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Acoustic Spectrum

Analog data take on continuous values

in some interval, the most familiar
example being audio, which, in the
form of acoustic sound waves, can be
perceived directly by human beings.
The acoustic spectrum for human
speech and for music (note log scales).
Frequency components of typical
speech may be found between
approximately 100 Hz and 7 kHz, and
has a dynamic range of about 25 dB (a
shout is approx 300 times louder than
Another common example of analog
data is video, as seen on a TV screen.
Audio Signals
Freq range 20Hz-20kHz (speech 100Hz-7kHz)
easily converted into electromagnetic signals
varying volume converted to varying voltage
can limit frequency range for voice channel to

where signal strength falls off with distance

depends on medium
received signal strength must be:
strong enough to be detected
sufficiently higher than noise to receive without error
so increase strength using amplifiers/repeaters
is also an increasing function of frequency
so equalize attenuation across band of frequencies used
eg. using loading coils or amplifiers

Attenuation of all frequency components is not same. Some frequencies are

passed without attenuation, some are weakened and some are blocked. This
dependence of attenuation of a channel on the frequency of a signal leads to
a new kind of distortion attenuation distortion.
Delay Distortion
• only occurs in guided media
• propagation velocity varies with frequency
• hence various frequency components arrive at different times
• particularly critical for digital data
• since parts of one bit spill over into others
• causing intersymbol interference

Delay distortion occurs because the velocity of propagation of a signal through a guided
medium varies with frequency.

For a bandlimited signal, the velocity tends to be highest near the center frequency and
fall off toward the two edges of the band.

Thus various frequency components of a signal will arrive at the receiver at different
times, resulting in phase shifts between the different frequencies.

Delay distortion is particularly critical for digital data, because some of the signal
components of one bit position will spill over into other bit positions, causing
intersymbol interference.

This is a major limitation to maximum bit rate over a transmission channel.

Additional signals inserted between transmitter and receiver
due to thermal agitation of electrons
uniformly distributed
white noise
Thermal noise is due to thermal agitation of electrons. It is present in all electronic devices and
transmission media and is a function of temperature. Thermal noise is uniformly distributed across
the bandwidths typically used in communications systems and hence is often referred to as white
noise. Thermal noise cannot be eliminated and therefore places an upper bound on
communications system performance, and. is particularly significant for satellite communication

signals that are the sum and difference of original frequencies sharing a medium

When more than one signal share a single transmission medium, intermodulation noise is
generated. For example, two signals f1 and f2 will generate signals of frequencies (f1 + f2) and (f1
- f2), which may interfere with the signals of the same frequencies sent by the transmitter.
Intermodulation noise is introduced due to nonlinearity present in any part of the communication
a signal from one line is picked up by another

irregular pulses or spikes
eg. external electromagnetic interference
short duration
high amplitude
a minor annoyance for analog signals
but a major source of error in digital data
a noise spike could corrupt many bits

Cross talk is a result of bunching several conductors together in a single cable. Signal
carrying wires generate electromagnetic radiation, which is induced on other
conductors because of close proximity of the conductors. While using telephone, it
is a common experience to hear conversation of other people in the background.
This is known as cross talk.

Impulse noise is irregular pulses or noise spikes of short duration generated by

phenomena like lightning, spark due to loose contact in electric circuits, etc.
Impulse noise is a primary source of bit-errors in digital data communication. This
kind of noise introduces burst errors.
Channel Capacity
The maximum rate at which data can be transmitted over a given communication channel, under given
conditions, is referred to as the channel capacity. There are four concepts here that we are trying to relate
to one another.

Bandwidth refers to the range of frequencies that a medium can pass without a loss of
one-half of the power (-3dB) contained in the signal. Figure shows the bandwidth of
a channel. The points Fl and Fh points correspond to -3bB of the maximum amplitude A.
• Data rate, in bits per second (bps), at which data can be communicated
• Bandwidth, as constrained by the transmitter and the nature of the transmission medium, expressed in
cycles per second, or Hertz
• Noise, average level of noise over the communications path
• Error rate, at which errors occur, where an error is the reception of a 1 when a 0 was transmitted or the
reception of a 0 when a 1 was transmitted
All transmission channels of any practical interest are of limited bandwidth, which arise from the physical
properties of the transmission medium or from deliberate limitations at the transmitter on the bandwidth to
prevent interference from other sources. Want to make as efficient use as possible of a given bandwidth.
For digital data, this means that we would like to get as high a data rate as possible at a particular limit of
error rate for a given bandwidth. The main constraint on achieving this efficiency is noise.
Nyquist Bandwidth
consider noise free channels
if rate of signal transmission is 2B then can carry signal with frequencies no greater than B
ie. given bandwidth B, highest signal rate is 2B
for binary signals, 2B bps needs bandwidth B Hz
can increase rate by using M signal levels
Nyquist Formula is: C = 2B log2M

where C is known as the channel capacity,

B is the bandwidth of the channel
and m is the number of signal levels used.
Baud Rate: The baud rate or signaling rate is defined as the number of distinct symbols
transmitted per second, irrespective of the form of encoding. For baseband digital
transmission m = 2. So, the maximum baud rate = 1/Element width (in Seconds) = 2B

Bit Rate: The bit rate or information rate I is the actual equivalent number of bits transmitted per
second. I = Baud Rate x Bits per Baud

