Computer networks

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COMPUTER NETWORKS CONTENTS: UNIT – I

01-32

Networking introduction – Categories of network – Protocols – Topology – Type of Topology – Signals – Type of signal – Transmission mode - Transmission Media – Type of Transmission media – Guided and Unguided media. UNIT – II

33-66

Encoding – Multiplexing – Transmission DTE –DCE interface – Modems – Types of modems – Error detection and correction – Line discipline – Flow control – Error control – Local Area networks – Ethernet. UNIT – III

67-82

Switching – Circuit switching – Packet switching – Message switching – Connection oriented and connectionless service – Integrated Service Digital Network – Applications of ISDN. UNIT – IV

83-104

Networking and Internetworking devices – Repeaters – Bridges – Routers – Gateways – Other network devices – Internet – History – Meaning – Advantages – Intranet – WWW. UNIT – V

105-128

Internet protocols – TCP/IP protocols – Domain Name System – File Transfer Protocol – TELNET –Simple Mail Transfer Protocols – HTTP – HTML usage – Web Browser – Common Gateway Interface.

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UNIT – I INTRODUCTION The term telecommunication means communication at a distance. The word data refers to information presented in whatever form is agreed upon by the parties creating and using the data. Data communications are the exchange of data between two devices via some form of transmission medium such as a wire cable.

The effectiveness of a data communication system depends on three fundamental characteristics: 

Delivery

Accuracy

Timeliness

Components A data communication system is made up of five components (Figure 1.1) 1. Message 2. Sender 3. Receiver 4. Medium 5. Protocol

Figure 1.1 Five components of data communication

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A network is a set of devices (often referred to as nodes) connected by communication links. A node can be a computer, printer, or any other device capable of sending and/or receiving data generated by other nodes on the network.

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Network Criteria: To be considered effective and efficient, a network must meet a number of criteria. The most important of these are performance, reliability and security.

The performance of a network depends on a number of factors, including the number of users, the type of transmission medium, the capabilities of the connected hardware, and the efficiency of the software.

Network reliability is measured by frequency of failure, the time it takes a link to recover from a failure, and the network’s robustness in a catastrophe.

Network security issues include protecting data from unauthorized access and viruses.

Applications:

Some of the network applications in different fields are the following: 

Marketing and sales

Financial services

Manufacturing

Electronic messaging

Directory services

Information services

Electronic data interchange(EDI)

Teleconferencing

Cellular telephone

Cable television

Line Configuration Line configuration defines the attachment of communication devices to a link. A Point-to-point line configuration provides a dedicated link between two devices. The entire capacity of the channel is reserved for transmission between those two devices.(figure 1.2a). A multipoint(also called multidrop)line configuration is one in which more than two specific devices share a single link. (figure 1.2b).

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Figure 1.2 Types of connections: point-to-point and multipoint

CATEGORIES OF NETWORKS

There are three primary categories: Local area networks (LAN), metropolitan area networks (MAN), and wide area networks(WAN).

Local area networks (LAN):

Local area networks, which are normally referred to simply as LANs are used to interconnect distributed communities of computer based data terminal equipments located within a single building or localized group of buildings.

Metropolitan Area Networks (MAN):

MAN is basically a bigger version of a LAN and normally uses similar technology. MAN is designed to extend over an entire city. It may be single network such as a cable television network or it may be means of connecting a number of LANs into a large network so that resources may be shared LAN-to-LAN as well as device-to-device. The high speed links between LANs within a MAN are made possible by fiber-optic connections. Wide Area Networks(WAN):

A wide area network is at the far end of the spectrum because it is for reaching system of networks that form a complex whole. One WAN is composed of two or more LANs that are

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connected across a distance of more than 30 miles. Large WANs may have many constituent LANs and MANs on different continents. The most well-known WAN is the Internet, which consists of thousands of LANs and MANs throughout the world. PROTOCOLS

In data communication, a protocol is a set of rules (conventions) that govern all aspects of information communication. A protocol defines what is communicated, how it is communicated, and when it is communicated. The key elements of a protocol are syntax, semantics, and timing.

Syntax refers to the structure or format of the data, meaning the order in which they are presented.

Semantics refers to the meaning of each section of bits.

Timing refers to two characteristics: when data should be sent and how fast they can sent.

TOPOLOGY Topology is the layout of the connections formed between computers. To some extent, the reliability and efficiency of a network is determined by its structure. Five basic topologies are possible : bus, star, ring, tree, and mesh.

Bus Topology

A network that uses a bus topology usually consists of a single, long cable to which computers are attached. Any computer attached to a bus can send a signal down the cable, and all computers receive the signal. Figure 1.3 illustrates the bus topology.

Figure 1.3 A bus topology connecting three stations

Star Topology

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The star topology is the oldest communications design method, with roots in telephone switching systems. However, the advances in network technology have made the star technology a good option for modern networks. A network uses a star topology if all computers attach to a central point. Figure 1.4 illustrates the star topology.

Figure 1.4 A star topology connecting four stations

Ring Topology The ring topology is a continuous path for data with no logical beginning or ending points and thus no terminators. Workstations and file servers are attached to the cable at points around the rings. Figure 1.5 illustrates the ring topology.

Figure 1.5 A ring topology connecting six stations

Tree Topology A tree topology is a variation of star. As in star, nodes in a tree are linked to a central hub that controls the traffic to the network. However, not every device plugs directly into the central hub. The majority of devices connect to a secondary hub that in turn, is connected to the central hub as shown in figure 1-6

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Figure 1.6 : Tree topology

Mesh Topology The mesh topology has a direct connections between every pair of devices in the network. This is an extreme design. Communication becomes very simple because there is no competition for common lines. If two devices want to communicate, they do so directly without involving other devices. Figure 1.7 illustrates the mesh topology.

Figure 1.7 A fully connected mesh topology (five devices) Combined Topologies Many computer networks used combinations of the various topologies. Figure 1.8 shows a possible combination.

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Figure 1.8 A hybrid topology: a star backbone with three bus networks

SIGNALS Information can be in the form of data, voice, picture and so on. To be transmitted, information must be transformed into electromagnetic signals.

Data can be analog or digital. The term analog data refers to information that is continuous; digital data refers to information that has discrete states. Analog data take on continuous values. Digital data take on discrete values.

Signals can be analog or digital. Analog signals can have an infinite number of values in a range; digital signals can have only a limited number of values. Figure 1.9 illustrates an analog and a digital signal.

Figure 1.9 Comparison of analog and digital signals

Periodic And Aperiodic Signals A periodic signal consists of a continuously repeated pattern. The period of a signal(T) is expressed in seconds.

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An aperiodic, or nonperiodic, signal has no repetitive pattern.

An aperiodic signal can be decomposed into an infinite number of periodic signals. A sinewave is the simplest periodic signal

In data communications, we commonly use periodic analog signals and nonperiodic digital signals. Periodic Analog Signals Periodic analog signals can be classified as simple or composite. A simple periodic analog signal, a sine wave, cannot be decomposed into simpler signals. A composite periodic analog signal is composed of multiple sine waves. Figure 1.10 shows a sine wave.

Figure 1.10 A sine wave We discuss a mathematical approach to sine waves: Example 1.1 The power in your house can be represented by a sine wave with a peak amplitude of 155 to 170 V. However, it is common knowledge that the voltage of the power in U.S. homes is 110 to 120 V. This discrepancy is due to the fact that these are root mean square (rms) values. The signal is squared and then the average amplitude is calculated. The peak value is equal to 2½ Ă— rms value.

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Figure

9

1.11 Two signals but different amplitudes

with

the

same

phase

and

frequency,

Example 1.2 The voltage of a battery is a constant; this constant value can be considered a sine wave, as we will

see

later.

For

example,

the

peak

value

of

an

AA

battery

is

normally

1.5 V.

Frequency and period are the inverse of each other.

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Figure

1.12 Two signals with but different frequencies

10

the

same

amplitude

and

phase,

Table 1.1 Units of period and frequency Example 1.3 The power we use at home has a frequency of 60 Hz. The period of this sine wave can be determined as follows:

Example 1.4

Express a period of 100 ms in microseconds.

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Solution From Table 1.1 we find the equivalents of 1 ms (1 ms is 10−3 s) and 1 s (1 s is 106 μs). We make the following substitutions:.

Example 1.5

The period of a signal is 100 ms. What is its frequency in kilohertz?

Solution

First we change 100 ms to seconds, and then we calculate the frequency from the period (1 Hz = 10−3 kHz).

Frequency is the rate of change with respect to time. Change in a short span of time means high frequency.Change over a long span of time means low frequency.

If a signal does not change at all, its frequency is zero.If a signal changes instantaneously, its frequency is infinite.

Phase describes the position of the waveform relative to time 0.

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Figure

1.13

Three sine waves but different phases

12

with

the

same

amplitude

and

frequency,

Example 1.6

A sine wave is offset 1/6 cycle with respect to time 0. What is its phase in degrees and radians?

Solution

We know that 1 complete cycle is 360째. Therefore, 1/6 cycle is

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Figure 1.14 Wavelength and period

Figure 1.15 The time-domain and frequency-domain plots of a sine wave

A complete sine wave in the time domain can be represented by one single spike in the frequency domain.

Example 1.7 The frequency domain is more compact and useful when we are dealing with more than one sine wave. For example, Figure 1.16 shows three sine waves, each with different amplitude and frequency. All can be represented by three spikes in the frequency domain.

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Figure 1.16 The time domain and frequency domain of three sine waves

A single-frequency sine wave is not useful in data communications; we need to send a composite signal, a signal made of many simple sine waves.

According to Fourier analysis, any composite signal is a combination of simple sine waves with different frequencies, amplitudes, and phases.

If the composite signal is periodic, the decomposition gives a series of signals with discrete frequencies; if the composite signal is nonperiodic, the decomposition gives a combination of sine waves with continuous frequencies.

Example 1.8

Figure 1.17 shows a periodic composite signal with frequency f. This type of signal is not typical of those found in data communications. We can consider it to be three alarm systems, each with a different frequency. The analysis of this signal can give us a good understanding of how to decompose signals.

Figure 1.17 A composite periodic signal

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Figure

1.18

Decomposition of frequency domains

15

a

composite

periodic

signal

in

the

time

and

Example 1.9

Figure 1.19 shows a nonperiodic composite signal. It can be the signal created by a microphone or a telephone set when a word or two is pronounced. In this case, the composite signal cannot be periodic, because that implies that we are repeating the same word or words with exactly the same tone.

Figure 1.19 The time and frequency domains of a nonperiodic signal The bandwidth of a composite signal is the difference between the highest and the lowest frequencies contained in that signal.

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Figure 1.20 The bandwidth of periodic and nonperiodic composite signals Example 1.10

If a periodic signal is decomposed into five sine waves with frequencies of 100, 300, 500, 700, and 900 Hz, what is its bandwidth? Draw the spectrum, assuming all components have a maximum amplitude of 10 V.

Solution

Let fh be the highest frequency, fl the lowest frequency, and B the bandwidth. Then

The spectrum has only five spikes, at 100, 300, 500, 700, and 900 Hz (see Figure 1.21).

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Figure 1.21 The bandwidth for Example 1.10

Example 1.11 A periodic signal has a bandwidth of 20 Hz. The highest frequency is 60 Hz. What is the lowest frequency? Draw the spectrum if the signal contains all frequencies of the same amplitude.

Solution

Let fh be the highest frequency, fl the lowest frequency, and B the bandwidth. Then

The spectrum contains all integer frequencies. We show this by a series of spikes (see Figure 1.22).

Figure 1.22 The bandwidth for Example 1.11

Example 1.12

A nonperiodic composite signal has a bandwidth of 200 kHz, with a middle frequency of 140 kHz and peak amplitude of 20 V. The two extreme frequencies have an amplitude of 0. Draw the frequency domain of the signal.

Solution

The lowest frequency must be at 40 kHz and the highest at 240 kHz. Figure 1.23 shows the frequency domain and the bandwidth.

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Figure 1.23 The bandwidth for Example 1.12

Digital Signals

In addition to being represented by an analog signal, information can also be represented by a digital signal. For example, a 1 can be encoded as a positive voltage and a 0 as zero voltage. A digital signal can have more than two levels. In this case, we can send more than 1 bit for each level.

Figure

1.24

Two digital signals: with four signal levels

one

with

two

signal

levels

and

the

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other


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Example 1.13

A digital signal has eight levels. How many bits are needed per level?

We calculate the number of bits from the formula

Each signal level is represented by 3 bits.

Example 1.14

Assume we need to download text documents at the rate of 100 pages per minute. What is the required bit rate of the channel?

Solution

A page is an average of 24 lines with 80 characters in each line. If we assume that one character requires 8 bits, the bit rate is

Example 1.15

A digitized voice channel, as we will see in Chapter 4, is made by digitizing a 4-kHz bandwidth analog voice signal. We need to sample the signal at twice the highest frequency (two samples per hertz). We assume that each sample requires 8 bits. What is the required bit rate? Solution

The bit rate can be calculated as

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Example 1.16

What is the bit rate for high-definition TV (HDTV)?

Solution

HDTV uses digital signals to broadcast high quality video signals. The HDTV screen is normally a ratio of 16 : 9. There are 1920 by 1080 pixels per screen, and the screen is renewed 30 times per second. Twenty-four bits represents one color pixel.

The TV stations reduce this rate to 20 to 40 Mbps through compression

Figure

1.25

The time and digital signals

frequency

domains

of

periodic

and

nonperiodic

TRANSMISSION MODE The term transmission mode is used to define the direction of signal flow between two linked devices. There are three types of transmission modes: simplex, half-duplex, and full-duplex

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Simplex:

In simplex mode, the communication is unidirectional. Only one of the two stations on a link can transmit; the other can only receive.

Half-duplex:

In half-duplex mode, each station can both transmit and receive, but not at the same time. When one device is sending, the other can only receive, and vice versa.

Full-Duplex:

In full-duplex mode (also called duplex), both stations can transmit and receive simultaneously.

Figure 1.26 Data flow (simplex, half-duplex, and full-duplex)

TRANSMISSION MEDIA

Computers and other telecommunication devices use signals to represent data. These signals are transmitted from one device to another in the form of electromagnetic energy. Electromagnetic signals can travel through a vacuum, air, or other transmission media. The following figure shows the transmission medium and physical layer.

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Figure 1.27 Transmission medium and physical layer

Transmission media can be divided into two broad categories: guided and unguided ( see figure 1.28)

Figure 1.28 Classes of transmission media

GUIDED MEDIA Guided media, which are those that provide a conduit from one device to another, include twisted-pair cable, coaxial cable, and fiber-optic cable.

Twisted-pair cable

A twisted pair consists of two conductors each surrounded by an insulating material. (see figure 1.29)

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Figure 1.29 Twisted-pair cable Twisted-pair cable comes in two forms: unshielded and shielded.

Unshielded Twisted-pair (UTP) cable

UTP cable is the most common type of telecommunication medium in use today. Advantages of UTP are its cost and ease of use. UTP is cheap, flexible and easy to install. Higher grades of UTP are used in many LAN technologies, including Ethernet and Token Ring.

Shielded Twisted-Pair (STP) Cable has a metal foil or braided-mesh covering that encases each pair of insulated conductors. Figure 1.30 shows UTP and STP cables.

Figure 1.30 UTP and STP cables UTP Connectors is most commonly connected to network devices via a type of snap-in plug like that used with telephone jacks. Connectors are either male (the plug) or female (the receptacle). The most frequently used of these plugs is an RJ45 connector with eight conductors, one for each wire of four twisted pairs ( see figure 1.31)

Figure 1.31 UTP connector Table 1.2 shows the categories of UTP cables.

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Table 1.2 Categories of unshielded twisted-pair cables

Coaxial Cable

Coaxial cable carries signals of higher frequency ranges than twisted-pair cable. Instead of having two wires, coax has a central core conductor of solid or standard wire enclosed in an insulating sheath, which is, in turn, encased in an outer conductor of metal foil, braid, or a combination of the two. The outer metallic wrapping serves both as a shield against noise and as the second conductor, which completes the circuit. The outer conductor is also enclosed in an insulating sheath, and the whole cable is protected by a plastic cover (see figure 1.32)

Figure 1.32 Coaxial cable

Each cable defined by RG ratings is adapted for a specialized function. The following are a few of the common ones:

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The most popular Coaxial cable connector is bayonet network connector (BNC), which pushes on and locks into place with a half turn. (see figure 1.33). Two other commonly used types of connectors are T-connectors and terminators.

Figure 1.33 BNC connectors Optical Fiber

Optical fiber, is made of glass or plastic and transmits signals in the form of light.

Light, a form of electromagnetic energy, travels at 300,000 kilometers/second , or approximately 186,000 miles/second, in a vacuum. This speed decreases as the medium through which the light travels becomes denser.

If a ray of light traveling through one substance suddenly enters another substance, its speed changes abruptly, causing the ray to change direction. This change is called refraction. When the angle of incidence becomes greater than the critical angle, a new phenomenon occurs called reflection.(see figure 1.33)

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Figure 1.33 Bending of light ray

Current technology supports two modes for propagating light along optical channels, each requiring fiber with different physical characteristics: multimode and single mode. (see figure 1.34)

Figure 1.34 Propagation modes

Multimode is so named because multiple beams from a light source move through the core in different paths. Single mode uses step-index fiber and a highly focused source of light that limits beams to a small range of angles, all close to the horizontal. (see figure 1.35)

Figure 1.35 Modes

Fiber sizes

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Optical fibers are defined by the ratio of the diameter of their core to the diameter of their cladding, both expressed in microns(micrometers). The common sizes are shown in table 1.3

Table 1.3 Fiber types Cable composition: Figure 1.36 shows the composition of a typical fiber-optic cable. A core is surrounded by cladding, forming the fiber. In most cases, the fiber is covered by a buffer layer that protects it from moisture. Finally, the entire cable is encased in an outer jacket.

Figure 1.36 Fiber construction

Fiber Optic Connectors All of the popular connectors are barrel shaped and come in male and female versions. The cable is equipped with a male connector that locks or threads into a female connector attached to the device to be connected.

Figure 1.37 Fiber-optic cable connectors

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Advantages of Optical Fiber 

Noise resistance

Less signal attenuation

Higher bandwidth

Disadvantages of Optical Fiber 

Cost

Installation/maintenance

Fragility

UNGUIDED MEDIA: WIRELESS

Unguided media transport electromagnetic waves without using a physical conductor. This type of communication is often referred to as wireless communication

The electromagnetic spectrum for wireless communication is shown below:

The section of the electromagnetic spectrum defined as radio communication is divided into eight ranges, called bands, each regulated by government authorities. These bands are rated from very low frequency (VLF) to extremely high frequency (EHF). Table 1.4 shows all eight bands.

