Prepared By :T.V.

GOPALA KRISHNA ,ASSOC PROF & HOD

Unit-1
COMPUTER NETWORKS:

Introduction: Network Topologies WAN, LAN, MAN. Reference models- The OSI Reference Model- the
TCP/IP Reference Model - A Comparison of the OSI and TCP/IP Reference Models. Physical Layer –
Introduction to physical layer-Data and Signals, Periodic analog signals, digital signals, transmission
impairment, ,Data rate limits, performance -Introduction to Guided Media- Twisted-pair cable, Coaxial
cable and Fiber optic cable and Unguided media: Wireless-Radio waves, microwaves, infrared.
Network Topologies

Network topologies refer to the arrangement or layout of the nodes (devices) and the communication
links in a computer network. Different topologies have distinct advantages and disadvantages in terms of
cost, scalability, performance, and fault tolerance. Here are some common network topologies:

1. Bus Topology: In a bus topology, all devices are connected to a common communication line
called a bus. Data is transmitted in both directions along the bus, and each device listens for its
specific address. It is a simple and cost-effective topology, but a single point of failure can disrupt
the entire network.

2. Star Topology: In a star topology, all devices are connected to a central device, such as a switch
or hub. The central device acts as a connection point for all the nodes in the network. If one
device or cable fails, only that particular connection is affected, leaving the rest of the network
operational. It offers better performance and scalability than a bus topology but requires more
cabling.

3. Ring Topology: In a ring topology, devices are connected in a closed loop, where each device is
connected to two neighboring devices, forming a ring. Data travels in one direction around the
ring, and each device acts as a repeater to amplify and pass on the signal. Ring topologies are
reliable, but a break in the ring can disrupt the entire network.

4. Mesh Topology: In a mesh topology, every device is connected to every other device in the
network. It offers high redundancy and fault tolerance since multiple paths exist between any two
devices. Mesh topologies can be fully meshed (direct connections between all devices) or
partially meshed (selected devices have direct connections). While it provides robustness, it
requires a significant number of connections, making it expensive and complex to implement.

5. Tree Topology: A tree topology combines characteristics of a bus and a star topology. Devices are
arranged in a hierarchical structure, resembling a tree, where central nodes connect to multiple
nodes, which, in turn, connect to more nodes. It offers scalability and allows for easy expansion
of the network, but a failure of the central node can bring down the entire network.

6. Hybrid Topology: Hybrid topologies are combinations of two or more basic topologies. For
example, a network may have a central star topology that connects to multiple smaller star or bus
1|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN
 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
topologies. Hybrid topologies are often used to achieve specific network requirements or to
accommodate complex network infrastructures.

It's important to note that the choice of network topology depends on various factors, including the
network size, cost constraints, performance requirements, and the desired level of fault tolerance and
scalability.

WAN, LAN, and MAN are terms used to describe different types of computer networks based on their
geographical coverage and the scope of their operation. Here's a brief explanation of each:

1. WAN (Wide Area Network): A Wide Area Network spans a large geographic area, connecting
multiple LANs and other devices over long distances. WANs often utilize public or private
telecommunication networks, such as leased lines, satellite links, or the Internet, to enable
communication between geographically dispersed locations. WANs are commonly used by
organizations to connect their branch offices, remote sites, or to provide internet access to users
across different regions.

2. LAN (Local Area Network): A Local Area Network is a network that covers a limited
geographical area, typically within a building, office, or campus. LANs are used to connect
computers, servers, printers, and other devices within a localized area. They are usually privately
owned and operated by an organization. Ethernet, Wi-Fi, or other high-speed technologies are
commonly used for data transmission within a LAN. LANs facilitate fast and reliable
communication among connected devices and enable resource sharing, such as file sharing and
networked printing.

3. MAN (Metropolitan Area Network): A Metropolitan Area Network covers a larger

geographical area than a LAN but smaller than a WAN, typically spanning a city or metropolitan
area. MANs are designed to interconnect multiple LANs within a city and provide high-speed
connectivity for organizations or institutions located in different parts of the area. MANs may use
a combination of fiber optic cables, microwave links, or other technologies to enable fast data
transmission across the metropolitan region. They are often used by universities, government
agencies, or corporations that require connectivity between multiple locations within a city.

To summarize, WANs are large-scale networks that connect multiple LANs over long distances, LANs
are local networks that cover a limited area like a building or campus, and MANs are intermediate
networks that span a larger geographical area than a LAN but smaller than a WAN, typically covering a
city or metropolitan region.

