CCN-module1.pdf

Computer Communication Network
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MODULE 1
SECTION :A,B,C,D
Topic Portion
Introduction
Chapter 1
Data Communications: Components, Representations, Data Flow,
Networks: Physical Structures, Network Types: LAN, WAN, Switching,
Internet
Network Models
Chapter 2
Protocol Layering: Scenarios, Principles, Logical Connections.
TCP/IP Protocol Suite: Layered Architecture, Layers in TCP/IP suite,
Description of layers
Encapsulation and Decapsulation, Addressing, Multiplexing and
Demultiplexing, The OSI Model: OSI Versus TCP/IP
Data-Link Layer
Chapter 11
Introduction: Nodes and Links, Services, Categories’ of link, Sublayers,
Link Layer addressing: Types of addresses, ARP.
Data Link Control (DLC) services: Framing, Flow and Error Control,
Data Link Layer Protocols: Simple Protocol, Stop and Wait protocol,
Piggybacking
TEXT BOOK: Data Communications and Networking, B Forouzan, 5th
Ed, McGrawHill
Education, 2016, ISBN: 1-25-906475-3

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MODULE 1
Data communications
When we communicate, we are sharing information. This sharing can be local or remote.
Between individuals,
local communication usually occurs face to face, while remote communication takes place over
distance.
The term telecommunication, which includes telephony, telegraphy, and television, means
communication at a distance (tele is Greek for “far”). 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. For data communications to occur, the
communicating devices must be part of a communication system made up of a combination of
hardware (physical equipment) and software (programs). The effectiveness of a data
communications system depends on four fundamental characteristics: delivery, accuracy,
timeliness, and jitter.
1. Delivery-The system must deliver data to the correct destination. Data must be received by
the intended device or user and only by that device or user.
2. Accuracy- The system must deliver the data accurately. Data that have been altered in
transmission and left uncorrected are unusable.
3. Timeliness. The system must deliver data in a timely manner. Data delivered late are useless.
In the case of video and audio, timely delivery means delivering data as they are produced, in
the same order that they are produced, and without significant delay. This kind of delivery is
called real-time transmission.
4. Jitter. Jitter refers to the variation in the packet arrival time. It is the uneven delay in the
delivery of audio or video packets. For example, let us assume that video packets are sent every
30 ms. If some of the packets arrive with 30-ms delay and others with 40-ms delay, an uneven
quality in the video is the result.

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Components
A data communications system has five components
1.Message-. The message is the information (data) to be communicated. Popular forms of
information include text, numbers, pictures, audio, and video.
2. Sender-. The sender is the device that sends the data message. It can be a computer,
workstation, telephone handset, video camera, and so on.
3.Receiver-. The receiver is the device that receives the message. It can be a computer,
workstation, telephone handset, television, and so on
4. Transmission medium-. The transmission medium is the physical path by which a message
travels from sender to receiver. Some examples of transmission media include twisted-pair
wire, coaxial cable, fiber-optic cable, and radio waves.
5. Protocol-. A protocol is a set of rules that govern data communications. It represents an
agreement between the communicating devices. Without a protocol, two devices may be
connected but not communicating, just as a person speaking French cannot be understood by a
person who speaks only Japanese.
Data Representation
Information today comes in different forms such as text, numbers, images, audio, and video.
Text -In data communications, text is represented as a bit pattern, a sequence of bits (0s or 1s).
Different sets of bit patterns have been designed to represent text symbols. Each set is called a
code, and the process of representing symbols is called coding. Today, the prevalent coding
system is called Unicode, which uses 32 bits to represent a symbol or character used in any
language in the world. The American Standard Code for Information Interchange (ASCII),
developed some decades ago in the United States, now constitutes the first 127 characters in
Unicode and is also referred to as Basic Latin.

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Numbers- are also represented by bit patterns. However, a code such as ASCII is not used to
represent numbers; the number is directly converted to a binary number to simplify
mathematical operations. Appendix B discusses several different numbering systems. Images
Images- are also represented by bit patterns. In its simplest form, an image is composed of a
matrix of pixels (picture elements), where each pixel is a small dot. The size of the pixel
depends on the resolution.
For example, an image can be divided into 1000 pixels or 10,000 pixels. In the second case,
there is a better representation of the image (better resolution), but more memory is needed
to store the image. After an image is divided into pixels, each pixel is assigned a bit pattern. The
size and the value of the pattern depend on the image. For an image made of only black and-
white dots (e.g., a chessboard), a 1-bit pattern is enough to represent a pixel. If an image is not
made of pure white and pure black pixels, we can increase the size of the bit pattern to include
gray scale. For example, to show four levels of gray scale, we can use 2-bit patterns. A black
pixel can be represented by 00, a dark gray pixel by 01, a light gray pixel by 10, and a white pixel
by 11. There are several methods to represent color images. One method is called RGB, so
called because each color is made of a combination of three primary colors: red, green, and
blue. The intensity of each color is measured, and a bit pattern is assigned toit. Another method
is called YCM, in which a color is made of a combination of three other primary colors: yellow,
cyan, and magenta.
Audio- Audio refers to the recording or broadcasting of sound or music. Audio is by nature
different from text, numbers, or images. It is continuous, not discrete. Even when we use a
microphone to change voice or music to an electric signal, we create a continuous signal.
Video- Video refers to the recording or broadcasting of a picture or movie. Video can either be
produced as a continuous entity (e.g., by a TV camera), or it can be a combination of images,
each a discrete entity, arranged to convey the idea of motion.
Data Flow
Communication between two devices can be simplex, half-duplex, or full-duplex as shown in
Figure
simplex mode- the communication is unidirectional, as on a one-way street. Only one of the
two devices on a link can transmit; the other can only receive (see Figure a).
example-Keyboards and traditional monitors are examples of simplex devices. The keyboard
can only introduce input; the monitor can only accept output.
simplex mode can use the entire capacity of the channel to send data in one direction

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fig: data flow
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 (see Figure
b). The half-duplex mode is like a one-lane road with traffic allowed in both directions. When
cars are traveling in one direction, cars going the other way must wait.
example-Walkie-talkies and CB (citizens band) radios are both half-duplex systems.
The half-duplex mode is used in cases where there is no need for communication in both
directions at the same time; the entire capacity of the channel can be utilized for each
direction.
Full-Duplex- In full-duplex mode (also called duplex), both stations can transmit and receive
simultaneously (see Figure c). The full-duplex mode is like a two-way street with traffic flowing
in both directions at the same time. In full-duplex mode, signals going in one direction share the
capacity of the link with signals going in the other direction.
This sharing can occur in two ways: Either the link must contain two physically separate
transmission paths, one for sending and the other for receiving; or the capacity of the channel
is divided between signals traveling in both directions.
NETWORKS
A network is the interconnection of a set of devices capable of communication. a device can be
a host (or an end system as it is sometimes called) such as a large computer, desktop, laptop,
workstation, cellular phone, or security system.

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A device can also be a connecting device such as a router, which connects the network to other
networks, a switch, which connects devices together, a modem (modulator-demodulator),
which changes the form of data, and so on.
These devices in a network are connected using wired or wireless transmission media such as
cable or air. When we connect two computers at home using a plug-and-play router, we have
created a network, although very small.
Network Criteria
A network must be able to meet a certain number of criteria. The most important of these are
performance, reliability, and security.
Performance- Performance can be measured in many ways, including transit time and response
time. Transit time is the amount of time required for a message to travel from one device to
another. Response time is the elapsed time between an inquiry and a response.
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
Performance is often evaluated by two networking metrics: throughput and delay.
Reliability- network reliability is measured by the frequency of failure, the time it takes a link
to recover from a failure, and the network’s robustness in a catastrophe.
Security- security issues include protecting data from unauthorized access, protecting data
from damage and development, and implementing policies and procedures for recovery from
breaches and data losses.
Physical Structures
Network attributes - Type of Connection and physical topology
Type of Connection
A network is two or more devices connected through links. A link is a communications pathway
that transfers data from one device to another.
There are two possible types of connections:
Point-to-Point -A point-to-point connection provides a dedicated link between two devices.
The entire capacity of the link is reserved for transmission between those two devices. Most

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point-to-point connections use an actual length of wire or cable to connect the two ends, but
other options, such as microwave or satellite links, are also possible (see Figure a).
example-When we change television channels by infrared remote control, we are establishing a
point-to-point connection between the remote control and the television’s control system.
Multipoint A multipoint (also called multidrop) connection is one in which more than two
specific devices share a single link (see Figure b).
fig: Type of connection
a multipoint environment, the capacity of the channel is shared, either spatially or temporally.
If several devices can use the link simultaneously, it is a spatially shared connection. If users
must take turns, it is a timeshared connection.
Physical Topology
The term physical topology refers to the way in which a network is laid out physically. Two or
more devices connect to a link; two or more links form a topology. The topology of a network is
the geometric representation of the relationship of all the links and linking devices (usually
called nodes) to one another.
There are four basic topologies possible: mesh, star, bus, and ring.