= Baud Rate x N = Baud Rate xlog2m

For binary encoding, the bit rate and the baud rate are the same; i.e., I = Baud Rate.
Example: Let us consider the telephone channel having bandwidth B = 4 kHz. Assuming
there is no noise, determine channel capacity for the following encoding levels:
(i) 2, and (ii) 128.
Ans: (i) C = 2B = 2x4000 = 8 Kbits/s
(ii) C = 2x4000xlog2128 = 8000x7 = 56 Kbits/s
Shannon Capacity Formula
consider relation of data rate, noise & error rate
faster data rate shortens each bit so bursts of noise affects more bits
given noise level, higher rates means higher errors
Shannon developed formula relating these to signal to noise ratio (in decibels)
SNRdb =10 log10 (signal/noise)
Capacity C=B log2(1+SNR)
theoretical maximum capacity
get lower in practise

Example: Suppose we have a channel of 3000 Hz bandwidth, we need an S/N ratio (i.e.
signal to noise ration, SNR) of 30 dB to have an acceptable bit-error rate. Then, the
maximum data rate that we can transmit is 30,000 bps.
In practice, because of the presence of different types of noises, attenuation and delay
distortions, actual (practical) upper limit will be much lower.
In case of extremely noisy channel, C = 0
Between the Nyquist Bit Rate and the Shannon limit, the result providing the smallest
channel capacity is the one that establishes the limit.
Bandwidth Utilization:
Multiplexing and Spreading
Bandwidth utilization is the wise use of
available bandwidth to achieve
specific goals.

Efficiency can be achieved by multiplexing;

privacy and anti-jamming can be achieved by

Whenever the bandwidth of a medium linking two devices is

greater than the bandwidth needs of the devices, the link can
be shared. Multiplexing is the set of techniques that allows the
simultaneous transmission of multiple signals across a single
data link. As data and telecommunications use increases, so
does traffic.
Dividing a link into channels
Categories of multiplexing
Frequency-division multiplexing

FDM is an analog multiplexing technique that combines analog signals.

FDM process
A generic depiction of an FDM system is shown in Stallings DCC8e
Figure 8.4. A number of analog or digital signals [mi(t), i = 1, n] are to
be multiplexed onto the same transmission medium. Modulation
equipment is needed to move each signal to the required frequency
band, and multiplexing equipment is needed to combine the modulated
signals. Each signal mi(t) is modulated onto a carrier fi; because multiple
carriers are to be used, each is referred to as a subcarrier. Any type of
modulation may be used. The resulting analog, modulated signals are
then summed to produce a composite baseband signal mb(t). Figure 8.4b
shows the result. The spectrum of signal mi(t) is shifted to be centered
on fi. For this scheme to work, fi must be chosen so that the bandwidths
of the various signals do not significantly overlap. Otherwise, it will be
impossible to recover the original signals. The composite signal may
then be shifted as a whole to another carrier frequency by an additional
modulation step. The FDM signal s(t) has a total bandwidth B = Sum
Bi . This analog signal may be transmitted over a suitable medium. At
the receiving end, the FDM signal is demodulated to retrieve mb(t),
which is then passed through n bandpass filters, each filter centered on fi
and having a bandwidth Bi, for 1 ≤ i ≤ n. In this way, the signal is again
split into its component parts. Each component is then demodulated to
recover the original signal.
FDM Voiceband
Illustrates a simple example of transmitting three voice signals
simultaneously over a medium. As was mentioned, the bandwidth of a
voice signal is generally taken to be 4 kHz, with an effective spectrum
of 300 to 3400 Hz (Figure 8.5a). If such a signal is used to amplitude-
modulate a 64-kHz carrier, the spectrum of Figure 8.5b results. The
modulated signal has a bandwidth of 8 kHz, extending from 60 to 68
kHz. To make efficient use of bandwidth, we elect to transmit only the
lower sideband. If three voice signals are used to modulate carriers at
64, 68, and 72 kHz, and only the lower sideband of each is taken, the
spectrum of Figure 8.5c results.
This figure points out two problems that an FDM system
must cope with. The first is crosstalk, which may occur if the spectra
of adjacent component signals overlap significantly. In the case of
voice signals, with an effective bandwidth of only 3100 Hz (300 to
3400), a 4-kHz bandwidth is adequate. The spectra of signals
produced by modems for voiceband transmission also fit well in this
bandwidth. Another potential problem is intermodulation noise. On a
long link, the nonlinear effects of amplifiers on a signal in one channel
could produce frequency components in other channels.
Assume that a voice channel occupies a bandwidth of 4 kHz. We need to combine three
voice channels into a link with a bandwidth of 12 kHz, from 20 to 32 kHz. Show the
configuration, using the frequency domain. Assume there are no guard bands.
We shift (modulate) each of the three voice channels to a different bandwidth, as shown in
Figure 6.6. We use the 20- to 24-kHz bandwidth for the first channel, the 24- to 28-kHz
bandwidth for the second channel, and the 28- to 32-kHz bandwidth for the third one. Then
we combine them as shown in Figure 6.6.
Five channels, each with a 100-kHz bandwidth, are to be multiplexed
together. What is the minimum bandwidth of the link if there is a need
for a guard band of 10 kHz between the channels to prevent
For five channels, we need at least four guard bands. This means
that the required bandwidth is at least
5 × 100 + 4 × 10 = 540 kHz,
as shown in Figure 6.7.
Four data channels (digital), each transmitting at 1 Mbps, use a satellite
channel of 1 MHz. Design an appropriate configuration, using FDM
The satellite channel is analog. We divide it into four channels, each
channel having a 250-kHz bandwidth. Each digital channel of 1 Mbps
is modulated such that each 4 bits is modulated to 1 Hz. One solution is
16-QAM modulation. Figure 6.8 shows one possible configuration
Analog hierarchy
Synchronous Time Division Multiplexing
Synchronous time division multiplexing can be used with
digital signals or analog signals carrying digital data. In this form
of multiplexing, data from various sources are carried in repetitive
frames. Each frame consists of a set of time slots, and each
source is assigned one or more time slots per frame. The effect is
to interleave bits of data from the various sources. The
interleaving can be at the bit level or in blocks of bytes or larger
quantities. For example, the multiplexer in Stallings DCC8e Figure
8.2b has six inputs that might each be, say, 9.6 kbps. A single line
with a capacity of at least 57.6 kbps (plus overhead capacity)
could accommodate all six sources.
Synchronous TDM is called synchronous not because
synchronous transmission is used, but because the time slots are
preassigned to sources and fixed. The time slots for each source
are transmitted whether or not the source has data to send.
Quadros TDM
• TDM Puro: débito mux-para-mux = débitos dos
PCs agregados

