make it suitable for

real-time communications, like voice communications.

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The disadvantages of the circuit switching is, the low

usage of circuit and long time in establishing circuit,

which make the mode unsuitable for data

communications with strong impulsiveness.

Duration of talk, because the dedicated path can not

share with other call, so that it is expensive.

p7

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Circuit switching refers to the switching mode that the switching equipment sets up a specially used circuit between the caller and the called before their communications, and the caller and the called occupy the circuit during the communication process until the communication is finished.

The advantage of the circuit switching is the real time and the short transmission time delay, which make it suitable for real-time communications like voice communications.

The disadvantages of the circuit switching is the low usage of circuit and long time in establishing circuit, which make the mode unsuitable for data communications with strong impulsiveness.

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SourceA C DB

Destination

Time

Message Switching

Time delayMSG

MSG

MSG

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In message switching, the transmitting side, need not

establish the circuit beforehand,

The basis for this switching is SAF (Store-And-Forward).

The switch can first store the received messages, and mails

in the buffer queue, and then calculate according to the

address information, in the mail heads to get the route. Once

the output line is determined and the line is idle, the stored

messages can be forwarded.

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The advantage: It can immediately send messages,

without waiting for the idleness of the receiving side.

Thus the circuits are very frequently used.

The disadvantage is the switch should be configured

with large-capacity memory, and the transmission time

delay is big and uncertain. Therefore, It is applicable only

for data transmission, but not for real-time

communications, for example voice communications.

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Message switching, called also as information switching, is used to switch the message information such as fax, mails, and text files. The basis for this switching is SAF (Store-And-Forward). In message switching, the transmitting side need not establish the circuit beforehand, but can send messages directly to the switching office of the receiving side at any time, no matter whether the receiving side is in the idle state.

The switch can first store the received messages and texts in the buffer queue and then calculate according to the address information in the text heads to get the route. Once the output line is determined and the line is idle, the stored messages can be forwarded. Switching equipment of the middle nodes of the telecom networks all adopt this mode for the receiving, storing and forwarding of texts until texts reach the destinations.

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It should be noted that in the text-switching network, a text is transmitted via only one path in the network, but different texts from the same source and to the same destination might be transmitted via different paths in the network.

The disadvantage of text switching is that the switch should be configured with large-capacity memory and the transmission time delay is big and uncertain. Therefore, this switching mode is applicable only for data transmission but not for real-time interactive communications, for example voice communications.

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2

A 3

4

5

a a

1

6

a aBA

(a) Text switching

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SourceA C DB

Destination

Time

Packetswitching

Time delayPKT1

PKT2

PKT3

PKT1

PKT2 PKT1

PKT3 PKT2

PKT3

Message

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Packet switching is the same as message switching, except a message is divided into a series of packets of limited lengths.

The advantage of packet switching, is the high rate in transmitting data. It enables better real-time than message switching. Packet switching enables interactive communications, (including voice communication) and makes high use of circuits. The transmission time delay of packet switching, is much less, than message switching, and its requirement for memory capacity, is also much less than message switching.

The disadvantage of packet switching, is the complicated handling process of node switches.

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In packet switching, a message will be divided into data packets of certain lengths, and each packet normally comprises hundreds or even thousands of bits. The packet data will be sent to packet switches together with addresses and suitable control information. As in text switching, the SAF technology is also adopted in the packets during packet switching.

The two switching modes differ in that the length of a packet is normally much less than the length of a text. In the switching network, packets of a text might reach the destination via different paths. And as the storage time delay of middle nodes is different from one another, the sequences of the arrivals of packets or the transmission of source nodes might be different.

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Therefore, after the packets reach the destination, they should be sequenced and de-packed before the correct data can be sent to users. Both text switching and packet switching adopt the error control technology called ARQ (Automatic Error Request) to handle the disturbance or other errors of the data when transmitted in the network.