Reference models:

Reference models provide a conceptual framework for understanding and designing computer networks
and communication protocols. They define the functions and interactions between different network
components and establish a common language for network engineers and researchers. Two well-known
reference models are the OSI model and the TCP/IP model:

2|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

The OSI Reference Model:

The OSI (Open Systems Interconnection) Reference Model is a conceptual framework that
standardizes the functions of a communication system or network into seven distinct layers. It was
developed by the International Organization for Standardization (ISO) in the late 1970s and early
1980s. The OSI model provides a structured approach for designing and implementing network
protocols and serves as a foundation for understanding how different network components interact.
Here's an overview of the seven layers of the OSI model:

1. Application Layer: The topmost layer is responsible for providing services directly to the end-
users or applications. It includes protocols for various network services such as file transfer,
email, remote login, and web browsing. Examples of protocols at this layer include HTTP, FTP,
SMTP, DNS, and DHCP.

2. Presentation Layer: The presentation layer is responsible for data representation and ensures
that information exchanged between systems is properly formatted, encoded, encrypted, or
compressed. It handles tasks such as data translation, data encryption, and data compression.

3. Session Layer: The session layer establishes, manages, and terminates connections between
applications or network devices. It provides mechanisms for session setup, maintenance, and
synchronization. This layer enables two endpoints to establish a session, exchange data, and
manage checkpoints or recovery in case of failures.

4. Transport Layer: The transport layer is responsible for reliable and efficient end-to-end delivery
of data. It ensures that data packets are delivered error-free, in sequence, and without losses or
duplications. The most common protocols at this layer are the Transmission Control Protocol
(TCP), which provides reliable, connection-oriented communication, and the User Datagram
Protocol (UDP), which offers connectionless, unreliable communication.

5. Network Layer: The network layer handles the addressing, routing, and logical connectivity of
data across different networks. It provides the necessary protocols and mechanisms to route data
packets between different networks. The Internet Protocol (IP) is the primary protocol used at this
layer.

6. Data Link Layer: The data link layer provides reliable and error-free communication between
directly connected network nodes. It is responsible for the framing of data packets, error
detection, flow control, and access to the physical medium. Ethernet is a widely used data link
layer protocol.

7. Physical Layer: The physical layer deals with the actual transmission and reception of raw bit
streams over the physical medium. It defines the electrical, mechanical, and functional
specifications of the physical medium, such as cables, connectors, and network interfaces. It
handles tasks like bit encoding, modulation, and signaling.

3|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
8. The OSI model divides the network functionality into discrete layers, allowing for modular
design, interoperability, and the independent development of protocols for each layer. It provides
a common framework for network engineers, allowing them to understand the interactions and
dependencies between different layers of the network stack.

TCP/IP Reference Model:

The TCP/IP (Transmission Control Protocol/Internet Protocol) Reference Model is a conceptual
framework for the design and implementation of network protocols used in the Internet and other
interconnected networks. It provides a set of protocols that allow devices to communicate and exchange
data over networks. The TCP/IP model, also known as the Internet Protocol Suite, consists of four layers.
Here's an overview of the layers in the TCP/IP model:

1. Application Layer: The application layer is the topmost layer in the TCP/IP model and is
responsible for providing network services to end-user applications. It includes protocols such as
HTTP (Hypertext Transfer Protocol), FTP (File Transfer Protocol), SMTP (Simple Mail Transfer
Protocol), DNS (Domain Name System), and DHCP (Dynamic Host Configuration Protocol).

2. Transport Layer: The transport layer is responsible for reliable end-to-end communication
between devices. It ensures the reliable delivery of data by providing services such as
segmentation, reassembly, flow control, and error recovery. The two main protocols at this layer
are TCP (Transmission Control Protocol), which offers reliable, connection-oriented
communication, and UDP (User Datagram Protocol), which provides connectionless, unreliable
communication.

3. Internet Layer: The internet layer, also known as the network layer, handles the logical
addressing, routing, and fragmentation of data packets. It is responsible for delivering data
packets across different networks. The Internet Protocol (IP) is the primary protocol used at this
layer. It enables the identification and addressing of devices on the network and ensures the
proper delivery of data packets from source to destination.