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Table 1.4 Bands

Wireless transmission waves

Figure 1.38 Wireless transmission waves

Radio waves are used for multicast communications, such as radio and television, and paging systems.

Microwaves are used for unicast communication such as cellular telephones, satellite networks, and wireless LANs.

Infrared signals can be used for short-range communication in a closed area using line-of-sight propagation.

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SUMMARY o

Data communication is the transfer of data from one device to another via some form of transmission medium.

o

A data communications system must transmit data to the correct destination in an accurate and timely manner.

o

The five components that make up a data communications system are the message, sender, receiver, medium, and protocol.

o

Text, numbers, images, audio, and video are different forms of information.

o

Data flow between two devices can occur in one of three ways: simplex, halfduplex, or full-duplex.

o

A network is a set of communication devices connected by media links.

o

In a point-to-point connection, two and only two devices are connected by a dedicated link. In a multipoint connection, three or more devices share a link.

o

Topology refers to the physical or logical arrangement of a network. Devices may be arranged in a mesh, star, bus, or ring topology.

o

A network can be categorized as a local area network (LAN), a metropolitan-area network (MAN), or a wide area network (WAN).

o

A LAN is a data communication system within a building, plant, or campus, or between nearby buildings.

o

A MAN is a data communication system covering an area the size of a town or city.

o

A WAN is a data communication system spanning states, countries, or the whole world.

o

An internet is a network of networks.

o

The Internet is a collection of many separate networks.

o

TCP/IP is the protocol suite for the Internet.

o

There are local, regional, national, and international Internet service providers (ISPs).

o

A protocol is a set of rules that governs data communication; the key elements of a protocol are syntax, semantics, and timing.

o

Data must be transformed into electromagnetic signals prior to transmission across a network.

o

Data and signals can be either analog or digital.

o

A signal is periodic if it consists of a continuously repeating pattern.

o

Each sine wave can be characterized by its amplitude, frequency, and phase.

o

Frequency and period are inverses of each other.

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A time-domain graph plots amplitude as a function of time.

o

A frequency-domain graph plots each sine wave’s peak amplitude against its

31

frequency. o

By using Fourier analysis, any composite signal can be represented as a combination of simple sine waves.

o

The spectrum of a signal consists of the sine waves that make up the signal.

o

The bandwidth of a signal is the range of frequencies the signal occupies. Bandwidth is determined by finding the difference between the highest and lowest frequency components.

o

Bit rate (number of bits per second) and bit interval (duration of 1 bit) are terms used to describe digital signals.

o

A digital signal is a composite signal with an infinite bandwidth.

o

Bit rate and bandwidth are proportional to each other.

o

The Nyquist formula determines the theoretical data rate for a noiseless channel.

o

The Shannon capacity determines the theoretical maximum data rate for a noisy channel.

o

Attenuation, distortion, and noise can impair a signal.

o

Attenuation is the loss of a signal’s energy due to the resistance of the medium.

o

The decibel measures the relative strength of two signals or a signal at two different points.

o

Distortion is the alteration of a signal due to the differing propagation speeds of each of the frequencies that make up a signal.

o

Noise is the external energy that corrupts a signal.

o

We can evaluate transmission media by throughput, propagation speed, and propagation time.

o

The wavelength of a frequency is defined as the propagation speed divided by the frequency.

o

Transmission media lie below the physical layer.

o

A guided medium provides a physical conduit from one device to another.

o

Twisted-pair cable, coaxial cable, and optical fiber are the most popular types of guided media.

o

Twisted-pair cable consists of two insulated copper wires twisted together. Twisting allows each wire to have approximately the same noise environment.

o

Twisted-pair cable is used in telephone lines for voice and data communications.

o

Coaxial cable has the following layers (starting from the center): a metallic rodshaped inner conductor, an insulator covering the rod, a metallic outer conductor (shield), an insulator covering the shield, and a plastic cover.

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Coaxial cable can carry signals of higher frequency ranges than twisted-pair cable.

o

Coaxial cable is used in cable TV networks and traditional Ethernet LANs.

o

Fiber-optic cables are composed of a glass or plastic inner core surrounded by cladding, all encased in an outside jacket.

o

Fiber-optic cables carry data signals in the form of light. The signal is propagated along the inner core by reflection.

o

Fiber-optic transmission is becoming increasingly popular due to its noise resistance, low attenuation, and high-bandwith capabilities.

o

Signal propagation in optical fibers can be multimode (multiple beams from a light source) or single-mode (essentially one beam from a light source).

o

In multimode step-index propagation, the core density is constant and the light beam changes direction suddenly at the interface between the core and the cladding.

o

In multimode graded-index propagation, the core density decreases with distance from the center. This causes a curving of the light beams.

o

Fiber-optic cable is used in backbone networks, cable TV networks, and Fast Ethernet networks.

o

Unguided media (usually air) transport electromagnetic waves without the use of a physical conductor.

o

Wireless data is transmitted through ground propagation, sky propagation, and line-of-sight propagation.

o

Wireless data can be classifed as radio waves, microwaves, or infrared waves.

o

Radio waves are omnidirectional. The radio wave band is under government regulation.

o

Microwaves are unidirectional; propagation is line of sight. Microwaves are used for cellular phone, satellite, and wireless LAN communications.

o

The parabolic dish antenna and the horn antenna are used for transmission and reception of microwaves.

o

Infrared waves are used for short-range communications such as those between a PC and a peripheral device.

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UNIT – II

ENCODING We must transform data into signals to send them from one place to another. Data stored in a computer are in the form of 0s and 1s. To be carried from one place to another, data are usually converted to digital signals. This is called digital to digital conversion or encoding digital data into a digital signal.

Some times, we need to convert an analog signal into a digital signal for several reasons, such as to decrease the effect of noise. This is called analog to digital conversion or digitizing an analog signal.

To send data from one place to another using the public telephone line, the digital signal produced by the computer should be converted to an analog signal. This is called digital to analog conversion or modulating a digital signal.

Often analog signal is sent over long distances using analog media. For example, voice or music from a radio station, which is naturally an analog signal, is transmitted through the air. However, the frequency of the voice or music is not appropriate for this kind of transmission; the signal should be carried by a higher frequency signal. This is called analog to analog conversion or modulating an analog signal.

DIGITAL-TO-DIGITAL CONVERSION

In this section, we see how we can represent digital data by using digital signals. The conversion involves three techniques: line coding, block coding, and scrambling. Line coding is always needed; block coding and scrambling may or may not be needed.

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Figure 2.1 Line coding and decoding

Figure 2.2 Signal element versus data element Of the many mechanisms for digital to digital encoding, we will discuss only those most useful for data communication. These fall into three broad categories: unipolar, polar, and bipolar.

Unipolar encoding is simple, with only one technique in use. Polar encoding has three subcategories, NRZ,RZ, and biphase, two of which have multiple variations of their own. The third option, bipolar encoding, has three variations : AMI, B8ZS, and HDB3.

Unipolar encoding uses only one level of value. Polar encoding uses two levels (positive and negative) of amplitude. In bipolar encoding, we use three levels: positive, zero and negative.

ANALOG-TO-DIGITAL CONVERSION

In analog to digital conversion, we are representing the information contained in a continuous wave form as a series of digital pulses (1s or 0s). Figure 2.3 shows the analog to digital converter.

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Figure 2.3: analog to digital converter

The first step in analog to digital conversion is called pulse amplitude modulation (PAM). This technique takes an analog signal, samples it, and generates a series of pulses based on the results of sampling. The term sampling means measuring the amplitude of the signal at equal intervals.

The method of sampling used in PAM is more useful to other areas of engineering than it is to data communication. However, PAM is the foundation of an important analog-to-digital conversion method called pulse code modulation (PCM). Figure 2.4 shows the three different sampling methods for PCM.

Figure 2.4 Three different sampling methods for PCM

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DIGITAL-TO-ANALOG CONVERSION

Digital-to-analog conversion is the process of changing one of the characteristics of an analog signal based on the information in digital data.

The digital data must be modulated on an analog signal that has been manipulated to look like two distinct values corresponding to binary 1 and binary 0. Figure 2.5 shows the digital to analog conversion.

Figure 2.5 Digital-to-analog conversion

Three mechanisms for modulating digital data into an analog signal:

Figure 2.6 Types of digital-to-analog conversion

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Aspects of Digital to Analog Conversion:

Bit rate is the number of bits per second. Baud rate is the number of signal elements per second.

In

the

analog

transmission

of

digital

data,

the

baud

rate

is

less

than

or equal to the bit rate.

ANALOG- TO- ANALOG CONVERSION

Analog-to-analog conversion is the representation of analog information by an analog signal. One may ask why we need to modulate an analog signal; it is already analog. Modulation is needed if the medium is bandpass in nature or if only a bandpass channel is available to us.

Analog to Analog modulation can be accomplished in three ways:

Figure 2.7 Types of analog-to-analog modulation

The total bandwidth required for AM can be determined from the bandwidth of the audio signal: BAM = 2B.

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Figure 2.8 Amplitude modulation

Figure 2.9 AM band allocation The

total

bandwidth

required

for

FM

can

be

determined

from

the

bandwidth

of the audio signal: BFM = 2(1 + β)B.

Figure 2.10 Frequency modulation

Figure 2.11 FM band allocation

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The total bandwidth required for PM can be determined from the bandwidth and maximum amplitude of the modulating signal: BPM = 2(1 + β)B.

Figure 2.12 Phase modulation

MULTIPLEXING Whenever the bandwidth of a medium linking two devices is greater than the bandwidth needs of the devices, the link can be shared. Multiplexing is the set of techniques that allows the simultaneous transmission of multiple signals across a single data link. As data and telecommunications use increases, so does traffic.

In a multiplexed system, n devices share the capacity of one link. Figure 2.13 shows the basic format of a multiplexed system.

Figure 2.13 Dividing a link into channels Signals are multiplexed using three basic techniques :

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Figure 2.14 Categories of multiplexing

Frequency-division multiplexing (FDM) is an analog technique that can be applied when the bandwidth of a link is greater than the combined bandwidths of the signals to be transmitted. Figure 2.15 gives a conceptual view of FDM.

Figure 2.15 Frequency-division multiplexing

FDM is an analog multiplexing technique that combines analog signals. Figure 2.16 is a conceptual time-domain illustration of the multiplexing process.

Figure 2.16 FDM process

Demultiplexing: The demultiplexer uses a series of filters to decompose the multiplexed signal into its constituent component signals. The individual signals are then passed to a demodulator that separates them

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from their carriers and passes them to the waiting receivers. Figure 2.17 is a time-domain illustration of FDM demultiplexing, again using three telephones as the communication devices.

Figure 2.17 FDM demultiplexing example Example 2.1 Assume that a voice channel occupies a bandwidth of 4 kHz. We need to combine three voice channels into a link with a bandwidth of 12 kHz, from 20 to 32 kHz. Show the configuration, using the frequency domain. Assume there are no guard bands. Solution We shift (modulate) each of the three voice channels to a different bandwidth, as shown in Figure 6.6. We use the 20- to 24-kHz bandwidth for the first channel, the 24- to 28-kHz bandwidth for the second channel, and the 28- to 32-kHz bandwidth for the third one. Then we combine them as shown in Figure 2.18.

Figure 2.18 Example 2.1

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Example 2.2 Five channels, each with a 100-kHz bandwidth, are to be multiplexed together. What is the minimum bandwidth of the link if there is a need for a guard band of 10 kHz between the channels to prevent interference? Solution For five channels, we need at least four guard bands. This means that the required bandwidth is at least 5 Ă— 100 + 4 Ă— 10 = 540 kHz, as shown in Figure 2.19.

Figure 2.19 Example 2.2 Example 1.3 Four data channels (digital), each transmitting at 1 Mbps, use a satellite channel of 1 MHz. Design an appropriate configuration, using FDM.

Solution The satellite channel is analog. We divide it into four channels, each channel having a 250-kHz bandwidth. Each digital channel of 1 Mbps is modulated such that each 4 bits is modulated to 1 Hz. One solution is 16-QAM modulation. Figure 2.20 shows one possible configuration.

Figure 2.20 Example 1.3

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WAVE-DIVISION MULTIPLEXING Wave-division multiplexing (WDM) is conceptually the same as FDM, except that the multiplexing and demultiplexing involve light signals transmitted through fiber optic channels. Figure 2.21 gives a conceptual view of a WDM multiplexer and demultiplexer.

Figure 2.21 Wavelength-division multiplexing

WDM is an analog multiplexing technique to combine optical signals. Figure 2.22 shows the prisms in WDM multiplexing and demultiplexing.

Figure 2.22 Prisms in wavelength-division multiplexing and demultiplexing

TIME-DIVISION MULTIPLEXING (TDM)

TDM

is

a

digital

multiplexing

technique

for

combining

several

low-rate

channels into one high-rate one.

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Figure 2.23 TDM

TDM can be implemented in two ways: synchronous TDM and asynchronous TDM.

In synchronous TDM, the data rate of the link is n times faster, and the unit duration is n times shorter.

Figure 2.24 Synchronous time-division multiplexing Like synchronous TDM, asynchronous TDM allows a number of lower-speed input lines to be multiplexed to a single higher speed line. Unlike synchronous TDM, however, in asynchronous TDM the total speed of the input lines can be greater than the capacity of the path.

DIGITAL DATA TRANSMISSION

The transmission of binary data across a link can be accomplished in either parallel or serial mode. In parallel mode, multiple bits are sent with each clock tick. In serial mode, 1 bit is sent with each clock tick. While there is only one way to send parallel data, there are three subclasses of serial transmission: asynchronous, synchronous, and isochronous. The following diagram shows data transmission and modes.

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Figure 2.25 Data transmission and modes Parallel Transmission Binary data, consisting of 1s and 0s, may be organized into groups of n bits each. Computers produce and consume data in groups of bits. By grouping, we can send data n bits at a time instead of one. This is called parallel transmission. Figure 2.26 shows how parallel transmission works for n=8.Typically, the eight wires are bundled in a cable with a connector at each end.

Figure 2.26 Parallel transmission Serial Transmission In serial transmission one bit follows another, so we need only one communication channel rather than n to transmit data between two communicating devices (see figure 2.27)

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Figure 2.27 Serial transmission In asynchronous transmission, we send 1 start bit (0) at the beginning and 1 or more stop bits (1s)

at

the

end

of

each

byte.

There

may

be

a

gap

between

each byte. Asynchronous here means “asynchronous at the byte level,� but the bits are still synchronized; their durations are the same.

Figure 2.28 Asynchronous transmission

In synchronous transmission, we send bits one after another without start or stop bits or gaps. It is the responsibility of the receiver to group the bits. Figure 2.29 gives a schematic illustration of synchronous transmission.

Figure 2.29 Synchronous transmission

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DTE – DCE INTERFACE Data Terminal Equipment(DTE) Data terminal equipment (DTE) is an end instrument that converts user information into signals for transmission or reconverts received signals into user information. A DTE device communicates with the data circuit-terminating equipment (DCE). The DTE/DCE classification was introduced by IBM.

A DTE is the functional unit of a data station that serves as a data source or a data sink and provides for the data communication control function to be performed in accordance with link protocol.

The data terminal equipment may be a single piece of equipment or an interconnected subsystem of multiple pieces of equipment that perform all the required functions necessary to permit users to communicate. A user interacts with the DTE (e.g. through a human-machine interface), or the DTE may be the user. Usually, the DTE device is the terminal (or a computer emulating a terminal), and the DCE is a modem. DTE is usually a male connector and DCE is a female connector.

A general rule is that DCE devices provide the clock signal (internal clocking) and the DTE device synchronizes on the provided clock (external clocking). D-sub connectors follow another rule for pin assignment. DTE devices usually transmit on pin connector number 2 and receive on pin connector number 3. DCE devices are just the opposite: pin connector number 2 receives and pin connector number 3 transmits the signals.

Data circuit-terminating equipment(DCE) A Data circuit-terminating equipment (DCE) is a device that sits between the data terminal equipment (DTE) and a transmission circuit. It is also called data communications equipment and data carrier equipment.

In a data station, the DCE performs functions such as signal conversion, coding, and line clocking and may be a part of the DTE or intermediate equipment. Interfacing equipment may be required to couple the data terminal equipment (DTE) into a transmission circuit or channel and from a transmission circuit or channel into the DTE. Although the terms are most commonly used with RS-232, several data communications standards define different types of interfaces between a

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DCE and a DTE. The DCE is a device that communicates with a DTE device in these standards. Standards that use this nomenclature include: Federal Standard 1037C, MIL-STD-188 ,RS-232 Certain ITU-T standards in the V series (notably V.24 and V.35) Certain ITU-T standards in the X series (notably X.21 and X.25)

A general rule is that DCE devices provide the clock signal (internal clocking) and the DTE device synchronizes on the provided clock (external clocking). D-sub connectors follow another rule for pin assignment. DTE devices usually transmit on pin connector number 2 and receive on pin connector number 3. DCE devices are just the opposite: pin connector number 2 receives and pin connector number 3 transmits the signals. Usually, the DTE device is the terminal (or computer), and the DCE is a modem

MODEMS

Modem (from modulator-demodulator) is a device that modulates an analog carrier signal to encode digital information, and also demodulates such a carrier signal to decode the transmitted information. The goal is to produce a signal that can be transmitted easily and decoded to reproduce the original digital data. Modems can be used over any means of transmitting analog signals, from driven diodes to radio. The most familiar example is a voiceband modem that turns the digital 1s and 0s of a personal computer into sounds that can be transmitted over the telephone lines of Plain Old Telephone Systems (POTS), and once received on the other side, converts those 1s and 0s back into a form used by a USB, Ethernet, serial, or network connection. Modems are generally classified by the amount of data they can send in a given time, normally measured in bits per second, or "bps". They can also be classified by Baud, the number of times the modem changes its signal state per second. Baud is NOT the modem's speed. The baud rate varies, depending on the modulation technique used. Original Bell 103 modems used a modulation technique that saw a change in state 300 times per second. They transmitted 1 bit for every baud, and so a 300 bit/s modem was also a 300-baud modem. However, casual computerists confused the two. A 300 bit/s modem is the only modem whose bit rate matches the baud rate. A 2400 bit/s modem changes state 600 times per second, but due to the fact that it transmits 4 bits for each baud, 2400 bits are transmitted by 600 baud, or changes in states.

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Faster modems are used by Internet users every day, notably cable modems and ADSL modems. In telecommunications, "radio modems" transmit repeating frames of data at very high data rates over microwave radio links. Some microwave modems transmit more than a hundred million bits per second. Optical modems transmit data over optical fibers. Most intercontinental data links now use optical modems transmitting over undersea optical fibers. Optical modems 9

routinely have data rates in excess of a billion (1x10 ) bits per second. One kilobit per second (kbit/s or kb/s or kbps) as used in this article means 1000 bits per second and not 1024 bits per second. For example, a 56k modem can transfer data at up to 56,000 bits per second over the phone line. Types of modems Cable modems: A cable modem is a type of modem that provides access to a data signal sent over the cable television infrastructure. Cable modems are primarily used to deliver broadband Internet access in the form of cable internet, taking advantage of the high bandwidth of a cable television network. They are commonly found in Australia, New Zealand, Canada, Europe, Costa Rica, and the United States. In the USA alone there were 22.5 million cable modem users during the first quarter of 2005, up from 17.4 million in the first quarter of 2004.