The advantage of packet switching is the high rate in transmitting data. It enables better real-time than text switching. Packet switching enables interactive communications (including voice communication) and makes high use of circuits. The transmission time delay of packet switching is much less than text switching, and its requirement for memory capacity is also much less than text switching. The disadvantage of packet switching is the complicated handling process of node switches.

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2

1 3 X.25

4 3 2 1

6 4

5

22 2

2

1 1

B 4 3 2 1

a a

(b) Packet switching

443 4

3

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Overview of Telecom Network

and Digital SPC SwitchTelecommunication network is composed by the telecom terminals and service provision points connected to switches via the transmission system. Switch is the core or hinge of the telecommunication network.

Telecommunication network

telecom terminals

transmission system

switches

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Telecom Network Types and Topology Structure

According to service types

telephone networks

telegraph networks

fax networks

CATV networks

data networks

ISDN networks

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According to signal forms

analog networks

data networks

mixed networks

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telecom networks

bearer network

switching networks

supporting networks

According to the usage

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Manage network

Synchronous networks

Signaling network

Switching networks

Bearer network

Supporting networks

Basic network

OMC

NMC OMC

OMC

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According to network topology

meshed networks

star networks

compound networks

tree networks

chain networks

loop networks

bus networks

p3

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non-hierarchical networks

According to network levels

hierarchical networks

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(A big area)

((B big area)

Local network

Toll netwo

rk

International office

Level-1 switching center (C1)

Tandem office (TM)）Terminal office

(C5)

Backbone route

Low call loss route

High-efficiency direct route

Fig. 1.1.5-1 Structure of the telephone network in China

Level-2 switching center (C2)Level-3 switching center (C3)Level-4 switching center (C4)

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Numbering system Telephone switching is a process to implement link connection according to addressing signals (dialing tone, number, occupancy, ringing, etc.) so that signal channels will be set up between subscribers in the telephone-switching network.

To enable the switching system to correctly and effectively select routes and called terminals, a reasonable numbering system is necessary.

Basic requirements for the number system are: unified numbering globally, minimum digits, regular numbering and convenience for upgrading and expansion.

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(1) Numbering according to national toll telephone subscribers

Toll prefix +toll area code + local telephone number (office number + subscriber number)

(2) Numbering of international toll call

International toll prefix + country code + national toll code + local telephone number

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No. Area No. Area

1234

5

North AmericaAfrica Europe EuropeSouth America and Cuba

67890

South Pacific (Australia)

CISNorth Pacific (East

Asia)Far East and Middle East

Standby

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(3) Local network subscriber and local network subscriber numbering

PQ (R)+ABCD, P=2~9, the range of Q, R, A, B and C is 0~9

Local network subscriber calling outer-network national subscriber

0+X1X2+PQ (R) ABCD

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Local network subscriber calling international subscriber

00+I1I2+X1X2+PQ(R)ABCD

I1I2…indicates the country codes, and X1X2…indicates the national

area codes.

Special service numbering: 1XX. X=0~9. Ordinary special services are as listed in Table 1.1.5-2.

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No. Special

service No. Special service

110112114117119

Police Local call fault Local call directory Timing Fire alarm

120121170174168

Instant securityWeather forecastInternational toll

automatic call charge queryIntra-network toll number

directory Information console

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Transmission As the digital switching adopts the 4-wire switching mode, 4-wire loops will be adopted in local office connection, inter-office connection and toll connection.

In addition, the time delay of the digital switch in transmitting voice signals is longer than the analog switch, and the affects of echoes caused by 4-wire loops on the transmission quality should be taken into consideration.

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Fixed attenuation value: local network connection and toll connection use the same tone attenuation value

Digital local end office

7dB

7dB

12dB 7dB 3dB

22dB

Fixed attenuation value mode

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Variable attenuation value: local network connection and toll connection use respectively different attenuation values as shown in Figure (a) local network connection and (b) toll connection.

