Data Comms Part 1

Course: IT 105Data 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

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 PCModem Modem

Cables

PC PC

Cables

(Modem not required)


TelecommunicationsFacilities

PC

PC

PC

PC

Modem

ModemModem

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)

DTEDCE DCE

DTE DTEDCE DCE

Communications Medium(earth’s atmosphere)

Radio signals

DTE - Data Terminal Equipment

DCE - Data Communications Equipment / Data Circuit Terminating EquipmentExample: modem

BLOCK DIAGRAM OF BASIC DATA COMMUNICATIONS SYSTEM

Telecommunications / Transmission

Facilities

DTE

DTE

DTE

DTEDCE

DCE

DCEDCE

DTE - Data Terminal Equipment

DCE - Data Communications Equipment / Data Circuit Terminating EquipmentExample: modem

DATA COMMUNICATIONS WITHOUT USING DCE

DTE

Communications Medium

DTE

Note: It is possible to have data communications without using a DCE in some instances, such as when DTE’s 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• Communications Medium

– Transmission path between DCEs, or between DTE and another DTE

– Could be copper cables, fiber optic cables, earth’s atmosphere, earth’s 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 sytems.

• 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

Common Terms Used In Data Communications• 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 subscriber’s

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 (25)– 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 such.• Typically used with “dumb terminals”

Asynchronous Data Transmission

DTE DTEDCE DCE

Medium

0 volts

Start Bit

StopBit

Data Bits

0 1Signal inside DTE

Asynchronous Data Transmission

• 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.

Asynchronous Data Transmission– 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

SpikeDetectEnable

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 FlagControl

Secondary Station Address

NR P/F FINS

NR P/F RR FI

InformationFrame

S I Frame

8 bits8 bits8 bits 8 bitsN bits 8 or 16 bits

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 realtime 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.

Common Error Detection Techniques• 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).

Common Error Detection Techniques

• 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.

– Single precision checksum – binary addition is performed on the data to produce checksum character. If a carryout occurs, the carry bit is ignored.

Common Error Detection Techniques– 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 checksumcharacter.

– 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.


• 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.


• 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.

Common Error Detection Techniques– 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 – X12 + X11 + X3 + X2 + X1 + X0 (1100000001111)• CRC ITU - X16 + X12 + X5 + X0 (10001000000100001)• CRC 16 - X16 + X15 + X2 + X0 (11000000000000101)

• Where X0 = 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) = X5 + X4 + X1 + X0 (110011)CRC P (x) = X4 + X3 + X0 (11001)

1. First, the data or message polynomial G(x) is multiplied by Xn-k

where n-k is the number of bits in BCC (CRC code).

X4 (X5 + X4 + X1 + X0 ) = X9 + X8 + X5 + X4 (1100110000)

2. The product is divided by P (x) (modulo-2 division – remainder is derived from XOR operation)

BCCR(x)remainder1001 11001 10000

11001

100001110011000011001

P(x)Q(x)


3. The remainder R (x) is appended to the data G (x) to give1100111001 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 inthe received data.

errornoisthereif0beshouldremainder 0 11001 11001

11001

100001110011100111001

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 +++

X2 X15 X16

Data Input

LSBMSB 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.)


MSB01000000000001001160100000000000011015

0010000000000001114

0111000000000001113

0101100000000001112

0100110000000001111

0100011000000001110

010000110000000119

010000011000000118

010000001100000117

010000000110000116

010000000011000115

010000000001100114

010000000000110113

010000000000011112

010000000000001011

LSB10000000000000000Start0123456789101112131415Shift 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


• ARQ (Automatic Retransmission Request) or Retransmission– 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


• ARQ (Automatic Retransmission Request) or Retransmission (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 theerror.

– 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.