DSL modem ADSL modem or DSL modem is a device used to connect a single computer or router to a DSL phone line, in order to use an ADSL service. Like other modems it is a type of transceiver. It is also called a DSL Transceiver or ATU-R. The acronym NTBBA (network termination broad band adapter, network termination broad band access) is also common in some countries.

Some ADSL modems also manage the connection and sharing of the ADSL service with a group of machines: in this case, the unit is termed a DSL router or residential gateway. DSL routers have a functional block which performs framing, while other functional blocks perform Asynchronous Transfer Mode Segmentation and Reassembly, IEEE 802.1D bridging and/or IP routing. Typical user interfaces are Ethernet and USB. Although an ADSL modem working as a bridge does not need an IP address, it may have one assigned for management purposes.

Radio modems Direct broadcast satellite, WiFi, and mobile phones all use modems to communicate, as do most other wireless services today. Modern telecommunications and data networks also make extensive use of radio modems where long distance data links are required. Such systems are an

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important part of the PSTN, and are also in common use for high-speed computer network links to outlying areas where fibre is not economical.

Even where a cable is installed, it is often possible to get better performance or make other parts of the system simpler by using radio frequencies and modulation techniques through a cable. Coaxial cable has a very large bandwidth, however signal attenuation becomes a major problem at high data rates if a digital signal is used. By using a modem, a much larger amount of digital data can be transmitted through a single piece of wire. Digital cable television and cable Internet services use radio frequency modems to provide the increasing bandwidth needs of modern households. Using a modem also allows for frequency-division multiple access to be used, making full-duplex digital communication with many users possible using a single wire.

Wireless modems come in a variety of types, bandwidths, and speeds. Wireless modems are often referred to as transparent or smart. They transmit information that is modulated onto a carrier frequency to allow many simultaneous wireless communication links to work simultaneously on different frequencies.

Transparent modems operate in a manner similar to their phone line modem cousins. Typically, they were half duplex, meaning that they could not send and receive data at the same time. Typically transparent modems are polled in a round robin manner to collect small amounts of data from scattered locations that do not have easy access to wired infrastructure. Transparent modems are most commonly used by utility companies for data collection.

Smart modems come with a media access controller inside which prevents random data from colliding and resends data that is not correctly received. Smart modems typically require more bandwidth than transparent modems, and typically achieve higher data rates. The IEEE 802.11 standard defines a short range modulation scheme that is used on a large scale throughout the world. WiFi and WiMax Wireless data modems are used in the WiFi and WiMax standards, operating at microwave frequencies.

WiFi is principally used in laptops for Internet connections (wireless access point) and wireless application protocol (WAP).

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Mobile modems & routers Modems which use mobile phone lines (GPRS,UMTS,HSPA,EVDO,WiMax,etc.), are known as Cellular Modems. Cellular modems can be embedded inside a laptop or appliance, or they can be external to it. External cellular modems are datacards and cellular routers. The datacard is a PC card or ExpressCard which slides into a PCMCIA/PC card/ExpressCard slot on a computer. The most famous brand of Radio modem datacards is the AirCard made by Sierra Wireless. Nowadays, there are USB cellular modems as well that use a USB port on the laptop instead of a PC card or ExpressCard slot. A cellular router may or may not have an external datacard ("AirCard") that slides into it. Most cellular routers do allow such datacards or USB modems, except for the WAAV, Inc. CM3 mobile broadband cellular router. Cellular Routers may not be modems per se, but they contain modems or allow modems to be slid into them. The difference between a cellular router and a cellular modem is that a cellular router normally allows multiple people to connect to it (since it can "route"), while the modem is made for one connection.

Most of the GSM cellular modems come with an integrated SIM cardholder. The CDMA (EVDO) versions do not use SIM cards, but use ESN (Electronic Serial Numbers) instead.

The cost of using a cellular modem varies from country to country. Some carriers implement "flat rate" plans for unlimited data transfers. Some have caps (or maximum limits) on the amount of data that can be transferred per month. Other countries have "per Megabyte" or even "per Kilobyte" plans that charge a fixed rate per Megabyte or Kilobyte of data downloaded; this tends to add up quickly in today's content-filled world, which is why many people are pushing for flat data

rates.

The

faster

data

rates

of

the

newest

cellular

modem

technologies

(UMTS,HSPA,EVDO,WiMax) are also considered to be "Broadband Cellular Modems" and compete with other Broadband modems below.

Error Detection and Correction Networks must be able to transfer data from one device to another with acceptable accuracy. For most applications, a system must guarantee that the data received are identical to the data transmitted. Any time data are transmitted from one node to the next, they can become corrupted in passage. Many factors can alter one or more bits of a message. Some applications require a mechanism for detecting and correcting errors.

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Types of Errors Single-Bit Error

The term single-bit error means that only 1 bit of a given data unit (such as a byte, character, or packet) is changed from 1 to 0 or from 0 to 1.

Figure 2.30 shows the effect of a single-bit error on a data unit.

Figure 2.30 : Single bit error Burst Error The term burst error means that 2 or more bits in the data unit have changed from 1 to 0 or from 0 to 1.

Figure 2.31 shows the effect of a burst error on a data unit. In this case, 0100010001000011 was sent, but 0101110101100011 was received. Note that a burst error does not necessarily mean that the errors occur in consecutive bits. The length of the burst is measured from the first corrupted bit to the last corrupted bit. Some bits in between may not have been corrupted.

Figure 2.31 Burst error of length 8

A burst error is more likely to occur than a single-bit error. The duration of noise is normally longer than the duration of 1 bit, which means that when noise affects data, it affects a set of bits. The number of bits affected depends on the data rate and duration of noise. For example, if we

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are sending data at 1 kbps, a noise of 1/100 s can affect 10 bits; if we are sending data at 1 Mbps, the same noise can affect 10,000 bits.

Redundancy

The central concept in detecting or correcting errors is redundancy. To be able to detect or correct errors, we need to send some extra bits with our data. These redundant bits are added by the sender and removed by the receiver. Their presence allows the receiver to detect or correct corrupted bits.

Detection Versus Correction

The correction of errors is more difficult than the detection. In error detection, we are looking only to see if any error has occurred. The answer is a simple yes or no. We are not even interested in the number of errors. A single-bit error is the same for us as a burst error. In error correction, we need to know the exact number of bits that are corrupted and more importantly, their location in the message. The number of the errors and the size of the message are important factors. If we need to correct one single error in an 8-bit data unit, we need to consider eight possible error locations; if we need to correct two errors in a data unit of the same size, we need to consider 28 possibilities. You can imagine the receiver’s difficulty in finding 10 errors in a data unit of 1000 bits.

Coding

Redundancy is achieved through various coding schemes. The sender adds redundant bits through a process that creates a relationship between the redundant bits and the actual data bits. The receiver checks the relationships between the two sets of bits to detect or correct the errors. The ratio of redundant bits to the data bits and the robustness of the process are important factors in any coding scheme. Figure 2.32 shows the general idea of coding.

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Figure 2.32 The structure of encoder and decoder

DATA LINK CONTROL

The most important functions in the data link layer are line discipline, flow control and error control. 

Line discipline coordinates the link systems. It determines which device can send and when it can send.

Flow control coordinates the amount of data that can be sent before receiving acknowledgement. It also provides the receiver’s acknowledgment of frames received intact, and so is linked to error control.

Error control means error detection and correction. It allows the receiver to inform the sender of any frames lost or damaged in transmission and coordinates the retransmission of those frames by the sender.

LINE DISCIPLINE Line discipline can be done in two ways: 1. ENQ/ACK (Enquiry Acknowledge) 2. Poll/Select

ENQ/ACK 

Typically used for point-to-point links

ENQ/ACK coordinates

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Which device may start a transmission

Determine if the recipient is ready

How ENQ/ACK works 1. Initiator transmits an ENQ 2. Receiver replies with an ACK (OK) or NACK (not OK) 3. If ACK received transmit data, end with EOT (End of Text)

Time diagrams for ENQ/ACK, connection reject and successful data Transmission

N.B. data transmission will depend on flow control mechanism POLL/SELECT LINE DISCIPLINE Used for topologies where one device is the primary station and the remaining are secondary stations • All exchanges must made through a the primary device • The primary device controls the link, while the secondary devices follows its instructions • Therefore, primary station initiates all sessions Poll - Primary asks secondary: Do you need to send? Select - Primary needs to send, notifies secondary Polling should be performed fairly This is a Don’t call us will call you strategy • Since multiple stations exist, there is a need for addresses

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– Each station must be uniquely identified

FLOW CONTROL

Flow control is the process of managing the rate of data transmission between two nodes to prevent a fast sender from over running a slow receiver. This should be distinguished from congestion control, which is used for controlling the flow of data when congestion has actually occurred. Flow control mechanisms can be classified by whether or not the receiving node sends feedback to the sending node. Flow control is important because it is possible for a sending computer to transmit information at a faster rate than the destination computer can receive and process them. This can happen if the receiving computers have a heavy traffic load in comparison to the sending computer, or if the receiving computer has less processing power than the sending computer. Flow control refers to a set of procedures used to restrict the amount of data the sender can send before waiting for acknowledgement.

Two methods have been developed to control the flow of data across communications link: stopand-wait and sliding window. Stop – and –wait In the stop and wait method of flow control, the sender sends one frame and waits for an acknowledgement before sending the next frame. Sender waits for ACK after each frame transmission:

Advantage: simplicity. Disadvantage: inefficiency (wait times).

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Sliding window

Sliding Window Protocol is a bi-directional data transmission protocol in the data link layer. It is used to keep a record of the frame sequences sent and their respective acknowledgements received by both the users. In transmit flow control, sliding window is a variable-duration window that allows a sender to transmit a specified number of data units before an acknowledgment is received or before a specified event occurs.

An example of a sliding window is one in which, after the sender fails to receive an acknowledgment for the first transmitted frame, the sender "slides" the window, i.e. resets the window, and sends a second frame. This process is repeated for the specified number of times before the sender interrupts transmission. In the sliding window method of flow control, several frames can be in transit at a time.

Conceptually, the sliding window of the sender shrinks from the left when frames of data are sent. The sliding window of the sender expands to the right when acknowledgments are received.

Conceptually, the sliding window of the receiver shrinks from the left when frames of data are received. The sliding window of the receiver expands to the right when acknowledgments are sent.

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ERROR CONTROL Error control in the data link layer is based on automatic repeat request(ARQ), which means retransmission of data in three cases: damaged frame, lost frame, and lost acknowledgment.

Error Control: ARQ (Automatic Repeat Request) schemes • if error(s) detected in received Frame, return NAK to Sender • NAK can be explicit or implicit (Sender’s Timeout timer expires) • Sender keeps a copy of each un-ACKed Frame to re-transmit if required • ACK received by Sender for Frame =discard copy • NAK received by Sender for Frame =decide how to re-transmit Frame • Sender starts Timeout timer for each Frame when it is transmitted • appropriate Timeout value = the expected delay for Sender to receive ACK for the Frame (in practice, set Timeout slightly larger than this…) • packet is not considered to be delivered successfully to the Receiver’s Network layer until the Sender knows this (by getting ACK for it) • 3 types of ARQ scheme: • Stop-and-wait ARQ – extension of Stop-and-wait flow control • Sliding window ARQ – extension of sliding window flow control: • Go-back-n ARQ – Receiver must get Frames in correct order • Selective repeat ARQ – correctly-received out-of-order Frames are stored at Receiver until they can be re-assembled into correct order

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LOCAL AREA NETWORKS A Local Area Network is a data communication system that allows a number of independent devices to communicate directly with each other in a limited geographic area. PROJECT 802 IEEE 802 refers to a family of IEEE standards dealing with local area networks and metropolitan area networks. More specifically, the IEEE 802 standards are restricted to networks carrying variable-size packets. (By contrast, in cell-based networks data is transmitted in short, uniformly sized units called cells. Isochronous networks, where data is transmitted as a steady stream of octets, or groups of octets, at regular time intervals, are also out of the scope of this standard.) The number 802 was simply the next free number IEEE could assign, though “802” is sometimes associated with the date the first meeting was held — February 1980.

The services and protocols specified in IEEE 802 map to the lower two layers (Data Link and Physical) of the seven-layer OSI networking reference model. In fact, IEEE 802 splits the OSI Data Link Layer into two sub-layers named Logical Link Control (LLC) and Media Access Control, so that the layers can be listed like this:

Data link layer LLC Sublayer MAC Sublayer Physical layer

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The IEEE 802 family of standards is maintained by the IEEE 802 LAN/MAN Standards Committee (LMSC). The most widely used standards are for the Ethernet family, Token Ring, Wireless LAN, Bridging and Virtual Bridged LANs. An individual Working Group provides the focus for each area.

IEEE 802.1 Bridging (networking) and Network Management IEEE 802.2 Logical link control (inactive) IEEE 802.3 Ethernet IEEE 802.4 Token bus (disbanded) IEEE 802.5 Defines the MAC layer for a Token Ring (inactive) IEEE 802.6 Metropolitan Area Networks (disbanded) IEEE 802.7 Broadband LAN using Coaxial Cable (disbanded) IEEE 802.8 Fiber Optic TAG (disbanded) IEEE 802.9 Integrated Services LAN (disbanded) IEEE 802.10 Interoperable LAN Security (disbanded) IEEE 802.11 Wireless LAN & Mesh (Wi-Fi certification) IEEE 802.12 demand priority (disbanded) IEEE 802.13 Not Used IEEE 802.14 Cable modems (disbanded) IEEE 802.15 Wireless PAN IEEE 802.15.1 (Bluetooth certification) IEEE 802.15.4 (ZigBee certification) IEEE 802.16 Broadband Wireless Access (WiMAX certification) IEEE 802.16e (Mobile) Broadband Wireless Access IEEE 802.17 Resilient packet ring IEEE 802.18 Radio Regulatory TAG IEEE 802.19 Coexistence TAG IEEE 802.20 Mobile Broadband Wireless Access IEEE 802.21 Media Independent Handoff IEEE 802.22 Wireless Regional Area Network

ETHERNET Ethernet is a family of frame-based computer networking technologies for local area networks (LANs). The name comes from the physical concept of the ether. It defines a number of wiring and signaling standards for the Physical Layer of OSI networking, through means of network access at the Media Access Control (MAC)/Data Link Layer, and a common addressing format.

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Ethernet is standardized as IEEE 802.3. The combination of the twisted pair versions of Ethernet for connecting end systems to the network, along with the fiber optic versions for site backbones, is the most widespread wired LAN technology. It has been in use from around 1980 to the present, largely replacing competing LAN standards such as token ring, FDDI, and ARCNET. Varieties of Ethernet Some early varieties 

10BASE5 -- the original standard uses a single coaxial cable into which you literally tap a connection by drilling into the cable to connect to the core and screen. Largely obsolete, though due to its widespread deployment in the early days, some systems may still be in use.

10BROAD36 -- Obsolete. An early standard supporting Ethernet over longer distances. It utilized broadband modulation techniques, similar to those empolyed in cable modem systems, and operated over coaxial cable.

1BASE5 -- An early attempt to standardize a low-cost LAN solution, it operates at 1 Mbit/s and was a commercial failure.

10Mbit/s Ethernet 

10BASE2 (also called ThinNet or Cheapernet) -- 50-ohm coaxial cable connects machines together, each machine using a T-adaptor to connect to its NIC. Requires terminators at each end. For many years this was the dominant Ethernet standard 10 Mbit/s.

10BASE-T -- runs over 4 wires (two twisted pairs) on a cat-3 or cat-5 cable. A hub or switch sits in the middle and has a port for each node. This is also the configuration used for 100BASE-T and Gigabit Ethernet. 10 Mbit/s.

FOIRL -- Fiber-optic inter-repeater link. The original standard for Ethernet over fibre.

10BASE-F -- A generic term for the new family of 10 Mbit/s Ethernet standards: 10BASEFL, 10BASE-FB and 10BASE-FP. Of these only 10BASE-FL is in widespread use. o

10BASE-FL -- An updated version of the FOIRL standard.

o

10BASE-FB -- Intended for backbones connecting a number of hubs or switches, it is now obsolete.

o

10BASE-FP -- A passive star network that required no repeater, it was never implemented

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Fast Ethernet 

100BASE-T -- A term for any of the three standard for 100 Mbit/s Ethernet over twisted pair cable. Includes 100BASE-TX, 100BASE-T4 and 100BASE-T2. o

100BASE-TX -- Uses two pairs, but requires cat-5 cable. Similar star-shaped configuration to 10BASE-T. 100 Mbit/s.

o

100BASE-T4 -- 100 Mbit/s Ethernet over Category 3 cabling (as used for 10BASE-T installations). Uses all four pairs in the cable. Now obsolete, as Category 5 cabling is the norm. Limited to half-duplex.

o

100BASE-T2 -- No products exist. 100 Mbit/s Ethernet over Category 3 cabling. Supports full-duplex, and uses only two pairs. It is functionally equivalent to 100BASE-TX, but supports old cable.

100BASE-FX -- 100 Mbit/s Ethernet over fibre.

Gigabit Ethernet 

1000BASE-T -- 1 Gbit/s over Cat 5e copper cabling.

1000BASE-SX -- 1 Gbit/s over fiber.

1000BASE-LX -- 1 Gbit/s over fiber. Optimized for longer distances over single-mode fiber.

1000BASE-CX -- A short-haul solution (up to 25m) for running 1 Gbit/s Ethernet over special copper cable. Predates 1000BASE-T, and now obsolete.

SUMMARY: o

Digital-to-analog modulation can be accomplished using the following: *Amplitude shift keying (ASK)—the amplitude of the carrier signal varies. *Frequency shift keying (FSK)—the frequency of the carrier signal varies. *Phase shift keying (PSK)—the phase of the carrier signal varies. *Quadrature amplitude modulation (QAM)—both the phase and amplitude of the carrier signal vary.

o

QAM enables a higher data transmission rate than other digital-to-analog methods.

o

Baud rate and bit rate are not synonymous. Bit rate is the number of bits transmit-ted per second. Baud rate is the number of signal units transmitted per second. One signal unit can represent one or more bits.

o

The minimum required bandwidth for ASK and PSK is the baud rate.

o

The minimum required bandwidth (BW) for FSK modulation is BW =f c1 .f c0 + N baud , where f c1 is the frequency representing a 1 bit, f c0 is the frequency representing a 0 bit, and N baud is the baud rate.