• Sem perdas de dados (similar à multiplexagem

de chamadas telefónicas)

Based on
Data Communications and Networking, 3rd EditionBehrouz A.
© McGraw-Hill Companies, Inc., 2004
Exemplo 1

4 ligações 1-Kbps são multiplexadas. A unidade é 1 bit.

Determine:(1) A duração de 1 bit antes da
multiplexagem, (2) O débito da ligação , (3) a duração
duma fatia de tempo (4) a duração dum quadro?
Podemos responder assim:
1. A duração do bit é 1/1 Kbps, ou 0.001 s (1 ms).
2. O débito da ligação é 4 Kbps.
3. A duração de cada fatia de tempo é 1/4 ms ou 250 µ s.
4. A duração de cada quadro é 1 ms.
Multiplexador/Desmultiplexador processa um PC de cada vez

Intercalação de caracter (byte)

A multiplexagem processa de cada vez um/mais caracteres de
•cada unidade
Intercalação de bit

A multiplexagem processa um bit de cada unidade de cada vez

Exemplo 2

Quatro canais são multiplexados usando o TDM.

Cada um deles envia 100 bytes/seg e é multiplexado
1 byte por canal. Mostre o quadro a viajar no canal,
a duração do quadro, a taxa de quadros e
o débito em bits para a ligação
Exemplo 2: Solução
Electrónico para
• Questão essencial da multiplexagem é a
sincronização entre comutador e o

• Cada amostra tem que ser entregue ao

destino correcto e no instante devido

• O distribuidor deve estar posicionado

na saída do destino i sempre que chega
amostra originária da fonte i (trama
Sinal PAM TDM com
Conceito de Canal
• Tudo se Virtual
passa como se cada para fonte
destino tivesse um caminho dedicado
onde transitam amostras do respectivo

• Este conceito aparece frequentemente

em outros contextos de
telecomunicações em especial na
comunicação de dados e comunicação
entre computadores
Representação de
Canal Virtual
Técnicas de TDM
•Na técnica descrita
• Símbolos sucedem-se regularmente no tempo

• Tramas contíguas sem interrupção

•TDM síncrono
Quando uma fonte deixa de transmitir os intervalos de tempo
lhe estão atribuídos têm que decorrer pois são esses
intervalos que identificam a fonte

• TDM assíncrono

•Não se exige a referida ordenação temporal nem a contiguidade

das tramas e pode-se usar o tempo desperdiçado.
Aplicações TDM
• Telefone Digital
• Comunicação de Dados
• Acesso a satélite
• Rádio Celular
TDM Síncrono
primeira forma apareceu com a digitalização
do sistema telefónico
• Começou com a preocupação de transmitir
canais de voz de qualidade telefónica
• Sistemas de multiplexagem TDM
• ATVsua estrutura mostrou-se desadequado para
digital, comunicação entre computadores
• Surgiram outras estruturas de multiplexagem
como a SDH e a SONET
• Usaremos o PCM como referência e
abordaremos aspectos de outros sistemas
Organização de
• Tramas
Os sistemas TDM digitais multiplexam os
canais na forma digital binária

• Fontes analógicas são previamente


• A cada amostra PAM passa a

corresponder um grupo de k bits (uma

• O ritmo unitário corresponde a uma canal

telefónico (4kHz) com 8 bits por amostra
rb = 2 xBxk = 2 x 4 x8 = 64Kbps
Organização das
• Tramas
Constituída pela multiplexagem no
tempo de N canais de K bits
• 2 organizações
• Canais entrelaçados
• Dígitos entrelaçados
Formas de
organização de
Alinhamento das
• Típica da tramas
multiplexagem síncrona

• Sincronização do equipamento
terminal de recepção tanto em
frequência como em fase à frequência
de símbolos que está a receber.