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Mesh Topology- In a mesh topology, every device has a dedicated point-to-point link to every
other device. The term dedicated means that the link carries traffic only between the two
devices it connects.
We need n (n – 1) physical links in a fully connected mesh network with n nodes. if each
physical link allows communication in both directions (duplex mode),we need n (n – 1) / 2
duplex-mode links.
To accommodate that many links, every device on the network must have n – 1 input/output
(I/O) ports (see Figure ) to be connected to the other n – 1 stations.
Advantages-
1.Use of dedicated links guarantees that each connection can carry its own data load, thus
eliminating the traffic problems that can occur when links must be shared by multiple devices.
2. A mesh topology is robust. If one link becomes unusable, it does not incapacitate the entire
system.
3. Privacy or security. When every message travels along a dedicated line, only the intended
recipient sees it. Physical boundaries prevent other users from gaining access to messages.
4. Point-to-point links make fault identification and fault isolation easy. Traffic can be routed to
avoid links with suspected problems. This facility enables the network manager to discover the
precise location of the fault and aids in finding its cause and solution.
A fully connected mesh topology (five devices)
example- of a mesh topology is the connection of telephone regional offices in which each
regional office needs to be connected to every other regional office.

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Star Topology
In a star topology, each device has a dedicated point-to-point link only to a central controller,
usually called a hub.
The devices are not directly linked to one another.A star topology does not allow direct traffic
between devices. The controller acts as an exchange: If one device wants to send data to
another, it sends the data to the controller, which then relays the data to the other connected
device
A star topology connecting four stations
Advantages
1. A star topology is less expensive than a mesh topology.
2. Each device needs only one link and one I/O port to connect . This factor makes it easy to
install and reconfigure. Far less cabling needs to be housed, and additions, moves, and
deletions involve only one connection: between that device and the hub.
3. robust. If one link fails, only that link is affected. All other links remain active. This factor also
lends itself to easy fault identification and fault isolation. As long as the hub is working, it can be
used to monitor link problems and bypass defective links.
Disadvantage
1. The dependency of the whole topology on one single point, the hub. If the hub goes down,
the whole system is dead.
2. more cabling is required in a star than in some other topologies (such as ring or bus).
The star topology is used in local-area networks (LANs), High-speed LANs often use a star
topology with a central hub.

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Bus Topology
A bus topology, is multipoint. One long cable acts as a backbone to link all the devices in a network
Nodes are connected to the bus cable by drop lines and taps. A drop line is a connection
running between the device and the main cable.
A tap is a connector that either splices into the main cable or punctures the sheathing of a
cable to create a contact with the metallic core.
As a signal travels along the backbone, some of its energy is transformed into heat. Therefore, it
becomes weaker and weaker as it travels farther and farther. For this reason there is a limit on
the number of taps a bus can support and on the distance between those taps.
Advantages
1. Easy to install.
2. bus uses less cabling than mesh or star topologies. Only the backbone cable stretches
through the entire facility. Each drop line has to reach only as far as the nearest point on the
backbone.
Disadvantages
1. Difficult reconnection and fault isolation.
2. Difficult to add new devices.
3. Signal reflection at the taps can cause degradation in quality. This degradation can be
controlled by limiting the number and spacing of devices connected to the given length of the
cable
4. Adding new devices require modification or replacement of the backbone.
5. a fault or break in the bus cable stops all transmission. The damaged area reflects signals
back in the direction of origin, creating noise in both directions.

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Bus topology was the one of the first topologies used in the design of early local area networks.
Traditional Ethernet LANs can use a bus topology, but they are less popular now .
Ring Topology
In a ring topology, each device has a dedicated point-to-point connection with only the two
devices on either side of it. A signal is passed along the ring in one direction, from device to
device, until it reaches its destination.
Each device in the ring incorporates a repeater. When a device receives a signal intended for
another device, its repeater regenerates the bits and passes them along
fig: A ring topology connecting six stations
Advantages
1. A ring is relatively easy to install and reconfigure. Each device is linked to only its immediate
neighbors (either physically or logically). To add or delete a device requires changing only two
connections.
2. Fault isolation is simplified.
Generally, in a ring a signal is circulating at all times. If one device does not receive a signal
within a specified period, it can issue an alarm. The alarm alerts the network operator to the
problem and its location.
Disadvantage
In a simple ring, a break in the ring (such as a disabled station) can disable the entire network.
This weakness can be solved by using a dual ring or a switch capable of closing off the break.
Ring topology was prevalent when IBM introduced its local-area network, Token Ring. Today,
the need for higher-speed LANs has made this topology less popular.

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NETWORK TYPES
Different types of networks
Local Area Network (LAN)-
fig: An isolated LAN in the past and today
A local area network (LAN) is usually privately owned and connects some hosts in a single
office, building, or campus. Depending on the needs of an organization,
A LAN can be as simple as two PCs and a printer in someone’s home office, or it can extend
throughout a company and include audio and video devices.
Each host in a LAN has an identifier, an address, that uniquely defines the host in the LAN. A
packet sent by a host to another host carries both the source host’s and the destination host’s
addresses.

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In the past, all hosts in a network were connected through a common cable, which meant that a
packet sent from one host to another was received by all hosts. The intended recipient kept the
packet; the others dropped the packet.
Today, most LANs use a smart connecting switch, which is able to recognize the destination
address of the packet and guide the packet to its destination without sending it to all other
hosts. The switch alleviates the traffic in the LAN and allows more than one pair to
communicate with each other at the same time if there is no common source and destination
among them.
Wide Area Network
A wide area network (WAN) is also an interconnection of devices capable of communication.
However, it. We see two distinct examples of WANs today: point-to-point WANs and switched
WANs.
Point-to-Point WAN
A point-to-point WAN is a network that connects two communicating devices through a
transmission media (cable or air).
fig: Point-to-Point WAN
Switched WAN
A switched WAN is a network with more than two ends. A switched WAN, is used in the
backbone of global communication today. We can say that a switched WAN is a combination of
several point-to-point WANs that are connected by switches..

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fig: A switched WAN
LAN VS WAN
1. A LAN is normally limited in size, spanning
an office, a building, or a campus.
2. A LAN interconnects hosts
3. A LAN is normally privately owned by the
organization that uses it
1. A WAN has a wider geographical span,
spanning a town, a state, a country, or even
the world
2. WAN interconnects connecting devices such
as switches, routers, or modems
3. a WAN is normally created and run by
communication companies and leased by an
organization that uses it.
Internetwork
Today, it is very rare to see a LAN or a WAN in isolation; they are connected to one another.
When two or more networks are connected, they make an internetwork, or internet.
example- Assume that an organization has two offices, one on the east coast and the other on
the west coast. Each office has a LAN that allows all employees in the office to communicate
with each other. To make the communication between employees at different offices possible,
the management leases a point-to-point dedicated WAN from a service provider, such as a
telephone company, and connects the two LANs. Now the company has an internetwork, or a
private internet (with lowercase i). Communication between offices is now possible.