Primary Methods Used for Error CorrectionHamming 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 2n codes, 2n must be equal to or greater than m + n + 1. Therefore, the number of Hamming bits is determined by the following expression:

2n > or = m + n + 1

Where: n = number of Hamming bitsm = number of bits in each data character

– A seven bit ASCII character requires four Hamming bits (24 > 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 n1 n2 1 n4 0 0 0 n8 0 0 1

− The Hamming bits are placed in bit positions 1, 2, 4, and 8 (allpowers 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

10 0 1Logic conditionsn89 10 11Bit positions

00 0 0Logic conditions

n45 6 7Bit positions

01 0 0 0 1Logic conditions

n23 6 7 10 11Bit positions

01 0 0 0 1Logic conditions

n13 5 7 9 11Bit 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 n1 and n8 would pass, but parity checks for n2 and n4 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 registerStatus word

register

UART

RS232 signal

converter

RS232 signal

converterFrom DCE

To DCE

CPU

Parallel data bus

Address Decoder

CRSTDSSWE

RDARRDE

Transmit buffer empty Transmit clock

Address bus

Error detection circuit

Transmit

Receive

.

.

Typical UART Transmitter Block Diagram



Transmit shift register

Transmit shift register empty logic circuit

Status word register

Data, parity, and stop bit logic

Controlregister

Stop Parity 7 / 8 data bits StartOutput circuit

Parallel data from CPUControl 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.

UART Transmitter Block Diagram• Status word register is an n-bit register that keeps track of the status of the

UART’s 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 Transmitter Block Diagram• 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 UARTStatus 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


Parity checkercircuit

Start bit Verification circuitReceive

Serial data

Receive data bits (RD0 to RD7)(Parallel data)

(to CPU)

Status word register

Status WordEnable (SWE)

Control register(for transmit and receive)

Receive clock pulse (RCP)

Receive data Enable (RDE)

ReceiveParity

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 actualtime 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 voltsStart Bit

StopBit

Data Bits

0 1

Clock pulses(16 times the incoming bit rate)

line must be 0 during 7 clock pulses after 1 to0 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 (fb) with a receiver clock 16 times faster than the incoming data.

The time of one bit (tb) 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 11000

1f1tb

b

hz 16,000(16)(1000)16fR bcl

dsmicrosecon 62.516,000

1R1t

clcl

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.

• It is located in the serial communications interface of DTEs.• 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 dataregister

Transmit syncregister

Transmit shift register

TransmitTiming

andcontrol

Multiplexer

ControlRegister

Receive Timing

andcontrol

Receive shift register


Comparator

Receive Syncregister

TDS

SCTSCSTCP

CSNDB1NDB2POENPBRR

RCPSCRRORRDA

RDARRSIRDE

Received data (To data bus of computer)

Transmit data / Sync (From data bus of computer)

TSS

TSO

RSS

TCP-transmit clock pulseTSS – Transmit sync strobeTDS- Transmit data strobeTSO-transmit serial outputTCS-transmit clock signalSCT-sync character transmitRSS-receive sync strobeRR-Receive restSCR-sync character receiveSCS-sync character signal RDA-receive data availableRPE-receive parity errorROR-receiver overrunRSI-receive serial inputRCP-Receive clock pulseRDAR-receive data available resetRDE-receive data enableCS-control strobeNDB1-number of data bits 1NDB2-number of data bits 2POE-parity odd or evenNPB-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 controlstatus, is used by the USRT to request the next character from the CPU.

USRT Transmit Operation

CPU USRTSync 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 USRTReceive 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.

• It is located in the serial communications interface of DTEs.• 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 carriersbetween two or more points in a communications system, using free space or the earth’s 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 PCModem 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


PC

PC

PC

PC

ModemModem

ModemModem

Mainframe

Modem

TYPICAL DIGITAL COMMUNICATIONS USING DIGITAL RADIO

Computer RadioTransmitter

Receiver Communications Medium(earth’s atmosphere)

Radio signals (Analog signals)

ComputerRadio

TransmitterReceiver

Digital SignalsDigital Signals

DIGITAL MODULATION



PC

PC

PC

PC

Modem

ModemModem

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 earth’s 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 transmittedas 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 Hartley’s law.• Note that there are exceptions in using the above formula as in the case

of frequency shift keying (FSK).