• Operação necessária cada vez que o

receptor entra em operação

• Inicial ou após interrupção

Receptor alinhado
Precisa de referência temporal periódica para verificar

isocronismo e detectar eventuais desvios de fase
Referência temporal consiste num padrão de bits organizado da

seguinte forma
Alinhamento agrupado: v bits consecutivos no ínicio de cada
• trama
Alinhamento distribuído: os v bits do padrão estão distribuídos
• na mesma trama ou em várias
• Padrão com baixasimulada
• Bloqueando canais da trama e
enviando sequência determinística
• Confirmando o correcto alinhamento
através de critérios diferentes
(ausência de padrão de alinhamento
em tramas alternadas)
• Transmissão de informação de controlo entre
equipamentos de multiplexagem

• Possui semântica própria

• Sinalização dentro do octeto (em banda)

• Sinalização fora do octeto

• Sinalização em canal comum (+ utilizada)

Sistemas de
• Proliferação
de sistemas de multiplexagem

• Ritmos de transmissão
• Número de canais por trama
• Método de sinalização
• Normas ITU
• Sistema Europeu
• Sistema Americano (AT&T)
Sistema de
• Multiplexagem
Referenciado abreviadamente como ritmo de
2Mbps resultante da composição da trama com
125us de duração por 32Europeu
canais básicos

•Canais 1-15;17-31 constituem os 30 canais de

informação utilizáveis

•Canal 0 destina-se à transmissão em tramas

alternadas do padrão de alinhamento da trama

•Canal 16- Usado para numeração das tramas

num alinhamento multitrama

•Sinalização fora do octecto

Estrutura da trama
PCM de 2 Mbps
Multitrama PCM de
2 Mbps
Algoritmo de
SP Europeu
Sistema de
• 24 canais +1 bit de sincronismo de
Primário Amerciano
trama =193 bits
• 193bits/125 us = 1544
• Sinalização dentro do octeto : último
bit de cada canal das tramas 6 e 12
Estrutura da trama
PCM de 1.5Mbps
Multitrama PCM
1.5Mbps e
canais de sinalização
TDM Link Control
• No headers and trailers

• Data link control protocols not needed

• Flow control

• Data rate of multiplexed line is fixed

•must carry on
If one channel receiver can not receive data, the others

• The corresponding source must be quenched

• This leaves empty slots

• Error control

Errors are detected and handled by individual channel
Data Link Control
on TDM
No flag or SYNC characters bracketing TDM frames

Must provide synchronizing mechanism

Added digit framing

One control bit added to each TDM frame

Looks like another channel - “control channel”

Identifiable bit pattern used on control channel

e.g. alternating 01010101…unlikely on a data channel

Can compare incoming bit patterns on each channel with
• sync pattern
Pulse Stuffing
•Problem - Synchronizing data sources
•Clocks in different sources drifting
•rational number
Data rates from different sources not related by simple

• Solution - Pulse Stuffing

•sum of incoming rates

Outgoing data rate (excluding framing bits) higher than

• Stuff extra dummy bits or pulses into each incoming

signal until it matches local clock

Stuffed pulses inserted at fixed locations in frame and
at demultiplexer
TDM of Analog and
Digital Sources
Digital Carrier
• Hierarchy Systems
of TDM
• USA/Canada/Japan use one system
• ITU-T use a similar (but different) system
• US system based on DS-1 format
• Multiplexes 24 channels
• Each frame has 8 bits per channel plus
one framing bit
• 193 bits per frame
Digital Carrier
• Systems (2)
For voice each channel contains one word of digitized data (PCM,
8000 samples per sec)
Data rate 8000x193 = 1.544Mbps

Five out of six frames have 8 bit PCM samples

Sixth frame is 7 bit PCM word plus signaling bit

Signaling bits form stream for each channel containing control
•and routing info
Same format for digital data

23 channels of data

7 bits per frame plus indicator bit for data or systems control

24th channel is sync

Mixed Data
• DS-1 can carry mixed voice and data

• 24 channels used
• No sync byte
• Can also interleave DS-1 channels
• Ds-2 is four DS-1 giving 6.312Mbps
TDM System
In frequency division multiplexing, all signals operate at
the same time with different frequencies, but in Time-
division multiplexing all signals operate with same
frequency at
different times. This is a base band transmission system,
where an electronic commutator sequentially samples all
data source and combines them to form a composite
base band signal, which travels through the media and is
being demultiplexed into appropriate independent
message signals by the corresponding commutator at
the receiving end. The
incoming data from each source are briefly buffered.
Each buffer is typically one bit or one character in length.
The buffers are scanned sequentially to form a
composite data
stream. The scan operation is sufficiently rapid so that
each buffer is emptied before more data can arrive.
Composite data rate must be at least equal to the sum of
the individual
data rates. The composite signal can be transmitted
directly or through a modem. The multiplexing operation
is shown
Synchronous time-division multiplexing

In synchronous TDM, the data rate

of the link is n times faster, and the unit duration is n times shorter.
Example 6.5

In Figure 6.13, the data rate for each input connection is

3 kbps. If 1 bit at a time is multiplexed (a unit is 1 bit),
what is the duration of (a) each input slot, (b) each output
slot, and (c) each frame?