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A regular telephone line uses frequencies between 600 and 3000 Hz for data communication.

o

ASK modulation is especially susceptible to noise.

o

Because it uses two carrier frequencies, FSK modulation requires more bandwidth than ASK and PSK.

o

PSK and QAM modulation have two advantages over ASK: *They are not as susceptible to noise. *Each signal change can represent more than one bit.

o

Trellis coding is a technique that uses redundancy to provide a lower error rate.

o

The 56K modems are asymmetric; they download at a rate of 56 Kbps and upload at 33.6 Kbps.

o

Analog-to-analog modulation can be implemented by using the following: * Amplitude modulation (AM) * Frequency modulation (FM) * Phase modulation (PM)

o

In AM radio, the bandwidth of the modulated signal must be twice the bandwith of the modulating signal.

o

In FM radio, the bandwith of the modulated signal must be 10 times the bandwidth of the modulating signal.

o

Multiplexing is the simultaneous transmission of multiple signals across a single data link.

o

Frequency-division multiplexing (FDM) and wave-division multiplexing (WDM) are techniques for analog signals, while time-division multiplexing (TDM) is for digital signals.

o

In FDM, each signal modulates a different carrier frequency. The modulated carriers are combined to form a new signal that is then sent across the link.

o

In FDM, multiplexers modulate and combine signals while demultiplexers decompose and demodulate.

o

In FDM, guard bands keep the modulated signals from overlapping and interfering with one another.

o

Telephone companies use FDM to combine voice channels into successively larger groups for more efficient transmission.

o

Wave-division multiplexing is similar in concept to FDM. The signals being multiplexed, however, are light waves.

o

In TDM, digital signals from n devices are interleaved with one another, forming a frame of data (bits, bytes, or any other data unit).

o

Framing bits allow the TDM multiplexer to synchronize properly.

o

Digital signal (DS) is a hierarchy of TDM signals.

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T lines (T-1 to T-4) are the implementation of DS services. A T-1 line consists of 24 voice channels.

o

T lines are used in North America. The European standard defines a variation called E lines.

o

Inverse multiplexing splits a data stream from one high-speed line onto multiple lowerspeed lines.

o

Errors can be categorized as a single-bit error or a burst error. A single-bit error has one bit error per data unit. A burst error has two or more bit errors per data unit.

o

Redundancy is the concept of sending extra bits for use in error detection.

o

Three common redundancy methods are parity check, cyclic redundancy check (CRC), and checksum.

o

An extra bit (parity bit) is added to the data unit in the parity check.

o

The parity check can detect only an odd number of errors; it cannot detect an even number of errors.

o

In the two-dimensional parity check, a redundant data unit follows n data units.

o

CRC, a powerful redundancy checking technique, appends a sequence of redundant bits derived from binary division to the data unit.

o

The divisor in the CRC generator is often represented as an algebraic poly-nomial.

o

Errors are corrected through retransmission and by forward error correction.

o

The Hamming code is an error correction method using redundant bits. The number of bits is a function of the length of the data bits.

o

In the Hamming code, for a data unit of m bits, use the formula 2 r >= m +r +1 to determine r, the number of redundant bits needed.

o

By rearranging the order of bit transmission of the data units, the Hamming code can correct burst errors.

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UNIT - III SWITCHING A switched network consists of a series of interlinked nodes, called switches. Switches are hardware and/or software devices capable of creating temporary connections between two or more devices lined to the switch but not to each other. In a switched network, some of these nodes are connected to the communicating devices. Others are used only for routing. Figure 3.1 shows a switched network.

Three methods of switching have been important: Circuit switching, packet switching and message switching.

Circuit Switching In telecommunications, a circuit switching network is one that establishes a fixed bandwidth circuit (or channel) between nodes and terminals before the users may communicate, as if the nodes were physically connected with an electrical circuit.

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The bit delay is constant during a connection, as opposed to packet switching, where packet queues may cause varying delay. Each circuit cannot be used by other callers until the circuit is released and a new connection is set up. Even if no actual communication is taking place in a dedicated circuit that channel remains unavailable to other users. Channels that are available for new calls to be set up are said to be idle. Virtual circuit switching is a packet switching technology that may emulate circuit switching, in the sense that the connection is established before any packets are transferred, and that packets are delivered in order. There is a common misunderstanding that circuit switching is used only for connecting voice circuits (analog or digital). The concept of a dedicated path persisting between two communicating parties or nodes can be extended to signal content other than voice. Its advantage is that it provides for non-stop transfer without requiring packets and without most of the overhead traffic usually needed, making maximal and optimal use of available bandwidth. The disadvantage of inflexibility tends to reserve it for specialized applications, particularly with the overwhelming proliferation of internet-related technology. Examples of circuit switched networks 

Public Switched Telephone Network (PSTN)

ISDN B-channel

Circuit Switched Data (CSD) and High-Speed Circuit-Switched Data (HSCSD) service in cellular systems such as GSM

X.21 (Used in the German DATEX-L and Scandinavian DATEX circuit switched data network)

Compared to datagram packet switching Since the first days of the telegraph it has been possible to multiplex multiple connections over the same physical conductor, but nonetheless each channel on the multiplexed link was either dedicated to one call at a time, or it was idle between calls. With circuit switching, and virtual circuit switching, a route is reserved from source to destination. The entire message is sent in order so that it does not have to be reassembled at the destination. Circuit switching can be relatively inefficient because capacity is wasted on connections which are set up but are not in continuous use (however momentarily). On the other hand, the connection is immediately available and capacity is guaranteed until the call is disconnected.

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Circuit switching contrasts with packet switching which splits traffic data (for instance, digital representation of sound, or computer data) into chunks, called packets, that are routed over a shared network. Packet switching is the process of segmenting a message/data to be transmitted into several smaller packets. Each packet is labeled with its destination and the number of the packet, precluding the need for a dedicated path to help the packet find its way to its destination. Each is dispatched and many may go via different routes. At the destination, the original message is reassembled in the correct order, based on the packet number. Datagram Packet switching networks do not require a circuit to be established and allow many pairs of nodes to communicate almost simultaneously over the same channel. ďƒ˜

A circuit-switched network is made of a set of switches connected by physical links, in which each link is divided into n channels.

Figure 3.2 A trivial circuit-switched network ďƒ˜

In circuit switching, the resources need to be reserved during the setup phase; the resources remain dedicated for the entire duration of data transfer until the teardown phase.

Example 3.1 As a trivial example, let us use a circuit-switched network to connect eight telephones in a small area. Communication is through 4-kHz voice channels. We assume that each link uses FDM to connect a maximum of two voice channels. The bandwidth of each link is then 8 kHz. Figure 8.4

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shows the situation. Telephone 1 is connected to telephone 7; 2 to 5; 3 to 8; and 4 to 6. Of course the situation may change when new connections are made. The switch controls the connections.

Figure 3.3 Circuit-switched network used in Example 3.1

Packet switching Packet switching is a network communications method that splits data traffic (digital representations of text, sound, or video data) into chunks, called packets, that are then routed over a shared network. To accomplish this, the original message/data is segmented into several smaller packets. Each packet is then labeled with its destination and the number of the packet. This precludes the need for a dedicated path to help the packet find its way to its destination. Each packet is dispatched and may go via different routes. At the destination, the original message/data is reassembled in the correct order, based on the packet number and other statistically determined factors. In each network node, packets are queued or buffered, resulting in variable delay. This contrasts with the other principal paradigm, circuit switching, which sets up a specific circuit with a limited number of constant bit rate and constant delay connections between nodes for exclusive use during the communication session. Packet mode or packet-oriented communication may be utilized with or without a packet switch, in the latter case directly between two hosts. Examples of that are point-to-point data links, digital video and audio broadcasting or a shared physical medium, such as a bus network, ring network, or hub network. Packet mode communication is a statistical multiplexing technique, also known as a dynamic bandwidth allocation method, where a physical communication channel is effectively divided into an arbitrary number of logical variable bit-rate channels or data streams. Each logical stream consists of a sequence of packets, which normally are forwarded by a network node asynchronously in a first-come first-serve fashion. Alternatively, the packets may be forwarded according to some scheduling discipline for fair queuing or differentiated and/or guaranteed

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Quality of service. In case of a shared physical media, the packets may be delivered according to some packet-mode multiple access scheme. The service actually provided to the user by networks using packet switching internally to the network can be datagrams (connectionless messages), and/or virtual circuit switching (also known as connection oriented). Some connectionless protocols are Ethernet, IP, and UDP; connection oriented protocols include X.25, Frame relay, Asynchronous Transfer Mode (ATM), Multiprotocol Label Switching (MPLS), and TCP. It is also entirely possible to have to weigh the various metrics against each other. For example, reducing the hop count could increase the latency to an unacceptable limit and some kind of balance would need to be found. For multi-parameter optimization, some form of optimization may be needed. Once a route is determined for a packet, it is entirely possible that the route may change for the next packet, thus leading to a case where packets from the same source headed to the same destination could be routed differently. Packet switching influenced the development of the Actor model of concurrent computation in which messages sent to the same address may be delivered in an order different from the order in which they were sent. Packet switching in networks Packet switching is used to optimize the use of the channel capacity available in digital telecommunication networks such as computer networks, to minimize the transmission latency (i.e. the time it takes for data to pass across the network), and to increase robustness of communication. The most well-known use of packet switching is the Internet and local area networks. The Internet uses the Internet protocol suite over a variety of data link layer protocols. For example, Ethernet and frame relay are very common. Newer mobile phone technologies (e.g., GPRS, I-mode) also use packet switching. X.25 is a notable use of packet switching in that, despite being based on packet switching methods, it provided virtual circuits to the user. These virtual circuits carry variable-length packets. In 1978, X.25 was used to provide the first international and commercial packet switching network, the International Packet Switched Service (IPSS). Asynchronous Transfer

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Mode (ATM) also is a virtual circuit technology, which uses fixed-length cell relay connection oriented packet switching. Datagram packet switching is also called connectionless networking because no connections are established. Technologies such as Multiprotocol Label Switching (MPLS) and the Resource Reservation Protocol (RSVP) create virtual circuits on top of datagram networks. Virtual circuits are especially useful in building robust failover mechanisms and allocating bandwidth for delaysensitive applications. MPLS and its predecessors, as well as ATM, have been called "fast packet" technologies. MPLS, indeed, has been called "ATM without cells"

[1]

. Modern routers, however, do not require these

technologies to be able to forward variable-length packets at multigigabit speeds across the network. 

In a packet-switched network, there is no resource reservation; resources are allocated. on demand

Figure 3.4 A datagram network with four switches (routers) 

A switch in a datagram network uses a routing table that is based on the destination address

The destination address in the header of a packet in a datagram network remains the same during the entire journey of the packet

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Figure 3.5 Delay in a datagram network 

Switching in the Internet is done by using the datagram approach to packet switching at the network layer.

VIRTUAL-CIRCUIT NETWORKS 

A virtual-circuit network is a cross between a circuit-switched network and a datagram network. It has some characteristics of both.

Figure 3.6 Virtual-circuit network

In virtual-circuit switching, all packets belonging to the same source and destination travel the same path; but the packets may arrive at the destination with different delays if resource allocation is on demand.

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Figure 3.7 Delay in a virtual-circuit network ďƒ˜

Switching at the data link layer in a switched WAN is normally implemented by using virtual-circuit techniques.

Message switching

In telecommunications, message switching was the precursor of packet switching, where messages were routed in their entirety, one hop at a time. It was first introduced by Leonard Kleinrock in 1961. Message switching systems are nowadays mostly implemented over packetswitched or circuit-switched data networks. Examples Hop-by-hop Telex forwarding and UUCP are examples of message switching systems. E-mail is another example of a message switching system. When this form of switching is used, no physical path is established in advance in between sender and receiver. Instead, when the sender has a block of data to be sent, it is stored in the first switching office (i.e. router) then forwarded later at one hop at a time. Each block is received in its entity form, inspected for errors and then forwarded or re-transmitted.

A form of store-and-forward network. Data is transmitted into the network and stored in a switch. The network transfers the data from switch to switch when it is convenient to do so, as such the data is not transferred in real-time. Blocking can not occur, however, long delays can happen. The source and destination terminal need not be compatible, since conversions are done by the message switching networks.

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A message switch is “transactional”. It can store data or change its format and bit rate, then convert the data back to their original form or an entirely different form at the receive end. Message switching multiplexes data from different sources onto a common facility. Store and forward delays Since message switching stores each message at intermediate nodes in its entirety before forwarding, messages experience an end to end delay which is dependent on the message length, and the number of intermediate nodes. Each additional intermediate node introduces a delay which is at minimum the value of the minimum transmission delay into or out of the node. Note that nodes could have different transmission delays for incoming messages and outgoing messages due to different technology used on the links. The transmission delays are in addition to any propagation delays which will be experienced along the message path. In a message-switching centre an incoming message is not lost when the required outgoing route is busy. It is stored in a queue with any other messages for the same route and retransmitted when the required circuit becomes free. Message switching is thus an example of a delay system or a queuing system. Message switching is still used for telegraph traffic and a modified form of it, known as packet switching, is used extensively for data communications. Connection-Oriented and Connectionless Services 

Connection-oriented

Requires a session connection (analogous to a phone call) be

established before any data can be sent. This method is often called a "reliable" network service. It can guarantee that data will arrive in the same order. Connection-oriented services set up virtual links between end systems through a network, as shown in Figure 3.8. Note that the packet on the left is assigned the virtual circuit number 01. As it moves through the network, routers quickly send it through virtual circuit 01. 

Connectionless

Does not require a session connection between sender and receiver.

The sender simply starts sending packets (called datagrams) to the destination. This service does not have the reliability of the connection-oriented method, but it is useful for periodic burst transfers. Neither system must maintain state information for the systems that they send transmission to or receive transmission from. A connectionless network provides minimal services.

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Figure 3.8 Connection-oriented methods may be implemented in the data link layers of the protocol stack and/or in the transport layers of the protocol stack, depending on the physical connections in place and the services required by the systems that are communicating. TCP (Transmission Control Protocol) is a connection-oriented transport protocol, while UDP (User Datagram Protocol) is a connectionless network protocol. Both operate over IP. The physical, data link, and network layer protocols have been used to implement guaranteed data delivery. For example, X.25 packet-switching networks perform extensive error checking and packet acknowledgment because the services were originally implemented on poor-quality telephone connections. Today, networks are more reliable. It is generally believed that the underlying network should do what it does best, which is deliver data bits as quickly as possible. Therefore, connection-oriented services are now primarily handled in the transport layer by end systems, not the network. This allows lower-layer networks to be optimized for speed. LANs operate as connectionless systems. A computer attached to a network can start transmitting frames as soon as it has access to the network. It does not need to set up a connection with the destination system ahead of time. However, a transport-level protocol such as TCP may set up a connection-oriented session when necessary. The Internet is one big connectionless packet network in which all packet deliveries are handled by IP. However, TCP adds connection-oriented services on top of IP. TCP provides all the upperlevel connection-oriented session requirements to ensure that data is delivered properly. MPLS is

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a relatively new connection-oriented networking scheme for IP networks that sets up fast labelswitched paths across routed or layer 2 networks. A WAN service that uses the connection-oriented model is frame relay. The service provider sets up PVCs (permanent virtual circuits) through the network as required or requested by the customer. ATM is another networking technology that uses the connection-oriented virtual circuit approach. Integrated Services Digital Network Integrated Services Digital Network or Isolated Subscriber Digital Network (ISDN), is a telephone system network. Prior to the ISDN, the phone system was viewed as a way to transport voice, with some special services available for data. The key feature of the ISDN is that it integrates speech and data on the same lines, adding features that were not available in the classic telephone system. There are several kinds of access interfaces to the ISDN defined: Basic Rate Interface (BRI), Primary Rate Interface (PRI) and Broadband-ISDN (B-ISDN). ISDN is a circuit-switched telephone network system, that also provides access to packet switched networks, designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in better voice quality than an analog phone. It offers circuitswitched connections (for either voice or data), and packet-switched connections (for data), in increments of 64 kbit/s. Another major use case is Internet access, where ISDN typically provides a maximum of 128 kbit/s in both upstream and downstream directions (which can be considered to be broadband speed, since it exceeds the narrowband speeds of standard analog 56k telephone lines). ISDN channels may use bonding to achieve a greater data rate, typically 3 or 4 BRIs (6 to 8 64 kbit/s channels) are bonded. ISDN should not be mistaken with any specific protocol, like Q.931. In a broad sense ISDN can be considered a digital communications medium existing on layers 1, 2, and 3 of the OSI model. ISDN is designed to provide access to voice and data services simultaneously. However, common use has reduced ISDN to be limited to Q.931 and related protocols, which are a set of protocols for establishing and breaking circuit switched connections, and for advanced call features for the user. They were introduced in the late 1980s. In a videoconference, ISDN provides simultaneous voice, video, and text transmission between individual desktop videoconferencing systems and group (room) videoconferencing systems.

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ISDN elements 

Integrated Services refers to ISDN's ability to deliver at minimum two simultaneous connections, in any combination of data, voice, video, and fax, over a single line. Multiple devices can be attached to the line, and used as needed. That means an ISDN line can take care of most people's complete communications needs at a much higher transmission rate, without forcing the purchase of multiple analog phone lines.

Digital refers to its purely digital transmission, as opposed to the analog transmission of plain old telephone service (POTS). Use of an analog telephone modem for Internet access requires that the Internet service provider's (ISP) modem converts the digital content to analog signals before sending it and the user's modem then converts those signals back to digital when receiving. When connecting with ISDN there is no digital to analog conversion.

Network refers to the fact that ISDN is not simply a point-to-point solution like a leased line. ISDN networks extend from the local telephone exchange to the remote user and includes all of the telecommunications and switching equipment in between.

The purpose of the ISDN is to provide fully integrated digital services to the users. These services fall under three categories: bearer services, supplementary services and teleservices. Basic Rate Interface The entry level interface to ISDN is the Basic Rate Interface (BRI), a 144 kbit/s service delivered over a pair of standard telephone copper wires. The 144 kbit/s rate is broken down into two 64 kbit/s bearer channels ('B' channels) and one 16 kbit/s signaling channel ('D' channel or Delta channel). BRI is sometimes referred to as 2B+D The Interface specifies three different network interfaces: 

The U interface is a two-wire interface between the exchange and the Network Terminating Unit which is usually the demarcation point in non-North American networks.

The T interface is a serial interface between a computing device and a Terminal Adapter, which is the digital equivalent of a modem.

The S interface is a four-wire bus that ISDN consumer devices plug into; the S & T reference points are commonly implemented as a single interface labeled 'S/T' on an NT1

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The R interface defines the point between a non-ISDN device and a terminal adapter (TA) which provides translation to and from such a device.