signal discreteor symbol ain drepresente bits ofnumber MlogN levels or voltage signal discrete ofnumber

mediumssion at transmi levels coding ofnumber M (hz)bandwidth channel B

sec)per (bitscapacity n informatioC sec)per (bits ratebit f :where

N B 2 Mlog B 2 C f

2

b

2b

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 discreteor symbol ain drepresente bits ofnumber MlogN levels or voltage signal discrete ofnumber

mediumssion at transmi levels coding ofnumber M (hz)bandwidth channel B

sec)per (bitscapacity n informatioC sec)per (bits ratebit f :where

Nf

Mlogf B

N B Mlog B C f

2

b

b

2

b

2b

Nyquist Bandwidth

• The baud rate can also be computed as:

mediumion transmissat the signal discreteor element signalingeach in bits ofnumber N

raten informatioratebit fb:where

Nfbbaud

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.

Amplitude Shift Keying (ASK)

Carriersignal

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.

Amplitude Shift Keying (ASK)• 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 analogω

amplitudecarrier dunmodulate2A

(volts) signal g)(modulatinn informatio digital(t)v wavekeyingshift amplitude v(t):where

t)]cos(ω2A(t)][v[1 v(t)

c

m

cm

t)cos(ωA

t)]cos(ω2A1)][[1 v(t)

c

c

volt0 t)]cos(ω2A1][[1 v(t) c

Amplitude Shift Keying (ASK)• 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.

ratebit fb1fbbaud

ratebit fb1fbbandwidthB

hz 10,0001

10,000B

baud 10,0001

10,000baud

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.

Frequency Shift Keying (FSK)

Carriersignal

0 volt

0 volt

0 volt

+5 v = 1 +5 v = 1

Modulated Signal


01 1

Frequency = 2 khzFrequency = 1 khz Frequency = 1 khz

1 0 1





Carriersignal

0 volt

0 volt

0 volt

+5 v = 1 +5 v = 1

Modulated Signal


0

1 1

Frequency = 1 khzFrequency = 2 khz Frequency = 2 khz

1

0

1




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)ωc = angular velocity of unmodulated carrier signal (rad / sec)

= radian carrier frequency=2fcfc = analog carrier center frequency = carrier unmodulated frequency (hz)vm(t) = binary digital modulating signal Δ ω = peak to peak change in radian output frequency (radian)

• Output carrier radian frequency (ωc ) shifts by an amount equal to + or – (Δ ω ) / 2, which is proportional to the amplitude and polarity of the binary input signal.

or )t]2Δωv [(ω cos Vc )( m(t)

c tv

deviationfrequency peak 2ΔωΔf

since Δf)t]v(f [2cos Vc )( m(t)c

tv

Frequency Shift Keying (FSK)• 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 + (Δ ω ) / 2, while binary 0 = -1 volt could produce - (Δ ω ) / 2.

Output carrier frequency deviates between ωc + (Δ ω ) / 2, and ωc - (Δ ω ) / 2.

Rate of change of the frequency shifts is equal to the rate of change of the binary signal.

Δf)t] (f [2 cos Vc )( c tv

Δf)t] - (f [2 cos Vc )( ctv


• 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 = Δf / fa

Where: MI = modulation index (unitless)Δf = peak frequency deviation (hz)fa = modulating frequency (hz)

Note:- Δf could be equal to ½ the frequency of Δ ω.- Under worst case condition (alternating 1s and 0s), fa = ½ bit rate (fb)


• In binary FSK modulator, Δf 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.

Δf = peak frequency deviation (hz)

2frequency spacencyMarkfreque

frequencymark frequencyRest

frequency spacefrequencyRest


For binary FSK, the modulation index (MI) can be expressed as:

• Where:

fbfsfm

2fb2

fsfm

MI

frequency spacefsfrequencymark fm

signalinput binary offrequency lfundamentafa2fb

ratebit input fb

deviationfrequency peak f2

fsfm

• For FSK, the modulation index is generally kept below1.0, to produce arelatively 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(f + fb) = minimum Nyquist bandwidth

where: f = peak frequency deviation fb = bit rate

ratebit fb1fbbaud


• 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

120Mbps

60Mhz80Mhzfb

fsfmMI

FSK Receiver

• The most common circuit used for demodulating binary FSK signals is the phased-lock-loop (PLL).