4. Network Interface Layer: The network interface layer, also called the link layer or the network
access layer, is responsible for the physical transmission of data packets over the network
medium. It defines the protocols and hardware necessary to transmit data over the physical
network. Ethernet, Wi-Fi, and other protocols operate at this layer, and they handle tasks such as
frame encoding/decoding, media access control, and physical addressing.

The TCP/IP model is a simpler and more streamlined model compared to the OSI model. It was
developed specifically for the TCP/IP protocol suite used in the Internet. The TCP/IP model's layers are
not as strictly separated as in the OSI model, with some functions overlapping between layers.
Nonetheless, the TCP/IP model provides a foundation for the design and implementation of modern
internet-based networks.

4|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
A Comparison of the OSI and TCP/IP Reference Models:

The OSI (Open Systems Interconnection) and TCP/IP (Transmission Control Protocol/Internet Protocol)
reference models are both conceptual frameworks that provide guidelines for designing and
implementing network protocols. While they share similarities in terms of their layered approach, there
are some key differences between the two models. Here's a comparison of the OSI and TCP/IP reference
models:

1. Number of Layers:

 OSI Model: The OSI model consists of seven layers: Application, Presentation, Session,
Transport, Network, Data Link, and Physical.

 TCP/IP Model: The TCP/IP model comprises four layers: Application, Transport, Internet,
and Network Interface (or Network Access).

2. Layer Functions:

 OSI Model: The OSI model provides a detailed breakdown of network functions, with
each layer having specific responsibilities such as addressing, routing, data representation,
and session management.

 TCP/IP Model: The TCP/IP model combines some of the functions of the OSI layers. For
example, the TCP/IP model's network layer incorporates aspects of the OSI model's
network and data link layers.

3. Development and Adoption:

 OSI Model: The OSI model was developed in the late 1970s and early 1980s. While it is a
comprehensive and well-structured model, it did not gain widespread adoption in practice.

 TCP/IP Model: The TCP/IP model was developed alongside the TCP/IP protocol suite,
which became the foundation for the modern Internet. It gained significant popularity and
became the de facto standard for networking.

4. Flexibility:

 OSI Model: The OSI model was designed to be flexible, allowing for the independent
development of protocols for each layer. This modular design facilitates interoperability
and the ability to replace or update protocols without affecting other layers.

 TCP/IP Model: The TCP/IP model is more rigid, with fewer layers and less strict
separation between functions. The protocols within the TCP/IP suite, such as IP, TCP, and
UDP, are tightly integrated and work together.

5. Practicality:

5|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 OSI Model: While the OSI model provides a comprehensive and theoretical framework, it
is often seen as overly complex and less practical for real-world implementations.

 TCP/IP Model: The TCP/IP model, being closely aligned with the protocols used in the
Internet, is considered more practical and widely adopted for network design and
implementation.

6. Popularity:

 OSI Model: The OSI model, despite not being extensively implemented, still serves as an
important reference and educational tool for understanding network architecture and
protocols.

 TCP/IP Model: The TCP/IP model is widely implemented and forms the basis of the
modern internet, making it the most prevalent reference model in networking.

In summary, the OSI model provides a comprehensive and layered approach to network design, while the
TCP/IP model is a more streamlined and practical model closely tied to the protocols used in the Internet.
The TCP/IP model's simplicity, alignment with real-world implementations, and widespread adoption
have made it the dominant reference model in modern networking.

Physical Layer:

The Physical Layer is the lowest layer of the OSI (Open Systems Interconnection) and TCP/IP reference
models. It deals with the physical transmission of raw bit streams over the physical medium, such as
copper cables, fiber optics, or wireless connections. The main focus of the Physical Layer is to provide
electrical, mechanical, and functional specifications for transmitting and receiving data between network
devices.

Here are some key characteristics and responsibilities of the Physical Layer:

1. Physical Media: The Physical Layer defines the characteristics of the physical media used for
transmitting data. This includes specifications such as the type of cable (e.g., twisted pair, coaxial,
optical fiber), connectors, pinouts, signaling voltages, and transmission rates.

2. Physical Encoding: The Physical Layer is responsible for converting digital data into a physical
signal suitable for transmission over the physical medium. This may involve encoding schemes
like Non-Return-to-Zero (NRZ), Manchester encoding, or Differential Manchester encoding.

3. Data Transmission: The Physical Layer manages the transmission of data bits over the physical
medium. It determines the timing, synchronization, and voltage levels for transmitting and
receiving signals. It also handles aspects such as modulation techniques (e.g., amplitude
modulation, frequency modulation) for analog signals.