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fig: An network made of two LAN and point-to-point dedicated WAN
When a host in the west coast office sends a message to another host in the same office, the
router blocks the message, but the switch directs the message to the destination. On the other
hand, when a host on the west coast sends a message to a host on the east coast, router R1
routes the packet to router R2, and the packet reaches the destination. Figure shows another
internet with several LANs and WANs connected. One of the WANs is a switched WAN with four
switches.
fig: A heterogeneous network made of four WANs and three LANs

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Switching
An internet is a switched network in which a switch connects at least two links together. A
switch needs to forward data from a network to another network when required. The two most
common types of switched networks are circuit-switched and packet-switched networks.
Circuit-Switched Network
In a circuit-switched network, a dedicated connection, called a circuit, is always available
between the two end systems; the switch can only make it active or inactive(continuous
communication between two telephone). FIG shows a very simple switched network that
connects four telephones to each end. We have used telephone sets instead of computers as
an end system because circuit switching was very common in telephone networks in the past,
fig : Circuit-Switched Network
The thick line connecting two switches is a high-capacity communication line that can handle
four voice communications at the same time; the capacity can be shared between all pairs of
telephone sets. The switches used in this example have forwarding tasks but no storing
capability.
Let us look at two cases.
In the first case, all telephone sets are busy; four people at one site are talking with four
people at the other site; the capacity of the thick line is fully used.
In the second case, only one telephone set at one side is connected to a telephone set at the
other side; only one-fourth of the capacity of the thick line is used. This means that a circuit-
switched network is efficient only when it is working at its full capacity; most of the time, it is
inefficient because it is working at partial capacity.

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The reason to make the capacity of the thick line four times the capacity of each voice line is
that we do not want communication to fail when all telephone sets at one side want to be
connected with all telephone sets at the other side
Packet-Switched Network
In a computer network, the communication between the two computers is done in blocks of
data called packets
This allows switches to function for both storing and forwarding because a packet is an
independent entity that can be stored and sent later. Fig shows a small packet-switched
network that connects four computers at one site to four computers at the other site.
fig: Packet-Switched Network
A router in a packet-switched network has a queue that can store and forward the packet.
Example- Now assume that the capacity of the thick line is only twice the capacity of the data
line connecting the computers to the routers.
If only two computers (one at each site) need to communicate with each other, there is no
waiting for the packets. However, if packets arrive at one router when the thick line is already
working at its full capacity, the packets should be stored and forwarded in the order they
arrived. The two simple examples show that a packet-switched network is more efficient than a
circuit switched network, but the packets may encounter some delays.
The Internet
An internet (note the lowercase i) is two or more networks that can communicate with each
other. The most notable internet is called the Internet (uppercase I ), and is composed of
thousands of interconnected networks.

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Figure 1.15 shows a conceptual (not geographical) view of the Internet. The figure shows the
Internet as several backbones, provider networks, and customer networks. At the top level, the
backbones are large networks owned by some communication companies such as Sprint,
Verizon (MCI), AT&T, and NTT. The backbone networks are connected through some complex
switching systems, called peering points. At the second level, there are smaller networks, called
provider networks, that use the services of the backbones for a fee. The provider networks are
connected to backbones and sometimes to other provider networks.
fig: The internet today
The customer networks are networks at the edge of the Internet that actually use the services
provided by the Internet. They pay fees to provider networks for receiving services. Backbones
and provider networks are also called Internet Service Providers (ISPs). The backbones are often
referred to as international ISPs; the provider networks are often referred to as national or
regional ISP

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Network Models
Protocol Layering
Protocol defines the rules that both the sender and receiver and all intermediate devices need
to follow to be able to communicate effectively.
When communication is simple, we may need only one simple protocol; when the
communication is complex, we may need to divide the task between different layers, in which
case we need a protocol at each layer, or protocol layering.
Let us develop two simple scenarios to better understand the need for protocol layering.
Scenarios
First Scenario
In the first scenario, communication is so simple that it can occur in only one layer. Assume
Maria and Ann are neighbors with a lot of common ideas. Communication between Maria and
Ann takes place in one layer, face to face, in the same language, as shown in Figure
fig: single layer protocol
Second Scenario
In the second scenario, we assume that Ann is offered a higher-level position in her company,
but needs to move to another branch located in a city very far from Maria. The two friends still
want to continue their communication and exchange ideas because they have come up with an
innovative project to start a new business when they both retire. They decide to continue their
conversation using regular mail through the post office. However, they do not want their ideas
to be revealed by other people if the letters are intercepted. They agree on an
encryption/decryption technique. The sender of the letter encrypts it to make it unreadable by
an intruder; the receiver of the letter decrypts it to get the original letter.

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Now we can say that the communication between Maria and Ann takes place in three layers,
as shown in Figure . We assume that Ann and Maria each have three machines (or robots) that
can perform the task at each layer.
fig: A three layer protocol
Assume that Maria sends the first letter to Ann. Maria talks to the machine at the third layer as
though the machine is Ann and is listening to her. The third layer machine listens to what Maria
says and creates the plaintext (a letter in English), which is passed to the second layer machine.
The second layer machine takes the plaintext, encrypts it, and creates the ciphertext, which is
passed to the first layer machine. The first layer machine, presumably a robot, takes the
ciphertext, puts it in an envelope, adds the sender and receiver addresses, and mails it
At Ann’s side, the first layer machine picks up the letter from Ann’s mail box, recognizing the
letter from Maria by the sender address. The machine takes out the ciphertext from the
envelope and delivers it to the second layer machine. The second layer machine decrypts the
message, creates the plaintext, and passes the plaintext to the third-layer machine. The third
layer machine takes the plaintext and reads it as though Maria is speaking.
Need for protocol layering
1) Protocol layering enables us to divide a complex task into several smaller and simpler tasks.

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For example, from fig, we could have used only one machine to do the job of all three
machines. However, if the encryption/ decryption done by the machine is not enough to
protect their secrecy, they would have to change the whole machine. In the present situation,
they need to change only the second layer machine; the other two can remain the same. This is
referred to as modularity. Modularity in this case means independent layers.
2) A layer (module) can be defined as a black box with inputs and outputs, without concern
about how inputs are changed to outputs. If two machines provide the same outputs when
given the same inputs, they can replace each other.
For example, Ann and Maria can buy the second layer machine from two different
manufacturers. As long as the two machines create the same ciphertext from the same
plaintext and vice versa, they do the job.
advantages
1) Protocol layering allows to separate the services from the implementation. Lower layer give
the services to the upper layer; we don’t care about how the layer is implemented.
For example, Maria may decide not to buy the machine (robot) for the first layer; she can do
the job herself. As long as Maria can do the tasks provided by the first layer, in both directions,
the communication system works.
2) Protocol layering in the Internet, is that communication does not always use only two end
systems; there are intermediate systems that need only some layers, but not all layers. If we did
not use protocol layering, we would have to make each intermediate system as complex as the
end systems, which makes the whole system more expensive.
Principles of Protocol Layering
First Principle The first principle dictates that if we want bidirectional communication, we need
to make each layer so that it is able to perform two opposite tasks, one in each direction.
For example, the third layer task is to listen (in one direction) and talk (in the other direction).
The second layer needs to be able to encrypt and decrypt. The first layer needs to send and
receive mail.
Second Principle The second principle that we need to follow in protocol layering is that the
two objects under each layer at both sites should be identical.

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For example, the object under layer 3 at both sites should be a plaintext letter. The object
under layer 2 at both sites should be a cipher text letter. The object under layer 1 at both sites
should be a piece of mail.
Logical Connections
After following the above two principles, we can think about logical connection between each
layer as shown in Figure . This means that we have layer-to-layer communication. Maria and
Ann can think that there is a logical (imaginary) connection at each layer through which they
can send the object created from that layer. We will see that the concept of logical connection
will help us better understand the task of layering we encounter in data communication and
networking.
fig: Logical connection between peer layer
TCP/IP PROTOCOL SUITE
TCP/IP is a protocol suite (a set of protocols organized in different layers) used in the Internet
today. It is a hierarchical protocol made up of interactive modules, each of which provides a
specific functionality. The term hierarchical means that each upper level protocol is supported
by the services provided by one or more lower level protocols. The original TCP/IP protocol
suite was defined as four software layers built upon the hardware. Today, however, TCP/IP is
thought of as a five-layer model. Figure shows both configurations.
Layered Architecture
To show how the layers in the TCP/IP protocol suite are involved in communication between
two hosts, we assume that we want to use the suite in a small internet made up of three LANs

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(links), each with a link-layer switch. We also assume that the links are connected by one
router, as shown in Figure
fig: layers in TCP/IP protocol suite
fig : Communication through an internet

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Assume that computer A communicates with computer B. As the figure shows, five
communicating devices in this communication: source host (computer A), the link-layer switch
in link 1, the router, the link-layer switch in link 2, and the destination host (computer B).
The source host needs to create a message in the application layer and send it down the layers
so that it is physically sent to the destination host. The destination host needs to receive the
communication at the physical layer and then deliver it through the other layers to the
application layer
The router is involved in only three layers; there is no transport or application layer in a router.
Although a router is always involved in one network layer, it is involved in n combinations of
link and physical layers in which n is the number of links the router is connected to. The reason
is that each link may use its own data-link or physical protocol.
For example, in the above figure, the router is involved in three links, but the message sent
from source A to destination B is involved in two links. Each link may be using different link-
layer and physical-layer protocols; the router needs to receive a packet from link 1 based on
one pair of protocols and deliver it to link 2 based on another pair of protocols.
A link-layer switch in a link, however, is involved only in two layers, data-link and physical.
Although each switch in the above figure has two different connections, the connections are in
the same link, which uses only one set of protocols. This means that, unlike a router, a link-
layer switch is involved only in one data-link and one physical layer.
Layers in the TCP/IP Protocol Suite
To better understand the duties of each layer, we need to think about the logical connections
between layers.