Phasecomparator

VoltageControlledOscillator

AmpAnalog 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 frequencyfb = binary bit raten = 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:

450 = 00900 = 011800 = 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=log2M or M = 2N

Phase Shift Keying (PSK)

Carriersignal

0 volt

0 volt

0 volt

Modulated Signal

(analog)



01 1

Frequency = 2 khz

1 0 1Phase 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 1800 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 PSKJ wave could be:

logic 0 = 1800

logic 1 = 00

• The baud rate is equal to the bit rate, since 1 bit is represented by one signaling element.

BPSK Modulator

Balanced Modulator

ReferenceCarrier

Oscillator

Bandpassfilter

BinaryData

In

AnalogPSK

Output

Balanced modulator is used to produced PSK signal

Balanced Modulator Used for Binary Phase Shift Keying

Digital binaryinput

ReferenceCarrierinput

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 (digitalinput controls the conduction of the diodes.)

• If input is logic 1 (positive voltage), D1 and D2 conducts, while D3 and D4 doesnot , 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 D2does not, thus BPSK output is 1800 out of phase with reference oscillator signal.

Cos ωc

- Cos ωc

00

reference+ or - 1800

Logic 1Logic 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 ωat) (sin ωct)

where: sin ωat = fundamental frequency of binary modulating signalsin ωct = unmodulated carrier

fa = fb/2


• The output phase of a BPSK balanced modulator can also be expressed as:

output = (1/2) [cos(ωc- ωa)t] - (1/2) [cos(ωc + ωa)t]

• Consequently, the minimum double-sided Nyquist bandwidth (fN) is

where: fa = fundamental frequency (Hz)fb = bit rate (bits per second)

(ωc + ωa) - (ωc- ωa) = 2 ωa

fN = 2fafN = 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(ωc- ωa)t] - (1/2) [cos(ωc + ωa)t] = (1/2) [cos(2π70 Mhz -2π5 Mhz)t] - (1/2) [cos(2π70 Mhz + 2π5 Mhz)t] = (1/2) [cos(2π65 Mhz)t] - (1/2) [cos(75 Mhz)t]

Lower side frequency (LSF) = 65 MhzUpper side frequency (USF) = 75 MhzMinimum Nyquist bandwidth = 75 Mhz – 65 Mhz = 10 MhzBaud = fb = 10 Megabaud

65 Mhz 70 Mhz 75 Mhz

BPSK Receiver

Balanced Modulator

CoherentCarrier

Recovery

Low passfilter

BPSKinput

Binary Data Output

Sinωct(recovered carrier)

+ or - sinωct

• 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 thecomplex demodulated signal.

BPSK Receiver (cont.)

• For a balanced modulator with input signal + sin ωct (logic 1), the output is:

output = (sin ωct) (sin ωct)= (1/2) [cos(ωc- ωc)t] - (1/2) [cos(ωc + ωc)t] = (1/2) [1 - cos2ωct] = 1/2 - (1/2)cos2ωct

• For a balanced modulator with input signal - sin ωct (logic 0), the output is:

output = (-sin ωct) (sin ωct)= - (1/2) [cos(ωc- ωc)t] + (1/2) [cos(ωc + ωc)t] = -(1/2) [1 - cos2ωct] = -1/2 + (1/2)cos2ωct

Filtered outDC component(= logic 1)