6|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
4. Signal Integrity: The Physical Layer ensures that the transmitted signal maintains its integrity
and quality during transmission. It deals with issues like signal attenuation, noise, distortion, and
interference that can affect the quality and reliability of the data being transmitted.

5. Bit Synchronization: The Physical Layer establishes and maintains synchronization between the
sender and receiver by defining how bits are grouped into frames and providing mechanisms for
clock synchronization.

6. Physical Addressing: In some network technologies, the Physical Layer may include physical
addressing schemes that identify network devices at the physical level, such as MAC (Media
Access Control) addresses used in Ethernet.

7. Network Interfaces: The Physical Layer defines the specifications and protocols required for
network interfaces, including network interface cards (NICs) and transceivers. These interfaces
connect network devices to the physical medium and ensure proper communication.

The Physical Layer forms the foundation for higher-level layers of the network stack, enabling the
transmission and reception of raw bit streams between network devices. Its responsibilities are primarily
concerned with the electrical, mechanical, and functional aspects of data transmission over the physical
medium.

Data and Signals: In the context of the Physical Layer, "data" refers to the information to be transmitted,
which can be in various forms such as text, numbers, images, or audio. Before data can be transmitted, it
needs to be converted into a suitable form called a "signal" that can be transmitted over the physical
medium.

Signals can be analog or digital, depending on how they represent and transmit data.

Periodic Analog Signals: Analog signals are continuous, varying signals that represent data using
continuous changes in voltage, frequency, or amplitude. Periodic analog signals are those that repeat
themselves over time, following a specific pattern.

Examples of periodic analog signals include sine waves, square waves, and triangle waves. These signals
have specific characteristics such as frequency, amplitude, and phase.

Frequency: Frequency refers to the number of complete cycles of a periodic signal that occur in one
second. It is measured in Hertz (Hz).

Amplitude: Amplitude represents the maximum value of the signal's strength or voltage. It indicates the
magnitude or intensity of the signal.

Phase: Phase describes the relative position of a signal with respect to a reference point in time. It is
usually measured in degrees or radians and determines the alignment of the signal with other signals.

7|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Digital Signals: Digital signals, unlike analog signals, represent data using discrete values or binary
digits (bits), typically 0s and 1s. These signals are characterized by discrete voltage levels or signal states,
where each state represents a bit (0 or 1).

Digital signals are more resilient to noise and interference compared to analog signals, making them
suitable for long-distance transmission and reliable data communication.

In digital signal transmission, data is encoded using modulation techniques such as Amplitude Shift
Keying (ASK), Frequency Shift Keying (FSK), or Phase Shift Keying (PSK). These techniques modify
the digital signal's amplitude, frequency, or phase to represent different binary states.

Digital signals are commonly used in modern communication systems, including computer networks, the
Internet, and telecommunication networks.

The Physical Layer plays a crucial role in transmitting and receiving both analog and digital signals,
ensuring their accurate transmission over the network medium. It establishes the necessary electrical,
mechanical, and functional specifications to enable successful data communication between network
devices.

Transmission Impairment: Transmission impairment refers to the degradation or distortion of a signal

during transmission over a network medium. Various factors can cause transmission impairments,
including attenuation, noise, distortion, and interference. These impairments can affect the quality and
reliability of the transmitted data.

Attenuation: Attenuation is the gradual loss of signal strength as it propagates through the medium. It is
influenced by the distance the signal travels, the characteristics of the medium (e.g., resistance,
capacitance), and the frequency of the signal. Attenuation can result in a weaker signal at the receiver,
leading to errors and reduced signal quality.

Noise: Noise refers to unwanted electrical or electromagnetic signals that interfere with the desired
signal. It can be introduced by external sources such as electromagnetic radiation, electrical devices, or
crosstalk from nearby communication channels. Noise can corrupt the original signal and make it more
difficult to accurately decode at the receiver.

Distortion: Distortion occurs when the shape or characteristics of the signal change during transmission.
It can be caused by factors like frequency-dependent attenuation, signal reflections, or interference.
Distortion can result in signal degradation, causing errors and making it challenging to extract the
original data accurately.

Interference: Interference refers to the presence of unwanted signals that disrupt or interfere with the
desired signal. It can be caused by external sources such as electromagnetic radiation from other devices
or intentional jamming. Interference can lead to signal degradation, increased error rates, and reduced
data integrity.