BRI-ISDN is very popular in Europe but is much less common in North America. Primary Rate Interface The other ISDN service available is the Primary Rate Interface (PRI) which is carried over an E1 (2048 kbit/s) in most parts of the world. An E1 is 30 'B' channels of 64 kbit/s, one 'D' channel of 64 kbit/s and a timing and alarm channel of 64 kbit/s. In North America PRI service is delivered on one or more T1s (sometimes referred to as 23B+D) of 1544 kbit/s (24 channels). A T1 has 23 'B' channels and 1 'D' channel for signalling (Japan uses a circuit called a J1, which is similar to a T1). In North America, NFAS allows two or more PRIs to be controlled by a single D channel, and is sometimes called "23B+D + n*24B". D-channel backup allows you to have a second D channel in case the primary fails. One popular use of NFAS is on a T3. PRI-ISDN is popular throughout the world, especially for connection of PSTN circuits to PBXs. Even though many network professionals use the term "ISDN" to refer to the lower-bandwidth BRI circuit, in North America by far the majority of ISDN services are in fact PRI circuits serving PBXs. Data Channel The bearer channel (B) is a standard 64 kbit/s voice channel of 8 bits sampled at 8 kHz with G.711 encoding. B-Channels can also be used to carry data, since they are nothing more than digital channels. Each one of these channels is known as a DS0 . Most B channels can carry a 64 kbit/s signal, but some were limited to 56K because they traveled over RBS lines. This was more of a problem in the past, and is not commonly encountered nowadays. X.25 X.25 can be carried over the B or D channels of a BRI line, and over the B channels of a PRI line. X.25 over the D channel is used at many point-of-sale (credit card) terminals because it

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eliminates the modem setup, and because it connects to the central system over a B channel, thereby eliminating the need for modems and making much better use of the central system's telephone lines. X.25 was also part of an ISDN protocol called "Always On/Dynamic ISDN", or AO/DI. This allowed a user to have a constant multi-link PPP connection to the internet over X.25 on the D channel, and brought up one or two B channels as needed. Frame Relay In theory, Frame Relay can operate over the D channel of BRIs and PRIs, but it is seldom, if ever, used. ISDN Applications Internet Access ISDN is a fast, accurate, economical conduit for all kinds of Internet-based information, including global email, file transfer, news groups, bulletin boards and interactive applications. Emerging applications include electronic commerce and low-speed video and radio broadcasting. Net surfers can even cruise virtual malls to shop the world for goods and services in a convenient and secure sales environment. Telecommuting or Work-at-Home Both for full-time telecommuters and for those who work at home, ISDN supports fast, reliable LAN access, enhanced voice services, corporate email, file transfer, remote printing and all the other functions available to those that work in an office environment. Video Conferencing ISDN supports distance learning, business training, administrative conferencing and a wide range of other video-based applications with high fidelity audio and broadcast-quality, full color, full motion images. ISDN can be configured as set-top TV to PC, and PC to PC video conferencing at either 128 Kbps or 384 Kbps.

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Branch Office Communications ISDN provides high-performance, cost-effective connections for transferring information and linking your multiple business locations to create a fast, smooth-running Wide Area Network (WAN). Standby Inter-Networking ISDN can help you minimize the risks of outages or overflows by providing reliable, high-speed back up and extra bandwidth on a cost-effective, dial-up basis. Enhanced Voice Services Businesses are under increasing pressure to handle inbound calls more efficiently. Qwest ISDN Services provide many advanced call management features – such as, multiple telephone numbers, shared telephone numbers, conferencing, transferring and caller name and number ID. And there's no need to miss crucial calls while cruising the Web, because ISDN enables you to access the Internet and place and receive calls simultaneously. Point-of-Sale Services ISDN delivers the speed and performance formerly associated only with costly private lines – at economical, public network prices. With ISDN, transaction times for processing credit card sales averages just four to six seconds. Compare that to analog telephone lines, which range from 30 to 60 seconds. That's especially important to queue-sensitive and high-volume retailers. Telemedia/Desktop Publishing ISDN offers the bandwidth and digital clarity to support such publishing-related functions as typesetting, processing color separations, creating printing plates, color scanning of photographs and page layout proofing. When turnaround speed is essential, ISDN is an ideal solution. Remote broadcasting ISDN can be a better way to broadcast from remote locations. Single Line Service can provide capability for traditional radio broadcasting – full duplex, broadcast-quality audio at 7.5 Khz on a single B channel – and for CD-quality stereo at 20 Khz on two B channels.

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Telemetry/Security With ISDN, a central surveillance site can visually observe numerous remote locations, while monitoring the status of alarms and access control devices. The B channels provide connectivity for video and the D channel accommodates the other alarm inputs. SUMMARY 

Switching is a method in which communication devices are connected to one another efficiently.

A switch is intermediary hardware or software that links devices together temporarily.

There are three fundamental switching methods: circuit switching, packet switching, and message switching.

In circuit switching, a direct physical connection between two devices is created by space-division switches, time-division switches, or both.

In a space-division switch, the path from one device to another is spatially separate from other paths.

A crossbar is the most common space-division switch. It connects n inputs to m outputs via n × m crosspoints.

Multistage switches can reduce the number of crosspoints needed, but blocking may result.

Blocking occurs when not every input has its own unique path to every output.

In a time-division switch, the inputs are divided in time, using TDM. A control unit sends the input to the correct output device.

The time-slot interchange and the TDM bus are two types of time-division switches.

Space- and time-division switches may be combined.

A telephone network is an example of a circuit-switched network.

A telephone system has three major components: local loops, trunks, and switching offices.

The United States is divided into more than 200 local exchange carriers (ILECs) and competitive local exchange carriers (CLECs). Inter-LATA services are handled by interexchange carriers (IXCs).

Telephone companies provide digital services such as switched/56 services and digital data services.

The AT&T monopoly was broken in 1984 through a government suit.

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UNIT_IV NETWORKING AND INTERNET WORKING DEVICES An internet is an interconnection of individual networks. To create an internet, we need internetworking devices called routers and gateways.

Networking and internetworking devices are divided into four categories : repeaters, bridges, routers, and gateways. 

an internet = a collection of connected networks which share a common set of rules for communication

reminder: the Internet = connected set of networks which all use IP

usually, additional devices are needed:

a broadcast LAN may need to extend further than its standard allows – use repeaters (also called signal regenerators)

the number of nodes required on the network may be too high, so the network may have to be subdivided – use bridges

two or more networks may have to be connected together – use routers (if the networks use the same Network layer protocol) or gateways (if the networks use different protocol stacks)

warning: terminology is not universally agreed on (especially by equipment manufacturers and vendors )

e.g. confusion between functionality of bridges vs. routers…

each type of internetworking device interacts with protocols at different layers of the OSI model

repeater – active only at the Physical layer

bridge – most active at the Datalink layer

router – links separate but similar LANs ⇒most active at the Network layer

gateway – provides “translation” service between incompatible LANs or applications ⇒active in all layers

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NOTE: each of these devices also operates in all layers below the one in which it is most active

Internetworking Devices: Repeater

Repeaters are Internetworking devices which operate at the physical layer. It deals with signal reproduction and retransmission of data. A simple amplifier will amplify not only the signal but also any noise accompanying the signal. But a repeater strips the digital data & saves it. It then reconstructs and retransmits the signal. The new signal is an exact duplicate of the original transmitted signal, able to travel over the new network segment. A repeater does not incorporate any changes to, or even analysis of, the addressing or structure of the data associated with other layers (higher). It simply reconditions received data & passes it on. 

an electronic device which regenerates (“cleans up”) incoming signals

allows the physical reach of a network to be extended

• a repeater does not filter frames, e.g. A’s Frame to B also received by C & D

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• a repeater copies and “refreshes” incoming bits – it does not amplify the signal

Internetworking Devices: Bridge Bridge is used to connect two LAN segments at the Data Link Layer. Bridges can determine the physical addresses of the source and destination stations involved. Once determined, bridges can permit or deny access to the new segment based on physical address. Unlike repeaters, Bridges are selective about the traffic they allow through. Bridges are usually used to divide a toobusy network into separate segments. After such a division, the bridge prevents traffic internal to one segment, from reaching other segments. As long as inter segment traffic is not too heavy, this effectively reduces traffic on each segment. Since the Bridges store & forward data, they can analyse address fields in the frames and forward the data based on the database contained in the Bridge.

operates in both the Physical and Datalink layers

a bridge knows the physical addresses of the connected nodes

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a bridge can divide a large network into smaller segments, or relay Frames between 2 originally unconnected LANs:

unlike a repeater, a bridge contains logic which allows it to keep traffic for each segment separate » bridge can filter traffic

helps in controlling traffic congestion, isolating problems, security…

Internetworking Devices: Bridge traffic filtering 

when a Frame arrives, the bridge not only regenerates the signal but also checks the destination address, and only forwards the Frame to the segment to which this address belongs:

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Frame relayed to entire upper segment

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Frame relayed to entire lower segment

Internetworking Devices: Bridge types 

simple bridge – links 2 segments; node addresses entered manually in bridge table

multi-port bridge – connects more than 2 segments:

transparent bridge (also called a learning bridge) – builds its tables of addresses automatically as it relays Frames (by noting the source address in each Frame)

if more than one bridge connects 2 LANs, a loop could be formed in the bridges’ forwarding tables ⇒Frames could circulate forever

transparent bridges learn the topology and build a loop-free spanning tree

source routing bridge – each sender learns the topology (using Discovery Frames) and decides the exact path of segments and bridges each of its Frames will take

Internetworking Devices: Bridge between different LAN types 

example of a bridge connecting a CSMA/CD LAN to a Token Bus LAN:

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problems in connecting different types of LAN:

different Frame formats

different payload sizes (e.g. 1500 bytes in Ethernet, 4500 bytes in Token Ring)

different data rates (e.g. 10 Mbps in Ethernet, 16 Mbps in Token Ring)

different bit order of addresses

presence or absence of priority bits

presence or absence of ACK/NAK

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Internetworking Devices: Router

Routers have access to information from all three lower OSI layers Physical, Data Link & Network).Routers offer more functionality as compared to Bridges, in being to do, routing and management of traffic & filtering the data across the network. Routers send information using logical address information. Logical addresses are assigned normally by the network administrators whereas physical addresses are assigned by the hardware manufacturer. Routers use one (or more) specific routing algorithms to calculate the best path. Paths may be calculated in real time, so that they can constantly adjust to changing network conditions. Routers are typically much more processing intensive than bridges. As a result, their processing speeds(measured in packets forwarded/sec) are not usually as high. 

a router operates in the Network, Datalink, and Physical layers

a router knows Network layer (“logical”) addresses

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routers relay Packets among interconnected networks

a router has links to 2 or more networks at the same time

a router link to one of its connected networks has an address on that network

if there is no router connected to both the sender’s network and receiver’s network, the router connected to the sender’s network transfers the Packet across one of its connected networks to another router which (hopefully) is “nearer” the receiver

Packets forwarded from one router to the next like this, until receiver is found

Internetworking Devices: Gateway Gateway: Gateway interconnects two or more subnetworks that use different protocols above the network layer. Gateways can connect any network to any other network. They provide full range of functionality from bit handling at the physical level up through framing, error detection routing, flow control, etc. Suppose a PC network using Netware is to be connected to an SNA network consisting of several terminals. Not only is the hardware different, but the entire structure of the data and many of the protocols used are different. A Gateway translates between the different Transport, Session, Presentation, & Application Layer Protocols, altering as much of the entire data message as needed.

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also called a Protocol Convertor

may have to operate in all 7 layers of the OSI Model

a gateway is usually a piece of software installed in a router

gateway software understands all the protocols used by networks to which the router is connected ⇒

adjustments to incoming packets could include changes to:

values in header and/or trailer fields

data rate

size of packet

entire format of packet

OTHER NETWORK DEVICES

Internetworking Devices: Multiprotocol Router 

can relay Packets from 2 or more Network layer protocols

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Brouters

Many modern routers are really Brouters. Brouters are essentially routers that can also bridge. A brouter will first check a packet to see if it supports the packet’s routing algorithm. If not, rather than simply dropping the packet, the packet is bridged using layer 2 information.

Switches

Ethernet Switches: When the number of nodes on a particular Ethernet segment goes up, the available bandwidth comes down for each node. This is because of the fact that Ethernet works on shared medium rule. An Ethernet Switch provides dedicated bandwidth to every port connection. It improves performance through micro-segmentation – reducing the number of users per segment and so increasing the available bandwidth of the LAN for each user. Delay from input to output of a switch is very less when compared to Bridges. It achieves this by routing the data to the port that connects the Node whose address is in the destination address portion of the Ethernet packet. Using this address, the switch can send the packet to the desired destination port only. This results in reduced traffic on the other ports and higher total throughput. In the traditional hubs and repeaters, the data packets are sent to all ports that is, similar to connecting nodes on the same segment. A switch automatically learns the identity of attached end-stations, so no configuration is necessary. This makes switches much simpler to install and use than routers. Since the decision making is hardware-based, it results in higher performance.

Port Switches: In this type of switch, every port is intended to connect to a single end station or a server. Port switching is merely an electronic patch panel function, not the genuine switching capability that provides a performance boost. Port switching lets administrators configure their networks to allocate any port to any backplane segment on their hub. Unlike true switching, it

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does not increase the bandwidth available to the users or servers. What is more, changing users from one backplane segment to another can be dangerous, particularly in routed networks. The path that the user’s data needs to take may be increased by moving from one segment to another, or data transmissions may be stopped altogether if a device has moved from one routed subnet to another. Port Switching is useful for data intensive applications such as Client-Server computing, imaging, CAD/CAM, desktop video conferencing and multimedia.

Segment switching: In this type of switch, each port is connected to a Ethernet segment directly or through Hub(if the nodes have a 10base T interface). Since each port of the Ethernet switch is connected to a segment, each port has to support large number of Ethernet addresses. Normally it is 256 to 1024. The advantage of the segment switch is that it increases the bandwidth available to the nodes, by providing “on-the-fly” 10Mbps link on a packet basis to the respective segments. The process of splitting a large segment into smaller segments but still maintaining the whole as a logical segment is called as “segmentation”.

Routing Switches

A new generation of switches that are a combination of a router and a bridge has recently appeared on the market. These routing switches use the network layer destination address to find the output link to which the packet should be forwarded. The process is faster because the network layer software in a regular router finds only the network address of the next station and the passes this information to the data link layer software to find the output link.

INTERNET The Internet is a global system of interconnected computer networks that interchange data by packet switching using the standardized Internet Protocol Suite (TCP/IP). It is a "network of networks" that consists of millions of private and public, academic, business, and government networks of local to global scope that are linked by copper wires, fiber-optic cables, wireless connections, and other technologies. The Internet carries various information resources and services, such as electronic mail, online chat, file transfer and file sharing, online gaming, and the inter-linked hypertext documents and other resources of the World Wide Web (WWW).

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Terminology The terms "Internet" and "World Wide Web" are often used in every-day speech without much distinction. However, the Internet and the World Wide Web are not one and the same. The Internet is a global data communications system. It is a hardware and software infrastructure that provides connectivity between computers. In contrast, the Web is one of the services communicated via the Internet. It is a collection of interconnected documents and other resources, linked by hyperlinks and URLs. History of the Internet Prior to the widespread internetworking that led to the Internet, most communication networks were limited by their nature to only allow communications between the stations on the network, and the prevalent computer networking method was based on the central mainframe method. In the 1960s, computer researchers, Levi C. Finch and Robert W. Taylor pioneered calls for a joined-up global network to address interoperability problems. Concurrently, several research programs began to research principles of networking between separate physical networks, and this led to the development of Packet switching. These included Donald Davies (NPL), Paul Baran (RAND Corporation), and Leonard Kleinrock's MIT and UCLA research programs. This led to the development of several packet switched networking solutions in the late 1960s and 1970s, including ARPANET and X.25. Additionally, public access and hobbyist networking systems grew in popularity, including UUCP and FidoNet. They were however still disjointed separate networks, served only by limited gateways between networks. This led to the application of packet switching to develop a protocol for inter-networking, where multiple different networks could be joined together into a super-framework of networks. By defining a simple common network system, the Internet protocol suite, the concept of the network could be separated from its physical implementation. This spread of inter-network began to form into the idea of a global inter-network that would be called 'The Internet', and this began to quickly spread as existing networks were converted to become compatible with this. This spread quickly across the advanced telecommunication networks of the western world, and then began to penetrate into the rest of the world as it became the de-facto international standard and global network. However, the disparity of growth led to a digital divide that is still a concern today. Following commercialisation and introduction of privately run Internet Service Providers in the 1980s, and its expansion into popular use in the 1990s, the Internet has had a drastic impact on culture and commerce. This includes the rise of near instant communication by e-mail, text based discussion forums, the World Wide Web. Investor speculation in new markets provided by these

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innovations would also lead to the inflation and collapse of the Dot-com bubble, a major market collapse. But despite this, Internet continues to grow. Creation A 1946 comic science-fiction story, A Logic Named Joe, by Murray Leinster laid out the Internet and many of its strengths and weaknesses. However, it took more than a decade before reality began to catch up with this vision. The USSR's launch of Sputnik spurred the United States to create the Advanced Research Projects Agency, known as ARPA, in February 1958 to regain a technological lead.

