ECE 513 - Data Communications Part 1

Course: IT 105
Data Communications
Prepared by: Armando V. Barretto
Course Objectives
Learn the fundamental principles used in Data Communications
Learn existing data communications devices and facilities
Learn to apply existing data communications technologies
Grading System
Grade = (Q1 + Q2 + Q3+ Q4 + 2 P.E + 2 F.E) / 4
Passing Grade >= 60
References:
Data Communications and Networking by Wayne Tomasi
Electronic Communications Systems by Wayne Tomasi
Data Communications and Networking by Behrouz A. Forouzan
Data Communications and Networking by William Stallings
Network Fundamentals by Mark Dye, Rick McDonald, and Antoon
Rufi
Wikipedia
Introduction to Data
Communication
Data Communications
The transmission, reception, and processing of digital
information (Wayne Tomasi).
Original source information can be human voice,
alphanumeric characters stored in databases, or other forms.
Process of transferring digital information between two or more
points (Wayne Tomasi).
Data generally are defined as information that is stored in
digital form.
Information are knowledge or intelligence.
Transmission of data between devices, using electronic,
electrical or light signals.
between computers
between computers and peripherals
between other electronic devices
Importance of Data Communications
Computers and peripherals need to communicate.
Computers need to communicate with other computers, such as in
computer networks.
Resources have to be shared
Different types of computers need to communicate
Other electronic equipment need to communicate with other electronic
equipment.
TYPICAL DATA COMMUNICATIONS NETWORK
PC PC Modem Modem
Cables
PC PC
Cables
(Modem not required)
Telecommunications
Facilities
PC
PC
PC
PC
Modem
Modem Modem
Mainframe
Modem
Front End
Processor (FEP)
FEP is a DTE which directs traffic
to and from many different
circuits, which could have different
parameters, such as codes, and data
formats.
LCU is a DTE that directs traffic
between one data communication
medium and a relatively few
terminals which uses the same
protocols, character codes, and
other parameters
Line control Unit
(LCU)
BLOCK DIAGRAM OF BASIC DATA
COMMUNICATIONS SYSTEM
DTE
Communications Medium
(Cable)
DTE DCE DCE
DTE
DTE
DCE
DCE
(earths atmosphere)
Radio signals
DTE - Data Terminal
Equipment
DCE - Data Communications Equipment /
Data Circuit Terminating
Equipment
Example: modem
BLOCK DIAGRAM OF BASIC DATA
COMMUNICATIONS SYSTEM
Telecommunications /
Transmission
Facilities
DTE
DTE
DTE
DTE
DCE
DCE
DCE DCE
DTE - Data Terminal
Equipment
DCE - Data Communications Equipment /
Data Circuit Terminating
Equipment
Example: modem
DATA COMMUNICATIONS WITHOUT USING DCE
DTE
DTE
Note:
It is possible to have data communications without using a DCE in
some instances, such as when DTEs are located close to one
another and DTEs have facilities to establish communications with
each other without using a DCE.
Data Terminal Equipment
Any digital device that generates, transmits, receives, or interprets
data messages.
Could be a computer, printer, POS (point of sale terminal), ATM
(Automatic Teller Machine), or other electronic devices.
Where information originates or terminate
Data Communications Equipment
Equipment that interfaces data terminal equipment to a
communications medium or channel, such as a telephone line.
Devices used to convert signal from a DTE into a signal which is
suitable for transmission in the communications medium or
transmission facilities
Input and/or output signals of a DCE could be digital or analog
depending on the DTE and the communications medium /
facilities used.
Also referred to as Data Circuit Terminating Equipment
Transmission path between DCEs, or between DTE and another
DTE
Could be copper cables, fiber optic cables, earths atmosphere,
earths surface, free space, or other suitable medium
Common Terms Used In Data Communications
Information
Knowledge or intelligence
Could be human voice, music, alphanumeric characters stored in
database of computers,etc.
Data
Information stored in digital form
Could be stored in computers using hard disks, magnetic tapes, and
nonvolatile memory
Data Transmission code
Information converted to a binary code
Data Communication Code
Used to represent characters and symbols
Includes character codes, character sets, symbol codes, character
languages
Bandwidth
Range of frequencies contained in a frequency spectrum
Equal to highest frequency minus the lowest frequency which
could be used / transmitted
Information theory
A highly theoretical study of the efficient use pf bandwidth to
propagate information through electronic communications systems.
Bit
Binary digit
Most basic digital symbol used to represent information
Could be a 1 or a 0
Mark
Refers to binary 1
Space
Refers to binary 0
Block / Frame / Packet
A group of bits transferred as a unit
Bit rate
Refers to rate of change of a digital information signal
Number of bits transmitted per second (bits per second)
Information Capacity
Measure of how much information can be propagated through a
communications system
Represents the number of independent symbols that can be carried
through a system in a given unit of time
Usually dependent on bandwidth and frequency used in the
communications system
Baud Rate
Refers to rate of change of signal in a transmission
(communications) medium after encoding and modulation have
occurred
Number of signaling element per second in a transmission medium
Equal to 1 / time of one output signaling element
May or may not be equal to Bits per second
Station or Node
An endpoint where subscribers gain access to the data
communications circuit
Communication Link
Path for transmission of signals between communicating devices
Data Communications Circuit
Provides a transmission path between locations used to transfer
digital information from one station to another.
Channel
Specific band of frequencies allocated to a particular service of
transmission
Ex.: 3 kilohertz voice channel, FM radio broadcast channel
Protocol
Formal set of conventions governing how communications should
take place in a communications system
Defines procedures that the systems involved in the
communications process will use
Protocol stack
The list of protocols used by a system
Network
Set of devices (sometimes called nodes or stations) interconnected
by communications media links
Data communications network
Systems of interrelated computers and computer equipment that
are interconnected to one another, for the purpose of transmitting
and / or receiving information.
Computer network
Two or more computers interconnected with one another for the
purpose of sharing resources such as printers, databases, files, and
backup devices
Analog Signal
Signal whose amplitude continuously varies in time
Ex.: voice signal
Digital signal
Signals which are discrete; their amplitude maintains a constant
level for a prescribed period of time.
Consists of on-off pulses
Ex.: Binary code of alphanumeric characters, signals being
processed by digital computers
Two Basic Types of Electronic Communication System
Analog Communication System
System in which signals are transmitted in analog form (a
continuously varying signal such as a sine wave)
Examples: AM and FM radio
Digital Communication System
System in which signals are transmitted and received in digital
form (discrete levels such as on (+ voltage) and off (0 volts)
pulses.
Example: T1 and E1 communication systems
Digital information converted into analog form prior to transmission,
can be transmitted through an analog communications system.
Analog information converted into digital form prior to transmission,
can be transmitted through a digital communications system.
Communications facilities could use a combination of analog
communications systems and digital communications systems.
Telephone system could use analog transmission from subscribers
premises to switching sites, while it could use digital transmission
between switching sites
Transmission Modes
Simplex
Transmissions can occur only in one direction
Also called one-way-only, receive only, or transmit-only systems
A location may be a transmitter or receiver but not both.
Example: AM and FM radio
Half Duplex
Transmissions can occur in both directions, but not at the same
time
Also called two-way-alternate, either-way, or over-and-out systems
A location may be a transmitter and a receiver but not both at the
same time.
Example: two way radios that use push-to-talk (PTT) buttons to
activate their transmitters such as citizens-band and police-band
radios
Transmission Mode
Full Duplex
Transmissions can occur in both directions at the same time.
Also called two-way-simultaneous, duplex, or both-way-lines.
A location can transmit and receive at the same time. However, the
station it is transmitting to must also be the station it is receiving
from.
Example: standard telephone system
Full Full Duplex
A station can transmit and receive simultaneously, but not
necessarily to and from the same station. One station can transmit
to a second station, and receive from a third station, at the same
time.
Some Codes Used in Data Communications
ASCII (American Standard Code for Information Interchange)
Seven bit code widely used today.
It is almost always transmitted with a parity bit in which case, it becomes an
8 bit code (if parity checking is used)
Extension of six-bit trans code
Extended ASCII code
Developed by IBM
Forbids using parity bit
All 8 bits can be used to represent characters.
Codes 00 hex to 7F hex are backward compatible to standard ASCII code.
BAUDOT
Named after Emil Baudot who invented the first constant length teleprinter.
Fixed length 5 bit code used for telegraph, and is also called Telex code.
Less characters / codes can be used (2
5
)
Uses figure shift and letter shift control characters to expand its capability to
58 characters.
Some Codes Used in Data Communications
EBCDIC (Extended Binary Coded Decimal Interchange Code)
Developed by IBM
Uses 8 bits (256 codes are possible)
Does not facilitate the use of parity bit
LSB is designated b7, MSB is designated b0, such that b7 is transmitted first
and b0 is transmitted last.
Morse Code (International Morse Code)
First character set which was developed by Samuel Morse.
Variable Length Source Code which uses dot, dash, and space symbols
Was used in telegraph, not suited for modern data communications
Literally requires reasoning ability of human brain to decode.
Bar Codes
Series of vertical black bars separated by vertical white bars
Examples are: Code 39, Universal Product Code, POSTNET Bar Code
Serial Data Transmission
Bits are transmitted one at a time (serial by bit)
One communication link is used for transmission
Slower compared to parallel data transmission
Less costly compared to parallel data transmission
Used for short or long distance communications
Example: Com 1 and Com 2 ports of PCs
Parallel Data Transmission
More than one bit are transmitted at the same time.
More than one communication link is used for transmission.
Faster compared to serial data transmission.
More costly compared to serial data transmission.
Used for short distance communications.
Example: parallel printer port (Centronics) of PCs
ASYNCHRONOUS AND
SYNCHRONOUS DATA
TRANSMISSION
Asynchronous Data Transmission
Uses serial data transmission.
Characters are transmitted and received one at a time.
Sometimes called start/stop transmission.
Uses start and stop bits.
Start bit is always a 0.
The logical status of the communication line when there is no data being
transmitted is always a 1 (idle line ones).
The 1 to 0 transition of the start bit activates a circuit at the receiver to
check if a valid start bit has arrived.
Stop bit is always a 1 (to make sure that there will be a 1 to 0 transition for the
next start bit).
Receiver clock is not synchronized with transmitter clock.
Frequencies of the transmitter and receiver clocks must be sufficiently close if
not the same. (bit rate or baud rate configuration must be the same). Otherwise,
clock slippage may occur.
If transmit clock is substantially lower than receive clock, underslipping
occurs. The reverse causes overslipping.
Framing characters individually with start and stop bits is sometimes said to
occur on a character by character basis.
Relatively slow if a lot of data are to be transmitted continuously.
More efficient for short messages.
Typically used with dumb terminals
DTE DTE DCE DCE
Medium
0 volts
Start
Bit
Stop
Bit
Data Bits
0 1
Signal inside DTE
While precise synchronism between transmitter and receiver is not required,
receiving station and transmitting station must have the same set up
regarding:
Number of bits for data
Transmission speed (bit rate)
Framing error results when transmission speed for transmitter and
receiver are not the same. This is because incorrect timing in the
sampling of received data results.
Framing is the process of deciding which groups of 8 bits constitute a
character.
Use of parity checking (odd, even, disabled)
If set up for parity are not the same, receiver could interpret that there
was a Parity Error even if there was none.
Flow control procedure to prevent data loss due to insufficient memory
used as buffer for received data.
Overrun error results when receiver buffer (temporary storage for
received data) becomes full and additional data are stored in it while
previously stored data have not yet been processed.
Data Overrun occurs when a a character arrive and it cannot be
handled by the receiver.
Typical Asynchronous Serial Data Receiver
Circuit which determines
when a valid start bit has
been detected
Shift register
Spike
Detect
Enable
Count 8
Ticks
To computer
data bus
Sampling Clock
(16 times the bit rate)
Circuit to detect received data being
0 (assuming a character is not being
assembled, i.e. start bit
Divide by
16
Flag
Bit
sampling
clock
Received data
Reset
Enable
for bit
sampling
clock
Typical Asynchronous Serial Data Receiver
16 x clock samples the incoming data line at 16 times the anticipated
bit rate, to detect the 1 to 0 transition (start bit).
Spike detection circuit counts 8 ticks of the 16 x clock and checks the
line to see if it is still in the 0 state.
If the line is still in the 0 state, a valid start bit has presumably
arrived.
If the line has returned to a 1 state, it is assumed that the initial 1 to
0 transition was due to noise.
If spike detection circuit has detected a valid start bit, circuit enables a
counter which divides the 16 x clock by 16 to produce bit sampling
clock which ticks once per bit time.
Bit sampling clock is used to store the succeeding incoming bits into
the shift register. Sampling is done at the center of each bit.
The contents of the shift register are transferred into the receiver buffer
(memory for received data) for further processing.
The flag is used to indicate that a character has arrived.
Synchronous Data Transmission
Uses serial data transmission.
Does not use start and stop bits.
Usually uses start and ending flags.
Receiver clock is synchronized with transmitter clock.
Usually, more than one character is transmitted in one packet.
Relatively fast if a lot of data are to be transmitted continuously.
Synchronous Data Transmission Protocols
Bisync (Binary Synchronous)
developed by IBM
Uses half duplex error control procedures
SDLC (Synchronous Data Link Control)
developed by IBM
Uses full duplex error control procedures
relatively faster compared to Bisync
HDLC (High Level Data Link Control)
Typical SDLC Frame
Ending
Flag
Frame
Check
Sequence
Payload
(User Data)
Starting Flag
Control
Secondary
Station
Address
NR P/F FI NS
NR P/F RR FI
Information
Frame
S I Frame
8 bits 8 bits 8 bits 8 bits N bits 8 or 16 bits
NR receive sequence number
NS send sequence number
P/F poll / final bit
FI - frame indicator
RR Receiver ready
ERROR DETECTION AND
CORRECTION
Error Detection And Correction
Data communications errors can be generally classified as:
Single bit only one bit in data string is in error
Multiple bit two or more nonconsecutive bits within a given data
string are in error.
Burst two or more consecutive bits within a data string are in
error
Error performance is the rate in which errors occur, which can be
describes as:
expected value pertains to probability of error as expected in a
system. Ex: P(e) 10
-4
means 1 bit is expected to be with error out
of 10000 bits transmitted
empirical value pertains to actual error performance of a system
which is called bit error rate (BER). Ex.: If 1 bit has an error out
of 1 million bits transmitted, then BER is equal to 10
-6.
Typically, BER is measured and compared with the probability of
error.
Error Detection And Correction
Error control can be divided into two general categories which are:
Error detection - process of monitoring data and determining
when transmission errors have occurred.
Error correction - Process of correcting the errors which occurred
during the transmission of data.
The common error detection techniques are:
Redundancy
Echoplex
Exact-count coding
Redundancy checking which includes:
Vertical redundancy check (Character parity check)
Checksum
Longitudinal redundancy check
Cyclic redundancy check
Common Error Detection Techniques
Redundancy is a form of error detection where each data unit is sent
multiple times, usually twice.
Receiver compares the two data units to detect errors.
When the data unit is a single character, it is called character
redundancy.
When data unit is a message, it is called message redundancy.
Echoplex (echo checking) used almost exclusively with data
communications systems involving human operators working in real
time at computer terminals or PCs.
Received data are retransmitted to the transmitting station and
displayed on the transmitting station screen, so that operators could
check if what they typed are correct.
Exact count coding - the number of binary 1s (and binary 0s) in each
character is the same.
Example is ARQ code.
Receiver detects error if the number of 1s (or 0s) is different from
the number of 1s (or 0s) supposed to be received.
Redundancy checking is the process of adding additional bits to data units
to check for transmission errors. It includes:
Vertical redundancy check (VRC or character parity checking)
Checksum
Longitudinal redundancy check (LRC)
Cyclic redundancy check (CRC)
Vertical Redundancy Check (VRC or Character Parity Check)
Probably the simplest error-detection scheme
Also called as character parity or simply parity check
Mostly used for asynchronous data communications
Each character has its own parity bit.
Parity check could be odd parity (total number of 1 is odd) or even parity
(total number of 1 is even).
Not efficient if a lot of data are to be transmitted continuously.
May be used in asynchronous or synchronous data transmission.
Errors will not be detected if even number of 1s or 0s are in error.
Other forms of parity include space parity (parity bit always a 0),
marking parity (parity bit always a 1), no parity (parity bit not sent or
checked), and ignored parity (parity bit is 0 and is ignored).
Checksum redundancy checking where the data is summed together to produce an error
checking character (Checksum).
Checksum is appended to the data at the end of the message.
Receiver adds the received data and compares result to received checksum to detect
errors.
The five primary ways of calculating a checksum are: Check character, single
precision, double precision, Honeywell and residue.
Check character checksum decimal value is assigned to each character, which are
added together to produce the checksum character. There are variations of check
character checksum.
character) (one bits 5 is length 8 32 - 40 is Checksum
40 is 10 of multiple highest Next
32 bits all of Sum
4220 - 85281 Code
character) per bits (5 code Bar POSTNET For Ex.
= =
=
Single precision checksum binary addition is performed on the data to
produce checksum character. If a carryout occurs, the carry bit is ignored.
Double precision checksum The same as single precision checksum except
that checksum character is two times longer than the character, so that
carryout can be included in the checksum character.
character) of that as length same bits, (8 checksum as sent is 01110100 and discarded is carry
carry with sum 101110100
01001111
01001100
01001100
01000101
01001000 Ex.
character) a of length times 2 bits, (16 checksum as sent is 01110100 00000001 and discarded not is carry
carry with sum 101110100 000000
01001111
01001100
01001100
01000101
01001000 Ex.
Direction of transmission
Direction of transmission
Honeywell checksum form of double precision checksum wherein the checksum
character is two times longer than the character. The checksum is based on
interleaving consecutive data words to form double length words.