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fig: Figure shows logical connections in our simple internet.
Using logical connections makes it easier to think about the duty of each layer. As the figure
shows, the duty of the application, transport, and network layers is end-to-end. However, the
duty of the data-link and physical layers is hop-to-hop, in which a hop is a host or router.
In other words, the domain of duty of the top three layers is the internet, and the domain of
duty of the two lower layers is the link.
Another way of thinking of the logical connections is to think about the data unit created from
each layer. In the top three layers, the data unit (packets) should not be changed by any router
or link-layer switch. In the bottom two layers, the packet created by the host is changed only by
the routers, not by the link-layer switches.
Fig shows the second principle discussed previously for protocol layering. We show the identical
objects below each layer related to each device.
fig: identical objects in the TCP/IP protocol suite
Note that, although the logical connection at the network layer is between the two hosts, we
can only say that identical objects exist between two hops in this case because a router may
fragment the packet at the network layer and send more packets than received .Note that the
link between two hops does not change the object.

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Description of Each Layer
Physical Layer
Physical layer is responsible for carrying individual bits in a frame across the link. Although the
physical layer is the lowest level in the TCP/IP protocol suite, the communication between two
devices at the physical layer is still a logical communication because there is another, hidden
layer, the transmission media, under the physical layer.
Two devices are connected by a transmission medium (cable or air). Transmission medium does
not carry bits, it carries electrical or optical signals. So the bits received in a frame from the
data-link layer are transformed and sent through the transmission media, but we can think that
the logical unit between two physical layers in two devices is a bit. There are several protocols
that transform a bit to a signal.
The physical layer of TCP/IP describes hardware standards such as IEEE 802.3, the specification
for Ethernet network media, and RS-232, the specification for standard pin connectors.
The following are the main responsibilities of the physical layer
Definition of Hardware Specifications, Encoding and Signaling, Data Transmission and
Reception,Topology and Physical Network Design
Data-link Layer
Internet is made up of several links (LANs and WANs) connected by routers. The data-link layer
is responsible for taking the datagram and moving it across the link.(node to node
communication)
The link can be a wired LAN with a link-layer switch, a wireless LAN, a wired WAN, or a wireless
WAN. We can also have different protocols used with any link type.
In each case, the data-link layer is responsible for moving the packet through the link. TCP/IP
does not define any specific protocol for the data-link layer. It supports all the standard and
proprietary protocols. The data-link layer takes a datagram and encapsulates it in a packet
called a frame.
Each link-layer protocol provide a different service like framing, Flow control, Error control
and congestion control.

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Network Layer
The network layer is responsible for creating a connection between the source computer and
the destination computer. The communication at the network layer is host-to-host. However,
since there can be several routers from the source to the destination, the routers in the path
are responsible for choosing the best route for each packet.
The network layer is responsible packetizing and routing and forwarding the packet through
possible routes. others services are error and flow control, congestion control.
The network layer in the Internet includes the main protocol, Internet Protocol (IP), that
defines the format of the packet, called a datagram at the network layer. IP also defines the
format and the structure of addresses used in this layer.
IP is also responsible for routing a packet from its source to its destination, which is achieved by
each router forwarding the datagram to the next router in its path.
IP is a connectionless protocol that provides no flow control, no error control, and no
congestion control services. This means that if any of these services is required for an
application, the application should rely only on the transport-layer protocol.
The network layer also includes unicast (one-to-one) and multicast (one-to-many) routing
protocols. A routing protocol does not take part in routing (it is the responsibility of IP), but it
creates forwarding tables for routers to help them in the routing process. The network layer
also has some auxiliary protocols that help IP in its delivery and routing tasks.
The Internet Control Message Protocol (ICMP) helps IP to report some problems when routing a
packet. The Internet Group Management Protocol (IGMP) is another protocol that helps IP in
multitasking. The Dynamic Host Configuration Protocol (DHCP) helps IP to get the network-layer
address for a host. The Address Resolution Protocol (ARP) is a protocol that helps IP to find the
link-layer address of a host or a router when its network-layer address is given.
Transport Layer
The logical connection at the transport layer is also end-to-end. The transport layer at the
source host gets the message from the application layer, encapsulates it in a transport layer
packet (called a segment or a user datagram in different protocols) and sends it, through the
logical (imaginary) connection, to the transport layer at the destination host.
The transport layer is responsible for giving services to the application layer: to get a message
from an application program running on the source host and deliver it to the corresponding
application program on the destination host.(process to process coommunication)

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There are more than one protocol in the transport layer, which means that each application
program can use the protocol that best matches its requirement. There are a few transport-
layer protocols in the Internet, each designed for some specific task.
The main protocol, Transmission Control Protocol (TCP), is a connection-oriented protocol that
first establishes a logical connection between transport layers at two hosts before transferring
data. It creates a logical pipe between two TCPs for transferring a stream of bytes. TCP provides
flow control (matching the sending data rate of the source host with the receiving data rate of
the destination host to prevent overwhelming the destination), error control (to guarantee that
the segments arrive at the destination without error and resending the corrupted ones), and
congestion control to reduce the loss of segments due to congestion in the network.
User Datagram Protocol (UDP), is a connectionless protocol that transmits user datagrams
without first creating a logical connection. In UDP, each user datagram is an independent entity
without being related to the previous or the next one (the meaning of the term connectionless).
UDP is a simple protocol that does not provide flow, error, or congestion control.
Its simplicity, which means small overhead, is attractive to an application program that needs
to send short messages and cannot afford the retransmission of the packets involved in TCP,
when a packet is corrupted or lost.
A new protocol, Stream Control Transmission Protocol (SCTP) is designed to respond to new
applications that are emerging in the multimedia.
Application Layer
As Figure shows, the logical connection between the two application layers is end to-end. The
two application layers exchange messages between each other as though there were a bridge
between the two layers. However, communication is done through all the layers.
Communication at the application layer is between two processes (two programs running at
this layer). To communicate, a process sends a request to the other process and receives a
response. Process-to-process communication is the duty of the application layer.
The application layer in the Internet includes many predefined protocols.
1) The Hypertext Transfer Protocol (HTTP) is a vehicle for accessing the World Wide Web
(WWW).
2) The Simple Mail Transfer Protocol (SMTP) is the main protocol used in electronic mail (e-
mail) service.

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3) The File Transfer Protocol (FTP) is used for transferring files from one host to another.
4)The Terminal Network (TELNET) and Secure Shell (SSH) are used for accessing a site remotely.
5) The Simple Network Management Protocol (SNMP) is used by an administrator to manage
the Internet at global and local levels.
6) The Domain Name System (DNS) is used by other protocols to find the network-layer address
of a computer.
7) The Internet Group Management Protocol (IGMP) is used to collect membership in a group.
Encapsulation and Decapsulation
One of the important concepts in protocol layering in the Internet is encapsulation/
decapsulation. Figure shows this concept for the small internet
fig: encapsulation/ decapsulation
We have not shown the layers for the link-layer switches because no encapsulation/
decapsulation occurs in this device. Figure show the encapsulation in the source host,
decapsulation in the destination host, and encapsulation and decapsulation in the router.
Encapsulation at the Source Host
At the source, we have only encapsulation.