Filtered outDC 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 ½ of the input bit rate.– The minimum bandwidth required is equal to ½ of the input bit

rate

QPSK Transmitter

Balanced Modulator

Linearsummer

Balanced Modulator

Band passfilter

900 phaseshift

ReferenceCarrier

OscillatorSin ωct

Inputbuffer

I

Q

Divide by 2Bit clock

Binary inputData fb

Logic 1 = +1 vLogic 0 = -1 v

Q channel fb/2

I channel fb/2

Logic 1 = +1 vLogic 0 = -1 v

+ or – sin ωct

+ or – cos ωct

QPSKoutput

sin ωct

Cos ωct

- Cos ωct

sin ωct- sin ωct00 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 900 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 ωct + cos ωct+ sin ωct - cos ωct - sin ωct + cos ωct - sin ωct - cos ωct

• The angular separation between any two adjacent phasorsin QPSK is 900

Bandwidth Considerations in QPSK• The highest fundamental frequency present at the data input of the I

or Q balanced modulator is equal to ¼ of the input data rate.• The output of the balanced modulators can be expressed as:

output = (sin ωat) (sin ωct)= [sin 2π(fb/4)t] [sin 2πfct]= (1/2) [cos 2π(fc- fb/4 )t] - (1/2) [cos 2π(fc + fb/4 )t]

• The output frequency spectrum extends from fc + fb/4 to fc- fb/4 • The minimum bandwidth (fN) is:

fN = [fc + fb/4] - [fc- fb/4] = fb/2

• The highest output rate of change (baud) is equal to fb/2.• The minimum bandwidth and baud are equal.

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 levelconverter


Product modulator

Product modulator

Referenceoscillator

Linearsummer

fb/3

fb/3

sin ωct

cos ωct


+900

8 PSKoutput

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 900 (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 ωct.– The output of the Q product modulator is –1.307 cos ωct.– The output of the linear summer is -.541 sin ωct –1.307 cos ωct.

or 1.41 sin (ωct - 112.50)• Separation between phasors is 450.• 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

010QIC

sin ωct(Reference

phase)

- sin ωct

+ cos ωct

- cos ωct

8 PSK

QIC000

QIC001

QIC010

QIC011

QIC100

QIC101

QIC110

QIC111

1.41V-112.50

1.41V-67.50

1.41V+112.50

1.41V+67.50

1.41V-157.50

1.41V-22.50

1.41V+157.50

1.41V+22.50

Q I C AmplitudePhase

(degrees)0 0 0 1.41 -112.50 0 1 1.41 -157.50 1 0 1.41 -67.50 1 1 1.41 -22.51 0 0 1.41 +112.51 0 1 1.41 +157.51 1 0 1.41 +67.51 1 1 1.41 45

8-PSK outputBinary input

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 MbaudBit rate in the I, Q, and C channels = 10 Mbps/ 3 = 3.33 MbpsMinimum 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 ¼ of incoming bit

rate.• The minimum Nyquist bandwidth is equal to ¼ of the incoming bit

rate.• The angular separation between adjacent phasors is only 22.50.• 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

Cfb/3




Product modulator

Product modulator

Referenceoscillator

Linearsummer

fb/3

fb/3

sin ωct

cos ωct


+900

8 QAMoutput

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 900 (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 Cdetermines 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 ωct.– The output of the Q product modulator is -.541 cos ωct.– The output of the linear summer is -.541 sin ωct -.541 cos ωct.

or 0.765 sin (ωct - 1350)• Separation between phasors is 900.• Resulting modulated signal could have

a peak amplitude of .765 or 1.848 volts.

000

001

101100 110

111

011010

QIC

sin ωct(Reference

phase)

- sin ωct

+ cos ωct

- cos ωct

8 QAM

QIC000

QIC001

QIC010

QIC011

QIC100

QIC101

QIC110

QIC111

0..765V-1350

0..765V-450

0..765V+1350

0..765V+450

1.848V-1350

1.848V-450

1.848V+1350

1.848V+450

Q I C AmplitudePhase

(degrees)0 0 0 0.765 -1350 0 1 1.848 -1350 1 0 0.765 -450 1 1 1.848 -451 0 0 0.765 1351 0 1 1.848 1351 1 0 0.765 451 1 1 1.848 45

8-QAM outputBinary input

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 ¼ of incoming bit

rate.• The minimum Nyquist bandwidth is equal to ¼ of the incoming bit

rate.