We can answer the questions as follows:
a. The data rate of each input connection is 1 kbps. This
means that the bit duration is 1/1000 s or 1 ms. The
duration of the input time slot is 1 ms (same as bit

Example 6.5 (continued)

b. The duration of each output time slot is one-third of

the input time slot. This means that the duration of the
output time slot is 1/3 ms.

c. Each frame carries three output time slots. So the

duration of a frame is 3 × 1/3 ms, or 1 ms. The
duration of a frame is the same as the duration of an
input unit.
Example 6.6

Figure 6.14 shows synchronous TDM with a data stream

for each input and one data stream for the output. The
unit of data is 1 bit. Find (a) the input bit duration, (b)
the output bit duration, (c) the output bit rate, and (d) the
output frame rate.
We can answer the questions as follows:
a. The input bit duration is the inverse of the bit rate:
1/1 Mbps = 1 μs.

b. The output bit duration is one-fourth of the input bit

duration, or ¼ μs.
Example 6.6 (continued)

c. The output bit rate is the inverse of the output bit

duration or 1/(4μs) or 4 Mbps. This can also be
deduced from the fact that the output rate is 4 times as
fast as any input rate; so the output rate = 4 × 1 Mbps
= 4 Mbps.

d. The frame rate is always the same as any input rate. So

the frame rate is 1,000,000 frames per second.
Because we are sending 4 bits in each frame, we can
verify the result of the previous question by
multiplying the frame rate by the number of bits per
Figure 6.14 Example 6.6
Example 6.7

Four 1-kbps connections are multiplexed together. A unit

is 1 bit. Find (a) the duration of 1 bit before multiplexing,
(b) the transmission rate of the link, (c) the duration of a
time slot, and (d) the duration of a frame.

We can answer the questions as follows:
a. The duration of 1 bit before multiplexing is 1 / 1 kbps,
or 0.001 s (1 ms).

b. The rate of the link is 4 times the rate of a connection,

or 4 kbps.
Example 6.5

In Figure 6.13, the data rate for each input connection is

3 kbps. If 1 bit at a time is multiplexed (a unit is 1 bit),
what is the duration of (a) each input slot, (b) each
output slot, and (c) each frame?

We can answer the questions as follows:
a. The data rate of each input connection is 1 kbps. This
means that the bit duration is 1/1000 s or 1 ms. The
duration of the input time slot is 1 ms (same as bit
Example 6.7 (continued)

c. The duration of each time slot is one-fourth of the

duration of each bit before multiplexing, or 1/4 ms or
250 μs. Note that we can also calculate this from the
data rate of the link, 4 kbps. The bit duration is the
inverse of the data rate, or 1/4 kbps or 250 μs.

d. The duration of a frame is always the same as the

duration of a unit before multiplexing, or 1 ms. We
can also calculate this in another way. Each frame in
this case has four time slots. So the duration of a
frame is 4 times 250 μs, or 1 ms.
Figure 6.15 Interleaving
Example 6.8

Four channels are multiplexed using TDM. If each

channel sends 100 bytes /s and we multiplex 1 byte per
channel, show the frame traveling on the link, the size of
the frame, the duration of a frame, the frame rate, and
the bit rate for the link.
The multiplexer is shown in Figure 6.16. Each frame
carries 1 byte from each channel; the size of each frame,
therefore, is 4 bytes, or 32 bits. Because each channel is
sending 100 bytes/s and a frame carries 1 byte from each
channel, the frame rate must be 100 frames per second.
The bit rate is 100 × 32, or 3200 bps.
Figure 6.16 Example 6.8
Example 6.9

A multiplexer combines four 100-kbps channels using a

time slot of 2 bits. Show the output with four arbitrary
inputs. What is the frame rate? What is the frame
duration? What is the bit rate? What is the bit duration?

Figure 6.17 shows the output for four arbitrary inputs.
The link carries 50,000 frames per second. The frame
duration is therefore 1/50,000 s or 20 μs. The frame rate
is 50,000 frames per second, and each frame carries 8
bits; the bit rate is 50,000 × 8 = 400,000 bits or 400 kbps.
The bit duration is 1/400,000 s, or 2.5 μs.
Figure 6.17 Example 6.9
Figure 6.18 Empty slots
Figure 6.19 Multilevel multiplexing
Figure 6.20 Multiple-slot multiplexing
Figure 6.21 Pulse stuffing
Figure 6.22 Framing bits
Example 6.10

We have four sources, each creating 250 characters per

second. If the interleaved unit is a character and 1
synchronizing bit is added to each frame, find (a) the
data rate of each source, (b) the duration of each
character in each source, (c) the frame rate, (d) the
duration of each frame, (e) the number of bits in each
frame, and (f) the data rate of the link.
We can answer the questions as follows:
a. The data rate of each source is 250 × 8 = 2000 bps = 2
Example 6.10 (continued)

b. Each source sends 250 characters per second;

therefore, the duration of a character is 1/250 s, or
4 ms.
c. Each frame has one character from each source,
which means the link needs to send 250 frames per
second to keep the transmission rate of each source.
d. The duration of each frame is 1/250 s, or 4 ms. Note
that the duration of each frame is the same as the
duration of each character coming from each source.
e. Each frame carries 4 characters and 1 extra
synchronizing bit. This means that each frame is
4 × 8 + 1 = 33 bits.
Example 6.11

Two channels, one with a bit rate of 100 kbps and

another with a bit rate of 200 kbps, are to be multiplexed.
How this can be achieved? What is the frame rate? What
is the frame duration? What is the bit rate of the link?