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1. At the application layer, the data to be exchanged is referred to as a message. A message
normally does not contain any header or trailer, but if it does, we refer to the whole as the
message. The message is passed to the transport layer.
2. The transport layer takes the message as the payload, the load that the transport layer
should take care of. It adds the transport layer header to the payload, which contains the
identifiers of the source and destination application programs that want to communicate plus
some more information that is needed for the end-to end delivery of the message, such as
information needed for flow, error control, or congestion control. The result is the transport-
layer packet, which is called the segment (in TCP) and the user datagram (in UDP). The
transport layer then passes the packet to the network layer.
3. The network layer takes the transport-layer packet as data or payload and adds its own
header to the payload. The header contains the addresses of the source and destination hosts
and some more information used for error checking of the header, fragmentation information,
and so on. The result is the network-layer packet, called a datagram. The network layer then
passes the packet to the data-link layer.
4. The data-link layer takes the network-layer packet as data or payload and adds its own
header, which contains the link-layer addresses of the host or the next hop (the router). The
result is the link-layer packet, which is called a frame. The frame is passed to the physical layer
for transmission.
Decapsulation and Encapsulation at the Router
At the router, we have both decapsulation and encapsulation because the router is connected
to two or more links.
1. After the set of bits are delivered to the data-link layer, this layer decapsulates the datagram
from the frame and passes it to the network layer.
2. The network layer only inspects the source and destination addresses in the datagram
header and consults its forwarding table to find the next hop to which the datagram is to be
delivered. The contents of the datagram should not be changed by the network layer in the
router unless there is a need to fragment the datagram if it is too big to be passed through the
next link. The datagram is then passed to the data-link layer of the next link.
3. The data-link layer of the next link encapsulates the datagram in a frame and passes it to the
physical layer for transmission.

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Decapsulation at the Destination Host
At the destination host, each layer only decapsulates the packet received, removes the payload,
and delivers the payload to the next-higher layer protocol until the message reaches the
application layer. It is necessary to say that decapsulation in the host involves error checking
Addressing
we have logical communication between pairs of layers in this model. Any communication that
involves two parties needs two addresses: source address and destination address. Although it
looks as if we need five pairs of addresses, one pair per layer, we normally have only four
because the physical layer does not need addresses; the unit of data exchange at the physical
layer is a bit, which definitely cannot have an address.
Figure 2.9 shows the addressing at each layer. At the application layer, we normally use names
to define the site that provides services, such as someorg.com, or the e-mail address, such as
somebody@coldmail.com.
fig: Addressing in the TCP/IP Protocol suite
At the transport layer, addresses are called port numbers, and these define the application-
layer programs at the source and destination. Port numbers are local addresses that distinguish
between several programs running at the same time.
At the network-layer, the addresses are global, with the whole Internet as the scope. A
network-layer address uniquely defines the connection of a device to the Internet.

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The link-layer addresses, sometimes called MAC addresses, are locally defined addresses, each
of which defines a specific host or router in a network (LAN or WAN).
Multiplexing and Demultiplexing
TCP/IP protocol suite uses several protocols at some layers, we have multiplexing at the source
and demultiplexing at the destination.
Multiplexing means that a protocol at a layer can encapsulate a packet from several next-
higher layer protocols (one at a time); demultiplexing means that a protocol can decapsulate
and deliver a packet to several next-higher layer protocols (one at a time). Figure shows the
concept of multiplexing and demultiplexing at the three upper layers.
fig: multiplexing and demultiplexing
To be able to multiplex and demultiplex, a protocol needs to have a field in its header to
identify to which protocol the encapsulated packets belong.
At the transport layer, either UDP or TCP can accept a message from several application-layer
protocols.
At the network layer, IP can accept a segment from TCP or a user datagram from UDP. IP can
also accept a packet from other protocols such as ICMP, IGMP, and so on.
At the data-link layer, a frame may carry the payload coming from IP or other protocols such as
ARP .
THE OSI MODEL
An ISO standard that covers all aspects of network communications is the Open Systems
Interconnection (OSI) model. It was first introduced in the late 1970s. An open system is a set of

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protocols that allows any two different systems to communicate regardless of their underlying
architecture.
The purpose of the OSI model is to show how to facilitate communication between different
systems without requiring changes to the logic of the underlying hardware and software. The
OSI model is not a protocol; it is a model for understanding and designing a network
architecture that is flexible, robust, and interoperable. The OSI model was intended to be the
basis for the creation of the protocols in the OSI stack. The OSI model is a layered framework
for the design of network systems that allows communication between all types of computer
systems.
It consists of seven separate but related layers, each of which defines a part of the process of
moving information across a network
fig: OSI model
OSI versus TCP/IP
When we compare the two models, we find that two layers, session and presentation, are
missing from the TCP/IP protocol suite. These two layers were not added to the TCP/IP protocol
suite after the publication of the OSI model. The application layer in the suite is usually
considered to be the combination of three layers in the OSI model, as shown in Figure .
Two reasons were mentioned for this decision. First, TCP/IP has more than one transport-layer
protocol. Some of the functionalities of the session layer are available in some of the transport-
layer protocols.

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Second, the application layer is not only one piece of software. Many applications can be
developed at this layer. If some of the functionalities mentioned in the session and
presentation layers are needed for a particular application, they can be included in the
development of that piece of software.
fig:TCP/IP and OSI model

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DATA-LINK LAYER
INTRODUCTION
The Internet is a combination of networks glued together by connecting devices (routers or
switches). If a packet is to travel from a host to another host, it needs to pass through these
networks.. Communication at the data-link layer is made up of five separate logical connections
between the data-link layers in the path.
Figure 9.1
fig
fig: communication at the data-link layer
The data-link layer at Alice’s computer communicates with the data-link layer at router R2. The
data-link layer at router R2 communicates with the data-link layer at router R4,
and so on. Finally, the data-link layer at router R7 communicates with the data-link layer at
Bob’s computer. Only one data-link layer is involved at the source or the destination, but two
data-link layers are involved at each router. The reason is that Alice’s and Bob’s computers are

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each connected to a single network, but each router takes input from one network and sends
output to another network. Note that although switches are also involved in the data-link-layer
communication, for simplicity we have not shown them in the figure.
Nodes and Links
Communication at the data-link layer is node-to-node. A data unit from one point in the
Internet needs to pass through many networks (LANs and WANs) to reach another point.
Theses LANs and WANs are connected by routers. It is customary to refer to the two end hosts
and the routers as nodes and the networks in between as links. Figure shows a simple
representation of links and nodes when the path of the data unit is only six nodes.
fig: links and nodes
The first node is the source host; the last node is the destination host. The other four nodes are
four routers. The first, the third, and the fifth links represent the three LANs; the second and
the fourth links represent the two WANs.
Services
The data-link layer is located between the physical and the network layers. The data link layer
provides services to the network layer; it receives services from the physical layer.
Services provided by the data-link layer.
The duty scope of the data-link layer is node-to-node. When a packet is travelling in the
Internet, the data-link layer of a node (host or router) is responsible for delivering a datagram
to the next node in the path. For this purpose, the data-link layer of the sending node needs to
encapsulate the datagram received from the network in a frame, and the data-link layer of the
receiving node needs to decapsulate the datagram from the frame. In other words, the data-

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link layer of the source host needs only to encapsulate, the data-link layer of the destination
host needs to decapsulate, but each intermediate node needs to both encapsulate and
decapsulate.
One may ask why we need encapsulation and decapsulation at each intermediate node. The
reason is that each link may be using a different protocol with a different frame format. Even if
one link and the next are using the same protocol, encapsulation and decapsulation are needed
because the link-layer addresses are normally different.
Analogy :-may help in this case. Assume a person needs to travel from her home to her friend’s
home in another city. The traveller can use three transportation tools. She can take a taxi to go
to the train station in her own city, then travel on the train from her own city to the city where
her friend lives, and finally reach her friend’s home using another taxi. Here we have a source
node, a destination node, and two intermediate nodes. The traveller needs to get into the taxi
at the source node, get out of the taxi and get into the train at the first intermediate node (train
station in the city where she lives), get out of the train and get into another taxi at the second
intermediate node (train station in the city where her friend lives), and finally get out of the taxi
when she arrives at her destination. A kind of encapsulation occurs at the source node,
encapsulation and decapsulation occur at the intermediate nodes, and decapsulation occurs at
the destination node. Our traveller is the same, but she uses three transporting tools to reach
the destination. Figure shows the encapsulation and decapsulation at the data-link layer.
For simplicity, we have assumed that we have only one router between the source and
destination. The datagram received by the data-link layer of the source host is encapsulated in
a frame. The frame is logically transported from the source host to the router. The frame is
decapsulated at the data-link layer of the router and encapsulated at another frame. The new
frame is logically transported from the router to the destination host. Note that, although we
have shown only two data-link layers at the router, the router actually has three data-link layers
because it is connected to three physical links.

