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Analog and Digital Communications
Concepts
Representing data as Analog Signals
Converting Analog Data-to-Analog Signals
During the early development stages of copper-based analog
telephone networks it was discovered that human voice
could be carried adequately from 300Hz to 3330 Hz. As a
result telephone networks were originally designed to
transmit voice conversations within a range of 3000 Hz.
Voice signals that generate a signal less than 300 Hz or
greater than 3330 Hz are discarded.
There are 2 basic ways in which analog data are represented
as analog signals:
 At their original frequency, called a baseband signal
 Or at a different frequency
When we pick up a telephone and speak into it, the
telephone network transmits our analog voice signal at face
value (somewhere between 300 and 3330 Hz). This face
value is an example of a baseband signal.
Alternatively, the telephone network can combine our signal
into another signal (called a carrier), and then transmit these
combined signals at a different frequency.
A carrier is a continuous signal that operates at a predefined
frequency. Changing a carrier so that it can represent data in
a form suitable for transmission is called modulation.
Characteristics of a carrier that can be modified are
Amplitude, Frequency and Phase.
Modifying the Amplitude of a wave is called
Amplitude Modulation
(Changing the signal’s strength)
Modifying the Frequency of a wave is called
Frequency Modulation
(Changing the signal’s pitch)
Modifying the Phase of a wave is called Phase
Modulation
(Temporarily delaying the natural flow of the
waveform)
Converting Digital Data-to-Analog signal
Transmitting digital data (output from a computer) across an
analog based communication network, is done by modifying
(modulating) a continuous signal (Carrier) at the sender end
so that the signal conforms to the digital data being
transmitted, and then converting the signal back into digital
form at the receiver.
The device that performs these functions is called a modem,
which is a contraction of the words modulator and
demodulator. Two modems are required- one at each end of
the transmission line- and both modems must use the same
modulation technique. A sending modem first produces a
continuous carrier signal and then modified the signal using
a specific modulation technique. When the modified signal
is received by the receiving modem, the signal is then
converted back into digital form.
When used in the context of converting digital data into
analog signals the modulations techniques are called:
 Amplitude-shift keying (ASK)
 Frequency-shift keying (FSK)
 Phase-shift keying (PSK)
ASK this process involves varying the signal voltage
while keeping the frequency constant. One amplitude
is used to represent a binary 0 and a second
amplitude is used to represent a binary 1. A sending
ASK modem, generates a continuous carrier and used
this unmodulated signal to represent a binary 0; the
modem also modulates this signal (using ASK) to
represent a binary 1.
FSK Alters the frequency (cycles per second) of a
signal so that it conforms to the digital data.
Amplitude is kept constant. One frequency is used to
represent a binary 0 and a second frequency is used
to represent a binary 1. A sending FSK modem
generates a continuous carrier and let’s the signals
standard frequency represent a binary 0. Whenever a
1 needs to be converted, the modem can either reduce
or increase the frequency.
PSK modifies the phase angle of a carrier wave
based on the digital data being transmitted. The
changes in phase angle are what convey the data in a
phase-modulated signal.
Changing the phase of a wave enables data
encoding with more than one bit of data at a
time. If a phase shift occurs at 0, 90, 180 and
270 0 then 2 bits of information (called dibits)
can be transmitted for each signal change.
These dibits pair 00, 01, 10, and 11
respectively. Similarly if phase shift occur at
8 different angles (0, 45, 90, 135, 180, 225,
270 and 3150 then these signal represent three
bits of information (tribits). These tribits are
respectively 000, 001, 010, 011, 100, 101,
110, and 111. As a result PSK can encode up
to three bits per baud.
QAM Phase shift can also be combined with
Amplitude modulation. One common strategy
employed is called Quadrature Amplitude
Modulation, which uses eight phase changes and two
amplitudes to create 16 different signal changes. A
QAM can encode between four and seven bits per
baud. A modified version of QAM called trelliscoded modulation (TCM) incorporates extra bits for
error-correction. Both QAM and TCM provide high
data transfer rates because they are able to
incorporate several bits per signal change.
Modem Demo
Representing Data as Digital Signals
As digital technology and computer data applications
emerged, analog technology was unable to separate data
from noise in a satisfactory manner. This led to the
introduction of digital signalling, which requires converting
analog signals to digital signals.
Converting Analog-Data-to-Digital signals
Representing Analog data as digital signals requires
converting the data’s corresponding analog signal, which is
in the form of a sine wave, into digital signal, which is
represented by 0’s and 1’s The most common approach is a
process known as pulse-code modulation (PCM), and
involves two steps: Sampling and coding.