We can allocate one slot to the first channel and two slots
to the second channel. Each frame carries 3 bits. The
frame rate is 100,000 frames per second because it
carries 1 bit from the first channel. The bit rate is
100,000 frames/s × 3 bits per frame, or 300 kbps.
Figure 6.23 Digital hierarchy
Table 6.1 DS and T line rates
Figure 6.24 T-1 line for multiplexing telephone lines
Figure 6.25 T-1 frame structure
Table 6.2 E line rates
Figure 6.26 TDM slot comparison
In spread spectrum (SS), we combine signals from
different sources to fit into a larger bandwidth, but
our goals are to prevent eavesdropping and jamming.
To achieve these goals, spread spectrum techniques
add redundancy.

Topics discussed in this section:

Frequency Hopping Spread Spectrum (FHSS)
Direct Sequence Spread Spectrum Synchronous (DSSS)
Figure 6.27 Spread spectrum
Figure 6.28 Frequency hopping spread spectrum (FHSS)
Figure 6.29 Frequency selection in FHSS
Figure 6.30 FHSS cycles
Figure 6.31 Bandwidth sharing
Figure 6.32 DSSS
Figure 6.33 DSSS example
Hierarquia de
• Para multiplexar um maior PDH
de canais
• Recorre-se à hierarquização de
estágios de multiplexagem
• Ritmo agregado mais elevado
Exemplo de
Multiplexagem PDH
Hierarquia de
multiplexagem PDH
Exemplo 6.1
Hierarquia de
• Infra-estruturas das PDH
operadoras estão
estruturadas segundo esta hierarquia

• Utente deve dispor de terminal

apropriado para o serviço que
pretende usar

• Se não for o caso tem que se

intercalar um DSU
Anisocronismo das
• tributárias
Os multiplexadores de nível inferior a um
determinado nível de multiplexagem são
designadas tributárias

Na construção do nível n de multiplexagem
de tramas de n-1 há o problema do

assincronismo das tributárias

•Fontes geograficamente distantes

•Frequências ligeiramente diferentes
•Frequências isócronas mas desfasadas
•Desfasamento provoca atrasos (jitter)
Estrutura da trama
PDH de 2ª ordem a 8
Formato da trama de
multiplexagem de 8
Arquitectura dos
Duplexidade dos
Formatos de
• 125 us de duração
Tramas básicas: blocosSDH
de 810 e
bytes com

• Coincide SONET
com o período PCM: são
produzidas 8000 tramas por segundo

• Sistema síncrono: as tramas são sempre

emitidas com informação útil ou não

• Tramas descritas com matriz 9*90

Formato de Tramas
Formatos de
tramas SDH e
•Ritmo binário bruto
810x8=6480 bits/125 us -> 51.84 Mbps (canal

•Trama correspondenre STS-1

•Todos os restantes ritmos são múltiplos do STS-1
3 1ºs bytes são reservados para gestão do sistema
da linha e da secção)

Restantes 87 colunas transportam dados do
- as origens/destinos ligados aos
equipamentos terminais. Esses dados SPE
(Synchronous Payload Envelope)
Hierarquias de
Multiplexagem SDH e
Hierárquica SDH
TDM Estatístico
• síncrona
Apropriada para transmissão digitalizada de
fontes que produzem tráfego a um ritmo
contínuo ou regular

• Existem fontes que não se comportam desta


• Computadores, Terminais de Dados

• Tráfego produzido de forma aleatória

Característica do

De natureza
• Multiplexador para tirar partido deveria

• Alocar dinamicamente as ranhuras

temporais das tramas de saída

•Em função da existência de informação

nos buffers de entrada e com o seu
estado de ocupação
Multiplexagem estatística ou assíncrona
Característica do
• Tráfego
Intermitente não Gerado
regular por
• CarácterComputador

• Com elevado factor de crista

• Relação elevada entre o ritmo binário

nominal e médio (>> 1)

• Muitos picos (bursts) de tráfego

• Muitos tempos mortos
Características do
trafego gerado por
dao intermitncia do trafego 1
λi − τ i Ž aleat—ria
< nœmero de bits transmitidos durante t d >
rb =
Factor de crista
rb td
fc = =
rb ∑ ri

Os equipamentos transmitem apenas durante uma frac‹o

do tempo
Características do
trafego gerado por
• computador
Unidade de dados (DU) de transmissão

• Equipamento nunca transmite apenas um bit

• Ou um byte ou um múltiplo do byte

• DU é um conjunto de bits emitidos

consecutivamente e interpretado como um todo

• DU pode ser de tamanho fixo ou variável

1 / λ − tempo m Ž dio entre transmiss›es
de DUs
• ι
Da figura anterior
λι − valor mŽdio de DUs produzidos por segundo
TDM Estatístico
Débito Mux-to-Mux < débito agregado dos terminais/hosts

• Fatias de tempo alocadas baseada em padrões de tráfego
Usar estatística para determinar alocação entre utilizadores

Tem que nviar o endereço da porta com os dados (sobrecarga)

• Potencia perdas em períodos de pico
Pode usar buffer de dados e controlo de fluxo para reduzir perdas