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fig: communication with only three nodes
Framing
The first service provided by the data-link layer is framing. The data-link layer at each node
needs to encapsulate the datagram (packet received from the network layer) in a frame before
sending it to the next node. The node also needs to decapsulate the datagram from the frame
received on the logical channel. Although we have shown only a header for a frame, frame may
have both a header and a trailer. Different data-link layers have different formats for framing.
Flow Control
If the rate of produced frames is higher than the rate of consumed frames, frames at the
receiving end need to be buffered while waiting to be consumed (processed). Definitely, we
cannot have an unlimited buffer size at the receiving side. We have two choices.
The first choice is to let the receiving data-link layer drop the frames if its buffer is full. The
second choice is to let the receiving data-link layer send a feedback to the sending data-link
layer to ask it to stop or slow down.
Different data-link-layer protocols use different strategies for flow control. Since flow control
also occurs at the transport layer, with a higher degree of importance.
Error Control
At the sending node, a frame in a data-link layer needs to be changed to bits, transformed to
electromagnetic signals, and transmitted through the transmission media. At the receiving
node, electromagnetic signals are received, transformed to bits, and put together to create a
frame. Since electromagnetic signals are susceptible to error, a frame is susceptible to error.

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The error needs first to be detected. After detection, it needs to be either corrected at the
receiver node or discarded and retransmitted by the sending node. Since error detection and
correction is an issue in every layer (node-to node or host-to-host).
Congestion Control
Although a link may be congested with frames, which may result in frame loss, most data-link-
layer protocols do not directly use a congestion control to alleviate congestion, although some
wide-area networks do. In general, congestion control is considered an issue in the network
layer or the transport layer because of its end-to-end nature.
Two Categories of Links
In a point-to-point link, the link is dedicated to the two devices; in a broadcast link, the link is
shared between several pairs of devices. For example, when two friends use the traditional
home phones to chat, they are using a point-to-point link; when the same two friends use their
cellular phones, they are using a broadcast link (the air is shared among many cell phone users).
Two Sublayers
The data-link layer is divided into two sublayers: data link control (DLC) and media access
control (MAC). LAN protocols actually use the same strategy. The data link control sublayer
deals with all issues common to both point-to-point and broadcast links; the media access
control sublayer deals only with issues specific to broadcast links. In other words, we separate
these two types of links at the data-link layer, as shown in fig
fig: dividing the data link layer into two sublayers
LINK-LAYER ADDRESSING
In a connectionless internetwork such as the Internet we cannot make a datagram reach its
destination using only IP addresses. The reason is that each datagram in the Internet, from the
same source host to the same destination host, may take a different path. The source and

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destination IP addresses define the two ends but cannot define which links the datagram
should pass through. We need to remember that the IP addresses in a datagram should not be
changed. If the destination IP address in a datagram changes, the packet never reaches its
destination; if the source IP address in a datagram changes, the destination host or a router can
never communicate with the source if a response needs to be sent back or an error needs to be
reported back to the source (ICMP ).
The above discussion shows that we need another addressing mechanism in a connectionless
internetwork: the link-layer addresses of the two nodes. A link-layer address is sometimes
called a link address, sometimes a physical address, and sometimes a MAC address. Since a
link is controlled at the data-link layer, the addresses need to belong to the data-link layer.
When a datagram passes from the network layer to the data-link layer, the datagram will be
encapsulated in a frame and two data-link addresses are added to the frame header. These two
addresses are changed every time the frame moves from one link to another. Figure
demonstrates the concept in a small internet.
fig: IP address and link layer addresses in a small internet
above Figure shows three links and two routers and also have only two hosts: Alice (source)
and Bob (destination). For each host, two addresses, the IP addresses (N) and the link-layer
addresses (L) are shown. Note that a router has as many pairs of addresses as the number of
links the router is connected to. We have shown three frames, one in each link. Each frame

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carries the same datagram with the same source and destination addresses (N1 and N8), but
the link-layer addresses of the frame change from link to link.
In link 1, the link-layer addresses are L1 and L2. In link 2, they are L4 and L5. In link 3, they are
L7 and L8. Note that the IP addresses and the link-layer addresses are not in the same order.
For IP addresses, the source address comes before the destination address; for link-layer
addresses, the destination address comes before the source. The datagrams and frames are
designed in this way, and we follow the design. We may raise several questions:
❑ The IP address of a router does not appear in any datagram sent from a source to a
destination, why do we need to assign IP addresses to routers? The answer is that in some
protocols a router may act as a sender or receiver of a datagram. For example, in routing
protocols a router is a sender or a receiver of a message. The communications in these
protocols are between routers.
❑ Why do we need more than one IP address in a router, one for each interface? The answer
is that an interface is a connection of a router to a link. We will see that an IP address defines a
point in the Internet at which a device is connected. A router with n interfaces is connected to
the Internet at n points. This is the situation of a house at the corner of a street with two gates;
each gate has the address related to the corresponding street.
❑ How are the source and destination IP addresses in a packet determined? The answer is that
the host should know its own IP address, which becomes the source IP address in the packet.
the application layer uses the services of DNS to find the destination address of the packet and
passes it to the network layer to be inserted in the packet.
❑ How are the source and destination link-layer addresses determined for each link? Again,
each hop (router or host) should know its own link-layer address, The destination link-layer
address is determined by using the Address Resolution Protocol.
❑ What is the size of link-layer addresses? The answer is that it depends on the protocol used
by the link. Although we have only one IP protocol for the whole Internet, we may be using
different data-link protocols in different links.
different link-layer protocols.
Three Types of addresses
Some link-layer protocols define three types of addresses: unicast, multicast, and broadcast.

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Unicast Address: Each host or each interface of a router is assigned a unicast address.
Unicasting means one-to-one communication. A frame with a unicast address destination is
destined only for one entity in the link.
Example: The unicast link-layer addresses in the most common LAN, Ethernet, are 48 bits (six
bytes) that are presented as 12 hexadecimal digits separated by colons; for example, the
following is a link-layer address of a computer.
A2:34:45:11:92:F1
Multicast Address: Some link-layer protocols define multicast addresses. Multicasting means
one-to-many communication. However, the jurisdiction is local (inside the link).
Example: the multicast link-layer addresses in the most common LAN, Ethernet, are 48 bits (six
bytes) that are presented as 12 hexadecimal digits separated by colons. The second digit,
however, needs to be an even number in hexadecimal. The following shows a multicast
address: A3:34:45:11:92:F1
Broadcast Address: Some link-layer protocols define a broadcast address. Broadcasting means
one-to-all communication. A frame with a destination broadcast address is sent to all entities in
the link.
Example : the broadcast link-layer addresses in the most common LAN, Ethernet, are 48 bits, all
1s, that are presented as 12 hexadecimal digits separated by colons. The following shows a
broadcast address: FF:FF:FF:FF:FF:FF
Address Resolution Protocol (ARP)
Anytime a node has an IP datagram to send to another node in a link, it has the IP address of
the receiving node. The source host knows the IP address of the default router. Each router
except the last one in the path gets the IP address of the next router by using its forwarding
table. The last router knows the IP address of the destination host. However, the IP address of
the next node is not helpful in moving a frame through a link; we need the link-layer address of
the next node. This is the time when the Address Resolution Protocol (ARP) becomes helpful.
The ARP protocol is one of the auxiliary protocols defined in the network layer, as shown in
Figure . It belongs to the network layer, it maps an IP address to a logical-link address. ARP
accepts an IP address from the IP protocol, maps the address to the corresponding link-layer
address, and passes it to the data-link layer.