Comparison of Different Modulation Techniques

= fb / 4= fb / 441616 QAM

= fb / 3= fb / 3388 QAM

= fb / 2= fb / 2244 QAM= fb / 5= fb / 553232 PSK

= fb / 5= fb / 553232 QAM

= fb / 4= fb / 441616 PSK

= fb / 3= fb / 3388 PSK

= fb / 2= fb / 2244 PSK (QPSK)

= fb= fb12Binary PSK(BPSK)

12FSK

= fb= fb12ASK

MinimumBandwidth

(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 / 2Lower Side frequency = carrier frequency – Bandwidth / 2 or LSF = fc – BW / 22N = 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) rateon transmissiefficiencybandwidth (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 pulsesprior 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 transmissionsystem (radio transmission), such as earth’s 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 TRANSMISSIONIN DATA COMMUNICATIONS

DTE DTEDCE(DSU / CSU)

Communications MediumBetween DCEs

(for long distance communications)

DCE(DSU / CSU)

Communications MediumBetween DTE and DCE

(short distance only)

Communications MediumBetween 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 unitCSU – Channel service unit


Network using digital transmission PC

PC

PC

PC

DCEDCE

DCEDCE

Digital signals Digital signals

Encodeddigital signals



Network using digital transmission

Voice Signal

(analog)

DCEDCE

Digital signals

Analog to digital

converter

Voice Signal

(analog)

Digital to analog converter

Digital signals


Encoding• Encoding (digital line encoding) – involves converting standard logic

levels to a form more suitable for transmission through digitalcommunications 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 0Binary digits


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 signalsbefore 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

AnalogSignal

(such as voice)

0 volt

0 volt

0 volt

0 volt

0 volt

SamplePulses

PulseWidth

Modulation

PulsePosition

Modulation

PulseAmplitudeModulation

PulseCode

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

Bandpassfilter

SampleAnd hold

Analog toDigital

converter

Digital to Analog

converter

Holdcircuit

Low passfilter

Receiver

Transmitter

Transmission medium

PCMPAM

•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

-3110

-2010

-1100

-0000

+0001

+1101

+2011

+3111

DecimalLevelMagnitudeSign

• 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 = Vlsb / 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 voltageVmin = 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 PCM code 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: 2n-1= DR

• 1 is subtracted from 2n 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.

log21)log(DRn

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 thefollowing: 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) = 2.3DR = Vmax / Vmin = 102.3

DR = 199.5

63.7log2

1)log(199.5n

(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 bitsmust 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 = 29 = 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 / (2n – 1) = 2.55 / (28– 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 ofnumber actual

bits ofnumber minimumefficiency 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: Vlsb= voltage level corresponding to the least significant bit= step size= resolution

Vlsb / 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.

2/2V

V voltagenoiseon quantizati

voltageminimumSQRlsb

lsb

/2VV

voltagenoiseon quantizati voltagemaximumSQR

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 2n levels are:• Level at a Time Coding

– Compares PAM signal to a ramp waveform while a binary counter isbeing 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 amplitudeVin = amplitude of the input signal at a particular instant of time = parameter used to define the amount of compressionVout = compressed output amplitude

The higher is, the more compression there is.• When approaches 0, Vout / Vin = Gain approaches 1, and there is no

compression.

PCMfor Companding lawμ

Gainμ)ln(1μVin/Vmax)ln(1 Vin)(Vmax /

VinVout

μ)ln(1μVin/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:

PCMfor Companding lawμ

Gainμ)ln(1μVin/Vmax)ln(1 Vin)(Vmax /

VinVout

30.25 Vmax1.750.5 Vmax1.260.75 Vmax

1VmaxGain (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

• A–law companding is a form of analog companding used in Europe, and its compression characteristics may be expressed as:

where: Vmax = maximum uncompressed analog input amplitudeVin = amplitude of the input signal at a particular instant of time= parameter used to define the amount of compressionVout = compressed output amplitude

GainlnA1

)V / ln(AV1V

VVV

GainlnA)(1V

)V / (AVVVV

lnA1)V / ln(AV1VV

lnA1V / AVVV

maxin

in

max

in

out

in

maxinmax

in

out

maxinmaxout

maxinmaxout

1VV

A1

A1

VV0

max

in

max

in

1VV

A1

A1

VV0

max

in

max

in

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 PCM is 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 resultingto 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.