• Nem sempre transparente aos terminais/hosts
Perdas e atrasos são possíveis

• Porquê usar o STDM
Mais económico (menos MUXs e linhas mais baratas) e
• eficiente
TDM síncrono vs
Formato de tramas
do TDM estatístico
• Recorre-se à teoria das filas de
espera para descrever o
comportamento do multiplexador
Classificação de Modelo de Filas

Processo de Chegada / Tempo de Serviço / Servidores / Max Ocupação

Intervalo entre Service times X 1 servidor K clientes

chegadas τ M = exponential c servidores Não especificado
M = exponencial D = deterministic se ilimitado
D = determinístico G = geral
G = geral Tempo de serviço:
Ritmo de chegada: µ = 1/ E[X]
λ = 1/ E[τ ]

Multiplexer Models: M/M/1/K, M/M/1, M/G/1, M/D/1

Modelos de Trunking: M/M/c/c, M/G/c/c
Actividade de utilizadores: M/M/∞, M/G/ ∞
Variáveis de um Sistema de Fila de Espera

N(t) = Nq(t) + Ns(t) N(t) = nº no sistema

Ns(t) Nq(t) = nº na fila
Nq(t) 1 Ns(t) = nº em serviço
λ (1−Pb) 2

c T = atraso total
X W = tempo de espera
λ Pb T=W+X
X = tempo de serviço
Classificação de Modelo de Filas

Processo de Chegada / Tempo de Serviço / Servidores / Max Ocupação

Intervalo entre Service times X 1 servidor K clientes

chegadas τ M = exponential c servidores Não especificado
M = exponencial D = deterministic se ilimitado
D = determinístico G = geral
G = geral Tempo de serviço:
Ritmo de chegada: µ = 1/ E[X]
λ = 1/ E[τ ]

Multiplexer Models: M/M/1/K, M/M/1, M/G/1, M/D/1

Modelos de Trunking: M/M/c/c, M/G/c/c
Actividade de utilizadores: M/M/∞, M/G/ ∞
Fila de Espera de
MUX estatístico

Os clienteschegama um ritmo mŽ dio de λ/segundode acordocom uma determinada estat’ stic

O servidorpossuiuma capacidadede C opera› es/segundo;
O n¼mŽ dio de opera› espor cliente vari‡ velaleat— ria
Tempode servi o = Σ = vari‡ vel
aleat— ria
Fila de esperaem equil’ brioρ = = λΣ <1
Probabilístico para
Modelo para MUX
DU com comprimento fixo de k bits
α .rbe α .r
Cada uma das N entradas tem um ritmo mŽdio de DU/s Entrada œnica de N. be DU/s
k k
Serviocom capacidade de rbs bits/seg efectuando = κ opera›es por cliente
α .rbe 1 κ rbe
λ = N. S= = ρ = S.λ = Nα .
k µΧ rbs rbs
Teorema de Pollaczek - Khinchin (valor mŽdio
do tempo de espera em sistemas M/D/1 ) t w =
2(1 − ρ )
Tempo mŽdio total no sistema t q = S +
2(1 − ρ )
Teorema de Litle (n¼de DU no sistema) n q = ρ +
2(1 − ρ )
Ocupação e atraso
médios no MUX
Probabilidade de
sobrelotação dos
Exemplo 6.2
• Considere-se um multiplexador estatístico com
entradas de de 64Kbps e saída de 120Kbps.
Suponha-se que os equipamentos ligados à entrada
transmitem unidades de dados (DUs) de
comprimento fixo igual a 1000 bits durante 40% do
tempo a uma ritmo aleatório (Poisson), isto é,
tempos entre DUs distribuídos segundo uma
exponencial negativa. Valores para esta situação
• k=1000 bits; rbe=64000 bps; rbs=128000; alfa=0.4
a)3 equipamentos ligados à entrada
b)4 equipamentos ligados à entrada
Pulse-Code Modulation
PULSE-CODE MODULATION (pcm) refers to a system in which the standard values of a
QUANTIZED WAVE (explained in the following paragraphs) are indicated by a series of coded
pulses. When these pulses are decoded, they indicate the standard values of the original
quantized wave. These codes may be binary, in which the symbol for each quantized element
will consist of pulses and spaces: ternary, where the code for each element consists of any one
of three distinct kinds of values (such as positive pulses, negative pulses, and spaces); or n-ary,
in which the code for each element consists of nay number (n) of distinct values. This discussion
will be based on the binary pcm system. All of the pulse-modulation systems discussed
previously provide methods of converting analog wave shapes to digital wave shapes (pulses
occurring at discrete intervals, some characteristic of which is varied as a continuous function of
the analog wave). The entire range of amplitude (frequency or phase) values of the analog wave
can be arbitrarily divided into a series of standard values. Each pulse of a pulse train [figure 2-
48, view (B)] takes the standard value nearest its actual value when modulated. The modulating
wave can be faithfully reproduced, as shown in views (C) and (D). The amplitude range has
been divided into 5 standard values in view (C). Each pulse is given whatever standard value is
nearest its actual instantaneous value. In view (D), the same amplitude range has been divided
into 10 standard levels. The curve of view (D) is a much closer approximation of the modulating
wave, view (A), than is the 5-level quantized curve in view (C). From this you should see that the
greater the number of standard levels used, the more closely the quantized wave approximates
the original. This is also made evident by the fact that an infinite number of standard levels
exactly duplicates the conditions of nonquantization (the original analog waveform).
Quantization levels. MODULATION