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fig: Position of ARP in TCP/IP protocol suite
Anytime a host or a router needs to find the link-layer address of another host or router in its
network, it sends an ARP request packet. The packet includes the link-layer and IP addresses of
the sender and the IP address of the receiver. Because the sender does not know the link-layer
address of the receiver, the query is broadcast over the link using the link-layer broadcast
address.
fig: ARP operation

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Every host or router on the network receives and processes the ARP request packet, but only
the intended recipient recognizes its IP address and sends back an ARP response packet. The
response packet contains the recipient’s IP and link-layer addresses.
The packet is unicast directly to the node that sent the request packet. In Figure (a) the system
on the left (A) has a packet that needs to be delivered to another system (B) with IP address N2.
System A needs to pass the packet to its data-link layer for the actual delivery, but it does not
know the physical address of the recipient. It uses the services of ARP by asking the ARP
protocol to send a broadcast ARP request packet to ask for the physical address of a system
with an IP address of N2. This packet is received by every system on the physical network, but
only system B will answer it, as shown in Figure (b). System B sends an ARP reply packet that
includes its physical address. Now system A can send all the packets it has for this destination
using the physical address it received.
Caching
Let us assume that there are 20 systems connected to the network (link): system A, system B,
and 18 other systems. We also assume that system A has 10 datagrams to send to system B in
one second.
a. Without using ARP, system A needs to send 10 broadcast frames. Each of the 18 other
systems need to receive the frames, decapsulate the frames, remove the datagram and pass it
to their network-layer to find out the datagrams do not belong to them.This means processing
and discarding 180 broadcast frames.
b. Using ARP, system A needs to send only one broadcast frame. Each of the 18 other systems
need to receive the frames, decapsulate the frames, remove the ARP message and pass the
message to their ARP protocol to find that the frame must be discarded. This means processing
and discarding only 18 (instead of 180) broadcast frames. After system B responds with its own
data-link address, system A can store the link-layer address in its cache memory. The rest of the
nine frames are only unicast. Since processing broadcast frames is expensive (time consuming),
the first method is preferable.
Packet Format
Figure shows the format of an ARP packet.
The hardware type field - defines the type of the link-layer protocol; Ethernet given the type 1.
The protocol type field - defines the network-layer protocol: IPv4 protocol is (0x0800).
The source hardware and source protocol addresses are variable-length fields defining the link-
layer and network-layer addresses of the sender.

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The destination hardware address and destination protocol address fields define the receiver
link-layer and network-layer addresses.
An ARP packet is encapsulated directly into a data-link frame. The frame needs to have a field
to show that the payload belongs to the ARP and not to the network-layer datagram
fig: ARP packet
Example : A host with IP address N1 and MAC address L1 has a packet to send to another host
with IP address N2 and physical address L2 (which is unknown to the first host). The two hosts
are on the same network. Figure shows the ARP request and response messages

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DATA LINK CONTROL
DLC SERVICES
The data link control (DLC) deals with procedures for communication between two adjacent
nodes—node-to-node communication—no matter whether the link is dedicated or broadcast.
Data link control functions include framing and flow and error control.
Framing Data
Transmission in the physical layer means moving bits in the form of a signal from the source to
the destination. The physical layer provides bit synchronization to ensure that the sender and
receiver use the same bit durations and timing.
The data-link layer, on the other hand, needs to pack bits into frames, so that each frame is
distinguishable from another.
Framing in the data-link layer separates a message from one source to a destination by adding a
sender address and a destination address. The destination address defines where the packet is
to go; the sender address helps the recipient acknowledge the receipt.
Although the whole message could be packed in one frame, that is not normally done. One
reason is that a frame can be very large, making flow and error control very inefficient. When a
message is carried in one very large frame, even a single-bit error would require the
retransmission of the whole frame. When a message is divided into smaller frames, a single-bit
error affects only that small frame.
Frame Size
Frames can be of fixed or variable size.
Fixed-size framing: there is no need for defining the boundaries of the frames; the size itself
can be used as a delimiter. An example of this type of framing is the (Asynchronous transfer
mode)ATM WAN, which uses frames of fixed size called cells.
Variable-size framing: prevalent in local-area networks. In variable-size framing, we need a
way to define the end of one frame and the beginning of the next.
Two approaches were used for this purpose: a character-oriented approach and a bit-oriented
approach.

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Character-Oriented Framing
In character-oriented (or byte-oriented) framing, data to be carried are 8-bit characters from a
coding system such as ASCII. The header, which normally carries the source and destination
addresses and other control information, and the trailer, which carries error detection
redundant bits, are also multiples of 8 bits.
To separate one frame from the next, an 8-bit (1-byte) flag is added at the beginning and the
end of a frame. The flag, composed of protocol-dependent special characters, signals the start
or end of a frame. Figure shows the format of a frame in a character-oriented protocol.
fig: A frame in a character-oriented protocol
Character-oriented framing was popular when only text was exchanged by the data-link layers.
The flag could be selected to be any character not used for text communication. Now, however,
we send other types of information such as graphs, audio, and video;
Any character used for the flag could also be part of the information. If this happens, the
receiver, when it encounters this pattern in the middle of the data, thinks it has reached the
end of the frame.
To fix this problem, a byte-stuffing strategy was added to character-oriented framing. In byte
stuffing (or character stuffing), a special byte is added to the data section of the frame when
there is a character with the same pattern as the flag. The data section is stuffed with an extra
byte. This byte is usually called the escape character (ESC) and has a predefined bit pattern.
Whenever the receiver encounters the ESC character, it removes it from the data section and

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treats the next character as data, not as a delimiting flag. Figure shows the situation.
fig: Byte stuffing and unstuffing
Byte stuffing by the escape character allows the presence of the flag in the data section of the
frame, but it creates another problem. What happens if the text contains one or more escape
characters followed by a byte with the same pattern as the flag?
The receiver removes the escape character, but keeps the next byte, which is incorrectly
interpreted as the end of the frame. To solve this problem, the escape characters that are part
of the text must also be marked by another escape character. In other words, if the escape
character is part of the text, an extra one is added to show that the second one is part of the
text.
Character-oriented protocols present another problem in data communications. The universal
coding systems in use today, such as Unicode, have 16-bit and 32-bit characters that conflict
with 8-bit characters. We can say that, in general, the tendency is moving toward the bit-
oriented protocols that we discuss next.
Bit-Oriented Framing
In bit-oriented framing, the data section of a frame is a sequence of bits to be interpreted by
the upper layer as text, graphic, audio, video, and so on. However, in addition to headers (and
possible trailers), we still need a delimiter to separate one frame from the other. Most
protocols use a special 8-bit pattern flag, 01111110, as the delimiter to define the beginning
and the end of the frame, as shown in fig

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fig: A frame in bit-oriented protocol
If the flag pattern appears in the data, need to somehow inform the receiver that this is not the
end of the frame. We do this by stuffing 1 single bit (instead of 1 byte) to prevent the pattern
from looking like a flag. The strategy is called bit stuffing. In bit stuffing, if a 0 and five
consecutive 1 bits are encountered, an extra 0 is added. This extra stuffed bit is eventually
removed from the data by the receiver. Note that the extra bit is added after one 0 followed by
five 1s regardless of the value of the next bit. This guarantees that the flag field sequence does
not inadvertently appear in the frame.
Bit stuffing is the process of adding one extra 0 whenever five consecutive 1s follow a 0 in the
data, so that the receiver does not mistake the pattern 0111110 for a flag.
Figure shows bit stuffing at the sender and bit removal at the receiver. Note that even if we
have a 0 after five 1s, we still stuff a 0. The 0 will be removed by the receiver. This means that if
the flag like pattern 01111110 appears in the data, it will change to 011111010 (stuffed) and is
not mistaken for a flag by the receiver. The real flag 01111110 is not stuffed by the sender and
is recognized by the receiver.
fig: Bit stuffing and unstuffing
Flow and Error Control
One of the responsibilities of the data-link control sublayer is flow and error control at the data-
link layer.

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Flow Control Whenever an entity produces items and another entity consumes them, there
should be a balance between production and consumption rates. If the items are produced
faster than they can be consumed, the consumer can be overwhelmed and may need to discard
some items. If the items are produced more slowly than they can be consumed, the consumer
must wait, and the system becomes less efficient. Flow control is related to the first issue. We
need to prevent losing the data items at the consumer site.
fig: Flow control at the data link layer
The figure shows that the data-link layer at the sending node tries to push frames toward the
data-link layer at the receiving node. If the receiving node cannot process and deliver the
packet to its network at the same rate that the frames arrive, it becomes overwhelmed with
frames. Flow control in this case can be feedback from the receiving node to the sending node
to stop or slow down pushing frames.
Buffers
Although flow control can be implemented in several ways, one of the solutions is normally to
use two buffers; one at the sending data-link layer and the other at the receiving data-link
layer. A buffer is a set of memory locations that can hold packets at the sender and receiver.
The flow control communication can occur by sending signals from the consumer to the
producer. When the buffer of the receiving data-link layer is full, it informs the sending data-
link layer to stop pushing frames.
Error Control
Since the underlying technology at the physical layer is not fully reliable, we need to implement
error control at the data-link layer to prevent the receiving node from delivering corrupted
packets to its network layer. Error control at the data-link layer is normally very simple and
implemented using one of the following two methods. In both methods, a CRC is added to the
frame header by the sender and checked by the receiver.
❑ In the first method, if the frame is corrupted, it is silently discarded; if it is not corrupted,
the packet is delivered to the network layer. This method is used mostly in wired LANs such as
Ethernet.