Digitizing an analog signal requires taking regular samples
of the amplitude of the signal’s waveform over time so that
the generated digital signal matches the corresponding
analog signal. According to a sampling theorem known as
Nyquist’s Rule, if an analog signal is sampled at regular
intervals and at twice the highest frequency on the line, then
the sample will be an exact representation of the original
signal.
Even though voice transmission requires 3000 Hz
bandwidth, phone companies allocate 4000 Hz and install
filters at the 300 Hz and 3300 Hz. Therefore actual sampling
is 8000 per second when converting voice to digital form.
Once a sample has been taken, it must be converted into
binary digit where 0 and 1 represent the absence or presence
of voltage, respectively. Determining whether a sample gets
coded 0 or 1 depends on where the sample was taken.
If an eight-bit sample is used, then the sound wave can be
partitioned into 256 (28) possible points. The first 128 points
(0 to 127) get coded 00000000, and the last 128 (128 to 255)
get coded 00000001. After the sampling and coding steps are
complete, the resulting digital codes are then transmitted as a
digital signal waveform.
Digitizing an analog signal via PCM is done using a device
called a codec (coder-decoder), which can be thought of as
the opposite of a modem. A codec converts analog data into
a digital signal; a modem on the other hand converts digital
data into an analog signal.
This is an example of what happens when a computer using
a modem communicates with another computer equipped
with a modem over the telephones lines.
The telephone company maintains analog lines from clients
to the local switching stations. In between switching stations
lines are digital. The signal from the computer at the sending
end has to be converted from digital to analog, when the
signal arrives at the switching station it is converted again
from analog to digital. Again when the signal arrives at the
second switching station is has to be converted again to
analog…. Finally at the destination computer the signal is
converted again from analog to digital… In total it required
4 conversions for these two computers to effectively
communicate.
This is the role of a modem (modulator / demodulator) to
convert a digital signal into an analog one at the sending end
and to convert it from analog to digital at the receiving end.
Converting Digital data-to-Digital Signals
Transmitting digital data across a digital network (Ethernet
LAN) requires representing the digital data as a digital
signal. Three common techniques are used for this task are:
 Manchester encoding
 Differential Manchester coding
 Non-Return to Zero, Invert on ones (NRZI)
Manchester encoding Used on Ethernet/802.3
networks. Its main benefit is error recovery. A 1 is
sent as a half-time-period low followed by a halftime-period high, and a 0 is sent at half-time-period
high followed by a half-time-period low.
Consequently, the end of the last transmitted bit is
easily determined immediately following the
transmission of the last bit.
Differential Manchester Encoding used on token
ring networks. Similar to Manchester encoding: each
bit-period is partitioned into two intervals and a
transition between high and low occurs during each
bit period. The difference in between the two
methods is in the interpretation of this transition. In
Differential Manchester Encoding the interpretation
of low-to-high or high to low is a function of the
previous bit-period. More specifically, the presence
of a transition at the beginning of a bit-period is
coded 0, and the absence of a transition at the
beginning of a bit-period is coded 1.
NRZI used on Fast Ethernet and FDDI (Fiber
Distributed Data Interface). Instead of user level
voltages to encode the data, encoding is based n
transitions from one voltage state to another. Data are
coded 0 if no transition occur, but are coded 1 at the
beginning of a transition.
An application of NRZI is an encoding strategy
known as 4B/5B (four-bits to five bits) method. The
4B/5B-encoding scheme takes data in four-bit codes
and maps it to corresponding five-bits codes. By
transmitting five-bits codes using NZRI, a logical 1bit is transmitted at least once every five sequential
data bits resulting in a signal transmission. This
4B/5B scheme makes it possible for a LAN to
operate at 125MHz and provides a data rate of
100Mb/sec. Note that the use of one extra bit for
every five translate to 20% overhead for every clock
encoding. In contrast, Manchester Encoding requires
50% bandwidth overhead for clock encoding because
it guarantees at least one signal transition for every
bit transmitted.
Digital Carrier Systems
T1 and DS circuits
Digital signalling and the T-carrier system were introduced
to resolve attenuation and noise amplification.
 Repeaters are being used to regenerate signals
 T-carrier system, which uses TDM (Time Division
Multiplexing) to support multiple channels in a
single digital signal, was the first systems designed to
implement digitized voice transmission.
T1, a product of T-carrier, describes the multiplexing of 24
separate voice channels into one single wideband digital
signal.