Local end

office

Toll office

7dB3.5dB

12dB 3dB

18.5dB

3.5dB

3.5dB 12dB 3dB

22dB

7dB

7dB

(a) Local (b) Toll connections

Variable attenuation value mode

Toll office

Local end

office

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Time Division Multiplexing (TDM)

When many signals are arranged in different “positions” in the same range (time, frequency/wavelength, space, energy or other ranges) according to certain rules, the process of transmission along a single bearer is called multiplexing.

Signals multiplexed in the originating terminal are transmitted to the receiving terminal via channels and then separated into the original individual signals.

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I=x2

x1

f1（x ·） f2（x）dx =0

The basis of multi-channel multiplexing is to use the orthogonality of signals. In mathematics, the orthogonality of signals can be expressed as:

0)2sin()2sin( 21 dttftf

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Multiplexing

FDM

frequency Division Multiplexing

TDM

Time Division Multiplexing

WDM

Wave Division Multiplexing

SDM

Space Division Multiplexing

CDM

Code Division Multiplexing

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f

code

f

code

f

codetime

time

time

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FDMA TDMA CDMA

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Digital SignalDigital Signal

DigitalDigitalsignalsignal

AnalogAnalogsignal

it is continuous or real numbers for time axes and amplitude axes

Digital signal is discrete for time axes and amplitude axes

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P23

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Multiplexing

Line coding

Digital transm

ission or sw

itch

Analog

Digital signals NRZHDB3

Line de-coding

De-m

ultiplexing

De-coding

Low

pass filtering

10

Digital signalsHDB3

NRZ

A/D

Low

pass filter

Sampling

Quantization

Coding

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The conversion of analog signals to binary digital signals comprises three handling processes, sampling, quantization, and coding, which are the same as in pulse code modulation (PCM).

Differential pulse code modulation(DPCM)

Adaptive Differential pulse code modulation(ADPCM)

Delte modulation (DM)

Adaptive Delte modulation (ADM)

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Sampling

Sampling—Discretion of Time

Sampling is to convert the analog signals with continuous time and amplitude into analog signals with discrete time and continuous amplitude.

These analog signals of the latter type are also called the pulse amplitude modulation (PAM) signals.

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Sampling waveSampling wave

+0.3 +0.2 -0.4 +0.1 0 +0.1 -0.4 +0.2 -0.5

Nq(tNq(t)) on-offon-off

Sam

plingS

ampling

Quantizatio

Quantizatio nn

Sp(t)Sp(t)

&(t)&(t)

S(t)S(t) Sq(t)Sq(t)

Sq(t)Sq(t)

S(t)S(t)

Sp(t)Sp(t)5.3 10.2 7.6 2.1 5.0 8.1 13.6 14.2 7.5

5 10 8 2 5 8 14 14 8

&(t)&(t)

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To ensure that PAM signals after sampling can be restored to the original signals without distortion in the receiving terminal, the sampling period should satisfy the Nyquist Theorem, which will be introduced hereafter.

Nyquist Theorem: Signal S(t) with the restricted frequency band of B Hz can be uniquely determined by sample value series with the TS= period if only fs≥2B.

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on-offon-off

F(t)Fs(t)=f(t).s(t)

S(t)

n

tjnn

seSts )(

2

2 )sin()(

1 2

2

s

ssT

sT

s

n

ntjn

n Tdtets

TS

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That is, to completely restore a signal from its sampling value without distortion, the sampling frequency must satisfy the following conditions:

fs≥2B(Hz), fs can also be called the Nyquist Frequency, or TS≤ a

nd Ts is called the Nyquist time interval.

In telephone communications, the voice frequency band is 300-3400Hz, and the actual sampling frequency fs is taken as 8000

Hz2B=23400Hz=6800Hz. This is to prevent the confusion of signals after sampling and enable protection zone in the spectrum.