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❑ In the second method, if the frame is corrupted, it is silently discarded; if it is not corrupted,
an acknowledgment is sent (for the purpose of both flow and error control) to the sender.
Combination of Flow and Error Control
Flow and error control can be combined. In a simple situation, the acknowledgment that is sent
for flow control can also be used for error control to tell the sender the packet has arrived
uncorrupted. The lack of acknowledgment means that there is a problem in the sent frame. We
show this situation when we discuss some simple protocols in the next section. A frame that
carries an acknowledgment is normally called an ACK to distinguish it from the data frame.
Connectionless and Connection-Oriented
A DLC protocol can be either connectionless or connection-oriented..
Connectionless Protocol
In a connectionless protocol, frames are sent from one node to the next without any
relationship between the frames; each frame is independent. Note that the term
connectionless here does not mean that there is no physical connection (transmission medium)
between the nodes; it means that there is no connection between frames. The frames are not
numbered and there is no sense of ordering. Most of the data-link protocols for LANs are
connectionless protocols.
Connection-Oriented Protocol
In a connection-oriented protocol, a logical connection should first be established between the
two nodes (setup phase). After all frames that are somehow related to each other are
transmitted (transfer phase), the logical connection is terminated (teardown phase). In this type
of communication, the frames are numbered and sent in order. If they are not received in
order, the receiver needs to wait until all frames belonging to the same set are received and
then deliver them in order to the network layer. Connection oriented protocols are rare in
wired LANs, but we can see them in some point-to-point protocols, some wireless LANs, and
some WANs.
DATA-LINK LAYER PROTOCOLS
Traditionally four protocols have been defined for the data-link layer to deal with flow and
error control: Simple, Stop-and-Wait, Go-Back-N, and Selective-Repeat.
The behavior of a data-link-layer protocol can be better shown as a finite state machine (FSM).
An FSM is thought of as a machine with a finite number of states. The figure shows a machine

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with three states. There are only three possible events and three possible actions. The machine
starts in state I. If event 1 occurs, the machine performs actions 1 and 2 and moves to state II.
When the machine is in state II, two events may occur. If event 1 occurs, the machine performs
action 3 and remains in the same state, state II. If event 3 occurs, the machine performs no
action, but move to state I.
fig: Connectionless and connection oriented service represented as FSMs
Simple Protocol
First protocol is a simple protocol with neither flow nor error control. We assume that the
receiver can immediately handle any frame it receives. In other words, the receiver can never
be overwhelmed with incoming frames. Figure shows the layout for this protocol.
fig: Simple Protocol
The data-link layer at the sender gets a packet from its network layer, makes a frame out of it,
and sends the frame. The data-link layer at the receiver receives a frame from the link, extracts
the packet from the frame, and delivers the packet to its network layer. The data-link layers of
the sender and receiver provide transmission services for their network layers.
FSMs The sender site should not send a frame until its network layer has a message to send.
The receiver site cannot deliver a message to its network layer until a frame arrives. We can
show these requirements using two FSMs. Each FSM has only one state, the ready state.

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sending machine remains in the ready state until a request comes from the process in the
network layer. When this event occurs, the sending machine encapsulates the message in a
frame and sends it to the receiving machine.
The receiving machine remains in the ready state until a Frame arrives from the sending
machine. When this event occurs, the receiving machine decapsulates the message out of the
frame and delivers it to the process at the network layer. Figure shows the FSMs for the simple
protocol.
fig: FSMs for the simple protocol
Example
Flow diagram shows an example of communication using this protocol. It is very simple. The
sender sends frames one after another without even thinking about the receiver.
fig: Flow diagram for above example

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Stop-and-Wait Protocol
Our second protocol is called the Stop-and-Wait protocol, which uses both flow and error
control.
In this protocol, the sender sends one frame at a time and waits for an acknowledgment before
sending the next one. To detect corrupted frames, we need to add a CRC to each data frame.
When a frame arrives at the receiver site, it is checked. If its CRC is incorrect, the frame is
corrupted and silently discarded. The silence of the receiver is a signal for the sender that a
frame was either corrupted or lost.
Every time the sender sends a frame, it starts a timer. If an acknowledgment arrives before the
timer expires, the timer is stopped and the sender sends the next frame (if it has one to send).
If the timer expires, the sender resends the previous frame, assuming that the frame was either
lost or corrupted. This means that the sender needs to keep a copy of the frame until its
acknowledgment arrives.
When the corresponding acknowledgment arrives, the sender discards the copy and sends the
next frame if it is ready. Figure shows the outline for the Stop-and-Wait protocol. Note that only
one frame and one acknowledgment can be in the channels at any time.
fig: Stop and Wait protocol

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fig: FSMs for Stop-and-Wait protocol.
FSMs Figure shows the FSMs for primitive Stop-and-Wait protocol.
We describe the sender and receiver states below.
Sender States. The sender is initially in the ready state, but it can move between the ready and
blocking state.
Ready State. When the sender is in this state, it is only waiting for a packet from the network
layer. If a packet comes from the network layer, the sender creates a frame, saves a copy of the
frame, starts the only timer and sends the frame. The sender then moves to the blocking state.
❑ Blocking State. When the sender is in this state, three events can occur:
a. If a time-out occurs, the sender resends the saved copy of the frame and restarts the timer.
b. If a corrupted ACK arrives, it is discarded.
c. If an error-free ACK arrives, the sender stops the timer and discards the saved copy of the
frame. It then moves to the ready state.

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Receiver
The receiver is always in the ready state. Two events may occur:
a. If an error-free frame arrives, the message in the frame is delivered to the network layer and
an ACK is sent.
b. If a corrupted frame arrives, the frame is discarded
Example
Flow diagram for this example is shown below.
The first frame is sent and acknowledged. The second frame is sent, but lost. After time-out, it
is resent. The third frame is sent and acknowledged, but the acknowledgment is lost. The frame
is resent. However, there is a problem with this scheme. The network layer at the receiver site
receives two copies of the third packet, which is not right. In the next section, we will see how
we can correct this problem using sequence numbers and acknowledgment numbers.
Sequence and Acknowledgment Numbers
Problem in above Example needs to be addressed and corrected. Duplicate packets, as much as
corrupted packets, need to be avoided. we need to add sequence numbers to the data frames
and acknowledgment numbers to the ACK frames. However, numbering in this case is very
simple. Sequence numbers are 0, 1, 0, 1, 0, 1, . . . ; the acknowledgment numbers can also be 1,
0, 1, 0, 1, 0, … In other words, the sequence numbers start with 0, the acknowledgment
numbers start with 1. An acknowledgment number always defines the sequence number of
the next frame to receive.

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fig: flow diagram for example
example
Figure below shows how adding sequence numbers and acknowledgment numbers can prevent
duplicates. The first frame is sent and acknowledged. The second frame is sent, but lost. After
time-out, it is resent. The third frame is sent and acknowledged, but the acknowledgment is
lost. The frame is resent
FSMs with Sequence and Acknowledgment Numbers
We can change the FSM in Figure(FSM for stop and wait protocol) to include the sequence and
acknowledgment numbers.

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Piggybacking
The two protocols discussed in this section are designed for unidirectional communication, in which
data is flowing only in one direction although the acknowledgment may travel in the other direction.
Protocols have been designed in the past to allow data to flow in both directions. However, to make the
communication more efficient, the data in one direction is piggybacked with the acknowledgment in the
other direction. In other words, when node A is sending data to node B, Node A also acknowledges the
data received from node B. Because piggybacking makes communication at the datalink layer more
complicated, it is not a common practice.

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