 A T1 frame consists of 193 bits – eight bits
per channel and one bit plus one bit for
framing. (168 for data, 24 for control and 1
for synchronization)
 Each voice channel is digitized using PCM
(pulse-code modulation) and has a data rate
of 64 kbps
 When multiplexed into a digital signal a voice
channel is referred to as a Digital signal at
level 0 (DS-0), thus DS-0 has a data rate of 64
kbps.
 A T1 circuit carries a DS-1 signal, which
consists of 24 DS-0 channels plus one 8 kbps
channel reserved for framing. This results in
an aggregate bandwidth of 1.544 Mbps.
o Data 56,000 bps per channel at 24
channels = 1,344,000 bps
o Control 8,000 bps per channel at 24
channels = 192,000 bps
o Framing 8,000 bps for frame
synchronization = 8,000 bps
 Two T1 lines are combined to from a T1C
circuit rated at 3.152 Mbps
 A T2 circuit (DS-2) consists of 4 multiplexed
T1 circuits and has an aggregate bandwidth of
6.312 Mbps
 A T3 link (DS-3) consists of 28 multiplexed
T1 circuits with an aggregate bandwidth of
44.736 Mbps

A T4 channel (DS-4), rated at 274.176 Mbps,
consists of 168 multiplexed T1 circuits.
NADH Lines have the same meaning in Australia, Japan as
it does in North America. However in Europe, South
America, Africa, part of Asia and Mexico, an analogous
service called E-1 is used in these locations. An E-1 carrier
supports thirty 64 kbps channels plus 2 signalling and
control channels. E-1 links can be multiplexed into higher
capacity lines.
A T1 circuit requires special termination equipment called
CSU/DSU
 A Channel Service Unit (CSU) regenerates
the signal, monitors the line for electrical
anomalies, provides proper electrical
termination, performs framing, and provides


remote loopback testing for diagnosing line
problems.
A Data (or Digital) Service Unit (DSU)
provides the interface to connect a remote
bridge, router, or switch. Also provide flow
control between the network and the CSU.
An E-1 circuit is terminated using a Network
Termination Unit (NTU), which provides
broadly similar CSU/DSU functionality.
Document on CSU/DSU
Fractional T1 (FT1)


Provides a fraction of a T1 capacity.
Achieved by combining multiple DS-0 channels
o 128 kbps => 2 DS-0 channels



o 256 kbps => 4 DS-0 channels
o 512 kbps => 8 DS-0 channels
When ordering FT1 you actually get a full T1
channel, but only pay for the number of DS-0
channels you use.
Less efficient than ISDN (Integrated Services Data
Networks, Chapter 12), in terms of bandwidth
available for data communication. In FT1 controls
are inband, meaning that the 8 kbps required for
control are amongst the 64 kbps bandwidth. On the
other hand on ISDN separate channels are used for
control and therefore provides a full 64kbps
bandwidth.
FT1 service is attractive to customers who do not
require a full T1 service but need more capacity than
an ISDN (64/128-kbps) line.
SONET and OC Circuits
SONET (Synchronous Optical NETworks) and SDH
(Synchronous Digital Hierarchy) are both International
standards that provide specification for high-speed digital
transmission via optical fibre. This involves converting
signals in electrical to optical form at the source and an
optical-to-electrical at the destination.
o SONET Developed by ANSI
o SDH developed by ITU-T, drafted after
SONET and incorporates it
o Both standards work at the physical layer of
the OSI model
o The building block of the SONET signal
hierarchy is STS-1 (51,84 Mbps) The line rate
is derived from the STS-1 frame and consists
of 810 eight-bit bytes transmitted at 8000 Hz.
o In the ANSI world, SONET’s terminology
include Optical Carrier level (OC-n) and
signals over copper Synchronous transport
Signal (STS-n)
o In the ITU-T world, the official term used is
Synchronous Transport Module (STM-n).
Advantages over copper based (T1) hierarchy
o Hundreds of thousands of simultaneous voice
and data transmission are possible
o Immune to EMI (Electro Magnetic
Interference)
o Fibre is available in either single mode or
multimode, therefore making it suitable for
being used for LAN connections or as the
backbone for a WAN.
o Bandwidth can be allocated on an as-needed
basis, routes can be dynamically reconfigured
o Can serve as the transport facility for any type
of network technology or service, including
ATM (Asynchronous Transport Mode), FDDI
(Fiber Distributed Data Interface), SMDS
(Switched Multimegabit Data Service) and
ISDN (Integrated Services Digital Network).
o Can support various topologies including
point-to-point, star, and ring.