8|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Data Rate Limits: The data rate, also known as the transmission rate or bit rate, represents the speed at
which data is transmitted over a network medium. It is measured in bits per second (bps) and indicates
the number of bits that can be transmitted in a given time frame.

Several factors can impose limits on the achievable data rate:

Bandwidth: The available bandwidth of the medium determines the maximum data rate that can be
achieved. Bandwidth refers to the range of frequencies that can be effectively transmitted over the
medium. The greater the bandwidth, the higher the potential data rate.

Signal-to-Noise Ratio (SNR): The SNR is the ratio of the strength of the desired signal to the level of
background noise present in the transmission. A higher SNR allows for a higher data rate, as it reduces
the chances of errors and increases the reliability of signal detection.

Channel Capacity: The channel capacity represents the maximum data rate that can be achieved without
error on a given communication channel. It is influenced by factors such as bandwidth, noise level, and
channel characteristics. The channel capacity can be determined using mathematical formulas such as the
Shannon capacity theorem.

Encoding Techniques: Different encoding techniques and modulation schemes can impact the
achievable data rate. Some modulation schemes are more efficient than others in terms of packing more
information into a given bandwidth, allowing for higher data rates.

Performance: The performance of a network is typically evaluated based on several key metrics:

Throughput: Throughput refers to the actual amount of data transmitted per unit of time, taking into
account any transmission errors or overhead. It represents the effective data rate achieved in practical
conditions.

Latency: Latency, also known as delay, is the time it takes for a data packet to travel from the source to
the destination. It includes propagation delay (the time it takes for a signal to travel over the medium) and
processing delays at various network devices. Lower latency is desirable, particularly in real-time
applications like voice and video communication.

Error Rate: The error rate measures the frequency of errors in the received data compared to the original
transmitted data. Lower error rates indicate better performance and data integrity.

Reliability: Reliability refers to the ability of the network to consistently and accurately transmit data
without errors or disruptions. A reliable network minimizes data loss, delays, and interruptions, ensuring
the

Introduction to Guided Media: Guided media refers to the physical media that provide a pathway for
transmitting signals in a network. These media guide and confine the signals along a specific path,
ensuring reliable and controlled transmission. Some commonly used guided media include twisted-pair
cable, coaxial cable, and fiber optic cable.

9|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Twisted-Pair Cable: Twisted-pair cable consists of pairs of insulated copper wires twisted
together. The twisting helps to reduce electromagnetic interference (EMI) from external sources.
Twisted-pair cables are widely used in Ethernet networks and telephone systems. There are two
main types of twisted-pair cables:

 Unshielded Twisted Pair (UTP): UTP cables are commonly used in Ethernet networks.
They have four pairs of twisted wires and are identified by categories such as Cat 5e, Cat
6, or Cat 6a, which indicate the performance and data transmission capabilities of the
cable.

 Shielded Twisted Pair (STP): STP cables have an additional metallic shield around the
twisted pairs, providing better protection against external interference. They are
commonly used in environments with high levels of electromagnetic interference.

2. Coaxial Cable: Coaxial cable consists of a central conductor surrounded by a dielectric insulating
layer, a metallic shield, and an outer protective jacket. The central conductor carries the signal,
while the shield provides protection against external interference. Coaxial cables are commonly
used in cable television (CATV) systems, broadband internet connections, and CCTV systems.
They offer higher bandwidth and better shielding compared to twisted-pair cables.

3. Fiber Optic Cable: Fiber optic cable uses thin strands of glass or plastic fibers to transmit data as
pulses of light. Each fiber consists of a core, which carries the light signals, surrounded by a
cladding layer that reflects the light back into the core. Fiber optic cables offer high bandwidth,
immunity to electromagnetic interference, and long-distance transmission capabilities. They are
widely used in high-speed data networks, telecommunications, and internet backbone
infrastructure.

 Single-mode fiber: Single-mode fiber has a small core diameter and allows only one
mode of light to propagate. It is suitable for long-distance transmissions and high-
bandwidth applications.

 Multi-mode fiber: Multi-mode fiber has a larger core diameter, enabling multiple modes
of light to propagate. It is commonly used for shorter-distance transmissions within
buildings or campuses.