Hartley’s Law

• Hartley’s Law can be expressed as:

levels coding ofnumber M (hz)bandwidth channelB

(fb) ratebit sec)per (bitscapacity channelC :whereMlog B 2C

:as expressed becan law sHartley' noise, of absence totalIn the

ion time transmiss t bandwidth channel B

(fb) ratebit capacity n informatioC :whereion time) transmissandbandwidth

the toalproportiondirectly iscapacity on (InformatiBt α C

2

Hartley’s Law

• When the binary coding system is used, the preceding equation reduces to:

• Hartley’s 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 Hartley’s Law, the following equation can be derived:

(hz)bandwidth channelB

sec)per (bitscapacity channelC :where B 2C

levels coding ofnumber M (sec) ion time transmiss t (hz)bandwidth channel B sec)per (bitscapacity n informatioC

(bits) t in timesent n informatio TotalH :where Mlog t B 2

CtH2

Hartley’s 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 Hartley’s 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 transmittbits ofnumber Total2(4)log level codingper bits ofnumber Total

secondper bits 16,000 (4)log (4,000) 2

ratebit Mlog B 2C

2

2

2

Shannon Limit for Information Capacity (Shannon’s law)• Shannon’s 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 Shannon’s limit for information capacity, digital transmission systems that have more than two output conditions (symbols) must be used. (example: 0001 = 450, 1001 = 1800)

C = B log2(1+ S/N)or

C = 3.32B log (1+ S/N)

Shannon Limit for Information Capacity (Shannon’s law)• 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 khztelephone 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 log2(1+ S/N)C=(3400-300)log2(1+1585) = 32,953 bits per second

Shannon Limit for Information Capacity (Shannon’s law)• 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 capacityb. 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 mW at 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 log2(1+ S/N)C=(4000) log2(1+631) = 37,216 bits per second

C = B log2(1+ S/N)C=(8000) log2(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 onesource 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

TDMMultiplexer

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

TDMMultiplexer

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

FrequencyDivision

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

FrequencyDivision

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

Antialiasingfilter

BalancedModulator

Band passfilter

Antialiasingfilter

BalancedModulator

Band passfilter

.

.

.

Channel 12

Channel 1

ChannelCombining

Network

SSBSC(60khz-64khz)

Basic Group(12 voice channels)

FDMAnalog Signals60Khz-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 frequencyn= channel numberfc

fc

Formation of a Supergroup

Bandpassfilter

BalancedModulatorGroup 1

Band passfilter

Bandpassfilter

BalancedModulatorGroup 5

Band passfilter

.

.

.

Group 5

Group1

ChannelCombining

Network


Supergroup(60 voice channels)

FDMAnalog Signals

312 khz-552 khz

Group Signals(60khz-108khz)

DSBSC(504Khz-720Khz)

Group signals(60khz-108khz)

DSBSC(312 khz-528khz)


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 frequencyn= 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 basebandsignal. (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

DAVFDM

DAVIDVSB 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 (WDM)

• 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 BComputer B

Computer AComputer A Computer CComputer C

Computer DComputer D

Leased Line

Leased Line

Leased LineLeased 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

TDMMultiplexer

Computer A

Computer B

Computer C

Digital Signals

Flow of data

Digital Signals

TDMMultiplexer

Computer D

Computer E

Computer F

Digital Signals

TSI

TimeSlot

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

ControlUnit

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 HP n HT P 2 HT T

H - HeaderT - TrailerP - Payload

PAD – Packet assembler/ disassembler

Packet Switching Approaches

1. Virtual circuit approacho 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 approacho 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 FacilitiesFacilityField