Residue checksum - binary addition is performed on the data to produce checksum
character. If a carryout occurs, the carry bit is added to the LSB of the sum.
80 65 23 34 42 46
Hex 80 65 is checksum
23
65 80 34
23 34 42
42 46 46 Ex.
carry with sum 101110100
01001111
byte) t significan least the to added carry was (checksum, 01110101 01001100
(carry) 1 01000101
01110100 01001000 Ex.
Transmission
Longitudinal Redundancy Check (LRC)
Also called message parity and horizontal redundancy check
Each bit position has a parity bit, and the parity bits are transmitted with the
data.
LRC character is sometimes called block check character (BCC), frame check
character (FCC), block check sequence (BCS), or frame check sequence (FCS)
Errors will not be detected if even number of 1s or 0s are in error.
Practical to be used in synchronous data transmission only.
Can be used together with VRC to make error detection more effective.
For single bit errors, VRC used together with LRC will identify which bit is in
error.
VRC and LRC can be combined
position bit each for parity odd using LRC 1 0 0 11110
01001111
01001100
01000101
01001000 Ex.
Cyclic redundancy check
Error detection technique wherein the data transmitted is processed
at the transmitter according to a set rule (such as division by a
polynomial).
The remainder (CRC) of the processing (dividing) is appended to
the data and transmitted with the data.
The receiver processes received data and CRC according to the set
rule, to detect errors.
Considered as a systematic code.
Probably the most reliable redundancy checking technique
Approximately 99.999 %of all transmission errors are detected.
Used in synchronous data transmission
Popular versions are:
CRC 12 - 12 bit redundancy code used for 6 bit characters
CRC ITU - 16 bit redundancy code which is a European
standard
CRC 16 - 16 bit redundancy code used for 8 bit codes such as
ASCII and EBCDIC, or 7 bit codes using parity
CRC 32
Cyclic redundancy check (continuation)
Cyclic block codes are often written (n,k) , where n = bit length of
transmission, and k = bit length of message (data).
Length of BCC (CRC code) = n k.
Block check character (BCC) is the remainder of a binary division
process.
A data message polynomial G(x) is divided by a unique generator
polynomial function P(x).
The quotient is discarded, and the remainder is truncated to 16 bits
and appended to the message as a BCC.
The generator polynomial must be a prime number.
With CRC generation, division is not accomplished by arithmetic
division but by modulo-2 division, where the remainder is derived
from an exclusive or operation.
Mathematically, CRC can be expressed as:
G(x) / P(x) = Q(x) + R(x)
Where: G (x) = message polynomial (data polynomial, dividend)
P (x) = Generator polynomial (Used as divisor)
Q (x) = quotient (discarded, not sent with data)
R (x) = remainder (appended to the data)
The generator polynomial P (x) for several common CRC standards are:
CRC 12 X
12
+ X
11
+ X
3
+ X
2
+ X
1
+ X
0
(1100000001111)
CRC ITU - X
16
+ X
12
+ X
5
+ X
0
(10001000000100001)
CRC 16 - X
16
+ X
15
+ X
2
+ X
0
(11000000000000101)
Where X
0
= 1
The number of bits in the CRC (BCC or BCS) code is equal to the
highest exponent of the generating polynomial P(x).
The exponents identify the bit positions in generating polynomial
P(x) that contain a logic 1.
Example: Determine the BCC or BCS (CRC) for the following data:
Data G (x) = X
5
+ X
4
+ X
1
+ X
0
(110011)
CRC P (x) = X
4
+ X
3
+ X
0
(11001)
1. First, the data or message polynomial G(x) is multiplied by X
n-k
where n-k is the number of bits in BCC (CRC code).
X
4
(X
5
+ X
4
+ X
1
+ X
0
) = X
9
+ X
8
+ X
5
+ X
4
(1100110000)
2. The product is divided by P (x) (modulo-2 division remainder is
derived from XOR operation)
) X has P(X) of because bits (4 BCC R(x) remainder 1001
11001
10000
11001
100001
1100110000 11001
4
= =
P(x)
Q(x)
3. The remainder R (x) is appended to the data G (x) to give
1100111001 which is transmitted to the receiver.
4. At the receiver, the received data together with the appended
remainder is divided by P(x) to produce no remainder. If there
is a remainder, the receiver interprets that there is an error in
the received data.
error no is there if 0 be should remainder 0
11001
11001
11001
100001
1100111001 11001

Data BCC
P(x)
The division is done in binary without carries or borrows.
X-OR operation is used to generate the remainder.
15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 + + +
X
2
X
15
X
16
Data Input
LSB MSB
XOR
XOR XOR
BCC output
CRC 16 generating circuit is shown below:
Data is serially applied to the register.
Initially, all bits at the shift register are 0.
As the data bits enter the circuit, They are X-Ored with the
corresponding bits indicated. (Ex. LSB is X-Ored with bit
entering the shift register.)
MSB 0 1 0 0 0 0 0 0 0 0 0 0 0 1 0 0 1 16
0 1 0 0 0 0 0 0 0 0 0 0 0 0 1 1 0 15
0 0 1 0 0 0 0 0 0 0 0 0 0 0 0 1 1 14
0 1 1 1 0 0 0 0 0 0 0 0 0 0 0 1 1 13
0 1 0 1 1 0 0 0 0 0 0 0 0 0 0 1 1 12
0 1 0 0 1 1 0 0 0 0 0 0 0 0 0 1 1 11
0 1 0 0 0 1 1 0 0 0 0 0 0 0 0 1 1 10
0 1 0 0 0 0 1 1 0 0 0 0 0 0 0 1 1 9
0 1 0 0 0 0 0 1 1 0 0 0 0 0 0 1 1 8
0 1 0 0 0 0 0 0 1 1 0 0 0 0 0 1 1 7
0 1 0 0 0 0 0 0 0 1 1 0 0 0 0 1 1 6
0 1 0 0 0 0 0 0 0 0 1 1 0 0 0 1 1 5
0 1 0 0 0 0 0 0 0 0 0 1 1 0 0 1 1 4
0 1 0 0 0 0 0 0 0 0 0 0 1 1 0 1 1 3
0 1 0 0 0 0 0 0 0 0 0 0 0 1 1 1 1 2
0 1 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 1
LSB 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 Start
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 Shift no.
BCC
BCC Bit no.
Data
CRC 16 Generating Circuit Output Example
There are two basic types of errors which are:
Lost Message one that never arrives at the destination or one that
arrives but is heavily damaged that it is unrecognizable.
Damaged Message one that is recognized at the destination but
contains one or more transmission error.
There are two basic strategies for handling transmission errors:
Error detecting codes transmitted message includes redundant
information to enable receiver to determine if an error occurred.
Ex: Parity bit, CRC, Checksum.
Error correcting codes transmitted message includes sufficient
extraneous information to enable receiver to detect error and which
bit/s is in error.
Primary Methods Used for Error Correction
The three primary methods used for error correction are:
Symbol Substitution
ARQ (Automatic Retransmission Request) or Retransmission
Forward Error Correction (FEC)
Symbol Substitution designed to be used in a human environment
where there is human being at a terminal to analyze the received data
and make decisions on its integrity.
Human being can decide to replace character with error, or request
for a retransmission for messages with error.
Form of selective retransmission
Not suitable for modern day data communications
Also called automatic repeat request
Receiver requests for retransmission of data with error
Could use acknowledgements (ACK) and negative
acknowledgements (NAK) to inform source of information if there
are errors or no errors respectively.
Could be inefficient because of overheads (fields which do not
contain user information)
Messages between 256 and 512 characters long are the optimum
size for ARQ error correction.
Commonly used in data communications
Could be the most reliable method of error correction
(continuation)
There are two basic types of ARQ:
1. Discrete ARQ uses acknowledgements to indicate
successful or unsuccessful transmission of data.
Receiver sends positive acknowledgement (ACK) when
it receives error-free message.
Receiver sends negative acknowledgement (NAK) when
it receives message with error.
If sending station does not receive an acknowledgement
after a predetermined length of time (timeout) , it
retransmits the message (retransmission after timeout)
2. Continuous ARQ (Selective Repeat) used when messages
are divided into smaller blocks that are sequentially
numbered and transmitted in succession without waiting for
acknowledgements between blocks.
Allows destination station to asynchronously request the
retransmission of a specific block/s of data.
Forward Error Correction (FEC)
Only error correction scheme that actually detects and correct
transmission errors without requiring a retransmission.
Redundant bits are added to the message before transmission.
Errors are detected by the receiver using the redundant bits.
When an error is detected, receiver uses the redundant bits to
detect which bits are in error, and to automatically correct the
error.
The number of redundant bits needed to correct errors is much
greater than the number of bits needed to simply detect errors.
Suitable for systems when acknowledgements are impractical or
impossible such as when simplex transmission are used to many
receivers or when communicating to faraway places , such as
deep-space vehicles.
Probably the most popular error correction code is the Hamming
Code.
Hamming Code
Hamming code is used for correcting transmission errors in
synchronous data streams.
It only correct single bit errors, and cannot detect errors which
occur in the hamming bits themselves.
Hamming bits (sometimes called error bits) are inserted into a
character at random locations.
Combination of data bits and hamming bits is called Hamming
Code.
Sender and receiver must agree on where the hamming bits are
placed.
To calculate the number of redundant Hamming bits necessary
for a given character length, a relationship between the character
bits and the Hamming bits must be established.
If a data unit contains m bits and n Hamming bits, the total
number of bits in one data unit is equal to m + n.
Hamming Code (continuation)
Since Hamming bits must be able to identify which bit is in
error, n Hamming bits must be able to indicate at least m + n +
1 different codes.
One code is used to indicate that no errors have occurred, and
the remaining m + n codes is used to indicate the bit position
where an error occurred.
Since n bits can produce 2
n
codes, 2
n
must be equal to or greater
than m + n + 1. Therefore, the number of Hamming bits is
determined by the following expression:
2
n
> or = m + n + 1
Where: n = number of Hamming bits
m = number of bits in each data character
A seven bit ASCII character requires four Hamming bits
(2
4
> 7 +4+1) that could be placed at the end of each character
bits or other locations. This results to 57 % increase in message
length.
The Hamming code uses parity to determine the logic condition
of each hamming bit.
Each Hamming bit equates to the even parity bit for a different
combination of data bits.
Example: The data unit for character A (ASCII) is 1000001.
Hamming bits can be placed as shown below:
Bit no. 1 2 3 4 5 6 7 8 9 10 11
Logic condition n
1
n
2
1 n
4
0 0 0 n
8
0 0 1
The Hamming bits are placed in bit positions 1, 2, 4, and 8 (all
powers of 2).
To determine the logic condition of the Hamming bits, the
following criteria are used:
Bit no. 1 2 3 4 5 6 7 8 9 10 11
Logic condition 0 0 1 0 0 0 0 1 0 0 1
1 0 0 1 Logic conditions
n
8
9 10 11 Bit positions
0 0 0 0 Logic conditions
n
4
5 6 7 Bit positions
0 1 0 0 0 1 Logic conditions
n
2
3 6 7 10 11 Bit positions
0 1 0 0 0 1 Logic conditions
n
1
3 5 7 9 11 Bit positions
Hamming bit
(even parity)
Data bits
If an error occurs in one data bit, one or more of the Hamming bits
will indicate a parity error.
To determine the data bit in error, simply add the numbers of the
parity bits that failed.
Example: If bit 6 is in error, the received bit sequence would be
00100101001.
Parity checks for n
1
and n
8
would pass, but parity checks for n
2
and
n
4
would fail.
To determine the bit position in error, (called syndrome) simply
add the positions of the Hamming bits that are in error.
n2 + n4 = 2 + 4 = 6 (Bit 6 has an error)
To correct bit 6, simply complement bit 6.
UART / USRT / USART
UART (Universal Asynchronous Receiver Transmitter)
An integrated circuit which performs many fundamental functions in
data communications such as:
parallel to serial and serial to parallel conversion
parity bit generation and detection
parity, overrun, and framing error detection
generation and detection of start and stop bits
Formatting of data at the transmitter and receiver
Providing transmit and receive status to the CPU such as when
data have been received or transmitted.
May allow voltage level conversion for the serial interface
although many systems have separate voltage level converters.
Providing means of achieving bit and character synchronization.
Designed for asynchronous transmission
It is located in the serial communications interface of DTEs.
Many DTEs now incorporate the UART functions in larger scale ICs
(UART is combined with other circuit components in one IC.)
Some UART packages include two or more UARTs in one IC.
UART Block Diagram connected to a CPU
RS232 signal converter converts TTL signals from UART to RS232 voltage levels (-3 v
to -25 v = logic 1, + 3 v to + 25 v = logic 0)
Internal registers
Transmit buffer
register
Receive buffer
register
Control
register
Status word
register
UART
RS232
signal
converter
RS232
signal
converter
From
DCE
To
DCE
CPU
Parallel data bus
Address
Decoder
CRS
TDS
SWE
RDAR
RDE
Transmit buffer empty
Transmit clock
Address
bus
Error detection
circuit
Transmit
Receive
.
.
Typical UART Transmitter Block Diagram
Transmit buffer register
Transmit buffer register
Transmit shift register
Transmit shift register empty logic circuit
Status word register
Data, parity, and
stop bit logic
Control
register
Stop Parity 7 / 8 data bits Start
Output
circuit
Parallel data from CPU
Control Word input
(8 bits)
Transmit buffer
empty (TBMT)
To CPU
TCP
(Transmit clock
pulse)
Status word enable
(SWE)
From CPU
Control
register
strobe
(CRS)
Transmit
end of
Character
(TEOC)
Transmit
Serial
Output
(TSO)
UART Transmitter Block Diagram
Control word or mode instruction word (stored in control register)
specifies parameters for the transmitter / receiver such as:
Number of data bits
Parity enable / disable
Odd or even parity
Receive clock baud rate factor
As an example, the control word can contain the following (not necessarily
the same for all UART):
NPB no parity bit ( 0 = parity enabled, 1 = parity disabled)
POE parity odd or even (0 = odd parity, 1 = even parity)
NSB1, NSB2 number of stop bits (01 = 1, 10 = 1.5, 11 = 2)
NDB1, NDB2 number of data bits (00 = 5, 01 = 6, 10 = 7, 11 = 8)
RC1, RC2- Receive clock baud rate factor (00= sync mode, 01 = 1X,
10 = 16X, 11 = 32X)- indicates how many times receive clock is faster
than transmit clock.
Maximum character length is 11 bits (data plus overhead bits), which when
used with ASCII is sometimes called full ASCII.
Status word register is an n-bit register that keeps track of the status of the
UARTs transmit and receive buffer registers. Typical status conditions are:
Transmit buffer empty (TBMT)
Receive parity error (RPE)
Receive framing error (RFE)
Receive data available (RDA)
Receive overrun (ROR)
Data set ready (DSR)
Transmit buffer is a temporary storage for data to be transmitted. Example
includes two transmit buffer to allow new data to be loaded into the UART
while the previous data is being loaded into the transmit shift register.
Transmit shift register does parallel to serial conversion.
Data, parity and stop bit logic generates desired parity and stop bits.
Transmit shift register empty logic circuit determines the status of transmit
shift register such as when shifting out of the transmit data has been
completed.
UART transmit operation basically includes:
Desired control word is loaded into control word register by means of
control word strobe (CWS) pulse from CPU.
Status word is read from Status word register by means of Status Word
Enable (SWE) pulse from CPU.
If transmit buffer is empty, parallel data is loaded into the said buffer
using Transmit Data Strobe (TDS) signal from CPU.
Data are transferred in parallel into transmit shift register when
Transmit End of Character (TEOC) is active , parity and other bits are
added, and then the bits are shifted out.
CPU
UART
Status word enable (SWE)
Parallel data (for transmission)
Transmit data strobe (TDS)
Control word and Control word strobe
Serial data
Typical UART Receiver Block Diagram
Receive shift register
Receive buffer register
Parity checker
circuit
Start bit
Verification circuit
Receive
Serial
data
Receive data bits (RD0 to RD7)
(Parallel data)
(to CPU)
Status word register
Status Word
Enable (SWE)
Control register
(for transmit and receive)
Receive clock pulse (RCP)
Receive data
Enable (RDE)
Receive
Parity
Error (RPE)
Receive
Framing
Error (RFE)
Receive
Overrun
Error (ROE)
Receive
Data Available
(RDA)
Receive data
available reset (RDAR)
Receiver and transmitter have the same number of stop bits, type of
parity check, and number of data bits. These parameters are loaded into
the control register as part of the Control Word.
When a valid start bit is received, the succeeding bits are clocked into
the receive shift register.
If parity is used, parity is checked in the parity checker circuit.
After a complete data character is loaded into the shift register, the
character is transferred in parallel into the receive buffer register, and
the receive data available (RDA) flag is set in the status word register.
The CPU reads the status register by activating status word enable
(SWE), and if RDA is active, the CPU reads the character from the
receive buffer register by activating receive data enable (RDE).
After data is read from the receive buffer register, the CPU issues a
receive data available reset (RDAR) to reset the RDA.
The status flags contains the following:
Receive parity error (RPE) indicates that received data has a
parity error.
Receive framing error (RFE) indicates that received data has a
framing error (Failure to determine correct boundary between each
bit or character such as when no stop bit has been received).
Receive overrun error (ROE) happens when received character
in the receiver buffer is written over by another received character,
which results to lost of data. This happens when there is
insufficient flow control for data received.
Received data available (RDA) indicates that there is a received
character in the receive buffer register.
Bit synchronization is achieved by establishing a timing reference at
the center of each bit.