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125μSt

A

Cycle T= 125μS f=8kHz

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AA

BB

CC

DD

AA

BB

CC

DD

第一帧第一帧第二帧第二帧

AABBCCDDAABBCCDD

同步同步时钟时钟

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AA

BB

CC

DD

AA

BB

CC

DD


AABBCCDDAABBCCDD


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AA

BB

CC

DD

AA

BB

CC

DD


AABBCCDDAABBCCDD


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D1(t)D1(t)

D2(t)D2(t)

D3(t)D3(t)

D4(t)D4(t)

DM(t)DM(t)

TDM

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Sampling is to convert the analog signals with continuous time and amplitude into analog signals with discrete time and continuous amplitude.

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Quantization Quantization is to discrete (or quantify) the continuous amplitude of sample values and convert the analog PAM signals with continuous amplitude into multi-system digital signals.

As ordinary digital communication systems and computers all adopt binary signals, multi-system digital signals are processed with binary coding to be converted in to binary digital signals.

As is described above, the sample value series after sampling are still analog PAM signals. To transmit the signals in the digital mode, the amplitude of PAM signals must be discrete.

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A 3.8 5.9 3.2 1.1 2.6 4.6 6.2 5.3 2.5

Q 3.5 5.5 3.5 1.5 2.5 4.5 6.5 5.5 2.5

0.3 0.4 -0.3 -0.4 0.1 0.1 -0.3 -0.2 0

S/N 12.6 14.8 3.6 2.8 26 2646 20.6 26.5

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the process of quantization is to round up the amplitude values of analog sampling signals. Obviously, the round up processing might cause certain errors, which is the so-called “Quantization Error”. The quantization error will ause some noises in human ears, which are normally called the quantization noise.

Quantization is normally of two types, even and non-even quantization.

Even quantization divides the quantization range evenly, i.e., quantization is implemented by adopting the equal quantization hierarchy distance.

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An important index affecting communication quality is the SNR (Signal-To-Noise Ratio).

For even quantization, SNR for small signals will be obviously more than that SNR for big signals.

This will cause the redundancy of SNR for big signals and shortage of SNR for small signals.

To overcome the problem with even quantization, non-even quantization is adopted in the quantization process of voice sampling in actual communications.

That is, different quantization distances are adopted for different signals to enable that small and big quantization distances can be adopted for small and big signals respectively. This will ensure similar SNR for big and small signals. The principle of realizing the non-even quantization is as shown in Fig.

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even quantization

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No-even quantization

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A/D D/AInput signal

Compressed digital bit

flow^Y（ t)Compres

s Even PCM coder

Even PCM decoder

Expand

Fig. 1.3.2-1 Compress and extend PCM transmission system

Output signalY

（ t)

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In this process, the realization of the non-even quantization in the transmitting side is to convert signal S(t) via a compressor with non-linear features.

This will expand small signals and compress big signals to get the compressed signals. These signals are then quantized via an even quantizer, which is equal to the non-even quantization for signals after sampling.

In the receiving side, signals after quantization are processed via an expander, which has the opposite features to the compressor. Small signals are compressed and big signals are expanded. The original PAM signals will be restored.

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It should be noted that the quantization process is a non-reversible process. That is, the quantization process will unavoidably introduce the above-mentioned quantization error, and the error will not be deleted via non-reversal.

Ordinary compression features are A-law (A=87.6) (adopted in Europe and China) and -law (=255) (adopted in North America and Japan). Both are logarithm compression laws.

For the Law companding rule,

Y= ln(1+x)

ln(1+) SGNx -1≤ x≤ 1

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SGNx =

1 X>0

0 X=0

-1 X<0

where SGN is the symbol value of x, i.e.,

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40

0

5

15

40

100

=255

Output

1.0

1.0

Input

Figure 1.3.2-2 µ-law characteristics

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For the companding law A, we have:

Y= 1+lnAx

1+lnA SGNx, 1A ≤ x≤ 1

Y=Ax

1+lnA SGNx, 0≤ x≤1A

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Output

1.0

1.0 Input

A=87.6

A=65

A=1

Figure 1.3.2-3 A-law characteristics

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Coding /decoding using Law A 13-broken-line method

The realization of the above continuous companding requires infinite quantizing levels, thus impossible.

