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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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Source

A C D B

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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Source

A C D B

Destination

Time
Message
Switching
Time delay
MSG
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
a
B
A
(a) Text switching
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Source

A C D B

Destination

Time
Packet
switching
Time delay PKT1
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
2
2
2
2
1
1
B
4 3 2 1
a
a
(b) Packet switching
4
4
3
4
3
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Overview of Telecom Network
and Digital SPC Switch

Telecommunication 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

1
2
3
4
5

North
America
Africa
Europe
Europe
South
America
and Cuba

6
7
8
9
0

South Pacific
(Australia)
CIS
North 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+X
1
X
2
..+PQ (R) ABCD
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Local network subscriber calling international subscriber

00+I
1
I
2
.+X
1
X
2
.+PQ(R)ABCD

I
1
I
2
indicates the country codes, and X
1
X
2
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

110
112
114
117
119

Police
Local call
fault
Local call
directory
Timing
Fire
alarm

120
121
170
174
168

Instant security
Weather forecast
International toll
automatic call charge query
Intra-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
W
W
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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

7dB
3.5dB
12dB
3dB
18.5dB
3.5dB
3.5dB
12dB
3dB
22dB
7dB
7dB
(a) Local
(b) Toll connections
Variable attenuation value mode
W W
W
W
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=
]
(
x
2
x
1
f
1
xf
2
xdx =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 ) 2 sin( ) 2 sin(
2 1
=
}
dt t f t f t t
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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
code
time
time
time
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FDMA TDMA CDMA
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2013/12/17
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2013/12/17
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Digital Signal


Digital
signal
Analog
signal
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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2013/12/17
Mul
tipl
exin
g
Li
ne
co
din
g

Digit
al
trans
missi
on or
switc
h
Anal
og
Digital signals

NRZH
DB3
Lin
e
de-
co
din
g
De-
mul
tipl
exi
ng
De-
cod
ing
Lo
w
pas
s
filt
eri
ng
1
0
Digital
signals
HDB3
NRZ
A/D
L
o
w

p
a
s
s

f
i
l
t
e
r

S
a
m
p
l
i
n
g


Q
u
a
n
t
i
z
a
t
i
o
n

C
o
d
i
n
g


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2013/12/17
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
SamplingDiscretion 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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2013/12/17
Sampling wave
+0.3 +0.2 -0.4 +0.1 0 +0.1 -0.4 +0.2 -0.5
Nq(t)
on-off
S
a
m
p
l
i
n
g

Q
u
a
n
t
i
z
a
t
i
o
n

Sp(t)
&(t)
S(t)
Sq(t)
Sq(t)
S(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)
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2013/12/17
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 T
S
= period if only
f
s
2B.
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2013/12/17
on-off
F(t)
Fs(t)=f(t).s(t)
S(t)

=
=
n
t jn
n
s
e S t s
e
) (
2
2
) sin(
) (
1 2
2
t e
t e
e
t
s
s s
T
s
T
s
n
n
t jn
n
T
dt e t s
T
S = =
}

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2013/12/17

That is, to completely restore a signal from its sampling value
without distortion, the sampling frequency must satisfy the
following conditions:
f
s
2B(Hz), f
s
can also be called the Nyquist Frequency, or T
S

and T
s
is called the Nyquist time interval.
In telephone communications, the voice frequency band is 300-
3400Hz, and the actual sampling frequency f
s
is taken as
8000Hz>2B=23400Hz=6800Hz. This is to prevent the
confusion of signals after sampling and enable protection zone
in the spectrum.
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125S
t
A
Cycle T= 125S f=8kHz
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A
B
C
D
A
B
C
D


A B C D
A B C D


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2013/12/17
A
B
C
D
A
B
C
D


A B C D
A B C D


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2013/12/17
A
B
C
D
A
B
C
D


A B C D
A B C D


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D1(t)
D2(t)
D3(t)
D4(t)
DM(t)
TDM
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2013/12/17
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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2013/12/17

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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2013/12/17
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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2013/12/17
even quantization
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2013/12/17
No-even quantization
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2013/12/17











A/D

D/A
Input 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
signal
Y

t)
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2013/12/17
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 -1x1
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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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2013/12/17








40

0
5
15
40
100
=255
Output
1.0
1.0
Input
Figure 1.3.2-2 -law characteristics
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2013/12/17
For the companding law A, we have:

Y=
1+lnAx(
1+lnA
SGNx,
1
A
x(1
Y=
Ax(
1+lnA
SGNx, 0 x(
1
A

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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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2013/12/17
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
1
7/8
6/8
5/8
4/8
*
+
4
^
1
X
1/2 1/4 1/8 1/16
Figure 1.3.3-1 13-broken-line segment diagram

-

-

-

-










11



O





=

=

1
2


-

-
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2013/12/17
13-broken-line segment diagram
INput
OUTput
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2013/12/17
broken.doc
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2013/12/17
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 18, each sub-segment shall have
164as 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 4
th
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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Signa
ling

timesl
ot

Frame length
125s
Time
slot
0 1-15 16
17-30
31
Time slot
length 3.9s

Frame synchronization
word
Polarity
code Segment code Intra-segment code

e

D
1

D
2

D
3

D
4

D
5

D
6

D
7

D
8



D
1

D
2

D
3

D
4

D
5

D
5

D
7

D
8

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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 synchronization
code

D1 D2 D3 D4 D5 D6 D7 D8
Even frame
Odd frame
Loss of frame
opposite alarm
code
F1
a b c d a b c d
Channel 16
signaling code
Channel 1
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=0
Out 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
Channel 15
signaling code
Channel 30
signaling 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 0and
+1 or 1 to indicate 1with +1 and 1 coming alternatively.
When four continuous 0 come out, they shall be replaced
with B00V or 000V.

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1.3.5-3 AMIHDB

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

+1 0 0 0 0 -1 0 +1 0 0 0 0 0 0
-1
t
AMI
1.3.5-2 HDB3


1.0
0.5
1.0 f
T
HDB3
AMI
p
HD
B3
t
+1 0 0 0 +1 -1 0 +1 -1 0 0 -1 0
0 +1
V
B
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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

NRZ
AMI
HDB3
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.

1CH
64Kb/s

24CH
1.554
Mb/s

96CH
6.321
Mb/s

672CH
44.736
Mb/s

4032CH
274.176
Mb/s

480CH
32.064
Mb/s

1440CH
97.728
Mb/s

5760CH
397.2
Mb/s
North America

Japan

30CH
2.048
Mb/s

120CH
8.448
Mb/s

480CH
34.368
Mb/s

1920CH
139.264
Mb/s

7680CH
565.992
Mb/s
Europe, China

7 6
24
G.72
3
G.73
4
G.74
3
G.74
6
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.754
G.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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PDH
The 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/s
Line
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/s
Line
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.