[2][3]

ARPA

created the Information Processing Technology Office (IPTO) to further the research of the Semi Automatic Ground Environment (SAGE) program, which had networked country-wide radar systems together for the first time. J. C. R. Licklider was selected to head the IPTO, and saw universal networking as a potential unifying human revolution. Licklider moved from the Psycho-Acoustic Laboratory at Harvard University to MIT in 1950, after becoming interested in information technology. At MIT, he served on a committee that established Lincoln Laboratory and worked on the SAGE project. In 1957 he became a Vice President at BBN, where he bought the first production PDP-1 computer and conducted the first public demonstration of time-sharing. At the IPTO, Licklider recruited Lawrence Roberts to head a project to implement a network, and Roberts based the technology on the work of Paul Baran, who had written an exhaustive study for the U.S. Air Force that recommended packet switching (as opposed to circuit switching) to make a network highly robust and survivable. After much work, the first two nodes of what would become the ARPANET were interconnected between UCLA and SRI International in Menlo Park, California, on October 29, 1969. The ARPANET was one of the "eve" networks of today's Internet. Following on from the demonstration that packet switching worked on the ARPANET, the British Post Office, Telenet, DATAPAC and TRANSPAC collaborated to create the first international packet-switched network service. In the UK, this was referred to as the International Packet Switched Service (IPSS), in 1978. The collection of X.25-based networks grew from Europe and the US to cover Canada, Hong Kong and Australia by 1981. The X.25 packet switching standard was developed in the CCITT (now called ITU-T) around 1976. X.25 was independent of the TCP/IP protocols that arose from the experimental work of DARPA on the ARPANET, Packet Radio Net and Packet Satellite Net during the same time period. Vinton Cerf and Robert Kahn developed the first description of the TCP protocols during 1973 and published a paper on the subject in May 1974. Use of the term "Internet" to describe a single global TCP/IP

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network originated in December 1974 with the publication of RFC 675, the first full specification of TCP that was written by Vinton Cerf, Yogen Dalal and Carl Sunshine, then at Stanford University. During the next nine years, work proceeded to refine the protocols and to implement them on a wide range of operating systems. The first TCP/IP-based wide-area network was operational by January 1, 1983 when all hosts on the ARPANET were switched over from the older NCP protocols. In 1985, the United States' National Science Foundation (NSF) commissioned the construction of the NSFNET, a university 56 kilobit/second network backbone using computers called "fuzzballs" by their inventor, David L. Mills. The following year, NSF sponsored the conversion to a higher-speed 1.5 megabit/second network. A key decision to use the DARPA TCP/IP protocols was made by Dennis Jennings, then in charge of the Supercomputer program at NSF. The opening of the network to commercial interests began in 1988. The US Federal Networking Council approved the interconnection of the NSFNET to the commercial MCI Mail system in that year and the link was made in the summer of 1989. Other commercial electronic e-mail services were soon connected, including OnTyme, Telemail and Compuserve. In that same year, three commercial Internet service providers (ISP) were created: UUNET, PSINET and CERFNET. Important, separate networks that offered gateways into, then later merged with, the Internet include Usenet and BITNET. Various other commercial and educational networks, such as Telenet, Tymnet, Compuserve and JANET were interconnected with the growing Internet. Telenet (later called Sprintnet) was a large privately funded national computer network with free dial-up access in cities throughout the U.S. that had been in operation since the 1970s. This network was eventually interconnected with the others in the 1980s as the TCP/IP protocol became increasingly popular. The ability of TCP/IP to work over virtually any pre-existing communication networks allowed for a great ease of growth, although the rapid growth of the Internet was due primarily to the availability of commercial routers from companies such as Cisco Systems, Proteon and Juniper, the availability of commercial Ethernet equipment for local-area networking and the widespread implementation of TCP/IP on the UNIX operating system. Growth Although the basic applications and guidelines that make the Internet possible had existed for almost a decade, the network did not gain a public face until the 1990s. On August 6, 1991, CERN, which straddles the border between France and Switzerland, publicized the new World Wide Web project. The Web was invented by English scientist Tim Berners-Lee in 1989.

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An early popular web browser was ViolaWWW, patterned after HyperCard and built using the X Window System. It was eventually replaced in popularity by the Mosaic web browser. In 1993, the National Center for Supercomputing Applications at the University of Illinois released version 1.0 of Mosaic, and by late 1994 there was growing public interest in the previously academic, technical Internet. By 1996 usage of the word Internet had become commonplace, and consequently, so had its use as a synecdoche in reference to the World Wide Web. Meanwhile, over the course of the decade, the Internet successfully accommodated the majority of previously existing public computer networks (although some networks, such as FidoNet, have remained separate). During the 1990s, it was estimated that the Internet grew by 100% per year, with a brief period of explosive growth in 1996 and 1997. This growth is often attributed to the lack of central administration, which allows organic growth of the network, as well as the nonproprietary open nature of the Internet protocols, which encourages vendor interoperability and prevents any one company from exerting too much control over the network Internet Protocol Suite The Internet Protocol Suite (commonly TCP/IP) is the set of communications protocols used for the Internet and other similar networks. It is named from two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were the first two networking protocols defined in this standard. Today's IP networking represents a synthesis of several developments that began to evolve in the 1960s and 1970s, namely the Internet and LANs (Local Area Networks), which, together with the invention of the World Wide Web by Tim Berners-Lee in 1989, have revolutionized computing. The Internet Protocol Suite, like many protocol suites, may be viewed as a set of layers. Each layer solves a set of problems involving the transmission of data, and provides a well-defined service to the upper layer protocols based on using services from some lower layers. Upper layers are logically closer to the user and deal with more abstract data, relying on lower layer protocols to translate data into forms that can eventually be physically transmitted. The TCP/IP model consists of four layers (RFC 1122). From lowest to highest, these are the Link Layer, the Internet Layer, the Transport Layer, and the Application Layer. COMMON USES E-mail Electronic mail, often abbreviated to e-mail, email, e-post or originally eMail, is a store-andforward method of writing, sending, receiving and saving messages over electronic

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communication systems. The term "e-mail" (as a noun or verb) applies to the Internet e-mail system based on the Simple Mail Transfer Protocol, to network systems based on other protocols and to various mainframe, minicomputer, or internet by a particular systems vendor, or on the same protocols used on public networks. World Wide Web Many people use the terms Internet and World Wide Web (or just the Web) interchangeably, but, as discussed above, the two terms are not synonymous. The World Wide Web is a huge set of interlinked documents, images and other resources, linked by hyperlinks and URLs. These hyperlinks and URLs allow the web servers and other machines that store originals, and cached copies, of these resources to deliver them as required using HTTP (Hypertext Transfer Protocol). HTTP is only one of the communication protocols used on the Internet. Web services also use HTTP to allow software systems to communicate in order to share and exchange business logic and data. Software products that can access the resources of the Web are correctly termed user agents. In normal use, web browsers, such as Internet Explorer, Firefox and Apple Safari, access web pages and allow users to navigate from one to another via hyperlinks. Web documents may contain almost any combination of computer data including graphics, sounds, text, video, multimedia and interactive content including games, office applications and scientific demonstrations. Through keyword-driven Internet research using search engines like Yahoo! and Google, millions of people worldwide have easy, instant access to a vast and diverse amount of online information. Compared to encyclopedias and traditional libraries, the World Wide Web has enabled a sudden and extreme decentralization of information and data. Using the Web, it is also easier than ever before for individuals and organisations to publish ideas and information to an extremely large audience. Anyone can find ways to publish a web page, a blog or build a website for very little initial cost. Publishing and maintaining large, professional websites full of attractive, diverse and up-to-date information is still a difficult and expensive proposition, however. Many individuals and some companies and groups use "web logs" or blogs, which are largely used as easily updatable online diaries. Some commercial organisations encourage staff to fill

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them with advice on their areas of specialization in the hope that visitors will be impressed by the expert knowledge and free information, and be attracted to the corporation as a result. One example of this practice is Microsoft, whose product developers publish their personal blogs in order to pique the public's interest in their work. Collections of personal web pages published by large service providers remain popular, and have become increasingly sophisticated. Whereas operations such as Angelfire and GeoCities have existed since the early days of the Web, newer offerings from, for example, Facebook and MySpace currently have large followings. These operations often brand themselves as social network services rather than simply as web page hosts. Advertising on popular web pages can be lucrative, and e-commerce or the sale of products and services directly via the Web continues to grow. In the early days, web pages were usually created as sets of complete and isolated HTML text files stored on a web server. More recently, websites are more often created using content management system (CMS) or wiki software with, initially, very little content. Contributors to these systems, who may be paid staff, members of a club or other organisation or members of the public, fill underlying databases with content using editing pages designed for that purpose, while casual visitors view and read this content in its final HTML form. There may or may not be editorial, approval and security systems built into the process of taking newly entered content and making it available to the target visitors. Remote access The Internet allows computer users to connect to other computers and information stores easily, wherever they may be across the world. They may do this with or without the use of security, authentication and encryption technologies, depending on the requirements. This is encouraging new ways of working from home, collaboration and information sharing in many industries. An accountant sitting at home can audit the books of a company based in another country, on a server situated in a third country that is remotely maintained by IT specialists in a fourth. These accounts could have been created by home-working bookkeepers, in other remote locations, based on information e-mailed to them from offices all over the world. Some of these things were possible before the widespread use of the Internet, but the cost of private leased lines would have made many of them infeasible in practice. An office worker away from his desk, perhaps on the other side of the world on a business trip or a holiday, can open a remote desktop session into his normal office PC using a secure Virtual

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Private Network (VPN) connection via the Internet. This gives the worker complete access to all of his or her normal files and data, including e-mail and other applications, while away from the office. This concept is also referred to by some network security people as the Virtual Private Nightmare, because it extends the secure perimeter of a corporate network into its employees' homes; this has been the source of some notable security breaches, but also provides security for the workers. Collaboration The low cost and nearly instantaneous sharing of ideas, knowledge, and skills has made collaborative work dramatically easier. Not only can a group cheaply communicate and test, but the wide reach of the Internet allows such groups to easily form in the first place, even among niche interests. An example of this is the free software movement in software development, which produced GNU and Linux from scratch and has taken over development of Mozilla and OpenOffice.org (formerly known as Netscape Communicator and StarOffice). Internet "chat", whether in the form of IRC "chat rooms" or channels, or via instant messaging systems, allow colleagues to stay in touch in a very convenient way when working at their computers during the day. Messages can be sent and viewed even more quickly and conveniently than via e-mail. Extension to these systems may allow files to be exchanged, "whiteboard" drawings to be shared as well as voice and video contact between team members. Version control systems allow collaborating teams to work on shared sets of documents without either accidentally overwriting each other's work or having members wait until they get "sent" documents to be able to add their thoughts and changes. File sharing A computer file can be e-mailed to customers, colleagues and friends as an attachment. It can be uploaded to a website or FTP server for easy download by others. It can be put into a "shared location" or onto a file server for instant use by colleagues. The load of bulk downloads to many users can be eased by the use of "mirror" servers or peer-to-peer networks. In any of these cases, access to the file may be controlled by user authentication; the transit of the file over the Internet may be obscured by encryption, and money may change hands before or after access to the file is given. The price can be paid by the remote charging of funds from, for example, a credit card whose details are also passed—hopefully fully encrypted—across the

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Internet. The origin and authenticity of the file received may be checked by digital signatures or by MD5 or other message digests. These simple features of the Internet, over a worldwide basis, are changing the basis for the production, sale, and distribution of anything that can be reduced to a computer file for transmission. This includes all manner of print publications, software products, news, music, film, video, photography, graphics and the other arts. This in turn has caused seismic shifts in each of the existing industries that previously controlled the production and distribution of these products. Internet collaboration technology enables business and project teams to share documents, calendars and other information. Such collaboration occurs in a wide variety of areas including scientific research, software development, conference planning, political activism and creative writing. Streaming media Many existing radio and television broadcasters provide Internet "feeds" of their live audio and video streams (for example, the BBC). They may also allow time-shift viewing or listening such as Preview, Classic Clips and Listen Again features. These providers have been joined by a range of pure Internet "broadcasters" who never had on-air licenses. This means that an Internetconnected device, such as a computer or something more specific, can be used to access on-line media in much the same way as was previously possible only with a television or radio receiver. The range of material is much wider, from pornography to highly specialized, technical webcasts. Podcasting is a variation on this theme, where—usually audio—material is first downloaded in full and then may be played back on a computer or shifted to a digital audio player to be listened to on the move. These techniques using simple equipment allow anybody, with little censorship or licensing control, to broadcast audio-visual material on a worldwide basis. Webcams can be seen as an even lower-budget extension of this phenomenon. While some webcams can give full-frame-rate video, the picture is usually either small or updates slowly. Internet users can watch animals around an African waterhole, ships in the Panama Canal, the traffic at a local roundabout or their own premises, live and in real time. Video chat rooms, video conferencing, and remote controllable webcams are also popular. Many uses can be found for personal webcams in and around the home, with and without two-way sound. YouTube, sometimes described as an Internet phenomenon because of the vast amount of users and how rapidly the site's popularity has grown, was founded on February 15, 2005. It is now the leading website for free streaming video. It uses a flash-based web player which streams video files in the format FLV. Users are able to watch videos without signing up; however, if users do

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sign up they are able to upload an unlimited amount of videos and they are given their own personal profile. It is currently estimated that there are 64,000,000 videos on YouTube, and it is also currently estimated that 825,000 new videos are uploaded every day. Voice telephony (VoIP)

VoIP stands for Voice over IP, where IP refers to the Internet Protocol that underlies all Internet communication. This phenomenon began as an optional two-way voice extension to some of the instant messaging systems that took off around the year 2000. In recent years many VoIP systems have become as easy to use and as convenient as a normal telephone. The benefit is that, as the Internet carries the actual voice traffic, VoIP can be free or cost much less than a normal telephone call, especially over long distances and especially for those with always-on Internet connections such as cable or ADSL. Thus, VoIP is maturing into a viable alternative to traditional telephones. Interoperability between different providers has improved and the ability to call or receive a call from a traditional telephone is available. Simple, inexpensive VoIP modems are now available that eliminate the need for a PC. Voice quality can still vary from call to call but is often equal to and can even exceed that of traditional calls. Remaining problems for VoIP include emergency telephone number dialing and reliability. Currently, a few VoIP providers provide an emergency service, but it is not universally available. Traditional phones are line-powered and operate during a power failure; VoIP does not do so without a backup power source for the electronics. Most VoIP providers offer unlimited national calling, but the direction in VoIP is clearly toward global coverage with unlimited minutes for a low monthly fee. VoIP has also become increasingly popular within the gaming world, as a form of communication between players. Popular gaming VoIP clients include Ventrilo and Teamspeak, and there are others available also. The PlayStation 3 and Xbox 360 also offer VoIP chat features. INTRANET An intranet is a private computer network that uses Internet protocols and network connectivity to securely share any part of an organization's information or operational systems with its

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employees. Sometimes the term refers only to the organization's internal website, but often it is a more extensive part of the organization's computer infrastructure and private websites are an important component and focal point of internal communication and collaboration.

An intranet is built from the same concepts and technologies used for the Internet, such as clients and servers running on the Internet Protocol Suite (TCP/IP). Any of the well known Internet protocols may be found in an intranet, such as HTTP (web services), SMTP (e-mail), and FTP (file transfer). There is often an attempt to employ Internet technologies to provide modern interfaces to legacy information systems hosting corporate data. An intranet can be understood as a private version of the Internet, or as a private extension of the Internet confined to an organization. The term first appeared in print on April 19, 1995, in Digital News & Review in an article authored by technical editor Stephen Lawton. Intranets differ from extranets in that the former are generally restricted to employees of the organization while extranets may also be accessed by customers, suppliers, or other approved parties.

[2]

Extranets extend a private network onto the Internet with special provisions for access,

authorization and authentication (see also AAA protocol). An organization's intranet does not necessarily have to provide access to the Internet. When such access is provided it is usually through a network gateway with a firewall, shielding the intranet from unauthorized external access. The gateway often also implements user authentication, encryption of messages, and often virtual private network (VPN) connectivity for off-site employees to access company information, computing resources and internal communications. Increasingly, intranets are being used to deliver tools and applications, e.g., collaboration (to facilitate working in groups and teleconferencing) or sophisticated corporate directories, sales and Customer relationship management tools, project management etc., to advance productivity. Intranets are also being used as corporate culture-change platforms. For example, large numbers of employees discussing key issues in an intranet forum application could lead to new ideas in management, productivity, quality, and other corporate issues. In large intranets, website traffic is often similar to public website traffic and can be better understood by using web metrics software to track overall activity. User surveys also improve intranet website effectiveness.

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Intranet user-experience, editorial, and technology teams work together to produce in-house sites. Most commonly, intranets are managed by the communications, HR or CIO departments of large organizations, or some combination of these. Because of the scope and variety of content and the number of system interfaces, intranets of many organizations are much more complex than their respective public websites. Intranets and their use are growing rapidly. According to the Intranet design annual 2007 from Nielsen Norman Group, the number of pages on participants' intranets averaged 200,000 over the years 2001 to 2003 and has grown to an average of 6 million pages over 2005–2007. Benefits of intranets 

Workforce productivity: Intranets can help users to locate and view information faster and use applications relevant to their roles and responsibilities. With the help of a web browser interface, users can access data held in any database the organization wants to make available, anytime and - subject to security provisions - from anywhere within the company workstations, increasing employees' ability to perform their jobs faster, more accurately, and with confidence that they have the right information. It also helps to improve the services provided to the users.

Time: With intranets, organizations can make more information available to employees on a "pull" basis (i.e., employees can link to relevant information at a time which suits them) rather than being deluged indiscriminately by emails.

Communication: Intranets can serve as powerful tools for communication within an organization, vertically and horizontally. From a communications standpoint, intranets are useful to communicate strategic initiatives that have a global reach throughout the organization. The type of information that can easily be conveyed is the purpose of the initiative and what the initiative is aiming to achieve, who is driving the initiative, results achieved to date, and who to speak to for more information. By providing this information on the intranet, staff have the opportunity to keep up-to-date with the strategic focus of the organization.

Web publishing allows 'cumbersome' corporate knowledge to be maintained and easily accessed throughout the company using hypermedia and Web technologies. Examples include: employee manuals, benefits documents, company policies, business standards, newsfeeds, and even training, can be accessed using common Internet standards (Acrobat files, Flash files, CGI applications). Because each business unit can update the online copy of a document, the most recent version is always available to employees using the intranet.

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Business operations and management: Intranets are also being used as a platform for developing and deploying applications to support business operations and decisions across the internetworked enterprise.

Cost-effective: Users can view information and data via web-browser rather than maintaining physical documents such as procedure manuals, internal phone list and requisition forms.

Promote common corporate culture: Every user is viewing the same information within the Intranet.

Enhance Collaboration: With information easily accessible by all authorised users, teamwork is enabled.

Cross-platform Capability: Standards-compliant web browsers are available for Windows, Mac, and UNIX.

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UNIT – V INTERNET PROTOCOLS The Internet Protocol Suite (commonly TCP/IP) is the set of communications protocols used for the Internet and other similar networks. It is named from two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were the first two networking protocols defined in this standard. Today's IP networking represents a synthesis of several developments that began to evolve in the 1960s and 1970s, namely the Internet and LANs (Local Area Networks), which, together with the invention of the World Wide Web by Tim Berners-Lee in 1989, have revolutionized computing. The Internet Protocol Suite, like many protocol suites, may be viewed as a set of layers. Each layer solves a set of problems involving the transmission of data, and provides a well-defined service to the upper layer protocols based on using services from some lower layers. Upper layers are logically closer to the user and deal with more abstract data, relying on lower layer protocols to translate data into forms that can eventually be physically transmitted. The TCP/IP model consists of four layers (RFC 1122). From lowest to highest, these are the Link Layer, the Internet Layer, the Transport Layer, and the Application Layer. Layers in the Internet Protocol Suite The concept of layers The TCP/IP suite uses encapsulation to provide abstraction of protocols and services. Such encapsulation usually is aligned with the division of the protocol suite into layers of general functionality. In general, an application (the highest level of the model) uses a set of protocols to send its data down the layers, being further encapsulated at each level. This may be illustrated by an example network scenario, in which two Internet host computers communicate across local network boundaries constituted by their internetworking gateways (routers).