Instead, usually the digital circuit segmenting is used to compand signals.

This is not only easier, but low in costs. Law A compressing uses the 13-broken-line method. Table 1.3.3-1 lists the slopes of each segment of the 13-broken-line method and the Law A compressing method.

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Y

17/8

6/8

5/8

4/8

1X

1/21/41/81/16

Figure 1.3.3-1 13-broken-line segment diagram

-

-

-

-

11

12

- -

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13-broken-line segment diagram

INput

OUTput

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broken.doc

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The 13- broken-line law A compressing coding rules:

1) A signal sample can be positive or negative, which shall be indicated by a bit. This bit is called the polarity bit. “1” indicates the positive polarity, and “0” indicates the negative.

2) The 13-line compressing law has 8 segments in phase I. All segments have different slopes, so 3 bits are needed to indicate 8 different segments, and they are called the segment bits. They also indicate the initial level of each segment.

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3) In each segment there are evenly distributed 16 sub-segments. As the lengths of segments are different from each other, after even distribution, the length of sub-segment of different segments is also different. Assuming that a division of the first segment is the minimum even quantizing a quantum value . Then in segments 1—8, each sub-segment shall have △1△ ，…， 64△ ， as shown in Table 1.3.3-2.

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No. for each segment

1 2 3 4 5 6 7 8

Each segment length

16 16 32 64 128 256 512 1024

Uniform quantizing level in each segment

△ △ 2△ 4△ 8△ 16△ 32△ 64△

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The code word format of each

voice signal sampling is as follows:

Polarity code Segment code Intra-segment code

D1 D2 D3 D4 D5 D6 D7 D8

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If input signals have a dynamic range from -2048mv—+2048mv, then the detailed table of ranges in each segment can be obtained as shown in Table 1.3.3-1. For instance, if the encoder input quantizing signal values are +135mv and -1250mv, then according to the encoding rule and Table 1.3.3-3,

we have their coding respectively as 1 100 0000 and 0 111 0011. There are many kinds of PCM encoders, but usually the step by step feedback comparison encoder is used.

table1.doc

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Decoding To restore digital signals into the original analog signals, digital signals should be decoded and filtered.

Decoding is the reverse process of coding.

It is to convert the received PCM coding signals into the quantization signals, which is as in the transmitting side.

This needs the calculation of the original quantization value (absolute value) according to the quantization distance value in correspondence to the field code in the code group as well as the section serial number value in correspondence to the 4-digit intra-field code.

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For example1 received code is

1 100 0000

Polarity code 1 represents positivesignal.

Field code 100 indicates in 4th section, that is 128

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The decoder outputs a staircase quantization signal, which can be converted smoothly into an analog signal by filtering the high-frequency weight.

The digitization of analog signals (or the analog/digital conversion (A/D)) and the reversed process (normally called the digital/analog (D/A) conversion) can be implemented according to the above procedures, and the A/D conversion is enabled by three processing modes simultaneously. With the development of large-scale integration technologies, the above processing (including A/D and D/A) are integrated on a special single-channel PCM transcoding chip, which can be Intel2914, TP3067 or MC145567.

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PCM primary format The PCM primary system is the basic system for digital multiplexing.

It consists of 30 voice channels, and its primary frame structure contains 32 time slots. 30 channel TS plus two TS for synchronization and signaling respectively.

TS0 normally serves as the frame synchronization TS and TS16 serves as the signaling TS.

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Signaling

timeslot

Frame length 125μs

Time slot

0 1-15 16 17-30 31

Time slot length 3.9μs

Frame synchronization word

Polarity code Segment code Intra-segment code

e 段内码

D1

D2

D3

D4

D5

D6

D7

D8

D1

D2

D3

D4

D5

D5

D7

D8

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As there is only 8bit signaling information in 30 channels, the multi-frame structure is normally adopted.