The start bit verification circuit detects the 1 to zero transition when a
start bit is received. Valid start bit indicates the beginning of a
character.
To minimize the risk of noise being interpreted as a valid start bit,
the receiver is clocked at a higher rate than the incoming data.
Assuming that clocking is 16 times the incoming bit rate, the start
bit verification circuit samples the start bit 7 times after a 1 to 0
transition, during which the line must remain at logic 0, so that the
circuit will interpret that a valid start bit has been found.
After a valid start bit has been found, the succeeding incoming bits
are sampled after every 16 clock pulses (at the middle of incoming
bits) after a valid start bit has been found.
Sampling for data bits (and parity if enabled) continues until the
stop bit has arrived.
The difference between the time a sample is taken and the actual
time of the center of a data bit is called sampling error.
The difference in time between the beginning of a start bit and
when it is detected is called detection error.
The maximum detection error is equal to the time of one receive
clock cycle. (That is if the receive clock rate equaled the receive
data rate.)
0 volts
Start
Bit
Stop
Bit
Data Bits
0 1
Clock pulses
(16 times the
incoming bit rate)
line must be 0
during 7 clock
pulses after 1 to
0 transition took
place, so that a
valid start bit
could be
detected.
Each incoming bit
is sampled every 16
clock pulses
starting from the
time that a valid
start bit has been
detected (near
center of each bit).
This makes
sampling rate equal
to incoming bit rate
These conditions assume that the
clock rate is 16 times the
incoming bit rate. Higher rates
could be used.
Using a clock rate higher than the
incoming bit rate also ensures
that circuit will detect the 1 to 0
transition as soon as possible..
Example: Determine the bit time, receive clock rate, and maximum
detection error for a UART receiving data at 1000 bps (f
b
) with a
receiver clock 16 times faster than the incoming data.
The time of one bit (t
b
) is the reciprocal of the bit rate, or
The receive clock rate is:
The time of one receive clock cycle is the reciprocal of the receive
clock rate (Rcl)
The maximum detection error is equal to the time of one receive
clock cycle, or 62.5 microseconds
ms 1
1000
1
f
1
t
b
b = = =
hz 16,000 (16)(1000) 16f R b cl = = =
ds microsecon 62.5
16,000
1
R
1
t
cl
cl = = =
USRT (Universal Synchronous Receiver Transmitter)
USRT (Universal Synchronous Receiver Transmitter) is an integrated
circuit designed for synchronous transmission.
It performs the same basic functions as a UART, except it is used for
synchronous transmission. Its functions include:
parallel to serial and serial to parallel conversion
parity bit generation and detection
parity, overrun, and framing error detection; and
Inserting and detecting unique data synchronization (SYNC)
characters
Formatting of data at the transmitter and receiver
Providing transmit and receive status to the CPU such as when
data have been received or transmitted.
May allow voltage level conversion for the serial interface
although many systems have separate voltage level converters.
Providing means of achieving bit and character synchronization.
Many DTEs now incorporate the USRT functions in larger scale ICs
(USRT is combined with other circuit components in one IC.)
USRT Block Diagram
Data bus
Transmit data
register
Transmit sync
register
Transmit shift register
Transmit
Timing
and
control
Multiplexer
Control
Register
Receive
Timing
and
control
Receive shift register
Receive buffer register
Comparator
Receive Sync
register
TDS
SCT
SCS
TCP
CS
NDB1
NDB2
POE
NPB
RR
RCP
SCR
ROR
RDA
RDAR
RSI
RDE
Received data (To data bus of computer)
Transmit data / Sync (From data bus of computer)
TSS
TSO
RSS
TCP-transmit clock pulse
TSS Transmit sync strobe
TDS- Transmit data strobe
TSO-transmit serial output
TCS-transmit clock signal
SCT-sync character transmit
RSS-receive sync strobe
RR-Receive rest
SCR-sync character receive
SCS-sync character signal
RDA-receive data available
RPE-receive parity error
ROR-receiver overrun
RSI-receive serial input
RCP-Receive clock pulse
RDAR-receive data available reset
RDE-receive data enable
CS-control strobe
NDB1-number of data bits 1
NDB2-number of data bits 2
POE-parity odd or even
NPB-parity or no parity
USRT Transmit Operation
USRT transmit operation basically includes:
Desired control word is loaded into control register by pulsing control
strobe (CS) pulse from CPU.
SYN character is loaded into the SYNC register by pulsing transmit
sync strobe (TSS).
At the beginning of each data transmission, one or more sync
characters are loaded into the transmit shift register and then
transmitted.
After each transmitted SYN character, the Syn character transmit
(SCT) is set to inform the CPU regarding the event.
Data is loaded into the transmit data register by pulsing the transmit
data strobe (TDS).
Data are transferred from transmit data register to the transmit shift
register, and then they are shifted out for transmission.
Characters are transferred from the transmit data register into the
transmit shift register provided that TDS pulses while data is being
shifted out. Otherwise, sync character will be loaded into the shift
register.
The transmit buffer empty (TBMT), which is a part of the control
status, is used by the USRT to request the next character from the CPU.
USRT Transmit Operation
CPU USRT
Sync character and transmit syn strobe (TSS)
Transmit clock pulse (TCP)
Transmit data strobe (TDS)
Control word and Control word strobe (CS)
TSO
(Transmit
serial data)
Syn character sent (SCS)
Parallel data (for transmission)
USRT Receive Operation
USRT receive operation basically includes:
Desired control word is loaded into control register by pulsing control
strobe (CS) pulse from CPU.
Receive sync character is loaded into the receive sync register by
pulsing receive sync strobe (RSS).
Receive rest (RR) is transmitted by CPU to put the USRT in search
mode for a sync character.
Once a valid sync character has been received, it is transferred into the
receive buffer register, and the USRT is place into character mode.
Status register is read
Data from shift register are transferred into the receive buffer, and then
transferred to the CPU in parallel.
CPU resets receive data available (RDA) for the next reception of data
from the serial input.
USRT Receive Operation
CPU
USRT
Receive sync character and receive syn strobe (RSS)
Control word and Control word strobe (CS)
RSI
(Receive
Serial input)
Receive rest (RR) makes USRT in search mode
Parallel data received
Sync character receive (SRR)
Receive data enable (RDE)
Receive data available (from status word)
Receive data available reset
USART (Universal Synchronous /Asynchronous Receiver
Transmitter)
USART (Universal Synchronous /Asynchronous Receiver Transmitter)
an integrated circuit which combines the functions of a UART and a
USRT.
IC can be programmed to handle both asynchronous and synchronous
transmission using the same input and output terminals.
More versatile.
Many DTEs now incorporate the USART functions in larger scale ICs
(USART is combined with other circuit components in one IC.)
Digital Communications
Digital Communications
Digital communications covers:
Digital radio - the transmittal of digitally modulated analog carriers
between two or more points in a communications system, using free
space or the earths atmosphere as the transmission medium.
Transmission of digitally modulated analog carriers through cables
such as copper wires.
Digital transmission - the transmittal of digital pulses between two or
more points in a communications system.
Digital transmission systems require a physical facility between the
transmitter and the receiver, such as copper cables.
The original source information may be analog which was converted to
digital (such as voice converted into digital signals), or a digital signal
(such as signals from computers)
Many conventional communications systems, which use analog
modulation techniques, are being replaced with more modern digital
communications systems.
Advantages of Digital Communications
More immune to noise compared to Analog Communications
Precise amplitude, frequency, or phase need not be ascertained to
determine the logic condition of the signal.
Transmission errors can be detected and corrected more easily and
accurately than it is possible with analog signals.
More resistant to additive noise, repeaters can be used.
Digital signals are better suited than analog signals for processing and
combining using technique called time division multiplexing.
It is much simpler to store, measure, and evaluate digital signals than
analog signals.
Transmission rate of digital signals can be easily changed to adopt to
different environments and to interface with different types of equipment.
Disadvantages of Digital Transmission Systems
Transmission of digitally encoded analog signals requires significantly
more bandwidth than simply transmitting the original analog signals.
Information signals which are analog must be converted to digital
signals prior to transmission in digital communications systems, and
converted back to their analog form at the receiver (if it is so required),
thus requires more circuits.
TYPICAL DIGITAL COMMUNICATIONS
PC PC Modem Modem
Cables
PC PC
Cables
(Modem not required, no digital modulation is done, short distance only)
(Digital signals from computers digitally modulate analog carriers)
Analog Signals
Digital Signals
Digital Signals
Digital Signals
Digital Signals
TYPICAL DIGITAL COMMUNICATIONS
Telecommunications
Facilities
PC
PC
PC
PC
Modem
Modem
Modem Modem
Mainframe
Modem
TYPICAL DIGITAL COMMUNICATIONS USING
DIGITAL RADIO
Computer Radio
Transmitter
Receiver
(earths atmosphere)
Radio signals (Analog signals)
Computer
Radio
Transmitter
Receiver
Digital Signals Digital Signals
DIGITAL MODULATION
Telecommunications
Facilities
PC
PC
PC
PC
Modem
Modem Modem
Mainframe
Modem
Front End
Processor (FEP)
FEP is a DTE which directs traffic
to and from many different
circuits, which could have different
parameters, such as codes, and data
formats.
LCU is a DTE that directs traffic
between one data communication
medium and a relatively few
terminals which uses the same
protocols, character codes, and
other parameters
Line control Unit
(LCU)
Modulation
Modulation
Process of transforming information signals from its original form to a
form that is more suitable for transmission.
Process of changing the properties of a relatively high frequency signal
(carrier signal) in accordance with the properties of the information signal
(modulating signal), which results to a modulated signal or modulated
wave.
Process of impressing relatively low frequency information signals onto a
high-frequency carrier signal.
Digital modulation
Process of modulating a relatively high frequency carrier using a digital
signal as the modulating signal.
Transmittal of digitally modulated analog signals (carriers).
Sometimes called digital radio because digitally modulated signals can be
propagated through earths atmosphere.
Modulation takes place in a circuit called modulator which is found in
a transmitter.
Modulation
A device used for modulation and demodulation is called modem or
data modem (contraction of modulator and demodulator).
Modem or data modem is a DCE used to interface a DTE to an analog
telephone circuit / line commonly called POTS (Plain Old Telephone
System)
At the receiver side, the received modulated signal is demodulated to
extract the original information signal.
Demodulation is the process of extracting the original information
signal from the modulated signal.
Reasons for Modulation
Not all signals, whether analog or digital, could be efficiently and
effectively transmitted through a particular transmission medium or
transmission / telecommunications facilities.
Example: digital signals could not be transmitted through plain old
telephone system.
Information signals often occupy the same frequency band and, if
transmitted in their original form, would interfere with each other.
Example: Voice signals from several persons could occupy the
same band of frequencies.
Through modulation and frequency division multiplexing,
modulated signals could occupy different frequency ranges and
could be transmitted using one communications medium at the
same time.
It is not advisable for low frequency analog signals to be transmitted
as is, because the antennas needed could be very large. The size of
the antennas is proportional to the wavelength of the signals. The
lower the frequency, the higher is the wavelength, and the larger is the
antenna needed.
Low frequency signals are more difficult to radiate.
Computers Connected to a Telephone Network
Telephone Network
PC
PC
PC
PC
DCE
(Modem)
DCE
(Modem)
DCE
(Modem)
DCE
(Modem)
Digital signals
Analog Signals
Computers Connected to a Telephone Network
Digital computers use and process digital signals.
The output and input signals of digital computers in their
communication ports are digital signals.
Digital signals, as is, could not be effectively transmitted through a
plain old telephone network, because the telephone network is
designed for voice signals which is an analog signal.
The digital signals must be converted into a form which is suitable
for transmission through the telephone network. This could be
accomplished by using the digital signals to MODULATE an analog
signal which could be transmitted through the telephone network.
Nyquist Bandwidth
According to H. Nyquist, binary digital signals can be propagated through an
ideal noiseless transmission medium at a rate equal to two times the
bandwidth of the medium.
The minimum bandwidth required to propagate a signal is called minimum
Nyquist Bandwidth or sometimes the minimum Nyquist frequency.
If only two signaling elements or levels are used (at the communications
medium), then the bit rate for a given bandwidth according to H. Nyquist is:
fb = 2 B = bit rate = 2 (bandwidth)
Above equation serves as a guide for the minimum bandwidth, and the actual
bandwidth necessary to propagate at a given rate depends on:
Encoding and modulation used
types of filters used
system noise
desired error performance
The ideal Nyquist bandwidth may or may not be the same as the actual
bandwidth required, and it is generally used for comparison purposes only.
Nyquist Bandwidth
If more than two levels are used for signaling or coding (at the
transmission medium), more than one bit may be transmitted at a time, and
it is possible to propagate a bit rate that exceeds 2B.
Using multilevel signaling, the Nyquist formulation for channel capacity
is:
Above equation is similar to Hartleys law.
Note that there are exceptions in using the above formula as in the case
of frequency shift keying (FSK).
signal discrete or symbol a in d represente bits of number M log N
levels or voltage signal discrete of number
medium ssion at transmi levels coding of number M
(hz) bandwidth channel B
sec) per (bits capacity n informatio C
sec) per (bits rate bit f : where
N B 2
M log B 2 C f
2
b
2 b
= =
=
=
=
=
=
=
= =
Nyquist Bandwidth
Note: based on book of Tomasi, Data Communications and Networking, if
more than two levels are used for signaling or coding (at the transmission
medium), the following equation for the bit rate can be used:
Above equation might be used by examiners.
Note that there are exceptions in using the above formula as in the case of
frequency shift keying (FSK).
signal discrete or symbol a in d represente bits of number M log N
levels or voltage signal discrete of number
medium ssion at transmi levels coding of number M
sec) per (bits capacity n informatio C
sec) per (bits rate bit f : where
N
f
M log
f
B
N B
M log B C f
2
b
b
2
b
2 b
= =
=
=
=
=
=
= =
=
= =
Nyquist Bandwidth
The baud rate can also be computed as:
medium ion transmiss at the
signal discrete or element signaling each in bits of number N
rate n informatio rate bit fb : where
N
fb
baud
=
= =
=
Types of Digital Modulation
Amplitude Shift Keying
Frequency Shift Keying
Phase Shift Keying
QAM (Quadrature Amplitude Modulation)
Trellis Code Modulation
Amplitude Shift Keying (ASK)
Process of changing the amplitude of a relatively high frequency
carrier signal in proportion to the instantaneous value of a digital
modulating signal.
Sometimes called digital amplitude modulation (DAM).
Similar to standard amplitude modulation except there are only two
output amplitudes possible.
Sometimes called on-off keying (OOK) because amplitude of
modulated wave could be on (with value other than 0) or off (0 volt).
Simplest digital modulation technique.
Relatively simple and inexpensive.
Low performance / quality form of modulation.
Seldom used for new modems, except when used in combination with
other digital modulation techniques.
Carrier
signal
0 volt
0 volt
0 volt
+5 v +5 v
1 1
Modulated
Signal
(analog signal)
at output of modem
Modulating signal
(digital signal)
from information
source
0
1
0 1
Note: conventions, voltage levels, and voltage polarities could vary.
Mathematically, amplitude shift keying could be expressed as:
The above equation is a normalized binary waveform, where +1 v = logic 1
and -1 volt = logic 0. Therefore, for a logic 1 input vm(t) = +1 volt, above
equation reduces to:
For a logic 0 input, above equation reduces to:
frequency radian carrier analog
amplitude carrier d unmodulate
2
A

(volts) signal g) (modulatin n informatio digital (t) v
wave keying shift amplitude v(t) : where
t)] cos(
2
A
(t)][ v [1 v(t)
c
m
c m
=
=
=
=
+ =
t) cos( A
t)] cos(
2
A
1)][ [1 v(t)
c
c
=
+ =
volt 0 t)] cos(
2
A
1][ [1 v(t) c = =
The rate of change of the ASK waveform (baud) is the same as the rate of
change of the binary input. Thus, the bit rate is equal to the baud rate.
The bit rate is also equal to the minimum Nyquist bandwidth.
Example: Determine the baud and minimum bandwidth necessary to pass a
10 kbps binary signal using ASK.
rate bit fb
1
fb
baud = = =
rate bit fb
1
fb
bandwidth B = = = =
hz 10,000
1
10,000
B
baud 10,000
1
10,000
baud
= =
= =
Frequency Shift Keying (FSK)
Process of changing the frequency of a relatively high frequency
carrier signal in proportion to the instantaneous value of a digital
modulating signal.
Similar to standard frequency modulation (FM) except that the
modulating signal is a binary signal that varies between two discrete
voltages, rather than a continuously changing analog waveform.
Peak amplitude of the carrier remains constant.
In binary FSK, the frequency deviation is constant and is always at
its maximum value.
Sometimes called binary FSK (BFSK) because modulating signal
could have two discrete levels.
More immune to noise than ASK because noise could be lessened at
the receiver.
Relatively simple and inexpensive.
Low performance / quality form of modulation compared to PSK or
QAM.