Introduction to Unguided Media: Unguided media, also known as wireless or wireless communication,
refers to the transmission of signals through the air or space without the use of any physical media. It
provides wireless connectivity and mobility in various communication systems. Some examples of
unguided media include:

1. Radio Waves: Radio waves are electromagnetic waves used for wireless communication. They
are commonly used in radio broadcasting, mobile communication (e.g., cellular networks), Wi-Fi
networks, and Bluetooth.

10 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Microwaves: Microwaves are higher-frequency electromagnetic waves than radio waves. They
are used for long-distance point-to-point communication, such as satellite communication and
microwave links.

3. Infrared: Infrared (IR) waves are electromagnetic waves with lower frequencies than visible
light. They are used for short-range communication, such as remote control devices and infrared
data transmission.

4. Light Waves: Light waves, including visible light, can be used for optical wireless
communication. This technology utilizes light signals to transmit data, commonly used in
applications like infrared (IR) communication and Li-Fi (Light Fidelity) for high-speed wireless
data transfer.

Unguided media offer the advantage of wireless connectivity and mobility, allowing devices to
communicate without the need for physical connections. However, they are more susceptible to
environmental factors, interference, and signal attenuation compared to guided media.

Wireless communication relies on various types of electromagnetic waves for transmitting signals
without the need for physical cables or wires. Here are three commonly used types of wireless
communication:

1. Radio Waves: Radio waves are a type of electromagnetic radiation with long wavelengths and
low frequencies. They are widely used for wireless communication, including radio broadcasting,
television broadcasting, mobile communication (e.g., cellular networks), and Wi-Fi networks.
Radio waves can travel long distances and penetrate obstacles, making them suitable for wide-
area coverage.

2. Microwaves: Microwaves have shorter wavelengths and higher frequencies compared to radio
waves. They are commonly used for point-to-point communication over long distances.
Microwaves are employed in microwave links, which transmit data between two fixed points,
such as satellite communication, microwave communication towers, and long-range wireless
backhaul for cellular networks. They are also used in radar systems and microwave ovens.

3. Infrared: Infrared (IR) radiation has wavelengths longer than visible light but shorter than
microwaves. It is often used for short-range wireless communication. Infrared communication is
commonly found in applications such as remote controls for televisions and other electronic
devices. It works by transmitting data in the form of modulated infrared light signals, which are
received and interpreted by the receiving device. Infrared communication is limited to line-of-
sight communication and has a shorter range compared to radio waves and microwaves.

Each of these wireless communication technologies has its own advantages and applications. Radio
waves provide long-range coverage and are suitable for wide-area networks. Microwaves are used for
point-to-point communication over long distances. Infrared is used for short-range communication within
a confined space. The choice of wireless technology depends on factors such as the required range, data
rate, and environmental considerations.

11 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

12 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

Unit-2
COMPUTER NETWORKS:

The Data Link Layer - Services Provided to the Network Layer – Framing – Error Control – Flow Control,
Error Detection and Correction – Error-Correcting Codes – Error Detecting Codes. Elementary Data Link
Protocols- A Utopian Simplex Protocol-A Simplex Stop and Wait Protocol for an Error free channel-A
Simplex Stop and Wait Protocol for a Noisy Channel, Sliding Window Protocols-A One Bit Sliding
Window Protocol-A Protocol Using Go-Back-N- A Protocol Using Selective Repeat.

The Data Link Layer:

The Data Link Layer is the second layer in the OSI (Open Systems Interconnection) reference model and
the TCP/IP protocol suite. It is responsible for providing reliable and error-free transmission of data
frames between adjacent network nodes over a shared communication medium. The primary functions of
the Data Link Layer include framing, physical addressing, error detection, and flow control.

Here are the key responsibilities and functions of the Data Link Layer:

1. Framing: The Data Link Layer breaks the stream of data from the Network Layer into
manageable units called frames. Each frame consists of a header, data payload, and a trailer. The
header contains control information such as synchronization bits, addressing, and error-checking
fields. The trailer typically contains a frame check sequence (FCS) used for error detection.

2. Physical Addressing: The Data Link Layer assigns unique physical addresses, known as Media
Access Control (MAC) addresses, to network interface cards (NICs) connected to the same local
network. MAC addresses are typically assigned by the manufacturer and are globally unique.
They enable the Data Link Layer to identify the source and destination devices within a local
network.