Quantization levels. TIMING

Quantization levels. QUANTIZED 5-LEVEL

Quantization levels. QUANTIZED 10-LEVEL

Figure 2-49 shows the relationship between decimal numbers, binary numbers, and a pulse-code waveform that
represents the numbers. The table is for a 16-level code; that is, 16 standard values of a quantized wave could
be represented by these pulse groups. Only the presence or absence of the pulses are important. The next step
up would be a 32-level code, with each decimal number represented by a series of five binary digits, rather than
the four digits of figure 2-49. Six-digit groups would provide a 64-level code, seven digits a 128-level code, and so
forth. Figure 2-50 shows the application of pulse-coded groups to the standard values of a quantized wave.
Figure 2-49. - Binary numbers and pulse-code equivalents.
Pulse-code modulation of a quantized wave (128 bits).

In figure 2-50 the solid curve represents the unquantized values of a modulating sinusoid. The dashed curve
is reconstructed from the quantized values taken at the sampling interval and shows a very close agreement
with the original curve. Figure 2-51 is identical to figure 2-50 except that the sampling interval is four times
as great and the reconstructed curve is not faithful to the original. As previously stated, the sampling rate of
a pulsed system must be at least twice the highest modulating frequency to get a usable reconstructed
modulation curve. At the sampling rate of figure 2-50 and with a 4-element binary code, 128 bits (presence
or absence of pulses) must be transmitted for each cycle of the modulating frequency. At the sampling rate
of figure 2-51, only 32 bits are required; at the minimum sampling rate, only 8 bits are required.
As a matter of convenience, especially to simplify the demodulation of pcm, the pulse trains actually
transmitted are reversed from those shown in figures 2-49, 2-50, and 2-51; that is, the pulse with the
lowest binary value (least significant digit) is transmitted first and the succeeding pulses have
increasing binary values up to the code limit (most significant digit). Pulse coding can be performed in a
number of ways using conventional circuitry or by means of special cathode ray coding tubes. One
form of coding circuit is shown in figure 2-52. In this case, the pulse samples are applied to a holding
circuit (a capacitor which stores pulse amplitude information) and the modulator converts pam to pdm.
The pdm pulses are then used to gate the output of a precision pulse generator that controls the
number of pulses applied to a binary counter. The duration of the gate pulse is not necessarily an
integral number of the repetition pulses from the precisely timed clock-pulse generator. Therefore, the
clock pulses gated into the binary counter by the pdm pulse may be a number of pulses plus the
leading edge of an additional pulse. This "partial" pulse may have sufficient duration to trigger the
counter, or it may not. The counter thus responds only to integral numbers, effectively quantizing the
signal while, at the same time, encoding it. Each bistable stage of the counter stores ZERO or a ONE
for each binary digit it represents (binary 1110 or decimal 14 is shown in figure 2-52). An electronic
commutator samples the 20, 21, 22, and 23 digit positions in sequence and transmits a mark or space bit
(pulse or no pulse) in accordance with the state of each counter stage. The holding circuit is always
discharged and reset to zero before initiation of the sequence for the next pulse sample.
Figure 2-52. - Block diagram of quantizer and pcm coder.
The pcm demodulator will reproduce the correct standard amplitude represented by the pulse-code group. However, it will reproduce
the correct standard only if it is able to recognize correctly the presence or absence of pulses in each position. For this reason, noise
introduces no error at all if the signal-to-noise ration is such that the largest peaks of noise are not mistaken for pulses. When the noise
is random (circuit and tube noise), the probability of the appearance of a noise peak comparable in amplitude to the pulses can be
determined. This probability can be determined mathematically for any ration of signal-to-average-noise power. When this is done for
105 pulses per second, the approximate error rate for three values of signal power to average noise power is:

17 dB - 10 errors per second

20 dB - 1 error every 20 minutes
22 dB - 1 error every 2,000 hours

Above a threshold of signal-to-noise ration of approximately 20 dB, virtually no errors occur. In all other systems of modulation, even
with signal-to-noise ratios as high as 60 dB, the noise will have some effect. Moreover, the pcm signal can be retransmitted, as in a
multiple relay link system, as many times as desired, without the introduction of additional noise effects; that is, noise is not cumulative
at relay stations as it is with other modulation systems.
The system does, of course, have some distortion introduced by quantizing the signal. Both the standard values selected and the
sampling interval tend to make the reconstructed wave depart from the original. This distortion, called QUANTIZING NOISE, is initially
introduced at the quantizing and coding modulator and remains fixed throughout the transmission and retransmission processes. Its
magnitude can be reduced by making the standard quantizing levels closer together. The relationship of the quantizing noise to the
number of digits in the binary code is given by the following standard relationship:
n is the number of digits in the binary code

Thus, with the 4-digit code of figure 2-50 and 2-51, the quantizing noise will be about 35 dB weaker than the peak signal which the channel will

The advantages of pcm are two-fold. First, noise interference is almost completely eliminated when the pulse signals exceed noise levels by a
value of 20 dB or more. Second, the signal may be received and retransmitted as many times as may be desired without introducing distortion into
the signal.