Carrier
signal
0 volt
0 volt
0 volt
+5 v = 1 +5 v = 1
Modulated
Signal
(analog signal)
at output of modem
0
1
1
Frequency = 2 khz Frequency = 1 khz Frequency = 1 khz
1 0 1
Modulating signal
(digital signal)
from information
source
Carrier
signal
0 volt
0 volt
0 volt
+5 v = 1 +5 v = 1
Modulated
Signal
(analog signal)
at output of modem
0
1
1
Frequency = 1 khz Frequency = 2 khz Frequency = 2 khz
1
0
1
Modulating signal
(digital signal)
from information
source
Frequency = 1.5 khz
- 5 v = 0
The general expression for a binary FSK signal could be written as:
Where: v(t) = binary FSK waveform
Vc = peak unmodulated carrier amplitude (volts)
c
= angular velocity of unmodulated carrier signal (rad / sec)
= radian carrier frequency=2tfc
fc = analog carrier center frequency = carrier unmodulated frequency (hz)
v
m
(t) = binary digital modulating signal
= peak to peak change in radian output frequency (radian)
A f = peak frequency deviation
Output carrier radian frequency (
c
) shifts by an amount equal to + or ( )
/ 2, which is proportional to the amplitude and polarity of the binary input
signal.
or )t]
2
v
[( cos Vc ) (
m(t)
c + = t v
deviation) frequency peak ( 2
2
f 2
since f)t] v (f [2 cos Vc ) ( m(t) c
t t
t
= =
+ = t v
The modulating signal in the preceding equation is a normalized binary
waveform where a logic 1= +1 , and a logic 0 = -1 .
For a logic 1, the preceding equation becomes:
For a logic 0, the preceding equation becomes:
Example:
Binary 1 = + 1 volt could produce + ( ) / 2, while binary 0 = -1 volt could
produce - ( ) / 2.
Output radian frequency deviates between
c
+ ( ) / 2, and
c
- ( ) / 2.
Rate of change of the frequency shifts is equal to the rate of change of the
binary signal.
f)t] (f [2 cos Vc ) ( c + = t v t
f)t] - (f [2 cos Vc ) ( c = t v t
FSK modulators are very similar to conventional FM modulators and
are very often voltage controlled oscillators (VCOs).
The highest modulating frequency (alternating 1and 0) is equal to
one half the input bit rate.
The rest frequency of the VCO is chosen such that it falls halfway
between the the mark (1) and space (0) frequencies.
The modulation index for FSK can be expressed as:
MI = f / fa
Where: MI = modulation index (unitless)
f = peak frequency deviation (hz)
fa = modulating frequency (hz)
Note:
- f could be equal to the frequency of .
- Under worst case condition (alternating 1s and 0s), fa = bit rate (fb)
In binary FSK modulator, f is the peak frequency deviation of the
carrier and is equal to the difference between the rest frequency and
either the mark or space frequency.
f = peak frequency deviation (hz)
2
frequency space ncy Markfreque
frequency mark frequency Rest
frequency space frequency Rest
=
=
=
For binary FSK, the modulation index (MI) can be expressed as:
Where:
fb
fs fm
2
fb
2
fs fm
MI

=
=
frequency space fs
frequency mark fm
signal input binary of frequency l fundamenta fa
2
fb
rate bit input fb
deviation frequency peak f
2
fs fm
=
=
= =
=
= A =
For FSK, the modulation index is generally kept below1.0, to produce a

relatively narrow band FM output spectrum.
For binary FSK, the baud is computed as:
FSK is an exception to the rule for computation of bandwidth, as the
minimum bandwidth is determined from the Bessel table or from the
formula:
B=2(Af + fb) = minimum Nyquist bandwidth
where: Af = peak frequency deviation
fb = bit rate
rate bit fb
1
fb
N
fb
baud = = = =
Example: For FSK modulator with space, rest, and mark frequency of 60,
70, 80 Mhz respectively, and an input bit rate of 20 Mbps, determine the
modulation index, minimum bandwidth required, and the baud rate.
From the Bessel table, a modulation index of 1 yields three significant
side frequencies. Thus the bandwidth required is:
Bandwidth = 2(fa)(3) = (2)(fb/2)(3) = 2(10)(3) = 60 Mhz
or Bandwidth = 2[(80 M 70 M) + 20 M] = 60 Mhz
Since the number of signaling element at the output of the modulator is
the same as the number of signaling element at the input of the modulator,
the baud rate = bit rate = 20 Megabaud
1
20Mbps
60Mhz 80Mhz
fb
fs fm
MI =
=
FSK Receiver
The most common circuit used for demodulating binary FSK signals is
the phased-lock-loop (PLL).
Phase
comparator
Voltage
Controlled
Oscillator
Amp
Analog FSK in
DC error
voltage
Binary Data Out
PLL
Minimum Shift Keying FSK
Minimum Shift Keying FSK (MSK) is a form of continuous-phase
frequency shift keying (CPFSK), wherein the mark and space
frequencies are synchronized with the input binary bit rate (there is
precise time relationship between the two).
Mark and space frequencies are separated from the center frequency by
an exact odd multiple of one-half of the bit rate, to ensure that there is
smooth phase transition in the modulated signal.
fm and fs = n(fb/2)
where: fm = mark frequency
fs = space frequency
fb = binary bit rate
n = any odd integer
MSK has better bit-error performance than conventional binary FSK.
Phase Shift Keying (PSK)
Process of changing the phase of a relatively high frequency carrier signal
in accordance to a digital modulating signal.
Peak amplitude of the carrier remains constant.
PSK can have different forms such as:
Binary phase shift keying (BPSK)
One phase change represents one bit.
M-ary phase shift keying other than BPSK, such as when a single
phase change could represent two or more bits. Ex:
45
0
= 00
90
0
= 01
180
0
= 11
With four possible output phases, M = 4, the number of bits =2
With eight possible output phases, M = 8, the number of bits =3
The number of bits (N) that can be represented with M possible output
phases is:
Sensitive to phase delay (or envelope delay) distortion, which results
when different frequencies propagates at different speeds in a channel.
N=log
2
M or M = 2
N
Phase Shift Keying (PSK)
Carrier
signal
0 volt
0 volt
0 volt
Modulated
Signal
(analog)
Modulating signal
(digital signal)
from information
source
0
1
1
Frequency = 2 khz
1 0 1
Phase change Phase change
Binary Phase Shift Keying (BPSK)
Input has two possible conditions (logic 0 or 1)
N=1 and M = 2
Two output phases are possible (one to represent logic 1 and the other
to represent logic 0).
Phase of carrier shifts between two angles that are 180
0
out of phase
Also called Phase Reversal Keying (PRK) and Biphase Modulation.
Form of suppressed carrier, square wave modulation of a continuous
wave (CW) signal.
The output phase of a PSK wave could be:
logic 0 = 180
0
logic 1 = 0
0
The baud rate is equal to the bit rate, since 1 bit is represented by one
signaling element.
BPSK Modulator
Balanced
Modulator
Reference
Carrier
Oscillator
Bandpass
filter
Binary
Data
In
Analog
PSK
Output
Balanced modulator is used to produced PSK signal
Balanced Modulator Used for Binary Phase Shift Keying
Digital binary
input
Reference
Carrier
input
BPSK
signal
D1
D2
D3
D4
A B
Balanced modulator acts like a phase reversing switch.
Digital binary input is much greater than the peak amplitude of carrier (digital
input controls the conduction of the diodes.)
If input is logic 1 (positive voltage), D1 and D2 conducts, while D3 and D4 does
not , thus BPSK output is in phase with reference oscillator signal.
If input signal is logic 0 (negative signal), D3 and D4 conducts, while D1 and D2
does not, thus BPSK output is 180
0
out of phase with reference oscillator signal.
Cos
c
- Cos
c
0
0
reference
+ or - 180
0
Logic 1 Logic 0
Constellation Diagram
(Signal state-space Diagram)
Bandwidth of BPSK Signals
The widest output bandwidth occurs when the input binary data are
alternating 1/0 sequence.
The fundamental frequency (fa) of an alternating 1/0 sequence is equal
to one half the bit rate.
where: fa = fundamental frequency (Hz)
fb = bit rate (bits per second)
A balanced modulator is a product modulator.
The output phase of a BPSK balanced modulator can be expressed as:
output = (sin
a
t) (sin
c
t)
where: sin
a
t = fundamental frequency of binary modulating signal
sin
c
t = unmodulated carrier
fa = fb/2
The output phase of a BPSK balanced modulator can also be expressed as:
output = (1/2) [cos(
c
-
a
)t] - (1/2) [cos(
c
+
a
)t]
Consequently, the minimum double-sided Nyquist bandwidth (f
N
) is
where: fa = fundamental frequency (Hz)
fb = bit rate (bits per second)
(
c
+
a
) - (
c
-
a
) = 2
a
=2(2tfa)
f
N
= 2fa
f
N
=2 (fb/2) = fb = bit rate
Example: For a BPSK modulator with a carrier frequency of 70 Mhz and an
input bit rate of 10 Mbps, determine the maximum and minimum upper and
lower side frequencies, draw the output spectrum, determine the minimum
Nyquist bandwidth, and calculate the baud.
output = (1/2) [cos(
c
-
a
)t] - (1/2) [cos(
c
+
a
)t]
= (1/2) [cos(270 Mhz -25 Mhz)t] - (1/2) [cos(270 Mhz + 25 Mhz)t]
= (1/2) [cos(265 Mhz)t] - (1/2) [cos(75 Mhz)t]
Lower side frequency (LSF) = 65 Mhz
Upper side frequency (USF) = 75 Mhz
Minimum Nyquist bandwidth = 75 Mhz 65 Mhz = 10 Mhz
Baud = fb = 10 Megabaud
65 Mhz 70 Mhz 75 Mhz
BPSK Receiver
Balanced
Modulator
Coherent
Carrier
Recovery
Low pass
filter
BPSK
input
Binary Data
Output
Sin
c
t
(recovered carrier)
+ or - sin
c
t
The coherent carrier recovery circuit detects and regenerates a carrier
signal that is both frequency and phase coherent with the original
transmit carrier.
The balanced modulator is a product detector and its output is the
product of its two inputs (BPSK and recovered carrier).
The low pass filter separates the recovered binary data from the
complex demodulated signal.
BPSK Receiver (cont.)
For a balanced modulator with input signal + sin
c
t (logic 1), the
output is:
output = (sin
c
t) (sin
c
t)
= (1/2) [cos(
c
-
c
)t] - (1/2) [cos(
c
+
c
)t]
= (1/2) [1 - cos2
c
t]
= 1/2 - (1/2)cos2
c
t
For a balanced modulator with input signal - sin
c
t (logic 0), the
output is:
output = (-sin
c
t) (sin
c
t)
= - (1/2) [cos(
c
-
c
)t] + (1/2) [cos(
c
+
c
)t]
= -(1/2) [1 - cos2
c
t]
= -1/2 + (1/2)cos2
c
t
Filtered out
DC component
(= logic 1)
Filtered out
DC component
(= logic 0)
Quatenary Phase Shift Keying
Quatenary Phase Shift Keying (QPSK) , or quadrature PSK, is a
form of angle modulated, constant amplitude digital modulation,
wherein M = 4, and N = 2.
Four output phases are possible.
2 bits (1 dibit) can be represented per phase.
The rate of change at the output of the PSK modulator (baud rate)
is equal to of the input bit rate.
The minimum bandwidth required is equal to of the input bit
rate
QPSK Transmitter
Balanced
Modulator
Linear
summer
Balanced
Modulator
Band pass
filter
90
0
phase
shift
Reference
Carrier
Oscillator
Sin
c
t
Input
buffer
I
Q
Divide
by 2
Bit clock
Binary input
Data f
b
Logic 1 = +1 v
Logic 0 = -1 v
Q channel f
b
/2
I channel f
b
/2
Logic 1 = +1 v
Logic 0 = -1 v
+ or sin
c
t
+ or cos
c
t
QPSK
output
sin
c
t
Cos
c
t
- Cos
c
t
sin
c
t - sin
c
t
0
0
reference
10
11
00
01
QPSK Transmitter
The dibits are split into two.
One bit is inputted to a balanced modulator, while the other
is inputted into the other balanced modulator.
Each balanced modulator produces BPSK signals which
are 90
0
out of phase.
The BPSK signals are combined in the linear summer.
The resultant phasor at the output of the linear summer
could have the following values:
+ sin
c
t + cos
c
t
+ sin
c
t - cos
c
t
- sin
c
t + cos
c
t
- sin
c
t - cos
c
t
The angular separation between any two adjacent phasors
in QPSK is 90
0
Bandwidth Considerations in QPSK
The highest fundamental frequency present at the data input of the I
or Q balanced modulator is equal to of the input data rate.
The output of the balanced modulators can be expressed as:
output = (sin
a
t) (sin
c
t)
= [sin 2(f
b
/4)t] [sin 2f
c
t]
= (1/2) [cos 2(f
c
- f
b
/4 )t] - (1/2) [cos 2(f
c
+ f
b
/4 )t]
The output frequency spectrum extends from f
c
+ f
b
/4 to f
c
- f
b
/4
The minimum bandwidth (f
N
) is:
f
N
= [f
c
+ f
b
/4] - [f
c
- f
b
/4] = f
b
/2
The highest output rate of change (baud) is equal to fb/2.
The minimum bandwidth and baud are equal in magnitude.
Eight PSK
Eight Phase PSK (8 PSK) is a form of angle modulated, constant
amplitude digital modulation, wherein M = 8, and N = 3.
Eight output phases are possible.
3 bits (1 tribit) can be represented per phase.
The rate of change at the output of the PSK modulator (baud rate)
is equal to 1/3 of the input bit rate.
The minimum bandwidth required is equal to 1/3 of the input bit
rate.
Eight PSK Transmitter
Input
data
Q
I C
I channel
Q channel
C
C
fb/3
Pulse amplitude modulation (PAM)
2 to 4 level
converter
2 to 4 level
converter
Product
modulator
Product
modulator
Reference
oscillator
Linear
summer
fb/3
fb/3
sin
c
t
cos
c
t
+90
0
8 PSK
output
Bit splitter
Note: Product modulators could be Balanced modulators
Incoming serial bit is converted into parallel three channel output.
I is used to modulate the original carrier (In phase).
Q is used to modulate a carrier which is 90
0
(quadrature) out of
phase with original carrier.
C is for control channel.
I and Q bits, together with C and C are converted into 4 level signals
using the 2 to 4 level converters (digital to analog converters).
The I or Q bit determines the polarity of the output analog signal,
while C and C determine the voltage level of the output analog signal.
For I or Q, logic 1 could be a positive voltage, while logic 0 could
be a negative voltage.
For C and C , logic 1 could be 1.307 volts, while logic 0 could be
.541 volt.
For a tribit input of Q = 0, I = 0, and C = 0, the output amplitude and phase of the
8 PSK modulator can be determined as follows:
The output of the I channel is -.541 volt.
The output of the Q channel is 1.307 volts.
The output of the I product modulator is -.541 sin
c
t.
The output of the Q product modulator is 1.307 cos
c
t.
The output of the linear summer is -.541 sin
c
t 1.307 cos
c
t.
or 1.41 sin (
c
t - 112.5
0
)
Separation between phasors is 45
0
.
Resulting modulated signal always has
a constant peak amplitude of 1.41 volts.
.541/1.307 are relative values and can
have other values as long as their
ratio is the same.
Tribit code between adjacent angles
change only by 1 bit. This is called
gray code or maximum distance code,
and it is used to reduce errors.
000
001
101
100
110
111
011
010
QIC
sin
c
t
(Reference
phase)
- sin
c
t
+ cos
c
t
- cos
c
t
8 PSK
QIC
000
QIC
001
QIC
010
QIC
011
QIC
100
QIC
101
QIC
110
QIC
111
1.41V
-112.5
0
1.41V
-67.5
0
1.41V
+112.5
0
1.41V
+67.5
0
1.41V
-157.5
0
1.41V
-22.5
0
1.41V
+157.5
0
1.41V
+22.5
0
Q I C Ampl i tude
Phase
(degrees)
0 0 0 1.41 -112.5
0 0 1 1.41 -157.5
0 1 0 1.41 -67.5
0 1 1 1.41 -22.5
1 0 0 1.41 +112.5
1 0 1 1.41 +157.5
1 1 0 1.41 +67.5
1 1 1 1.41 45
8-PSK output Bi nary i nput
Note: phase changes on sine wave are not representative of actual phase change,
and is for discussion purposes only
Bandwidth Considerations for Eight PSK
The bit rate in the I,Q, and C channels is 1/3 of the bit rate (fb) of the input
binary signal.
The baud rate at the output of the modulator is equal to 1/3 of the bit rate
(fb) of the input binary signal.
The minimum bandwidth (fn) required is equal to fb/3.
The highest fundamental frequency (fa) in the I, Q, or C channel is equal
to 1/6 of the bit rate (fb) of the input binary signal.
Example: Given input data rate of 8 PSK modulator is 10 Mbps,
Baud rate = 10 Mbps / 3 = 3.33 Mbaud
Bit rate in the I, Q, and C channels = 10 Mbps/ 3 = 3.33 Mbps
Minimum Nyquist bandwidth = 10 Mbps / 3 = 3.33 Mhz
16 Phase PSK
16 phase PSK is an M-ary encoding technique where M=16, and there
are 16 different phases possible at the output of the modulator.
Peak amplitude of the modulated carrier is constant.
Input data are grouped into four (quadbits).
The output rate of change (baud rate) is equal to of incoming bit
rate.
The minimum Nyquist bandwidth is equal to of the incoming bit
rate.
The angular separation between adjacent phasors is only 22.5
0
.
Highly susceptible to phase impairments during transmission.
Differential Phase Shift Keying (DPSK)
Differential Phase Shift Keying (DPSK) is a form of digital modulation
wherein the binary input information is contained in the difference between
two successive signaling elements rather than the absolute phase.
With DPSK, it is not necessary to recover a phase-coherent carrier.
A received signaling element is delayed by one signaling element time slot
and then compared to the next received signaling element.
The difference in the phase of the two signaling elements determines the
logic condition of the data.
Quadrature Amplitude Modulation (QAM)
QAM is a form of digital modulation where the digital information is
contained in both the amplitude and phase of the transmitted carrier.