Each multi-frame is composed by 16 single frames. Each channel can be allocated with 4 information bits in 2ms, and the signaling rate will be 2Kb/s, which is the information bit of channel associated signaling in the 30/32PCM multi-frame structure.

The basic formats of multi-frame and single-frame of PCM30/32 are as shown in Fig. 1.3.4-1.

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16 frame,125 s 16=2ms

F0 F15

0 1 15 16 17 30 31

32TS,256bit,125s,1 frame

Reserved for the international

(now fixed as 1)

Synchronization TS

Channel TS TS1-TS15 Channel TS TS17-TS31

F0

Frame synchronizationcode

偶帧

D1 D2 D3 D4 D5 D6 D7 D8Even frame

Odd frame

Loss of frame opposite alarm

code

F1 a b c d a b c d

Channel 16 signaling code


1 1 A1 1 1 1 1 1

Reserved for the international

(now fixed as 1)

F15 a b c d a b c d

Synchronization: A1=0; A2=0Out of frame: A1=1; A2=1

Fig.1.3.4-1 Frame structure of PCM30/32

1 0 0 1 1 0 1 1 0 0 0 0 1 A2 1 1

Multi-frame synchronization code

Multi-frame synchronization code



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As is shown in the figure, in the 125s sampling period, each channel sends the 8bit voice code group for one time by turn, and each channel occupies one TS.

30 channels and the synchronization and signaling TSs form a single frame.

TS0 is used to transmit frame synchronization codes, and TS16 is used to transmit signaling codes of the channels (e.g. occupation, called subscriber pick-up, calling subscriber hang-up, forced disconnection, etc.).

In a single frame, featuring data of a PCM 30-channel system is as follows:

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In a single frame, the characteristic data of the 30-channel PCM system is as follows:

Voice frequency band 300-3400Hz

Sampling frequency 8000Hz

Frame cycle 125s

Coded bits for each sample value 8bit

Rate per channel 64kb/s

Time slot No. in each frame 32

Bits in each frame 256

Channels in each frame 30

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Duration of each time slot 3.9s

Bit duration 0.488s

Total data rate 2.048Mb/s

Compressing law Law A A=87.6

Signaling capacity

channel associated signaling 2Kb/s

common channel signaling 64Kb/s

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In a multi-frame, the 30-channel PCM system shall have the following characteristic data :

Multi-frame frequency 500Hz

Multi-frame cycle 2ms (0.125mS x 16=2mS)

Time slots in each multi-frame 3216=512

Bits in each multi-frame 25616=4096

512*8=4096

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Line coding

The main purposes of line coding are listed as follows:

To match sent signals well with the channel;

To easily extract the clock signal;

To eliminate DC, and both the high and low frequency components shall be as small as possible; and

To introduce error detection, and limit the error code increments.

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In a digital switch, there are two commonly used line codes: AMI(alternate mark inversion) and HDB3(3-order high density bi-polarity code).

Previous “V” code polarity

Numbers of “1” in last substitution

Even Odd

V+ 000V- B-00V-

V- 000V+ B+00V+

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AMI code waveform

Binary code

NRZ

AMI

t

t

1 0 0 0 1 0 1 1 0 0 0 0 1(a)

(b)

(c)

Average is 0 for AMI code

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When there are less than or equal to 3 “0”, they shall be converted in the AMI rule. That is, use 0 to indicate “0” ， and +1 or –1 to indicate “1” ， with +1 and –1 coming alternatively.

When four continuous “0” come out, they shall be replaced with B00V or 000V.