3. Error Detection and Handling: The Data Link Layer includes mechanisms for error detection to
ensure the integrity of transmitted data. This can involve techniques such as cyclic redundancy
check (CRC) or checksum calculations. If errors are detected in a received frame, the Data Link
Layer can request retransmission of the frame or take other appropriate error-handling actions.

4. Flow Control: Flow control mechanisms in the Data Link Layer manage the flow of data
between network nodes to avoid overwhelming the receiving device with more data than it can

1|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
handle. Flow control helps prevent data loss and ensures efficient and reliable communication.
Techniques such as buffering, sliding window protocols, and acknowledgment mechanisms are
used for flow control.

5. Media Access Control (MAC): The Data Link Layer also includes the MAC sublayer, which
handles access to the shared communication medium in network architectures such as Ethernet.
The MAC layer manages how devices contend for the right to transmit data over the shared
medium and resolves potential collisions. Various MAC protocols, such as CSMA/CD (Carrier
Sense Multiple Access with Collision Detection) in Ethernet, are used for efficient medium
access.

The Data Link Layer serves as an interface between the Network Layer and the Physical Layer, providing
reliable data transfer services over the underlying physical medium. It establishes and manages the
logical link between directly connected devices and ensures error-free transmission and proper flow
control. Different data link protocols, such as Ethernet, Point-to-Point Protocol (PPP), and Wi-Fi, operate
at this layer, catering to specific network technologies and requirements.

Services Provided to the Network Layer:

The Data Link Layer provides several services to the Network Layer above it. These services help
facilitate reliable and efficient communication between network nodes. Here are the main services
provided by the Data Link Layer to the Network Layer:

1. Framing: The Data Link Layer encapsulates network layer packets into frames by adding headers
and trailers. This framing process allows the Network Layer to transmit data in discrete units
(frames) over the physical medium. The Data Link Layer is responsible for breaking down the
data received from the Network Layer into manageable frames for transmission.

2. Physical Addressing: The Data Link Layer assigns unique physical addresses, known as MAC
(Media Access Control) addresses, to each network interface card (NIC) connected to the local
network. These addresses are used for identifying the source and destination devices within the
local network. The Data Link Layer uses MAC addresses to determine where to send frames
within the local network.

3. Error Detection and Handling: The Data Link Layer includes mechanisms for error detection to
ensure the integrity of the transmitted frames. It can use techniques like cyclic redundancy check
(CRC) or checksum calculations to detect errors during transmission. If errors are detected in a
received frame, the Data Link Layer can request retransmission of the frame or take other
appropriate error-handling actions.

2|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
4. Flow Control: Flow control mechanisms in the Data Link Layer manage the flow of data
between network nodes. The Data Link Layer ensures that the receiving device can handle the
incoming data without being overwhelmed. Flow control helps prevent data loss and regulates the
rate of data transmission. Techniques like buffering, sliding window protocols, and
acknowledgment mechanisms are used for flow control.

5. Access Control: The Data Link Layer manages access to the shared communication medium in
network architectures like Ethernet. It implements MAC (Media Access Control) protocols that
govern how devices contend for the right to transmit data over the shared medium. These
protocols coordinate access to the medium to avoid collisions and ensure efficient and fair
transmission.

By providing these services, the Data Link Layer shields the Network Layer from the complexities of the
underlying physical medium and ensures reliable, error-free, and efficient transmission of data. It acts as
a bridge between the Network Layer and the Physical Layer, enabling seamless communication between
network nodes.

Framing:
Framing is a fundamental function performed by the Data Link Layer in network communication. It
involves dividing the stream of data received from the Network Layer into manageable units called
frames. These frames serve as discrete units of transmission over the physical medium. Framing provides
a way to delineate the boundaries of data within a transmission, allowing the receiver to correctly
interpret and extract the data.

Here are the key aspects of framing:

1. Frame Structure: A frame consists of a header, data payload, and sometimes a trailer. The
header contains control information necessary for the proper handling and processing of the
frame, such as synchronization bits, addressing information, and error-checking fields. The data
payload carries the actual data from the Network Layer. The trailer typically contains additional
control information, such as a frame check sequence (FCS) used for error detection.

2. Synchronization: Synchronization bits or patterns are added to the frame's header to assist the
receiver in correctly identifying the start and end of each frame. These synchronization patterns
help establish frame boundaries, ensuring that the receiver can properly extract the data and
maintain synchronization with the transmitter.