QAM uses a combination of amplitude shift keying and phase shift
keying.
Amplitude and phase of carrier changes.
Capable of having relatively high information rate.
The degree of bandwidth compression is the same as that of PSK.
B = baud = fb / N = bit rate / number of bits per baud
8-QAM
8-QAM is an M-ary encoding technique wherein M = 8, and wherein
the peak amplitude of the modulated signal is not constant.
Eight outputs are possible.
3 bits (1 tribit) can be represented per output.
8-QAM Transmitter
Input
Data
fb
Q
I C
I channel
Q channel
C
C
fb/3
2 to 4 level
converter
2 to 4 level
converter
Product
modulator
Product
modulator
Reference
oscillator
Linear
summer
fb/3
fb/3
sin
c
t
cos
c
t
+90
0
8 QAM
output
Bit splitter
Note: - Circuit is similar to 8 PSK transmitter except for the omission of the inverter.
- Product modulators could be Balanced modulators
8 QAM Transmitter
Incoming serial bit is converted into parallel three channel output.
I is used to modulate the original carrier (In phase).
Q is used to modulate a carrier which is 90
0
(quadrature) out of phase with
original carrier.
C is for control channel.
I and Q bits, together with C are converted into 4 level signals using the 2 to 4
level converters (digital to analog converters).
The I or Q bit determines the polarity of the output analog signal, while C
determines the voltage level of the output analog signal.
For I or Q, logic 1 could be a positive voltage, while logic 0 could be a
negative voltage.
For C, logic 1 could be 1.307 volts, while logic 0 could be .541 volt.
Magnitude of I and Q PAM signals are always the same.
8-QAM Transmitter
For a tribit input of Q = 0, I = 0, and C = 0, the output amplitude and
phase of the 8 PSK modulator can be determined as follows:
The output of the I channel is -.541 volt.
The output of the Q channel is -.541 volt.
The output of the I product modulator is -.541 sin
c
t.
The output of the Q product modulator is -.541 cos
c
t.
The output of the linear summer is -.541 sin
c
t -.541 cos
c
t.
or 0.765 sin (
c
t - 135
0
)
Separation between phasors is 90
0
.
Resulting modulated signal could have
a peak amplitude of .765 or 1.848 volts.
000
001
101
100 110
111
011
010
QIC
sin
c
t
(Reference
phase)
- sin
c
t
+ cos
c
t
- cos
c
t
8 QAM
QIC
000
QIC
001
QIC
010
QIC
011
QIC
100
QIC
101
QIC
110
QIC
111
0..765V
-135
0
0..765V
-45
0
0..765V
+135
0
0..765V
+45
0
1.848V
-135
0
1.848V
-45
0
1.848V
+135
0
1.848V
+45
0
Q I C Ampl i tude
Phase
(degrees)
0 0 0 0.765 -135
0 0 1 1.848 -135
0 1 0 0.765 -45
0 1 1 1.848 -45
1 0 0 0.765 135
1 0 1 1.848 135
1 1 0 0.765 45
1 1 1 1.848 45
8-QAM output Bi nary i nput
8 QAM Bandwidth Considerations
The bit rate in the I,Q, and C channels is 1/3 of the bit rate (fb) of the
input binary signal.
The baud rate at the output of the modulator is equal to 1/3 of the bit
rate (fb) of the input binary signal.
The minimum bandwidth (fn) required is equal to fb/3.
The highest fundamental frequency (fa) in the I or Q channel is equal
to 1/6 of the bit rate (fb) of the input binary signal.
16 QAM
16 QAM is an M-ary encoding technique where M=16, and both the
phase and amplitude of the carrier are varied.
Input data are grouped into four (quadbits).
The output rate of change (baud rate) is equal to of incoming bit
rate.
The minimum Nyquist bandwidth is equal to of the incoming bit
rate.
Comparison of Different Modulation Techniques
= fb / 4 = fb / 4 4 16 16 QAM
= fb / 3 = fb / 3 3 8 8 QAM
= fb / 2 = fb / 2 2 4 4 QAM
= fb / 5 = fb / 5 5 32 32 PSK
= fb / 5 = fb / 5 5 32 32 QAM
= fb / 4 = fb / 4 4 16 16 PSK
= fb / 3 = fb / 3 3 8 8 PSK
= fb / 2 = fb / 2 2 4 4 PSK (QPSK)
= fb = fb 1 2 Binary PSK(BPSK)
1 2 FSK
= fb = fb 1 2 ASK
Minimum
Bandwidth
(Hz)
Baud Rate
(baud)
N
(Number of bits
per Signaling
Element or per
symbol)
M
(Possible Outputs=
Number of Signaling
Element or Symbol)
Modulation
fb indicates a magnitude equal to the bit rate.
Upper side frequency = carrier frequency + Bandwidth / 2 or USF = fc + BW / 2
Lower Side frequency = carrier frequency Bandwidth / 2 or LSF = fc BW / 2
2
N
= M
LSF
fc
USF
BW
Trellis Code Modulation (TCM)
Modulation which combines encoding and modulation to reduce the
probability of error.
Controlled redundancy are introduced into the bit stream to reduce the
likelihood of transmission errors.
Could be used for faster data transmission compared to ASK, FSK and
PSK.
Could be used for data transmission rates in excess of 56 kbps over
standard telephone lines.
Uses convolutional tree codes, which combines encoding and
modulation to reduce the probability of error.
Controlled redundancy are introduced in the bit steam.
Bandwidth Efficiency
Digital modulation schemes where N=1 achieve bandwidth compression
(i.e. less bandwidth is required to propagate a given bit rate.)
Bandwidth efficiency is used to compare the performance of one digital
modulation technique to another.
Also called information density or spectral efficiency
Equal to ratio of transmission bit rate to the minimum bandwidth
required for a particular digital modulation scheme.
Equal to number of bits that can be propagated per second, per one
hertz of bandwidth.
(hz) bandwidth minimum
(bps) rate on transmissi
efficiency bandwidth =
(bits per cycle)
Carrier Recovery
Carrier recovery is the process of extracting a phase-coherent reference
carrier from a received signal.
This is needed at the receiver for extracting the original information
from the modulated signal.
Also called phase referencing.
With PSK and QAM, the carrier is suppressed at the balanced
modulators and therefore are not transmitted with the modulated signal.
Other carrier recovery techniques are used such as:
Square loop
Costas loop
Remodulator
Modems
Modems may use two wires or four wires, although most modems
nowadays use two wires.
For two wire modems, transmit and receive are both present in the
same wires.
For 4 wire modems, transmit is present in one set of wires, and the
receive is at the other set of wires.
Modems may be hard wired or acoustically coupled
For hard wired modems, output and input of modems are directly
connected to the communication circuits.
For acoustically coupled modems, the mouthpiece of a telephone
set is placed in cups containing a speaker (for transmit) and a
microphone (for receive).
Modems may be configured to be in origin (caller) or answer (called)
mode.
Typical Modems
Bell 103
300 bps on two or four wires, FSK
Mark = 2225 hz (answer), Mark = 1270 hz (origin)
Space = 2025 hz (answer), Space = 1070 hz (origin)
Answer tone 2225 hz
Origin tone = 1270 hz
V.21
1200/1800 bps, FSK
Mark = 980 hz and 1650 hz
Space = 1180 hz and 1850 hz
Bell 209A
9600 bps, QAM
Bell 303D
230.4 kbps, VSB AM (vestigial sideband)
V.22
1200 bps
Compatible with Bell 212
Note, modems connected to one another must be using the same
modem standard.
Digital Transmission
Digital Transmission
Transmittal of digital signals between two or more points in a
communications system.
Transmitted signals can be binary or any other form of discrete-level
digital pulses.
Original information signal may be digital (data from computers), or
analog (such as voice) which have been converted to digital pulses
prior to transmission.
Physical facilities, such as pair of wires or fiber optic cables, are
required to interconnect various points within the system.
Digital pulses cannot be propagated through a wireless transmission
system (radio transmission), such as earths atmosphere or free space
(vacuum).
Examples of communication systems which require digital
transmission are T1 and E1 communications systems.
Digital transmission systems use Channel Service Units or Digital
(Data) Service Units to interface DTEs to digital transmission
channels / media such as T1 or E1 lines.
DIGITAL TRANSMISSION
IN DATA COMMUNICATIONS
DTE DTE DCE
(DSU / CSU)
Between DCEs
(for long distance communications)
DCE
(DSU / CSU)
Between DTE and DCE
(short distance only)
Between DTE and DCE
(short distance only)
Signal between DTE and DCE
(digital)
Signal between DTE and DCE
(digital)
0 volt
0 volt 0 volt
Signal between DCEs
(encoded digital signal)
DSU Data service unit
CSU Channel service unit
Network using
digital transmission
PC
PC
PC
PC
DCE
DCE
DCE DCE
Digital signals
Digital signals
Encoded
digital signals
Encoded
digital signals
Network using
digital transmission
Voice
Signal
(analog)
DCE DCE
Digital signals
Analog to
digital
converter
Voice
Signal
(analog)
Digital
to analog
converter
Digital signals
Encoded
digital signals
Encoding
Encoding (digital line encoding) involves converting standard logic
levels to a form more suitable for transmission through digital
communications systems.
The reasons for encoding are:
Some signals, whether the source information is digital or analog,
needs to be transmitted through a digital communications medium
or system.
(Analog information signals can be converted to digital signals
prior to transmission in a digital communications system.)
Digital communication systems are now widely used.
Factors which must be considered in selecting line-encoding format
are:
Transmission line voltages and DC component
Duty cycle
Bandwidth consideration
Clock and framing bit recovery
Error detection
Ease of detection and decoding
Line Encoding Formats
Unipolar nonreturn to zero (UPNRZ)
Bipolar nonreturn to zero (BPNRZ)
Unipolar return to zero (UPRZ)
Bipolar return to zero (BPRZ)
Bipolar-return-to-zero alternate-mark- conversion (BPRZ AMI)
Line Encoding Formats
O volt
O volt
O volt
O volt
O volt
+V
+V
+V
+V
+V
-V
-V
-V
UPNRZ
UPRZ
BPRZ
BPRZ AMI
BPNRZ
1 1
0 0 1 0 1 0
Binary digits
from information
source
Pulse Modulation
Pulse Modulation
It is not really a type of modulation but rather a form of digitally
coding analog signals.
Analog information signals must first be converted into digital signals
before they can be used in digital communications systems.
Pulse Modulation consists of sampling analog information signals
and then converting those samples into discrete pulses , and then
transporting the pulses from a source to a destination over a
transmission medium.
Predominant Methods Of Pulse Modulation
Pulse Width Modulation (PWM)
Pulse Position Modulation (PPM)
Pulse Amplitude Modulation (PAM)
Pulse Code Modulation (PCM)
0 volt
8 bit word 8 bit word 8 bit word 8 bit word
Analog
Signal
(such as voice)
0 volt
0 volt
0 volt
0 volt
0 volt
Sample
Pulses
Pulse
Width
Modulation
Pulse
Position
Modulation
Pulse
Amplitude
Modulation
Pulse
Code
Modulation
PREDOMINANT METHODS OF PULSE MODULATION
Time
Maximum Amplitude
Minimum Amplitude
Pulse Width Modulation (PWM)
Also called Pulse Duration Modulation (PDM) or Pulse Length
Modulation (PLM)
Some authors consider PWM, together with PPM, as a type of Pulse
Time Modulation (PTM).
Pulse modulation wherein the width of a constant amplitude pulse is
varied proportional to the amplitude of the analog signal at the time
it is sampled.
Widest pulse could represent the highest amplitude of analog signal,
while narrowest pulse could represent the minimum amplitude of the
analog signal.
Used in special purpose communications systems (usually for
military) but is seldom used for commercial applications.
Produced signal has varying power due to varying pulse width, which
could be considered a disadvantage.
PWM still works if synchronization between transmitter and receiver
fails, whereas pulse-position modulation does not.
Pulse Position Modulation (PPM)
Pulse modulation wherein the position of a constant-width and
constant-amplitude pulse is varied according to the amplitude of the
sample of the analog signal.
Some authors consider PPM, together with PWM, as a type of Pulse
Time Modulation (PTM).
The rightmost pulse could represent the maximum amplitude of the
analog signal, while the leftmost pulse could represent the minimum
amplitude of the analog signal.
Transmitter must send synchronizing pulses to operate timing circuits
in the receiver, which could be considered a disadvantage compared
to PWM.
Transmitter requires constant power output which could be considered
an advantage compared to PWM.
Used in special purpose communications systems (usually for
military) but is seldom used for commercial applications.
Pulse Amplitude Modulation (PAM)
Pulse modulation wherein the amplitude of a constant-width, constant
position pulse is varied according to the amplitude of the analog
signal.
The information signal is sampled at regular intervals, and each sample is
made proportional to the amplitude of the information signal.
The maximum amplitude of the pulse could represent the maximum
amplitude of the analog signal, while the pulse with minimum
amplitude could represent the minimum amplitude of the analog signal.
There are two types of PAM, namely:
Double polarity pulses could have positive and negative values.
Single polarity pulses could have either positive or negative values
only.
Used as an intermediate form of modulation with PSK, QAM, and PCM
Seldom used by itself.
PAM signals could be used to frequency modulate a carrier signal (PAM-
FM)
PAM signals could also be used to generate a Pulse Code Modulation
(PCM) signals.
Pulse Code Modulation (PCM)
Pulse modulation wherein the analog signal is sampled and then the sample is
converted to a serial n-bit binary code.
Each code has the same number of bits and requires the same length of time for
transmission.
Resulting signal has fixed length and fixed amplitude.
Is a binary system, wherein a pulse or lack of pulse could represent a 1 or 0.
Not really a type of modulation but rather a form of digitally coding analog
signals.
Commonly used in digital transmission systems.
PCM is the most prevalent form of pulse modulation, especially within the
public switched telephone network, because with PCM it is easy to combined
digitized voice and digital data into a single, high speed digital signal and
transmit it over metallic cables or fiver optic cables.
PCM signal may be transmitted as is, may be encoded into another digital signal
for use in digital transmission system (such as T1 or E1), or may be used to
modulate a carrier.
PCM is much better for noise immunity, as it is less affected by variations in
pulse shape, pulse amplitude, and pulse timing.
Simplified PCM System
Bandpass
filter
Sample
And hold
Analog to
Digital
converter
Digital to
Analog
converter
Hold
circuit
Low pass
filter
Receiver
Transmitter
Transmission medium
PCM PAM
PAM Pulse amplitude modulation
PCM Pulse code modulation
A codec (coder/decoder) could perform the PCM encoding and decoding
PCM signals can also be used to modulate a carrier (sine wave) signal and then
transmitted using analog transmission.
PAM Analog output
Analog
input
PCM
Digital transmission
(digital signals are transmitted)
PCM Sampling
Sample and hold circuit in PCM transmitter is used to sample
periodically the continually changing analog signal and convert the
sample to a series of constant amplitude PAM levels.
If the input to the analog to digital converter (ADC) is changing while
it is converting, aperture distortion will result.
The Nyquist sampling theorem establishes the minimum sampling rate
(fs) that can be used by a PCM system.
The theorem states that: For a sample to be reproduced accurately
at the receiver, each cycle of the analog signal must be sampled at
least twice. Thus, the sampling rate (fs) must be at least twice the
highest input frequency (fa).
If fs is less than twice the highest input frequency, aliasing or
foldover distortion will result.
The faster is the sampling rate, the better will be the quality of the
converted analog signal at the receiver.
fs >= 2fa
PCM Coding
Binary codes used for PCM are n-bit codes, where n may be any
positive integer greater than 1.
The codes currently used for PCM are sign-magnitude codes, where
the most significant bit (MSB) is the sign bit and the remaining bits are
used for magnitude.
With 2 magnitude bits, four codes are possible for positive numbers,
and four codes are possible for negative numbers (Total of 8 possible
codes).
Folded Binary Code for PCM
-3 1 1 0
-2 0 1 0
-1 1 0 0
-0 0 0 0
+0 0 0 1
+1 1 0 1
+2 0 1 1
+3 1 1 1
Decimal Level Magnitude Sign
Two codes are assigned to 0 volts.
Magnitude of minimum step size (resolution) is 1 volt.
Highest magnitude voltage is + 3 volts or 3 volts.
Voltages between +0.5 and +1.5 will be converted to 101.
Maximum input voltage is 3.5.
Resolution = min. step size
= 1 volt
Above .5 to 3.5 v
Above 1.5 to 2.5 v
Above 0.5 to 1.5 v
0 to +0.5 v
0 to 0.5 v
Below -0.5 to -1.5 v
Range
Folded Binary Code for PCM
Assigning PCM codes to absolute magnitudes is called quantizing.
Magnitude of minimum step size is called resolution, which is equal in
magnitude of the least significant bit.
Resolution is also the minimum voltage other than 0 volt which could be
decoded by the digital to analog converter.
The smaller the magnitude of the step size, the better (smaller) is the
resolution, and the more accurately the quantization interval will resemble the
actual analog sample.
Each code has a quantization range equal to + or one half the resolution,
except for the codes for 0 volt.
The maximum input voltage to the system is equal to the voltage of the
highest magnitude code plus one half of the voltage of the resolution or
minimum step size.
If magnitude of a sample exceeds the highest quantization interval, overload
distortion (also called peak limiting) occurs.
PCM Analog to Digital Conversion
0 v
-3 v
-2 v
-1 v
+ 3 v
+2 v
+1 v
0 v
-3 v
-2 v
-1 v
+ 3 v
+2 v
+1 v
+ 2 volts
+ 2.6 volts
- 1 volt
110
(1.5 v to 2.5 v)
001
(-.5 v to 1.5 v)
111
(2.5 v to 3.5 v)
(with quantization error or noise)
PCM codes
PAM signals
Analog signal
Sample Pulses
Quantization Error in PCM
Quantization error (Qe) results when the magnitude of the sample
(PAM signal) is rounded off to the nearest valid PCM code.