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Encapsulation of application data descending TCP/IP

stack

connected

via

operating two

on

routers

two and

hosts

through the protocol stack.

the

corresponding layers used at each hop

The major functional groups of protocols and methods are the Application Layer, the Transport Layer, the Internet Layer, and the Link Layer (RFC 1122). It should be noted that this model was not intended to be a rigid reference model into which new protocols have to fit in order to be accepted as a standard. The following table provides some examples of the protocols grouped in their respective layers.

DNS, TFTP, TLS/SSL, FTP, Gopher, HTTP, IMAP, IRC, NNTP, POP3, SIP, SMTP, SNMP, SSH, Telnet, Echo, RTP, PNRP, rlogin, ENRP Application Routing protocols like BGP and RIP which run over TCP/UDP, may also be considered part of the Internet Layer.

Transport

TCP, UDP, DCCP, SCTP, IL, RUDP, RSVP

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IP (IPv4, IPv6) ICMP, IGMP, and ICMPv6 Internet OSPF for IPv4 was inititally considered IP layer protocol since it runs per IP-subnet, but has been placed on the Link since RFC 2740.

Link

ARP, RARP, OSPF (IPv4/IPv6), IS-IS, NDP

TCP/IP Networking Protocols The TCP/IP suite of protocols is the set of protocols used to communicate across the internet. It is also widely used on many organizational networks due to its flexiblity and wide array of functionality provided. Microsoft who had originally developed their own set of protocols now is more widely using TCP/IP, at first for transport and now to support other services. TCP/IP by Layer Link Layer 

SLIP - Serial Line Internet Protocol. This protocol places data packets into data frames in preparation for transport across network hardware media. This protocol is used for sending data across serial lines. There is no error correction, addressing or packet identification. There is no authentication or negotiation capabilities with SLIP. SLIP will only support transport of IP packets.



CSLIP - Compressed SLIP is essentially data compression of the SLIP protocol. It uses Van Jacobson compression to drastically reduce the overhead of packet overhead. This may also be used with PPP and called CPPP.

PPP - Point to Point Protocol is a form of serial line data encapsulation that is an improvement over SLIP which provides serial bi-directional communication. It is much like SLIP but can support AppleTalk, IPX, TCP/IP, and NetBEUI along with TCP/IP which is supported by SLIP. It can negociate connection parameters such as speed along with the ability to support PAP and CHAP user authentication.

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Ethernet - Ethernet is not really called a protocol. There are also many types of ethernet. The most common ethernet which is used to control the handling of data at the lowest layer of the network model is 802.3 ethernet. 802.3 ethernet privides a means of encapsulating data frames to be sent between computers. It specifies how network data collisions are handled along with hardware addressing of network cards.

Network Layer 

ARP - Address Resolution Protocol enables the packaging of IP data into ethernet packages. It is the system and messaging protocol that is used to find the ethernet (hardware) address from a specific IP number. Without this protocol, the ethernet package could not be generated from the IP package, because the ethernet address could not be determined.

IP - Internet Protocol. Except for ARP and RARP all protocols' data packets will be packaged into an IP data packet. IP provides the mechanism to use software to address and manage data packets being sent to computers.

RARP - Reverse address resolution protocol is used to allow a computer without a local permanent data storage media to determine its IP address from its ethernet address.

Transport Layer 

TCP - A reliable connection oriented protocol used to control the management of application level services between computers. It is used for transport by some applications.

UDP - An unreliable connection less protocol used to control the management of application level services between computers. It is used for transport by some applications which must provide their own reliability.

ICMP - Internet control message protocol (ICMP) provides management and error reporting to help manage the process of sending data between computers. (Management). This protocol is used to report connection status back to computers that are trying to connect other computers. For example, it may report that a destination host is not reachable.

IGMP - Internet Group Management Protocol used to support multicasting. IGMP messages are used by multicast routers to track group memberships on each of its networks.

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Application Layer 

FTP - File Transfer Protocol allows file transfer between two computers with login required.

TFTP - Trivial File Transfer Protocol allows file transfer between two computers with no login required. It is limited, and is intended for diskless stations.

NFS - Network File System is a protocol that allows UNIX and Linux systems remotely mount each other's file systems.

SNMP - Simple Network Management Protocol is used to manage all types of network elements based on various data sent and received.

SMTP - Simple Mail Transfer Protocol is used to transport mail. Simple Mail Transport Protocol is used on the internet, it is not a transport layer protocol but is an application layer protocol.

HTTP - Hypertext Transfer Protocol is used to transport HTML pages from web servers to web browsers. The protocol used to communicate between web servers and web browser software clients.

BOOTP - Bootstrap protocol is used to assign an IP address to diskless computers and tell it what server and file to load which will provide it with an operating system.

DHCP - Dynamic host configuration protocol is a method of assigning and controlling the IP addresses of computers on a given network. It is a server based service that automatically assigns IP numbers when a computer boots. This way the IP address of a computer does not need to be assigned manually. This makes changing networks easier to manage. DHCP can perform all the functions of BOOTP.

BGP - Border Gateway Protocol. When two systems are using BGP, they establish a TCP connection, then send each other their BGP routing tables. BGP uses distance vectoring. It detects failures by sending periodic keep alive messages to its neighbors every 30 seconds. It exchanges information about reachable networks with other BGP systems including the full path of systems that are between them. Described by RFC 1267, 1268, and 1497.

EGP - Exterior Gateway Protocol is used between routers of different systems.

IGP - Interior Gateway Protocol. The name used to describe the fact that each system on the internet can choose its own routing protocol. RIP and OSPF are interior gateway protocols.

RIP - Routing Information Protocol is used to dynamically update router tables on WANs or the internet. A distance-vector algorithm is used to calculate the best route for a packet. RFC 1058, 1388 (RIP2).

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OSPF - Open Shortest Path First dynamic routing protocol. A link state protocol rather than a distance vector protocol. It tests the status of its link to each of its neighbors and sends the acquired information to them.

POP3 - Post Office Protocol version 3 is used by clients to access an internet mail server to get mail. It is not a transport layer protocol.

IMAP4 - Internet Mail Access Protocol version 4 is the replacement for POP3.

Telnet is used to remotely open a session on another computer. It relies on TCP for transport and is defined by RFC854.

Bandwidth Control 

BAP - Bandwidth Allocation Protocol is a bandwidth control protocol for PPP connections. It works with BACP.

BACP - Bandwidth Allocation Control Protocol.

TCP/IP by Function Packaging and Low Level 

IP - Internet Protocol. Except for ARP and RARP all protocols' data packets will be packaged into an IP data packet. IP provides the mechanism to use software to address and manage data packets being sent to computers.

SLIP - Serial Line Internet Protocol. This protocol places data packets into data frames in preparation for transport across network hardware media. This protocol is used for sending data across serial lines. There is no error correction, addressing or packet identification. There is no authentication or negotiation capabilities with SLIP. SLIP will only support transport of IP packets.

CSLIP - Compressed SLIP is essentially data compression of the SLIP protocol. It uses Van Jacobson compression to drastically reduce the overhead of packet overhead. This may also be used with PPP and called CPPP.

PPP - Point to Point Protocol is a form of serial line data encapsulation that is an improvement over SLIP which provides serial bi-directional communication. It is much like SLIP but can support AppleTalk, IPX, TCP/IP, and NetBEUI along with TCP/IP which is supported by SLIP. It can negociate connection parameters such as speed along with the ability to support PAP and CHAP user authentication.

Ethernet - Ethernet is not really called a protocol. There are also many types of ethernet. The most common ethernet which is used to control the handling of data at the lowest layer of the network model is 802.3 ethernet. 802.3 ethernet privides a means of

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encapsulating data frames to be sent between computers. It specifies how network data collisions are handled along with hardware addressing of network cards. Transport and Basic Functions 

TCP - A reliable connection oriented protocol used to control the management of application level services between computers. It is used for transport by some applications.

UDP - An unreliable connection less protocol used to control the management of application level services between computers. It is used for transport by some applications which must provide their own reliability.

Network Management 

SNMP - Simple Network Management Protocol is used to manage all types of network elements based on various data sent and received.

ICMP - Internet control message protocol provides management and error reporting to help manage the process of sending data between computers. (Management). This protocol is used to report connection status back to computers that are trying to connect other computers. For example, it may report that a destination host is not reachable. This protocol is required for basic TCP/IP operations.

ARP - Address Resolution Protocol enables the packaging of IP data into ethernet packages. It is the system and messaging protocol that is used to find the ethernet (hardware) address from a specific IP number. Without this protocol, the ethernet package could not be generated from the IP package, because the ethernet address could not be determined. protocol is used to report connection status back to computers that are trying to connect other computers. For example, it may report that a destination host is not reachable. This protocol is required for basic TCP/IP operations.

Host Management 

BOOTP - Bootstrap protocol is used to assign an IP address to diskless computers and tell it what server and file to load which will provide it with an operating system.

DHCP - Dynamic host configuration protocol is a method of assigning and controlling the IP addresses of computers on a given network. It is a server based service that automatically assigns IP numbers when a computer boots. This way the IP address of a computer does not need to be assigned manually. This makes changing networks easier to manage. DHCP can perform all the functions of BOOTP.

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RARP - Reverse address resolution protocol is used to allow a computer without a local permanent data storage media to determine its IP address from its ethernet address.

Mail Protocols 

SMTP - Simple Mail Transfer Protocol is used to transport mail. Simple Mail Transport Protocol is used on the internet, it is not a transport layer protocol but is an application layer protocol.

POP3 - Post Office Protocol version 3 is used by clients to access an internet mail server to get mail. It is not a transport layer protocol.

IMAP4 - Internet Mail Access Protocol version 4 is the replacement for POP3.

Multicasting Protocols 

IGMP - Internet Group Management Protocol used to support multicasting. IGMP messages are used by multicast routers to track group memberships on each of its networks.

Routing Protocols 

BGP - Border Gateway Protocol. When two systems are using BGP, they establish a TCP connection, then send each other their BGP routing tables. BGP uses distance vectoring. It detects failures by sending periodic keep alive messages to its neighbors every 30 seconds. It exchanges information about reachable networks with other BGP systems including the full path of systems that are between them. Described by RFC 1267, 1268, and 1497

EGP - Exterior Gateway Protocol is used between routers of different systems.

IGP - Interior Gateway Protocol. The name used to describe the fact that each system on the internet can choose its own routing protocol. RIP and OSPF are interior gateway protocols.

RIP - Routing Information Protocol is used to dynamically update router tables on WANs or the internet.

OSPF - Open Shortest Path First dynamic routing protocol. A link state protocol rather than a distance vector protocol. It tests the status of its link to each of its neighbors and sends the acquired information to them.

Domain Name System The Domain Name System (DNS) is a hierarchical naming system for computers, services, or any resource participating in the Internet. It associates various information

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with domain names assigned to such participants. Most importantly, it translates humanly meaningful domain names to the numerical (binary) identifiers associated with networking equipment for the purpose of locating and addressing these devices world-wide. An often used analogy to explain the Domain Name System is that it serves as the "phone book" for the Internet by translating human-friendly computer hostnames into IP addresses. For example, www.example.com translates to 208.77.188.166. The Domain Name System makes it possible to assign domain names to groups of Internet users in a meaningful way, independent of each user's physical location. Because of this, World-Wide Web (WWW) hyperlinks and Internet contact information can remain consistent and constant even if the current Internet routing arrangements change or the participant uses a mobile device. Internet domain names are easier to remember than IP addresses such as 208.77.188.166(IPv4) or 2001:db8:1f70::999:de8:7648:6e8 (IPv6). People take advantage of this when they recite meaningful URLs and e-mail addresses without having to know how the machine will actually locate them. The Domain Name System distributes the responsibility for assigning domain names and mapping them to Internet Protocol (IP) networks by designating authoritative name servers for each domain to keep track of their own changes, avoiding the need for a central register to be continually consulted and updated. In general, the Domain Name System also stores other types of information, such as the list of mail servers that accept email for a given Internet domain. By providing a world-wide, distributed keyword-based redirection service, the Domain Name System is an essential component of the functionality of the Internet. Other identifiers such as RFID tags, UPC codes, International characters in email addresses and host names, and a variety of other identifiers could all potentially utilize DNS. The Domain Name System also defines the technical underpinnings of the functionality of this database service. For this purpose it defines the DNS protocol, a detailed specification of the data structures and communication exchanges used in DNS, as part of the Internet Protocol Suite (TCP/IP). The context of the DNS within the Internet protocols may be seen in the following diagram. The DNS protocol was developed and defined in the early 1980's and published by the Internet Engineering Task Force .

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Structure The domain name space The domain name space consists of a tree of domain names. Only one node or leaf in the tree has zero or more resource records, which hold information associated with the domain name. The tree sub-divides into zones beginning at the root zone. A DNS zone consists of a collection of connected nodes authoritatively served by an authoritative nameserver. (Note that a single nameserver can host several zones.) Administrative responsibility over any zone may be divided, thereby creating additional zones. Authority is said to be delegated for a portion of the old space, usually in form of sub-domains, to another nameserver and administrative entity. The old zone ceases to be authoritative for the new zone. Parts of a domain name A domain name usually consists of two or more parts (technically a label), which is conventionally written separated by dots, such as example.com. 

The rightmost label conveys the top-level domain (for example, the address www.example.com has the top-level domain com).

Each label to the left specifies a subdivision, or subdomain of the domain above it. Note: “subdomain” expresses relative dependence, not absolute dependence. For example: example.com is a subdomain of the com domain, and www.example.com is a subdomain of the domain example.com. In theory, this subdivision can go down 127 levels. Each label can contain up to 63 octets. The whole domain name may not exceed a total length of 253 octets. In practice, some domain registries may have shorter limits.

A hostname refers to a domain name that has one or more associated IP addresses; ie: the 'www.example.com' and 'example.com' domains are both hostnames, however, the 'com' domain is not.

DNS servers The Domain Name System is maintained by a distributed database system, which uses the clientserver model. The nodes of this database are the name servers. Each domain or subdomain has one or more authoritative DNS servers that publish information about that domain and the name servers of any domains subordinate to it. The top of the hierarchy is served by the root nameservers: the servers to query when looking up (resolving) a top-level domain name (TLD).

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Domain Registration The right to use a domain name is delegated by domain name registrars which are accredited by the Internet Corporation for Assigned Names and Numbers (ICANN), the organization charged with overseeing the name and number systems of the Internet. In addition to ICANN, each toplevel domain (TLD) is maintained and serviced technically by a sponsoring organization, the TLD Registry. The registry is responsible for maintaining the database of names registered within the TLDs they administer. The registry receives registration information from each domain name registrar authorized to assign names in the corresponding TLD and publishes the information using a special service, the whois protocol. Registrars usually charge an annual fee for the service of delegating a domain name to a user and providing a default set of name servers. Often this transaction is termed a sale or lease of the domain name, and the registrant is called an "owner", but no such legal relationship is actually associated with the transaction, only the exclusive right to use the domain name. More correctly authorized users are known as "registrants" or as "domain holders". ICANN publishes a complete list of TLD registries and domain name registrars in the world. One can obtain information about the registrant of a domain name by looking in the WHOIS database held by many domain registries. For most of the more than 240 country code top-level domains (ccTLDs), the domain registries hold the authoritative WHOIS (Registrant, name servers, expiration dates, etc.). For instance, DENIC, Germany NIC, holds the authoritative WHOIS to a .DE domain name. Since about 2001, most gTLD registries (.ORG, .BIZ, .INFO) have adopted this so-called "thick" registry approach, i.e. keeping the authoritative WHOIS in the central registries instead of the registrars. For .COM and .NET domain names, a "thin" registry is used: the domain registry (e.g. VeriSign) holds a basic WHOIS (registrar and name servers, etc.). One can find the detailed WHOIS (registrant, name servers, expiry dates, etc.) at the registrars. Some domain name registries, also called Network Information Centres (NIC), also function as registrars, and deal directly with end users. But most of the main ones, such as for .COM, .NET, .ORG, .INFO, etc., use a registry-registrar model. There are hundreds of Domain Name Registrars that actually perform the domain name registration with the end user (see lists at ICANN or VeriSign). By using this method of distribution, the registry only has to manage the relationship with the registrar, and the registrar maintains the relationship with the end users, or 'registrants' -- in some cases through additional layers of resellers.

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In the process of registering a domain name and maintaining authority over the new name space created, registrars store and use several key pieces of information connected with a domain: 

Administrative contact. A registrant usually designates an administrative contact to manage the domain name. The administrative contact usually has the highest level of control over a domain. Management functions delegated to the administrative contacts may include management of all business information, such as name of record, postal address, and contact information of the official registrant of the domain and the obligation to conform to the requirements of the domain registry in order to retain the right to use a domain name. Furthermore the administrative contact installs additional contact information for technical and billing functions.

Technical contact. The technical contact manages the name servers of a domain name. The functions of a technical contact include assuring conformance of the configurations of the domain name with the requirements of the domain registry, maintaining the domain zone records, and providing continuous functionality of the name servers (that leads to the accessibility of the domain name).

Billing contact. The party responsible for receiving billing invoices from the domain name registrar and paying applicable fees.

Name servers. Domains usually need at least two authoritative name servers that perform name resolution for the domain. If they are not automatically provided by the registrar, the domain holder must specify domain names and IP addresses for these servers.

File Transfer Protocol File Transfer Protocol (FTP) is a network protocol used to transfer data from one computer to another through a network such as the Internet. FTP is a file transfer protocol for exchanging and manipulating files over a TCP computer network. A FTP client may connect to a FTP server to manipulate files on that server. Connection methods FTP runs exclusively over TCP. It defaults to listen on port 21 for incoming connections from FTP clients. A connection to this port from the FTP Client forms the control stream on which commands are passed to the FTP server from the FTP client and on occasion from the FTP client to the FTP server. FTP uses out-of-band control, which means it uses a separate connection for control and data. Thus, for the actual file transfer to take place, a different connection is required

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which is called the data stream. Depending on the transfer mode, the process of setting up the data stream is different. Port 21 for control(or program), port 20 for data. In active mode, the FTP client opens a dynamic port, sends the FTP server the dynamic port number on which it is listening over the control stream and waits for a connection from the FTP server. When the FTP server initiates the data connection to the FTP client it binds the source port to port 20 on the FTP server. In order to use active mode, the client sends a PORT command, with the IP and port as argument. The format for the IP and port is "h1,h2,h3,h4,p1,p2". Each field is a decimal representation of 8 bits of the host IP, followed by the chosen data port. For example, a client with an IP of 192.168.0.1, listening on port 49154 for the data connection will send the command "PORT 192,168,0,1,192,2". The port fields should be interpreted as p1Ă—256 + p2 = port, or, in this example, 192Ă—256 + 2 = 49154. In passive mode, the FTP server opens a dynamic port, sends the FTP client the server's IP address to connect to and the port on which it is listening (a 16-bit value broken into a high and low byte, as explained above) over the control stream and waits for a connection from the FTP client. In this case, the FTP client binds the source port of the connection to a dynamic port. To use passive mode, the client sends the PASV command to which the server would reply with something similar to "227 Entering Passive Mode (127,0,0,1,192,52)". The syntax of the IP address and port are the same as for the argument to the PORT command. In extended passive mode, the FTP server operates exactly the same as passive mode, however it only transmits the port number (not broken into high and low bytes) and the client is to assume that it connects to the same IP address that was originally connected to. Extended passive mode was added by RFC 2428 in September 1998. While data is being transferred via the data stream, the control stream sits idle. This can cause problems with large data transfers through firewalls which time out sessions after lengthy periods of idleness. While the file may well be successfully transferred, the control session can be disconnected by the firewall, causing an error to be generated. The FTP protocol supports resuming of interrupted downloads using the REST command. The client passes the number of bytes it has already received as argument to the REST command and restarts the transfer. In some commandline clients for example, there is an often-ignored but valuable command, "reget" (meaning "get again") that will cause an interrupted "get" command to be continued, hopefully to completion, after a communications interruption.