Length

PacketIdentifier Address

Length

Source/ Destination

AddressData

Endof

PacketFraming

ErrorDetecting

Code

Address Field PayloadLogical Channel Field

and Format IdentifierMessageNumber Trailer

LCGN Logical Channel Group NumberGFI 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)

Guided Transmission / Communications Media • 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– Used for electrical signals– 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.– Used for electrical signals– 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 LANMostly Digital600 Mbps600 Mhz7





T1 / slow LANAnalog / Digital2 Mbps<2 Mhz2

Telephone, slow LAN

Analog / slow digital

<100 KbpsVery low1

UseDigital / AnalogData RateBandwidthCategory

Guided Transmission / Communications Media

• 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.– Used for electrical signals– 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– Used for electrical signals– 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.

Guided Transmission / Communications Media– 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 mode12577/125

Multimode, graded index

125100100/125


12562.562.5/125


1255050/125

ModeCladding Diameter (μm)

Core Diameter(μm)

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:

– Earth’s surface (including land and sea)– Earth’s 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 earth’s 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 earth’s atmosphere.

– Distance of propagation is limited by earth’s curvature.– Includes direct wave (Line of sight) and ground reflected wave– Used for very high frequency signals

• Sky wave propagation– Signal travels through the earth’s atmosphere, and, upon reaching the

ionosphere, is refracted or reflected back towards the earth’s 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 RS232port 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 toanother.

• 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 subscriber’s 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• Initially intended to carry voice converted to digital signals• Circuit is fixed• Uses regenerative repeaters• 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

AT&T North American Digital Hierarchy

336560.160T5

168274.176T4

2846.304T3

46.312T2

23.152T1C

11.544T1

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

ServicesOffered

CCITT (E1 series) Digital Hierarchy

64139.264 MbpsE4

1634.368 MbpsE3

48.448 MbpsE2

12.048 MbpsE1

No. of E1Line speed Category

E1 can be multiplexed to form E2, E2 can be multiplexed to form E3, and so on.

SONET and SDH

• SONET stands for Synchronous Optical Network• SDH stands for Synchronous Digital Hierarchy• SONET was developed for US, SDH was developed for Europe.• SONET and SDH were originally developed for optical transmission

(fiber optics)• 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 TerminalMultiplexer

Line TerminalMultiplexer

T1 / E1 T1 / E1R

Repeater

R

Repeater

Fiber Optic Cables

SONET / SDH Transmission Facilities

OC1

OC3

OC12

OC24 1.244 Gbps

51.84 Mbps

155.52 Mbps

622.08 Mbps

SONET FRAME /SIGNAL RATE

24

1

3

12

MULTIPLEOF T3

SDH and SONET Signals Comparison

SONET - Synchronous Optical NetworkOC - Optical CarrierSTM - Synchronous Transport Mode

STM-0

STM-1

STM-4

SDHEQUIVALENT

OC48 2.488 Gbps 48 STM-16

OC192 9.6 Gbps 192

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

• 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.25PAD

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

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 Frame

Ending FlagFrame Check

Sequence

Payload(User Data)

Starting Flag

Control andAddress

16, 24 or 32 bits 8 bits8 bits8 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

Controland

Address

C/R EA DLCI FECN BECN DE EADLCI

Byte 2Byte 1

12345 to 8123 to 8 Bit

Frame Relay Frame Address and Control Field Format

Address and Control Field with 2 bytes



12 Bit No.

Bit No.

Bit No.

1

1

2

2

345678

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 = 0C/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)

• 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 Network Operation

ATMSwitch

ATMSwitch

ATMSwitch

UNI

UNI

UNI

UNI

User Device A

User Device EUser Device C

User Device B

User Device D

PhysicalMedium

Virtual Path 1

Virtual Path 2

Virtual Channel 30Virtual Channel 40

Virtual Channel 50Virtual Channel 60

Virtual Paths and Virtual Channels on ATM Networks

ATM Switch

ATM SwitchUser 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 SwitchVP 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

PTICode

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)(also called narrowband ISDN)

• 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.

Data Comms Part 1

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