Because of quantization error, the converted analog signal at the
receiver will not be the same as the analog signal at the transmitter
side.
It is equivalent to additive noise because it alters the signal amplitude.
It may add to or subtract from the original signal.
It is also called quantization noise, and its maximum magnitude is
equal to one-half the minimum step size (Vlsb / 2).
Maximum Qe = V
lsb
/ 2
Dynamic Range for PCM
Dynamic range is the ratio of the largest possible magnitude to the smallest
possible magnitude that can be decoded by the digital to analog converter
(DAC), and it can be expressed as:
Where: DR = dynamic range (unitless)
Vmax = largest possible magnitude of voltage
Vmin = smallest possible magnitude of voltage
= resolution
= minimum step size
DR = Vmax / Vmin
= Vmax / resolution
= Vmax / min. step size
= decimal equivalent of maximum magnitude of PCM code
DR in decibels = 20 log (Vmax / Vmin)
Number of Bits Needed for PCM
The decimal equivalent of the minimum binary code for magnitude (after 0) is
always a 1. Thus:
DR = decimal equivalent of maximum binary code for magnitude / 1
= decimal equivalent of maximum binary code for magnitude
The minimum number of bits (excluding sign bit) required for PCMcode can
be computed as follows:
Where: n = minimum total number of bits in PCM code (excluding sign bit)
DR = absolute value of dynamic range
The above equation was derived from:
2
n
-1= DR
1 is subtracted from 2
n
to take into account the code used for 0 volt.
The total number of bits required for the PCM code including the sign bit can
be computed as follows:
Total number of bits for PCM including sign bit = n + 1.
log2
1) log(DR
n
+
=
Example: A PCM system has the following parameters: A maximum analog
input frequency of 4 khz, a maximum decoded voltage at the receiver of + or
2.55 volts, and a minimum dynamic range of 46 db. Determine the
following: minimum sample rate, minimum number of bits used in the PCM
code, resolution, and quantization error.
Solution:
fs = 2 fa = (2)(4,000) = 8,000 samples per second
46 db = 20 log (Vmax / Vmin)
46 / 20 = log (Vmax / Vmin) = 2.3
DR = Vmax / Vmin = 10
2.3
DR = 199.5
63 . 7
log2
1) log(199.5
n =
+
=
(number of bits needed for the
magnitude of the positive or negative
PCM codes)
The closest whole number greater than 7.63 is 8. Therefore 8 bits
must be used for the magnitude and 1 bit must be used for the sign. The total
number of bits needed for the PCM code is 9, and the total number of PCM
codes can be computed as:
Total number of PCM codes = 2
9
= 512 (255 are positive codes, 255 are
negative codes, and two codes are
for 0.)
The actual dynamic range can be computed as follows:
DR = 20 log 255 = 48.13 db
The actual resolution can be computed as:
Resolution = minimum step size = Vmax / (2
n
1) = 2.55 / (2
8
1)
= 0.01 volt
The maximum quantization error (Qe) = resolution / 2 = 0.01 / 2 = 0.005 v
Coding Efficiency for PCM
Coding efficiency is the ratio of the minimum number of bits required to
achieve a certain dynamic range to the actual number of PCM bits used.
It is a numerical indication of how efficiently a PCM code is utilized.
Coding efficiency could be computed as follows:
In the preceding example, the coding efficiency can be computed as:
coding efficiency = (8.63 / 9)(100) = 95.89 %
100 x
bit) sign (including bits of number actual
bits of number minimum
efficiency coding =
Signal to Quantization Noise Ratio in PCM
Signal voltage to quantization noise voltage ratio (SQR) is the ratio of the
input signal voltage to the quantization noise voltage (Vlsb / 2).
The worst possible signal voltage to quantization noise voltage ratio (SQR)
occurs when the input signal is at its minimum amplitude (101 or 001), and it
can be computed as:
Where: V
lsb
= voltage level corresponding to the least significant bit
= step size
= resolution
V
lsb
/ 2 = maximum quantization noise voltage
The SQR for the maximum input voltage can be computed as:
Percentage of error decreases as the magnitude of input signal increases.
2
/2 V
V
voltage noise on quantizati
voltage minimum
SQR
lsb
lsb
= = =
/2 V
V
voltage noise on quantizati
voltage maximum
SQR
lsb
max
= =
Linear Versus Nonlinear PCM Codes
PCM with linear codes use uniform magnitude change between successive
steps.
PCM with nonlinear codes use non-uniform magnitude change between
successive steps. This is called nonlinear or nonuniform encoding.
With voice transmission, low amplitude signals are more likely to occur.
If more codes are used for low amplitude signals, accuracy would
increase where it is needed, but fewer codes would be used for high
amplitude signals, thus resulting to lower SQR for high amplitude
signals.
Dynamic range will also increase as the ratio of Vmax to Vmin will also
increase.
Idle Channel Noise in PCM
Idle channel noise is random thermal noise which is inputted to the PAM
sampler during times when there is no analog input signal.
Idle channel noise is converted to a PAM signal just as if it were an analog
input signal.
To reduce idle channel noise, midtread quantization can be used.
With midtread quantization, the first interval is made larger in amplitude
than the rest of the steps. (With midrise quantization, the lowest magnitude
positive and negative codes, have the same voltage range as all the other
codes.)
Because of midtread quantization, input noise can be quite large and still
be quantized as a positive or negative zero PCM code, thus reducing
thermal noise.
In folded binary PCM, most of the residual noise is inherently eliminated
by the decoder.
The disadvantage of midtread quantization is larger possible magnitude for
Qe in the lowest quantization interval.
Coding Methods Used to Quantize PAM Signals
Three coding methods used to quantize PAM signals into 2
n
levels are:
Level at a Time Coding
Compares PAM signal to a ramp waveform while a binary counter is
being advanced at a uniform rate.
When the ramp waveform equals or exceeds the PAM sample, the counter
contains the PCM code.
Generally limited to slow speed applications
Digit at a Time Coding
Determines each digit of the PCM code sequentially.
Analogous to a weight balance where known reference weights are used to
determine unknown weights.
Provides compromise between speed and complexity.
Word at a Time Coding
Uses logic gates to sense the highest threshold circuit and to produce the
approximate PCM code.
More complex but are more suitable for high speed applications.
Impractical for large values of n.
Companding is the process of compressing, and then expanding.
The higher amplitude analog signals are compressed (amplified less than
the low amplitude signals) prior to transmission, then expanded
(amplified more than the low amplitude signals) at the receiver.
law companding is a form of analog companding used in the US and
Japan, and its compression characteristics may be expressed as:
where: Vmax = maximum uncompressed analog input amplitude
Vin = amplitude of the input signal at a particular instant of time
= parameter used to define the amount of compression
Vout = compressed output amplitude
The higher is, the more compression there is.
When approaches 0, Vout / Vin = Gain approaches 1, and there is no
compression.
PCM for Companding law
Gain
) ln(1
Vin/Vmax) ln(1 Vin) (Vmax /
Vin
Vout
) ln(1
Vin/Vmax) ln(1 (Vmax)
Vout
=
+
+
=
+
+
=
Example: For a compressor with = 255, determine the gain for the
following values of Vin: Vmax, 0.75 Vmax, 0.5 Vmax, and 0.25 Vmax.
Solution: Substituting the above values of Vin into the equation below, the
following values were computed:
PCM for Companding law
Gain
) ln(1
Vin/Vmax) ln(1 Vin) (Vmax /
Vin
Vout
=
+
+
=
3 0.25 Vmax
1.75 0.5 Vmax
1.26 0.75 Vmax
1 Vmax
Gain (Vout / Vin) Vin
Note: As the input voltage decreases, the gain increases.
Early Bell systems used =100 and 7 bit PCM code.
Newer Bell systems use =255 and 8 bit PCM code.
A-law Companding for PCM
Alaw companding is a form of analog companding used in Europe, and
its compression characteristics may be expressed as:
where: Vmax = maximum uncompressed analog input amplitude
Vin = amplitude of the input signal at a particular instant of time
A = parameter used to define the amount of compression
Vout = compressed output amplitude
Gain
lnA 1
) V / ln(AV 1
V
V
V
V
Gain
lnA) (1 V
) V / (AV V
V
V
lnA 1
) V / ln(AV 1
V V
lnA 1
V / AV
V V
max in
in
max
in
out
in
max in max
in
out
max in
max out
max in
max out
=
(
+
+
=
=
+
=
+
+
=
+
=
1
V
V
A
1
A
1
V
V
0
max
in
max
in
s s
s s
1
V
V
A
1
A
1
V
V
0
max
in
max
in
s s
s s
Vocoder
Vocoder is a special voice encoder/decoder used in PCM.
Used to encode the minimum amount of speech information
necessary to reproduce a perceptible message with fewer bits than
those needed by conventional encoders/decoders.
Decoded waveform often vaguely resembles the original
information signal.
Delta Modulation PCM
Uses a single bit PCM code to achieve digital transmission of analog signals.
The single bit transmitted indicates whether the present sample is larger or
smaller in magnitude than the previous sample.
1 indicates that current sample is larger than previous one.
0 indicates that current sample is smaller than previous one.
When analog input signal changes at a faster rate than the DAC can keep up
with, slope overload occurs (slope of analog signal is greater than what the
delta modulator can maintain.)
Increasing clock frequency or minimum step size reduces slope overload.
When original analog signal has relatively constant amplitude, the
reconstructed signal has variations not present on original signal. This is called
granular noise.
Adaptive Delta Modulation PCM
Adaptive Delta Modulation PCMis a delta modulation system where
the step size of the DAC is automatically varied depending on the
amplitude characteristics of the analog input signal.
After a predetermined number of consecutive 1s or 0s (slope of
DAC output is lower than slope of analog signal), the step size is
automatically increased to minimize slope overload.
When alternating sequence of 1s or 0s is occurring (possibility of
granular noise is high), DAC automatically reverts to its minimum
step size.
Differential PCM
Binary code proportional to the difference in the amplitude of two
successive samples is transmitted.
In conventional PCM encoded speech waveform, there are many
successive samples whose amplitudes are the same, thus resulting
to redundant transmission of codes.
Range of sample differences is typically less than the range of
individual sample amplitudes in conventional PCM, thus fewer bits are
transmitted.
Smaller bandwidth is likewise needed.
Hartleys Law
Hartleys Law can be expressed as:
levels coding of number M
(fb) rate bit sec) per (bits capacity channel C : where
M log B 2 C
: as expressed be can law s Hartley' noise, of absence total In the
ion time transmiss t
bandwidth channel B
(fb) rate bit capacity n informatio C : where
ion time) transmiss and bandwidth
the to al proportion directly is capacity on (Informati Bt C
2
=
=
= =
=
=
=
= =
Hartleys Law
When the binary coding system is used, the preceding equation reduces
to:
Hartleys law implies the following:
Bandwidth required to transmit information at a given rate is
proportional to the information rate.
In the absence of noise, the greater the number of levels in the coding
system, the greater the information rate that may be sent through a
channel.
Extending Hartleys Law, the following equation can be derived:

sec) per (bits capacity channel C : where
B 2 C
=
=
=
levels coding of number M (sec) ion time transmiss t
(hz) bandwidth channel B sec) per (bits capacity n informatio C
(bits) t in time sent n informatio Total H : where
M log t B 2
Ct H
2
= =
= =
=
=
=
Hartleys Law
Example: Given a bandwidth of 4 khz, and a number of coding level
(signal level) of 4, what is the maximum bit rate using Hartleys law.
How many bits are transmitted per coding level? If data is transmitted
continuously for 3 seconds, what is the total number of bits transmitted?
bits 48,000 (1600)(3) Ct H ed transmitt bits of number Total
2 (4) log level coding per bits of number Total
second per bits 16,000
(4) log (4,000) 2
rate bit M log B 2 C
2
2
2
= = = =
= =
=
=
= =
Shannon Limit for Information Capacity (Shannons law)
Shannons Law (also known as Shannon-Hartley theorem) takes into
account the effects of noise in the information capacity of a system.
Information capacity represents the number of independent symbols
that can be carried through the system in a given unit of time.
Information capacity is usually expressed in bits per second (bps).
The Shannon limit for information capacity is :
Where:
C= information capacity (bps)
B = Bandwidth (hz)
S / N = signal to noise power ratio at input of the receiver
To achieve Shannons limit for information capacity, digital
transmission systems that have more than two output conditions
(symbols) must be used. (example: 0001 = 45
0
, 1001 = 180
0
)
C = B log
2
(1+ S/N)
or
C = 3.32B log (1+ S/N)
It would be incorrect to assume that doubling the bandwidth of a noise-
limited channel will automatically double its capacity.
Doubling the bandwidth will also double the noise power, while
the signal power remains the same. This results in the reduction of
S/N ratio, and thus the information capacity will not be doubled.
The Shannon-Hartley theorem represents a fundamental limitation.
The only consequence of trying to exceed the Shannon limit would be
an unacceptable error rate.
Example: Calculate the information capacity of a standard 4 khz
telephone channel with a 32 db signal to noise ratio.
Standard telephone channel occupy the frequency range 300 to 3400
hz. The actual S/N ratio is antilog of 32 / 10 = 1585
C = B log
2
(1+ S/N)
C=(3400-300)log
2
(1+1585) = 32,953 bits per second
Example: system has a bandwidth of 4 khz and a signal to noise ratio
of 28 db at the input to the receiver. Calculate:
a. Its information capacity
b. The information capacity of the channel if its bandwidth is
doubled, while the transmitted signal power remains constant.
a. S/N = antilog (28/10) = 631
b. If the signal to noise ratio in the 4 khz channel is 631:1, this can be
interpreted as a noise power of 1 mWat some point in the channel
where the signal power is 631 mW. The signal power is not changed
when the bandwidth is doubled, while the noise power is doubled. We
thus have:
C = B log
2
(1+ S/N)
C=(4000) log
2
(1+631) = 37,216 bits per second
C = B log
2
(1+ S/N)
C=(8000) log
2
(1+631/2) = 66,448 bits per second
MULTIPLEXING
Multiplexing / Demultiplexing
Multiplexing
Process of combining information from several sources into a single
composite information signal
The transmission of information (in any form) from more than one
source to more than one destination over the same transmission
medium / facility
Done at the transmitter side
Demultiplexing
Process of separating individual information from a composite
information signal created during multiplexing.
Done at the receiver side
Multiplexing / Demultiplexing
Domains in Which Multiplexing Can Be Accomplished
Space
Phase
Time
Frequency
Wavelength
The most Predominant Methods of Multiplexing
Time Division Multiplexing (TDM)
Frequency Division Multiplexing (FDM)
Wavelength Division Multiplexing (WDM)
Code Division Multiplexing (CDM)
Time Division Multiplexing
Time is subdivided for use by different sources of information or
channels.
Transmissions from multiple sources occur on the same
communications medium / facility but not at the same time.
Transmissions from multiple sources are interleaved in the time
domain.
Interleaving could be: bit interleaved, byte interleaved, or sample
interleaved.
Multiplexing technique used for digital signals such as PCM and data
from computers.
Used for digital transmission systems such as T1 series and E1 series.
Time Division Multiplexing
TDM
Multiplexer
Computer A
Computer B
Computer C
Digital Signals
Flow of data
Digital Signals
1. Signals from each computer are transmitted at the output of the TDM multiplexer
one at a time.
2. TDM multiplexer has buffer (memory) to prevent lost of data from computers.
3. Digitized voice (PCM) could also be an input signal to a TDM multiplexer.
4. At the receiver side, the multiplexed signals are demultiplexed and distributed to
individual destinations.
Computer D
Computer E
Computer F
Digital Signals
TDM
Multiplexer
Statistical Time Division Multiplexing
Used widely.
Also uses time division multiplexing but time allotment for each input
could be different.
Could automatically prioritize channels depending on channel
transmission requirements.
More efficient transmission of data.
Commonly used for data communications (information signals from
computers are transmitted)
Time Division Multiplexing Used on PCM
With Pulse Code Modulation Time Division Multiplexing (PCM-
TDM), two or more voice channels are sampled, converted to PCM
codes, and then time division multiplexed onto a single metallic or
optical fiber cable.
For a sampling rate of 8000 samples per sec (2x4000) and 8 bits per
sample, the transmission speed is 64,000 bits per second per voice
channel.
If 24 voice channels are time division multiplexed, and fed to a T1
line, a T1 frame will consists of 192 bits per frame from the voice
channels (24 channels per frame x 8 bits per channel)
Each T1 frame will have an additional bit (framing bit) which is used
for synchronization.
Thus the total number of bits for each frame will be 193.
The transmission speed of a T1 line is 1.544 Mbps (193 bits per frame
x 8000 frames per sec.
Frequency Division Multiplexing
Multiple sources of information that originally occupied the same
frequency spectrum are each converted to a different frequency band,
and transmitted simultaneously over one transmission medium /
facility.
Conversion to different frequency band before transmission is
necessary to prevent the signals from interfering with each other.
Available frequency spectrum in the transmission medium is
subdivided for use by different sources of information.
Used for analog signals (input and output signals are all analog)
Frequency Division Multiplexing
Frequency
Division
Multiplexer
Analog
Signal A
Analog
Signal B
Analog
Signal C
Analog Signals
Analog Signals
Analog
Signal A
Analog
Signal B
Analog
Signal C
Analog Signals
Frequency
Division
Multiplexer
0 4000 Hz
0 4000 Hz
0 4000 Hz
0 4000 Hz (from analog signal A)
4000 Hz 8000 Hz, (from analog signal B)
8000 Hz 12000 Hz (from analog signal C)
0 4000 Hz
0 4000 Hz
0 4000 Hz
Available frequency spectrum in the transmission medium is subdivided for
use by different sources of information.