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Resuming uploads is not as easy. Although the FTP protocol supports the APPE command to append data to a file on the server, the client does not know the exact position at which a transfer got interrupted. It has to obtain the size of the file some other way, for example over a directory listing or using the SIZE command. In ASCII mode (see below), resuming transfers can be troublesome if client and server use different end of line characters. The objectives of FTP, as outlined by its RFC, are: 1. To promote sharing of files (computer programs and/or data). 2. To encourage indirect or implicit use of remote computers. 3. To shield a user from variations in file storage systems among different hosts. 4. To transfer data reliably, and efficiently. Telnet Telnet (Telecommunication network) is a network protocol used on the Internet or local area network (LAN) connections. It was developed in 1969 beginning with RFC 15 and standardized as IETF STD 8, one of the first Internet standards. The term telnet also refers to software which implements the client part of the protocol. Telnet clients are available for virtually all platforms. Most network equipment and OSes with a TCP/IP stack support some kind of Telnet service server for their remote configuration (including ones based on Windows NT). Because of security issues with Telnet, its use has waned as it is replaced by the use of SSH for remote access. "To telnet" is also used as a verb meaning to establish or use a Telnet or other interactive TCP connection, as in, "To change your password, telnet to the server and run the passwd command". Most often, a user will be telnetting to a Unix-like server system or a simple network device such as a router. For example, a user might "telnet in from home to check his mail at school". In doing so, he would be using a telnet client to connect from his computer to one of his servers. Once the connection is established, he would then log in with his account information and execute operating system commands remotely on that computer, such as ls or cd. On many systems, the client may also be used to make interactive raw-TCP sessions. It is commonly believed that a telnet session which does not include an IAC (character 255) is

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functionally identical. This is not the case however due to special NVT (Network Virtual Terminal) rules such as the requirement for a bare CR (ASCII 13) to be followed by a NULL (ASCII 0). Protocol details Telnet is a client-server protocol, based on a reliable connection-oriented transport. Typically this protocol used to establish a connection to TCP port 23, where a getty-equivalent program (telnetd) is listening, although Telnet predates TCP/IP and was originally run on NCP. Initially, On March 5th, 1973, a meeting was held at UCLA where "New Telnet" was defined in two NIC The protocol has many extensions, some of which have been adopted as Internet standards. IETF standards STD 27 through STD 32 define various extensions, most of which are extremely common. Other extensions are on the IETF standards track as proposed standards Simple Mail Transfer Protocol Simple Mail Transfer Protocol (SMTP) is a de facto standard for electronic mail (e-mail) transmissions across the Internet. The protocol in widespread use today is also known as extended SMTP (ESMTP) and defined in RFC 5321. While electronic mail server software uses SMTP to send and receive mail messages, user-level client mail applications typically only use SMTP for sending messages to a mail server for relaying. For receiving messages, client applications usually use either the Post Office Protocol (POP) or the Internet Message Access Protocol (IMAP) to access their mail box accounts on a mail server. Description SMTP is a relatively simple, text-based protocol, in which one or more recipients of a message are specified (and in most cases verified to exist) along with the message text and possibly other encoded objects. The message is then transferred to a remote server using a procedure of queries and responses between the client and server. Either an end-user's e-mail client, a.k.a. MUA (Mail User Agent), or a relaying server's MTA (Mail Transport Agents) can act as an SMTP client. An e-mail client knows the outgoing mail SMTP server from its configuration. A relaying server typically determines which SMTP server to connect to by looking up the MX (Mail eXchange) DNS record for each recipient's domain name. Conformant MTAs (not all) fall back to a simple A record in the case of no MX. (Relaying servers can also be configured to use a smart host.)

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The SMTP client initiates a TCP connection to server's port 25 (unless overridden by configuration). It is quite easy to test an SMTP server using the netcat program. SMTP is a "push" protocol that cannot "pull" messages from a remote server on demand. To retrieve messages only on demand, which is the most common requirement on a single-user computer, a mail client must use POP3 or IMAP. Another SMTP server can trigger a delivery in SMTP using ETRN. It is possible to receive mail by running an SMTP server. POP3 became popular when single-user computers connected to the Internet only intermittently; SMTP is more suitable for a machine permanently connected to the Internet. A simple aid to memory is "Send Mail To People." Hypertext Transfer Protocol Hypertext Transfer Protocol (HTTP) is a communications protocol for the transfer of information on the Internet. Its use for retrieving inter-linked text documents (hypertext) led to the establishment of the World Wide Web. HTTP development was coordinated by the World Wide Web Consortium and the Internet Engineering Task Force (IETF), culminating in the publication of a series of Request for Comments (RFCs), most notably RFC 2616 (June 1999), which defines HTTP/1.1, the version of HTTP in common use. HTTP is a request/response standard between a client and a server. A client is the end-user, the server is the web site. The client making a HTTP request - using a web browser, spider, or other end-user tool - is referred to as the user agent. The responding server - which stores or creates resources such as HTML files and images - is called the origin server. In between the user agent and origin server may be several intermediaries, such as proxies, gateways, and tunnels. HTTP is not constrained to using TCP/IP and its supporting layers, although this is its most popular application on the Internet. Indeed HTTP can be "implemented on top of any other protocol on the Internet, or on other networks. HTTP only presumes a reliable transport; any protocol that provides such guarantees can be used." Typically, an HTTP client initiates a request. It establishes a Transmission Control Protocol (TCP) connection to a particular port on a host (port 80 by default; see List of TCP and UDP port numbers). An HTTP server listening on that port waits for the client to send a request message. Upon receiving the request, the server sends back a status line, such as "HTTP/1.1 200 OK", and a message of its own, the body of which is perhaps the requested file, an error message, or some other information.

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HTTP uses TCP and not UDP because much data must be sent for a webpage, and TCP provides transmission control, presents the data in order, and provides error correction. See the difference between TCP and UDP. Resources to be accessed by HTTP are identified using Uniform Resource Identifiers (URIs) (or, more specifically, Uniform Resource Locators (URLs)) using the http: or https URI schemes. Request methods HTTP defines eight methods indicating the desired action to be performed on the identified resource. HEAD Asks for the response identical to the one that would correspond to a GET request, but without the response body. This is useful for retrieving meta-information written in response headers, without having to transport the entire content. GET Requests a representation of the specified resource. By far the most common method used on the Web today. Should not be used for operations that cause side-effects (using it for actions in web applications is a common misuse). POST Submits data to be processed (e.g. from an HTML form) to the identified resource. The data is included in the body of the request. This may result in the creation of a new resource or the updates of existing resources or both. PUT Uploads a representation of the specified resource. DELETE Deletes the specified resource. TRACE Echoes back the received request, so that a client can see what intermediate servers are adding or changing in the request. OPTIONS Returns the HTTP methods that the server supports for specified URL. This can be used to check the functionality of a web server by requesting instead of a specific resource.

CONNECT Converts the request connection to a transparent TCP/IP tunnel, usually to facilitate SSLencrypted communication (HTTPS) through an unencrypted HTTP proxy

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HTTP versions HTTP has evolved into multiple, mostly backwards-compatible protocol versions. RFC 2145 describes the use of HTTP version numbers. The client tells in the beginning of the request the version it uses, and the server uses the same or earlier version in the response. HTTP/0.9 (1991) Deprecated. Supports only one command, GET, which does not specify the HTTP version. Does not support headers. Since this version does not support POST, the information a client can pass to the server is limited by the URI length. HTTP/1.0 (May 1996) This is the first protocol revision to specify its version in communications and is still in wide use, especially by proxy servers. HTTP/1.1 (1997-1999) Current version; persistent connections enabled by default and works well with proxies. Also supports request pipelining, allowing multiple requests to be sent at the same time, allowing the server to prepare for the workload and potentially transfer the requested resources more quickly to the client. HTTP/1.2 The initial 1995 working drafts of the document PEP – an Extension Mechanism for HTTP (which proposed the Protocol Extension Protocol, abbreviated PEP) were prepared by the World Wide Web Consortium and submitted to the Internet Engineering Task Force. PEP was originally intended to become a distinguishing feature of HTTP/1.2. In later PEP working drafts, however, the reference to HTTP/1.2 was removed. The experimental RFC 2774, HTTP Extension Framework, largely subsumed PEP. It was published in February 2000. The major changes between HTTP/1.0 and HTTP/1.1 include the way HTTP handles caching; how it optimizes bandwidth and network connections usage, manages error notifications; how it transmits messages over the network; how internet addresses are conserved; and how it maintains security and integrity. HTML USAGE HTML, an initialism of HyperText Markup Language, is the predominant markup language for Web pages. It provides a means to describe the structure of text-based information in a document — by denoting certain text as links, headings, paragraphs, lists, and so on — and to supplement that text with interactive forms, embedded images, and other objects. HTML is written in the form

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of tags, surrounded by angle brackets. HTML can also describe, to some degree, the appearance and semantics of a document, and can include embedded scripting language code (such as JavaScript) which can affect the behavior of Web browsers and other HTML processors. Files and URLs containing HTML often have a .html or .htm filename extension. Web Browser A web browser is a software application which enables a user to display and interact with text, images, videos, music, games and other information typically located on a Web page at a website on the World Wide Web or a local area network. Text and images on a Web page can contain hyperlinks to other Web pages at the same or different website. Web browsers allow a user to quickly and easily access information provided on many Web pages at many websites by traversing these links. Web browsers format HTML information for display, so the appearance of a Web page may differ between browsers. Some of the Web browsers currently available for personal computers include Internet Explorer, Mozilla Firefox, Safari, Netscape, Opera, Avant Browser, Konqueror, Google Chrome, Flock, Arachne, Epiphany, K-Meleon and AOL Explorer. Web browsers are the most commonly used type of HTTP user agent. Although browsers are typically used to access the World Wide Web, they can also be used to access information provided by Web servers in private networks or content in file systems. Protocols and standards Web browsers communicate with Web servers primarily using HTTP (hypertext transfer protocol) to fetch webpages. HTTP allows Web browsers to submit information to Web servers as well as fetch Web pages from them. The most commonly used HTTP is HTTP/1.1, which is fully defined in RFC 2616. HTTP/1.1 has its own required standards that Internet Explorer does not fully support, but most other current-generation Web browsers do. Pages are located by means of a URL (uniform resource locator, RFC 1738 ), which is treated as an address, beginning with http: for HTTP access. Many browsers also support a variety of other URL types and their corresponding protocols, such as gopher: for Gopher (a hierarchical hyperlinking protocol), ftp: for FTP (file transfer protocol), rtsp: for RTSP (real-time streaming protocol), and https: for HTTPS (an SSL encrypted version of HTTP). The file format for a Web page is usually HTML (hyper-text markup language) and is identified in the HTTP protocol using a MIME content type. Most browsers natively support a variety of

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formats in addition to HTML, such as the JPEG, PNG and GIF image formats, and can be extended to support more through the use of plugins. The combination of HTTP content type and URL protocol specification allows Web page designers to embed images, animations, video, sound, and streaming media into a Web page, or to make them accessible through the Web page. Early Web browsers supported only a very simple version of HTML. The rapid development of proprietary Web browsers led to the development of non-standard dialects of HTML, leading to problems with Web interoperability. Modern Web browsers support a combination of standardsand defacto-based HTML and XHTML, which should display in the same way across all browsers. No browser fully supports HTML 4.01, XHTML 1.x or CSS 2.1 yet. Currently many sites are designed using WYSIWYG HTML generation programs such as Adobe Dreamweaver or Microsoft FrontPage. Microsoft FrontPage often generates non-standard HTML by default, hindering the work of the W3C in developing standards, specifically with XHTML and CSS (cascading style sheets, used for page layout). Dreamweaver and other more modern Microsoft HTML development tools such as Microsoft Expression Web and Microsoft Visual Studio conform to the W3C standards. Some of the more popular browsers include additional components to support Usenet news, IRC (Internet relay chat), and e-mail. Protocols supported may include NNTP (network news transfer protocol), SMTP (simple mail transfer protocol), IMAP (Internet message access protocol), and POP (post office protocol). These browsers are often referred to as Internet suites or application suites rather than merely Web browsers.

Common Gateway Interface The Common Gateway Interface (CGI) is a standard protocol for interfacing external application software with an information server, commonly a web server. The task of such an information server is to respond to requests (in the case of web servers, requests from client web browsers) by returning output. Each time a request is received, the server analyzes what the request asks for, and returns the appropriate output. The two simplest ways for the server to do this, are the following: 

if the request identifies a file stored on disk, return the contents of that file;



if the request identifies an executable command and possibly arguments, run the command and return its output

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CGI defines a standard way of doing the second. It defines how information about the server and the request is passed to the command in the form of arguments and environment variables, and how the command can pass back extra information about the output (such as the type) in the form of headers. Implementation From the Web server's point of view, certain locators, e.g. http://www.example.com/wiki.cgi, are defined as corresponding to a program to execute via CGI. When a request for the URL is received, the corresponding program is executed. Data is passed into the program using environment variables. This is in contrast to typical execution, where Command-line arguments are used. In the case of HTTP PUT or POSTs, the user-submitted data is provided to the program via the standard input. Web servers often have a cgi-bin directory at the base of the directory tree to hold executable files called with CGI. The program returns the result to the web server in the form of standard output, prefixed by a header and a blank line. Header format The header is encoded in the same way as a HTTP header and must include the MIME type of the document returned. The headers are generally forwarded with the response back to the user, supplemented by the web server. SUMMARY 

Domain Name System (DNS) is a client-server application that identifies each host on the Internet with a unique user-friendly name.

DNS organizes the name space in a hierarchical structure to decentralize the responsibilities involved in naming.

DNS can be pictured as an inverted hierarchial tree structure with one root node at the top and a maximum of 128 levels.

Each node in the tree has a domain name.

A domain is defined as any subtree of te domain name space.

The name space information is distributed among DNS servers. Each server has jurisdiction over its zone.

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A root server's zone is the entire DNS tree.

A primary server creates, maintains, and updates information about its zone.

A secondary server gets its information from a primary server.

The domain name space in the Internet is divided into three sections: generic domains, country domains, and inverse domains.

There are seven traditional generic labels, each specifying an organization type. Recently some new labels have been added.

Each country domain specifies a country.

The inverse domain finds a domain name for a given IP address. This is called addressto-name resolution.

Name servers, computers that run the DNS server program, are organized in a hierarchy.

The DNS client, called a resolver, maps a name to an address or an address to a name.

In recursive resolution, the client may send its request to multiple servers before getting an answer.

In iterative resolution, the client may send its request to multiple servers before getting an answer.

A fully qualified doman name (FQDN) is a domain name consisting of labels beginning with the host and going back through each level to the root node.

A partially qualified domain name (PQDN) is a domain name that does not include all the levels between the host and the root node.

There are two types of DNS messages: queries and responses.

There are two types of DNS records: question records and resource records.

Dynamic DNS (DDNS) automatically updates the DNS master file.

DNS uses the services of UDP for messages of less than 512 bytes; otherwise, TCP is used.

The protocol that supports email on the Internet is called Simple Mail Transfer Protocol (SMTP).

The UA prepares the message, creates the envelope, and puts the message in the envelope.

The email address consists of two parts: a local address (user mailbox) and a domain name. The form is localname@domainname.

The MTA transfers the email across the Internet.

SMTP uses commands and responses to transfer messages between an MTA client and an MTA server.

The steps in transferring a mail message are connection establishment, message transfer, and connection termination.

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Multipurpose Internet Mail Extension (MIME) is an extension of SMTP that allows the transfer of multimedia and other non-ASCII messages.

Post Office Protocol, version 3 (POP3) and Internet Mail Access Protocol, version 4 (IMAP4) are protocols used by a mail server in conjunction with SMTP to receive and hold email for hosts.

File transfer protocol (FTP) is a TCP/IP client-server application for copying files from one host to another.

FTP requires two connections for data transfer: a control connection and a data connection.

FTP employs ASCII for communication between dissimilar systems.

Prior to the actual transfer of files, the file type, data structure, and transmission mode are defined by the client through the control connection.

Responses are sent from the server to the client during connection establishment.

There are three types of file transfer: o

A file is copied from the server to the client.

o

A file is copied from the client to the server.

o

A list of directories or file names is sent from the server to the client.

Most operating systems provide a user-friendly interface between FTP and the user.

Anonymous FTP provides a method ofr the general public to access files on remote sites.

The Hypertext Transfer Protocol (HTTP) is the main protocol used to access data on the World Wide Web (WWW).

The World Wide Web is a repository of information spread all over the world and linked together.

Hypertext and hypermedia are documents linked to one another through the con-cept of pointers.

Browsers interpret and display a Web document.

A browser consists of a controller, client programs, and interpreters.

A Web document can be classified as static, dynamic, or active.

A static document is one in which the contents are fixed and stored in a server. The client can make no changes in the server document.

Hypertext Markup Language (HTML) is a language used to create static Web pages.

Any browser can read formatting instructions (tags) embedded in an HTML document.

A dynamic Web document is created by a server only at a browser request.

The Common Gateway Interface (CGI) is a standard for creating and handling dynamic Web documents.

A CGI program with its embedded CGI interface tags can be written in a language such as C, C++, shell script, or Perl.

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The server sends the output of the CGI program to the browser.

The output of a CGI program can be text, graphics, binary data, status codes,

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instructions, or an address of a file. 

An active document is a copy of a program retrieved by the client and run at the client site.

Java is a combination of a high-level programming language, a run-time environment, and a class library that allows a programmer to write an active document and a browser to run it.

Java is used to created applets (small application programs).

Java is an object-oriented typed language with a rich library of classes.

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