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图 1.3.5-3 AMI 码和 HDB 码的功率谱

1 0 0 0 0 1 0 1 0 0 0 0 0 0 1

原二进数据 +1 0 0 0 0 -1 0 +1 0 0 0 0 0 0

-1

t

AMI

图 1.3.5-2 HDB3 码编码波形

1.0

0.5 1.0fT

HDB3

AMI

p

HDB3

t +1 0 0 0 +1 -1 0 +1 -1 0 0 -1 0 0 +1

V码

B码

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码型示意图

0 1 0 0 1 0 0 0 0 0 1 0 0 1 0 0 0 0 1 1 1 0 0 0 0 0 1 0原码

NRZ 码

AMI 码

HDB3 码

0 0 0 V- B4 0 0 V+ 0 0 0 V-

negative

positive

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Primary groups consist of 30 PCM HWs of voice signals at the rate of 64Kb/s added by the digital multiplexer shown in Figure 1.2.3-1 with the synchronization and signaling information.

1CH64Kb/s

24CH1.554Mb/s

96CH6.321Mb/s

672CH44.736Mb/s

4032CH274.176

Mb/s

480CH32.064Mb/s

1440CH97.728Mb/s

5760CH397.2Mb/s

（ North America ）

（ Japan ）

30CH2.048Mb/s

120CH8.448Mb/s

480CH34.368Mb/s

1920CH139.264

Mb/s

7680CH565.992

Mb/s（ Europe, China ）

7 6

24

G.723 G.734

G.743G.746

30

5

4

3 4

4 4 4

G.752 G.752

G.732 G.742 G.751 G.751 （ order-5 group ）G.735 G.744 G.753 G.754G.736 G.745 （ order-3 group ）（ order-4 group ）G.737 （ order-2 group ）(order-1 group )

Figure 1.2.3-3 CCITT PDH structure

G.752

Order-0 group

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So the 30/32-PCM signals are often called the primary signals.

The primary multiplexer/demultiplexer consists of the voice frequency unit, the receiver/sender timing synchronization unit, the logical receiver/sender unit and the interface unit. The interface unit converts the 2.048Mb/s NRZ codes into the HDB3 codes for output. And the received HDB3 codes are converted into the NRZ codes and then sent to the receiving logic. The 2.048Mb/s(often called the E1 rate) primary frame format is shown in Figure 1.2.3-2.

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Besides the above primary (first order) groups, there are even higher order digital multiplexing groups: second order, third order, and fourth order groups.

In these groups, each tributary signal comes from a different signal source controlled by a different crystal oscillator. In the transmission network, the clock at each node is independent from others.

In the transmission network, the clock at each node is independent from others. The accuracy of frequencies can have a small deviation within a given nominal value range. Though they have the same nominal bit rate, they are not accurately synchronized.

So they are called quasi-synchronization signals.

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PDHThe digital multiplexing groups they have formed are called the plesiochronous digital hierarchy (PDH).

The PDH structure recommended by CCITT is as shown in Figure 1.2.3-3.

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140Mb/sLine

terminating equipment

140

34

34

8

8 8

2 2

34Mb/s

8Mb/s

2Mb/s

Figure 1.2.3-4 CCITT PDH ADM

2Mb/s

140Mb/sLine

terminating equipment

140

34

34

8

E2/E1

E3/E2

E4/E3

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Besides, the capacity of network management is very limited. Finally, as the European and American systems are not compatible, this brings a lot of inconvenience to international interconnection.

To solve the above problem, America has developed the synchronization fiber optic network (SONET). On basis of SONET, CCITT has laid down new standards for the synchronous digital hierarchy (SDH) in 1988, and published the recommendations G.707, G.708 and G.709. In 1990, a revision was made on the 1988 version.

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SDH is based on the complete network clock synchronization. Its transmission capacity is divided into the two parts of bearer and overhead. The main purpose of overhead is for network management and bearer alignment.

The first level of SDH (corresponding to the synchronization transfer mode STM-1) has a bit rate of 155.520Mb/s. The two levels stipulated below are respectively STM-4(622.080Mb/s) and STM-16(2488.32Mb/s), which have reflected the achievements and future trends of the rapid development of the high bit rate and large capacity fiber optic communication.

[1] fundamental of telecommunic

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