AT&T Frequency Division Multiplexing Hierarchy
Message Channel (1 voice channel - 0 to 4 Khz)
Message channel is the basic building block of the FDM hierarchy.
Message channel may now be used for non-voice signals such as data.
Basic voice band (VB) circuit is called 3002 channel, which can be
subdivided into 24 narrower 3001 (telegraph) channels.
Basic Group (12 voice band channels)
Consists of 12 frequency division multiplexed (FDM) message channels.
12 channel modulating block is called A-type (analog) channel bank.
Basic Supergroup (5 basic groups)
Consists of 5 frequency division multiplexed basic groups.
Basic Mastergroup (10 basic supergroups)
Jumbogroup (6 mastergroup or 3600 voice channels)
Multijumbogroup (7200 voice channels)
Superjumbogroup (10,800 voice channels)
Formation of a Group
Antialiasing
filter
Balanced
Modulator
Band pass
filter
Antialiasing
filter
Balanced
Modulator
Band pass
filter
.
.
.
Channel 12
Channel 1
Channel
Combining
Network
SSBSC
(60khz-64khz)
Basic Group
(12 voice channels)
FDM
Analog Signals
60Khz-108Khz
Voice
Analog
Signal
(300 3000 Hz)
DSBSC
(60khz-68khz)
Voice
Analog
Signal
(300 3000 Hz)
DSBSC
(104khz-112khz)
SSBSC
(104khz-108khz)
Each voice signal occupies 4 khz in the FDM multiplexed signal, whereas
the input voice signal range from 300 3000 hz.
For output of
balance modulator:
fc =112 khz- 4n
Where:
fc= carrier frequency
n= channel number
fc
fc
Formation of a Supergroup
Bandpass
filter
Balanced
Modulator
Group 1
Band pass
filter
Bandpass
filter
Balanced
Modulator
Group 5
Band pass
filter
.
.
.
Group 5
Group1
Channel
Combining
Network
SSBSC
(504khz-552khz)
Supergroup
(60 voice channels)
FDM
Analog Signals
312 khz-552 khz
Group Signals
(60khz-108khz)
DSBSC
(504Khz-720Khz)
Group signals
(60khz-108khz)
DSBSC
(312 khz-528khz)
SSBSC
(312khz-360khz)
Each voice signal occupies 4 khz in the FDM multiplexed signal, whereas
the input voice signal range from 300 3000 hz.
For output of
balance modulator:
fc =372+48n
Where:
fc= carrier frequency
n= channel number
fc
fc
Hybrid Data
With hybrid data, it is possible to combine digitally encoded signals
with FDM signals and transmit them as one composite baseband
signal. (digital signals are first converted to analog through digital
modulation)
The four primary types of hybrid data are:
Data under voice (DUV)
Data above voice (DAV)
Data above video (DAVID)
Data in voice (DIV)
DUV
FDM
DAV FDM
DAVID VSB Video
Wavelength Division Multiplexing (WDM)
Also called wave-division multiplexing.
Used in optical transmission.
Multiple digital signals using different wavelengths are transmitted through
one fiber optic cable.
Signals does not interfere because of different wavelengths /
frequencies.
Optical filters are used to separate the signals at the receiver.
Code Division Multiplexing (CDM)
Code division multiplexing (CDM) allows signals from a series of
independent sources to be transmitted at the same time over the same
frequency band.
Orthogonal codes are used to spread each signal over a large, common
frequency band.
At the receiver, the appropriate orthogonal code is then used again to recover
the particular signal intended for a particular user.
The key principle of CDM is spread spectrum. Spread spectrum is a means of
communication with the following features:
Each information-bearing signal is transmitted with a bandwidth in excess
of the minimum bandwidth necessary to send the information.
The bandwidth is increased by using a spreading code that is independent
of the information.
The receiver has advance knowledge of the spreading code and uses this
knowledge to recover the information from the received, spread-out
signal.
Spread spectrum and CDM are currently being used in an ever-increasing
number of commercial cellular telephone systems.
SWITCHING
Interconnecting Computers Using Leased Lines
Computer B
Computer B
Computer A
Computer A Computer C
Computer C
Computer D
Computer D
Leased
Line
Leased Line
Leased Line
Leased Line
Leased
Line
Leased Line
Number of lines needed = (N(N-1)) / 2
Where N is the number of computers
Switch
Switch is a hardware and/or software capable of creating temporary
connections between two or more devices linked to the switch but not
to each other.
Several switches can be used in creating temporary connections
between devices.
Some switching methods commonly used are:
Circuit switching
Packet switching
Message Switching
Circuit Switching
Circuit switching is a switching method wherein direct physical
connection/s between two or more devices is/are created.
A circuit switch is a device with n inputs and m outputs.
The number of inputs does not have to match the number of outputs.
Circuit switching can use either:
Space division switch
Time division switch
Space division switching is a switching method wherein the paths in the
circuit are separated from each other spatially. An example is the crossbar
switch.
Crossbar switch connects n inputs to m outputs in a grid, using
electronic microswitches at each crosspoint.
Number of crosspoints needed = (n)(m)
Impractical if too many crosspoints are needed.
Multistage Switch is used to prevent large number of crosspoints
at each crossbar switch.
Circuit Switching
Time Division Switching ( TDM) is a switching method wherein time division
multiplexing is used.
The two popular methods used in time division switching are:
Time slot interchange
TDM bus
The following are considerations regarding multistage switching:
Blocking (input cannot connect to intended output) can occur if there is no
available path between switches in a multistage switch configuration.
Example is telephone network congestion
Increasing the number of switches / path in multistage switching will increase
the cost of the system.
Space switching and TDM switching can be combined to take advantage of each
switching method.
Space division switching is instantaneous.
Time division multiplexing needs no crosspoints which could be inefficient.
Example: Time Space Time
Space Time - Space
Time Slot Interchange
TDM
Multiplexer
Computer A
Computer B
Computer C
Digital Signals
Flow of data
Digital Signals
TDM
Multiplexer
Computer D
Computer E
Computer F
Digital Signals
TSI
Time
Slot
Interchange
Signals from Computers A,B, and C could be switched to
computers D,E, or F using the TSI
Smart Switching Multiplexers
Smart
Switching
Multiplexer
Computer A
Computer B
Computer C
Digital Signals
Flow of data
Digital Signals
Smart
Switching
Multiplexer
Computer D
Computer E
Computer F
Digital Signals
Signals from Computers A,B, and C could be switched to
computers D,E, or F using Smart Switching Multiplexers
TDM Bus
Control
Unit
Bus
(Set of wires)
A
B
C
D
E
F
Input and output lines are connected to high speed bus through input
and output switches. Switches are during allotted time slot.
Switches
Switches
Packet Switching
Networks using this are also called hold and forward network because
packets can be stored in switches for a short period of time.
Data to be transmitted are cut into packets.
Headers and trailers, such as source and destination addresses, are
attached to the packets of data.
One communication medium may be shared by different information
sources.
Network consists of switches which route the packets.
Packets may take alternative routes in the network, thus reliability is
improved.
Original information is reconstructed at the receiver.
Advantages of packet switching are:
Better reliability because of alternate routes
More efficient use of network facilities (shared facilities)
More flexible
Could be more cost effective instead of using leased lines
Typical service charge is fixed rate plus charge per packet transmitted.
May be more costly if many packets are to be transmitted continuously.
Example is X.25 packet switching and frame relay.
Packet Switching Network
PAD
Switch
(Cebu)
Switch
(Cebu)
Switch
(Manila)
Switch
(Manila)
Switch
(HongKong)
Switch
(HongKong)
Switch
(Baguio)
Switch
(Baguio)
Switch
(New York)
Switch
(New York)
Switch
(Singapore)
Switch
(Singapore)
Computer C
(New York)
Computer A
(Manila)
Terminal
(Baguio)
Computer B
(Singapore)
User Data
... P1 H P n H T P 2 H T T
H - Header
T - Trailer
P - Payload
PAD Packet assembler
/ disassembler
Packet Switching Approaches
1. Virtual circuit approach
o A virtual circuit (logical connection) is established.
o A virtual circuit identifier is used to identify where the data
should be transmitted.
o Virtual circuit may be permanent virtual circuit or switched
virtual circuit.
2. Datagram approach
o No virtual circuit (logical connection) is created.
o Each packet has its own destination and source addresses.
o Packets are routed based on addresses / identifiers in each
packet.
o Each packet is treated independently of all the others.
o Packets could take alternative routes.
o Packets are referred to as datagrams.
Typical X.25 Packet Format
(Virtual Circuit Packet Switching)
LCGN GFI LCN Facilities
Facility
Field
Length
Packet
Identifier
Address
Length
Source/
Destination
Address
Data
End
of
Packet
Framing
Error
Detecting
Code
Address
Field
Payload
Logical Channel Field
and Format Identifier
Message
Number
Trailer
LCGN Logical Channel Group Number
GFI General Format Identifier
LCN Logical Channel Number
Facilities
Field
Message Switching
Networks using this are also called store and forward networks.
Entire message is transmitted, stored in switch/switches, and
forwarded to destination. (store and forward) when it is convenient to
do so.
Message can be routed through any number of switches.
Messages are delivered when it is convenient to do so.
There could be substantial delay in routing the message.
Message format can be converted by the switches.
Example: Text messages
TRANSMISSION MEDIA
Guided Transmission / Communications Media
Guided Transmission Media provides a conduit from one device to
another, wherein the signals are confined as they are propagated.
Some of the guided transmission media are:
Unshielded cable (no twist)
Shielded cable (no twist)
Unshielded twisted pair cable (UTP)
Shielded twisted pair cable (STP)
Coaxial cable
Fiber optic cable
Waveguide
Unshielded Cable (No Twist)
Uses copper
Conductors have the same physical characteristics
Less expensive
More susceptible to noise
Used for electrical signals
Used for analog or digital signals (slow speed, such as 9600 bps)
Shielded Cable (No Twist)
Uses copper
Conductors have the same physical characteristics.
Conductors are inside another conductor or foil which acts as shield.
Shield can be connected to ground to protect inside conductors from
noise.
Less susceptible to noise than unshielded cables
More expensive and bulky than unshielded cables
Used for analog or digital signals (slow speed, such as 9600 bps)
Unshielded Twisted Pair Cable
Uses copper
Both conductors have the same physical characteristics
Two insulated conductors are twisted with one another.
Relatively less susceptible to noise if receiver has differential
amplifiers.
The greater the number of twist, the better noise rejection is, but cable
is more expensive.
Examples: CAT 1, CAT2, CAT 3, CAT 4, CAT 5, CAT 6, CAT 7
Many applications for this cable use RJ45 connectors.
CAT Cables
Mostly LAN Mostly Digital 600 Mbps 600 Mhz 7
T1 / slow LAN Analog / Digital 2 Mbps <2 Mhz 2
Telephone,
slow LAN
Analog / slow
digital
<100 Kbps Very low 1
Use Digital / Analog Data Rate Bandwidth Category
Shielded Twisted Pair Cable
Uses copper
Both conductors have the same physical characteristics
Two insulated conductors are twisted with one another.
Twisted cables are inside another conductor or foil which acts as
shield.
Shield can be connected to ground to protect twisted pairs from
noise.
Relatively less susceptible to noise if receiver has differential
amplifiers.
The greater the number of twist, the better noise rejection is, but
cable is more expensive.
More expensive and bulky than unshielded twisted pair cables
Coaxial Cable (Coax)
Uses copper
Has inner conductor and outer conductor surrounding the inner
conductor
Outer conductor can be connected to ground to protect inner
conductor from noise
Can be used for signals having higher frequencies than those
transmitted using twisted pair cables
Some standard coaxial cables being used are:
RG59 (75 ohms, used in cable TV)
RG58 (50 ohms, used in Thinwire Ethernet)
RG11 (50 ohms, used in Thickwire Ethernet)
Many applications for this cable use BNC connectors
Fiber Optic Cable
Made up of glass or plastic
Used for optical signals
Immune from many noises such as lightning, surges due to motors,
other interference from electrical signals.
Propagation of optical signals may be:
Multimode
Multiple beams of light move in different paths.
Can have two forms: Step index and Graded index
Single Mode
Beams of light travel almost in the direction to which the
fiber is folded.
Multimode Step Index Fiber
The density of the core remains constant from the center to the
edges.
Beam of light moves in a straight line until it reaches the
interface of the core and the cladding, where the light
changes direction.
Multimode Graded Index Fiber
The density of the core varies from its center to its interface with the
cladding.
Density of the core is highest at the center and decreases
gradually to its lowest at the edges / interface with the cladding.
Beam of light does not move in a straight line but instead in a
curved pattern, because of the varying density of the core.
Single Mode Fiber
Uses a core with constant density (step mode), and a highly focused
light source.
Light travels through a core with much smaller diameter than
multimode fibers.
Light travels almost in the direction to which the fiber is folded.
Propagation of beams is almost identical, and delays are
negligible.
All beams arrive at the destination almost together.
All beams can be recombined with almost no distortion to the
signal.
Typical Fiber Types
Single mode 125 7 7/125
Multimode, graded
index
125 100 100/125
Multimode, graded
index
125 62.5 62.5/125
Multimode, graded
index
125 50 50/125
Mode Cladding
Diameter (m)
Core Diameter
(m)
Type
Fiber Optic Cable Connectors
Subscriber channel connector (for cable TV)
Straight tip connector (for networking devices)
MT-RJ (same size as RJ45)
Waveguide
Uses hollow conductive tubes, usually rectangular in cross section,
but sometimes circular or elliptical.
Used to confine the propagation of radio waves inside hollow
tubes.
Used for signals around 1 Ghz and above (except signals using
fiber optics)
Parallel wire transmission, such as those using coaxial cables,
could not efficiently be used for frequencies above 1 Ghz,
because of radiation losses and skin effect.
Used for microwave transmission.
Unguided Transmission Media
Signals are not confined in a particular transmission medium.
Unguided transmission media include the following:
Earths surface (including land and sea)
Earths atmosphere (including lower and upper layers)
Space (vacuum)
Electrical signals are converted to radio signals
Three types of signal propagation can be used:
Ground wave (surface wave) propagation
Sky wave propagation
Space wave propagation (includes line of sight and reflected wave
propagation)
Three Ways of Propagating Electromagnetic Waves
Ground wave (surface wave) propagation
Signal travels through the earths surface
Used for relatively low radio frequency signals
Provides good coverage for frequencies below 1.5 Mhz
Ground losses increase rapidly with frequency.
Space wave propagation
Signal travels in almost a straight line from one point to another in the earths
atmosphere.
Distance of propagation is limited by earths curvature.
Includes direct wave (Line of sight) and ground reflected wave
Used for very high frequency signals
Sky wave propagation
Signal travels through the earths atmosphere, and, upon reaching the
ionosphere, is refracted or reflected back towards the earths surface.
Used for relatively high radio frequency signals.
TELECOMMUNICATIONS
FACILITIES
Telephone Network
Widely used
Offers flexibility.
Wireless service improves user mobility.
Relatively cheap.
Different service providers are available.
Dial up connection or leased line could be used.
Metered services could be expensive.
Echo suppressors, which are used for long distance telephone calls, must
be disabled to allow full duplex data communications.
Echo suppressors eliminate the weaker signal traveling in either
direction, to suppress echoes in the transmission line.
If echo suppressors are enabled, a signal coming from one of the
DCEs will be suppressed.
Disabling of echo suppressors could be done by a tone generated by
a DCE as activated by the DTE. The secondary channel of an RS232
port could be used to produce the tone needed to disable the echo
suppressor.
A 2025 hz tone applied for around 300 ms could disable echo
suppressors.
If a gap in data transmission of 100 ms or more occurs, echo
suppressors will be reactivated.
Leased Lines
Circuit is dedicated to the subscriber (fixed)
Available for low, medium and high speed applications
Less noise and interference
Quality of the line could be negotiated.
Line can be conditioned for better transmission.
Adaptive equalization (frequency response is automatically adjusted) or
preset equalization (frequency response is set prior to transmission) can be
used.
Delay due to setting up a call is eliminated
Cost could be cheaper.
Advisable to be used if transmission is done most of the time
Flexibility is reduced
Payment is fixed regardless of usage.
Planning for disaster recovery and expansion is critical.
Leased lines could be:
Voice Grade Line
T1/E1or higher aggregates
SONET/SDH
Could use different communications facilities such as microwave radio..
Telegraphy / Telex
The first successful practical data communications system, which was
invented by Samuel F. B. Morse in 1982.
Telegraphy is a form of communication that employs typewriter like
machines, and is used to send written messages from one point to
another.
A user lodges a written message for transmission at telegraph or post
office.
The message is subsequently transmitted to the office nearest to the
addressee, and delivered in typewritten form.
Telex combines the above system with subscriber dialing techniques.
Machines are placed at subscribers site.
Machines are linked to switching system which enables a
subscriber to send and receive messages to and from another
subscriber.
Telegraphy, telex, and facsimile are referred to as recorded services,
because they provide printed record.
T1 And E1 Lines
Used for digital services
T1 series is for U.S.
E1 series is for Europe
The E-carrier standards form part of the Plesiochronous Digital
Hierarchy (PDH)
G.703 is an ITU standard for transmitting voice or data over digital carriers
such as T1 and E1.
ITU Recommendation G.7043 "Virtual concatenation of plesiochronous
digital hierarchy (PDH) signals" enhances the frame structures of PDH
signals of 1544, 2048 and 44 736 kbit/s.
Initially intended to carry voice converted to digital signals
Circuit is fixed.
Unchannelized services now being offered
T1 speed is 1.544 Mbit/sec, E1 speed is 2.048 Mbit/sec
Could use different telecommunications facilities along the transmission path
No packet switching capability
Initial development of T1 was not based on standards
T1 series is not compatible with E1 series
The basic unit of the T-carrier system is the DS0, which has a transmission
rate of 64 kbits/s, and is commonly used for one voice circuit.
AT&T North American Digital Hierarchy
336 560.160 T5
168 274.176 T4
28 46.304 T3
4 6.312 T2
2 3.152 T1C
1 1.544 T1
No.
of T1
Line speed
(Mbps)
Category
-T1 can be multiplexed to form T2, T2 can be multiplexed to form T3, and so on.
-Muldems (Multiplexers / Demultiplexers) are used to upgrade to higher levels.
-Muldem designation (such as M12) identify the input and output digital signals.
-M12 interfaces DS-1 to DS-2, and M23 interfaces DS-2 to DS-3.
-Digital signals are routed at central locations called digital cross connects.
-Other signals such as picturephone and TV signals can be used as input signals.
DS-5
DS-4
DS-3
DS-2
DS-1C
DS-1
Signal
Voice/ pictphone/TV
Voice/ pictphone/TV
Voice/ pictphone/TV
Voice/ picturephone
Voice
Voice
Services
Offered
CCITT (E1 series) Digital Hierarchy
E1 can be multiplexed to form E2, E2 can be multiplexed to form E3, and so on.
564.992 Mbit/s E5
139.264 Mbit/s E4
34.368 Mbit/s E3
8.448 Mbit/s E2
2.048 Mbit/s E1
64 kbit/s E0
Rate Signal
SONET and SDH
SONET stands for Synchronous Optical Network.
SONET standard was defined by Telcordia and American National
Standards Institute (ANSI) standard T1.105.
SDH stands for Synchronous Digital Hierarchy
ITU standard G.707,G.783, G.784, and G.803.
The SDH standard was originally defined by the European
Telecommunications Standards Institute (ETSI)
SONET was developed for US, SDH was developed for Europe.
SONET and SDH were originally developed for optical transmission
(fiber optics)
SONET and SDH were developed to replace the Plesiochronous Digital
Hierarchy (PDH).
SONET and SDH were the choice for transporting Asynchronous
Transfer Mode (ATM) frames.
Has very high bandwidth which could support future transmission
requirements (higher than T1 and E1 series)
Multiplexing system is similar to conventional time division multiplexing.
Can be used to carry multiple T1 or E1 signals
Basic SDH / SONET Transmission Diagram
Line Terminal
Multiplexer
Line Terminal
Multiplexer
T1 / E1 T1 / E1
R
Repeater
R
Repeater
Fiber Optic Cables
SONET / SDH Transmission
Facilities
SDH and SONET Signals Comparison
SONET - Synchronous Optical Network
OC - Optical Carrier
STM - Synchronous Transport Mode
159,252,480 153,944,064 STM-1024 STS-3072 OC-3072
39,813,120 38,486,016 STM-256 STS-768 OC-768
9,953,280 9,621,504 STM-64 STS-192 OC-192
2,488,320 2,405,376 STM-16 STS-48 OC-48
1,244,160 1,202,688 STS-24 OC-24
622,080 601,344 STM-4 STS-12 OC-12
155,520 150,336 STM-1 STS-3 OC-3
51,840 50,112 STM-0 STS-1 OC-1
Line Rate
(Kbit/s)
Payload
bandwidth
(Kbit/s)
SDH level
and Frame
Format
SONET
Frame
Format
SONET
Optical
Carrier
Level
Terrestrial Microwave Communications
Used in point to point communications.
Provides high bandwidth.
Low power consumption.
Typical distance between stations is 20 to 30 miles.
Transmitter and receiver must have line of sight.
Careful planning with regards to obstruction must be done.
Weather condition and reflective surfaces could affect reliability.
Satellite Communications
Uses the same transmission techniques as Terrestrial Microwave
Communications
Designed to cover a very wide area and to reach isolated places.
Terrestrial microwave distance limitation is overcome
Many Channels could be used at the same time because of high bandwidth.
Geosynchronous (altitude:19,000 to 25,000 mi.) and non-geosynchronous
satellites (altitude: lower than 19,000 miles) could be used.
Not advisable for delay sensitive applications
Not advisable for half duplex error control.
Security of data being transmitted should be considered.
X.25 Packet Switching Network
An ITU-T standard protocol suite for packet switched wide area
network communication.
X.25 was originally defined by the International Telegraph and
Telephone Consultative Commitee (CCITT, now ITU-T) in a series of
drafts and finalized in a publication known as The Orange Book in
1976.
Uses packet switching techniques
Data to be transmitted are cut into packets
Addresses and error detection fields are attached to packets of data to
be transmitted.
Packets may take alternative routes in the network
Better reliability
More efficient use of network facilities
More flexible
Could be more cost effective
X.25 Packet Switching Network
X.25
PAD
X.25
Switch
(Cebu)
X.25
Switch
(Cebu)
X.25
Switch
(Manila)
X.25
Switch
(Manila)
X.25
Switch
(HongKong)
X.25
Switch
(HongKong)
X.25
Switch
(Baguio)
X.25
Switch
(Baguio)
X.25
Switch
(New York)
X.25
Switch
(New York)
X.25
Switch
(Singapore)
X.25
Switch
(Singapore)
Host C
(New York)
Host A
(Manila)
Terminal
(Baguio)
Host B
(Singapore)
Packets of User Data
Typical X.25 Packet Format
LCGN GFI LCN Facilities
Facility
Field
Length
Packet
Identifier
Address
Length
Source/
Destination
Address
Data
End
of
Packet
Framing
Error
Detecting
Code
Address
Field
Payload
Logical Channel Field
and Format Identifier
Message
Number
Trailer
LCGN Logical Channel Group Number
GFI General Format Identifier
LCN Logical Channel Number
Facilities
Field
Frame Relay
Uses packet switching technology similar to X.25 packet switching but
has a lot less overhead because of the following:
Has shorter control and address fields
Uses much less Operation and Maintenance procedure compared to
X.25
Less error detection and correction procedures
Less flow control procedures
Less traffic and congestion control
No diagnostics procedures used in X.25
Does not use receive acknowledgement procedures
Less overhead and procedures result to faster data transmission
Does not use receive acknowledgement procedures.
Does not use send and receive sequence numbers.
Relies heavily on reliability of communications facilities and user
devices.
Frames with error are normally discarded by the network.
Frame Relay Standards
Approved Q.933 Standard
T1.617
(previously known
as T1.6fr) Access Signaling
Approved Q.922 Annex A Standard
T1.618
(previously known
as T1.6ca) Core Aspects
Approved I.233 Standard T1.606
Service
Description
Status ITU Standard Status ANSI Standard Description
Frame Relay Frame
Ending Flag
Frame
Check
Sequence
Payload
(User Data)
Starting
Flag
Control and
Address
16, 24 or 32 bits 8 bits 8 bits 8 bits N bits
Frame Relay Frame Structure
Starts and ends with a flag
Address and control fields are combined
Address and Control field could have 2, 3 or 4 bytes
Frame size is not fixed
Maximum length of Payload Field is defined by service provider
Always has Frame Check Sequence (FCS) character
Frame Relay Frame With 2 Byte Control and Address
Field
Ending
Flag
Frame
Check
Sequence
Payload
(User Data)
Starting
Flag
Control
and
Address
C/R EA DLCI FECN BECN DE EA DLCI
Byte 2 Byte 1
1 2 3 4 5 to 8 1 2 3 to 8 Bit
Frame Relay Frame Address and Control Field Format
Address and Control Field with 2 bytes
1 2 Bit No.
Bit No.
Bit No.
1
1
2
2
3 4 5 6 7 8
3
3
4
4
5
5
6
6
7
7
8
8
EA = 1
EA = 0
EA = 0
EA = 1
EA = 0
EA = 0
EA = 0
EA = 1
EA = 0 C/R
C/R
C/R
DE
DE
DE
D/C
D/C
DLCI
DLCI
DLCI
DLCI
DLCI
DLCI
DLCI
FECN
FECN
FECN
BECN
BECN
BECN
DLCI or Control Field
DLCI or Control Field
Byte 1
Byte 1
Byte 1
Byte 2
Byte 2
Byte 2
Byte 3
Byte 3
Byte 4
Frame Relay Frame
Starting Flag - indicates beginning of frame
Ending flag - indicates end of frame
EA Extended Address - Address Extension set to 0 if more octets
follow the header of the frame, and set to 1 to indicate the end of the
header.
C/R - Command / Response indicates if the frame is a command or
response.
DE - Discard Eligibility set to 1 to indicate that the frame is more
eligible for discarding in case there is a problem in the network.
FECN - Forward Explicit Congestion Notification used to inform
destination of frame that there is congestion in the network. (Set to 1 if
there is congestion)
BECN - Backward Explicit Congestion Notification used to inform the
source of traffic that there is congestion in the network. (Set to 1 if there
is congestion)
DLCI - Data Link Control Identifier, Data Link Connection Identifier
serves as the address of the connection.
Asynchronous Transfer Mode (ATM)
Cell-based switching technique that uses asynchronous time division
multiplexing.
It encodes data into small fixed-sized cells (53 bytes) and provides data link
layer services that run over OSI Layer 1 physical links.
This differs from other technologies based on packet-switched networks (such
as the Internet Protocol or Ethernet, in which variable sized packets (known as
frames when referencing Layer 2) are used.
ATM exposes properties from both circuit switched and small packet switched
networking, making it suitable for wide area data networking as well as real-
time media transport.
ATM uses a connection oriented model and establishes a virtual circuit
between two endpoints before the actual data exchange begins.
ATM is a core protocol used over the SONET / SDH backbone of the
Integrated Service Digital Network.
Source: Wikipedia
Asynchronous Transfer Mode (ATM)
Design to support all types of signals (existing and future applications),
initially through use of fiber optics.
Platform for Broadband ISDN (BISDN).
User devices are connected to ATM switches using ATM Network Interface
Cards
A virtual connection is established between user devices.
Quality of service could be negotiated during connection establishment.
Connection could be Permanent virtual circuit or switch virtual circuit.
More than one virtual connection could be established using one physical link.
Data to be transmitted are first cut into fixed length cells (48 bytes)
Headers (5 bytes) are added into the data.
Contains abbreviated addresses
Cell loss priority bit, others
Fixed length cells (53 bytes) are transmitted using the addresses on the cell
headers.
Cells can take different routes.
Cells with error on the header could be discarded by receiver.
ATM Technology
Connection oriented protocol
Capable of handling all types of signals
Design to support existing and future applications
Uses statistical multiplexing
Simpler addressing scheme
Used in Broadband ISDN
Could use existing communications technology
Could be used for local area networks
more than one station could use the network at the same time
negotiable and flexible bandwidth allocation
could be used as network backbone
faster transmission of data
Easier routing of data
Could be used to interconnect existing local area networks
ATM networks could provide LAN emulation so it could interconnect
traditional LANs
Could use Mutiprotocol Over ATM
ATM Technology
ATM supports different types of services via Adaptation Layers (AAL).
Standardized AALs include AAL1, AAL2, AAL3, AAL4, and AAL5.
AAL1 is used for constant bit rate (CBR) services and circuit emulation.
Synchronization is also maintained at AAL1.
AAL2 through AAL4 are used for variable bit rate (VBR) services
AAL5 for data.
ATM ITU Standards
Defines the ATM Operation and Maintenance
(OAM) functions. ITU-T I.610
Defines the ATM Adaptation Layer protocols. ITU-T I.363
Defines the ATM Layer functions. ITU-T I.361
ATM Network Operation
ATM
Switch
ATM
Switch
ATM
Switch
UNI
UNI
UNI
UNI
User Device A
User Device E User Device C
User Device B
User Device D
Physical
Medium
Virtual Path 1
Virtual Path 2
Virtual Channel 30
Virtual Channel 40
Virtual Channel 50
Virtual Channel 60
Virtual Paths and Virtual Channels on ATM Networks
ATM Switch
ATM Switch
User Device User Device
Virtual Path Switch and Virtual Channel Switch
VC Switch
User Device A
VC Switch
User Device B
VPI = 1 VCI = 5
VPI = 2 VCI = 5
VPI = 3 VCI = 6
VPI = 4 VCI = 7
VPI = 5 VCI = 7
VP Switch
VP Switch
GFC
Data
VPI
VPI VCI
VCI
VCI PTI CLP
HEC
Byte
1
2
3
4
5
6 to 53
ATM Cell Structure at UNI
GFC - Generic Flow Control (4 bits)
VPI - Virtual Path Identifier (8 bits)
VCI - Virtual Channel Identifier (16 bits)
PTI - Payload Type Identifier (3 bits)
CLP - Cell Loss Priority (1 bit)
HEC - Header Error Control (8 bits)
48 Bytes Payload
5 Bytes Header
Data
VPI
VPI VCI
VCI
VCI PTI CLP
HEC
Byte
1
2
3
4
5
6 to 53
ATM Cell Structure between ATM switches
VPI - Virtual Path Identifier (12 bits)
VCI - Virtual Channel Identifier (16 bits)
PTI - Payload Type Identifier (3 bits)
CLP - Cell Loss Priority (1 bit)
HEC - Header Error Control (8 bits)
48 Bytes
5 Bytes
ATM Cell Structure
Generic Flow Control (GFC) - used at the UNI to control the flow of
traffic in the network.
Virtual Path Identifier (VPI) - used as one of the addresses of the
virtual connection. Many virtual paths could exist in one physical line.
Virtual Channel Identifier (VCI) - used as one of the addresses of the
virtual connection. Many VCIs could exist in one VPI
Payload Type Identifier (PTI) - used to identify the type of cell
whether it is for OAM, data, etc.
Cell Loss Priority (CLP) - used to indicate if the cell is of lower
priority
Header Error Control (HEC) - used to detect errors on the header of
the cell.
Data - User data which may also contain identifiers on type of data
being transmitted, and CRC for data
PTI
Code
Indication
User data cell, congestion not experienced, SDU type 0
ATM Payload Type Identifier Field
000
001 User data cell, congestion not experienced, SDU type1
010 User data cell, congestion experienced, SDU type 0
011 User data cell, congestion experienced, SDU type1
100 Segment OAM F5 flow related cell
101 End to End OAM F5 flow related cell
110 Reserved for future traffic control and resource
management
111 Reserved for future use
Integrated Services Digital Network (ISDN)
A set of communications standards for simultaneous digital
transmission of voice, video, data, and other network services over the
traditional circuits of the public switched telephone network
It was first defined in 1988 in the CCITT Red book.
The key feature of ISDN is that it integrates speech and data on the
same lines, adding features that were not available in the classic
telephone system.
There are several kinds of access interfaces to ISDN defined as Basic
Rate Interface (BRI), Primary Rate Interface (PRI) and Broadband
ISDN (B-ISDN).
ISDN is a circuit switch telephone network system and is designed to
allow digital transmission of voice and data over ordinary telephone
copper wires.
The signaling channel (D channel) uses Q.931 for signaling.
Intended to provide worldwide telecommunications support for voice,
data, video, and facsimile information within the same network.
Intended to integrate a wide range of services into one network.
Supports switched and non-switched connections.
64 kbps digital connection is the building block of ISDN
The three basic types of channels are:
B channel: 64 kbps (for voice and data)
D channel: 16 or 64 kbps ( for signaling information)
H channel: 384, 1536, 1920 kbps (data, video, fax, high quality audio)
The basic rate interface (BRI) includes 2 B channels and 1 D channel.
(2B+D).
Two entry devices used to connect DTEs to ISDN are:
Terminal Equipment Type 1 (TE1) supports standard ISDN interface.
Terminal equipment Type 2 (TE2) for non-ISDN such as RS232
Terminal Adapter (TA) translates incompatible protocols.
(also called narrowband ISDN)
The interface specifies the following network interfaces:
The U interface is a two-wire interface between the exchange and
a network terminating unit, which is usually the demarcation point
in non-North American networks.
The T interface is a serial interface between a computing device
and a terminal adapter, which is the digital equivalent of a
modem.
The S interface is a four-wire bus that ISDN consumer devices
plug into; the S & T reference points are commonly implemented
as a single interface labeled 'S/T' on an NT1.
The R interface defines the point between a non-ISDN device and
a terminal adapter (TA) which provides translation to and from
such a device.
Source: Wikipedia
Digital Subscriber Line (DSL)
Digital Subscriber Line (DSL) is a family of technologies that
provides digital data transmission over the wires of a local telephone
network.
DSL originally stood for digital subscriber loop.
In telecommunications marketing, the term Digital Subscriber Line is
widely understood to mean Asynchronous Digital Subscriber Line
(ADSL), the most commonly installed technical variety of DSL.
DSL service is delivered simultaneously with regular telephone on
the same telephone line. This is possible because DSL uses a higher
frequency. These frequency bands are subsequently separated by
filtering.
The data throughput of consumer DSL services typically ranges from
256 Kb/s to 24 Mbit/s in the direction to the customer (downstream),
depending on DSL technology, line conditions, and service-level
implementation.
In ADSL, the data throughput in the upstream direction, (i.e. in the
direction to the service provider) is lower, hence the designation of
asymmetric service.
In Symmetric Digital Subscriber Line (SDSL) service, the
downstream and upstream data rates are equal.
Source: Wikipedia

ECE 513 - Data Communications Part 1

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For FSK, the modulation index is generally kept below1.0, to produce a

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