3. Delimiting: Delimiting techniques are employed to mark the boundaries between frames within
the data stream. These techniques may include special bit patterns or specific control characters
that indicate the beginning and end of each frame. Delimiting ensures that the receiver can
distinguish individual frames when receiving continuous data from the sender.

4. Error Detection: The framing process may include error detection mechanisms, such as cyclic
redundancy check (CRC) or checksum calculations. These mechanisms enable the receiver to

3|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
verify the integrity of the received frame and detect any transmission errors that may have
occurred during transmission.

Framing plays a crucial role in reliable data transmission over the network. It allows the Data Link Layer
to break down the data received from the Network Layer into manageable frames, each with its own
control information. This structure facilitates error detection, synchronization, and proper handling of
data at the receiver's end. Different protocols and technologies may employ various framing techniques
depending on their specific requirements and characteristics.

Error control is an important aspect of data communication to ensure the accuracy and reliability of
transmitted data. It encompasses techniques for both error detection and error correction. Two key
components of error control are flow control and error detection/correction using error-correcting codes
and error-detecting codes.

Flow Control: Flow control mechanisms manage the flow of data between the sender and receiver to
prevent data loss or overwhelming the receiver with more data than it can handle. Flow control ensures
that the sender transmits data at a rate that matches the receiver's capacity. This coordination helps
maintain proper synchronization and efficient data transfer. Flow control techniques include:

1. Stop-and-Wait: The sender sends one frame and waits for acknowledgment from the receiver
before sending the next frame.

2. Sliding Window: The sender can send multiple frames without waiting for individual
acknowledgments. The receiver acknowledges the received frames, allowing the sender to send
new frames within a specified window size.

Error Detection and Correction: Error detection and correction techniques are employed to identify and,
in some cases, correct errors that may occur during data transmission. These techniques help ensure data
integrity and minimize the impact of transmission errors. Two common methods used for error control
are error-correcting codes and error-detecting codes:

1. Error-Correcting Codes: Error-correcting codes add extra bits to the transmitted data to allow
the receiver to detect and correct errors. These codes introduce redundancy into the data, which
enables the receiver to reconstruct the original message even if some errors are present. Popular
error-correcting codes include Hamming codes, Reed-Solomon codes, and Bose-Chaudhuri-
Hocquenghem (BCH) codes.

2. Error-Detecting Codes: Error-detecting codes are used to identify the presence of errors in the
received data but do not provide error correction capabilities. These codes add redundancy to the
data, allowing the receiver to determine if errors occurred during transmission. If an error is
detected, the receiver can request retransmission or take appropriate action. Common error-
detecting codes include parity checks, cyclic redundancy check (CRC), and checksums.

4|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Both error-correcting and error-detecting codes employ mathematical algorithms and techniques to
introduce redundancy and enable error detection and correction. The specific code used depends on
factors such as the error rate, data transmission characteristics, and desired level of reliability.

By implementing flow control and error control mechanisms, data communication systems can ensure
accurate and reliable transmission, even in the presence of errors or varying data transfer rates. These
techniques play a crucial role in maintaining data integrity and overall system performance.

Elementary Data Link Protocols:

Elementary Data Link Protocols are simple and basic protocols that operate at the Data Link Layer of the
OSI model. They provide essential functionalities for reliable data transmission over a communication
channel. Here are two commonly used elementary data link protocols:

1. Stop-and-Wait Protocol: The Stop-and-Wait Protocol is a simple and widely used elementary
protocol that ensures reliable transmission of data between a sender and a receiver. In this
protocol, the sender transmits one frame of data and waits for an acknowledgment (ACK) from
the receiver before sending the next frame. The receiver, upon receiving a frame, sends an ACK
back to the sender to confirm successful receipt. If the sender does not receive the ACK within a
specified time period, it assumes that the frame was lost or damaged and retransmits the frame.

Advantages of the Stop-and-Wait Protocol include its simplicity and ease of implementation. However, it
is not very efficient for high-speed or long-distance communication, as the sender must wait for
acknowledgment after each frame, leading to reduced throughput.

2. Sliding Window Protocol: The Sliding Window Protocol is another elementary data link
protocol that allows the sender to transmit multiple frames before receiving acknowledgments. It
utilizes a sliding window mechanism to control the flow of data between the sender and receiver.

In this protocol, the sender maintains a window of allowed, unacknowledged frames. It sends multiple
frames within the window and waits for acknowledgments from the receiver. The receiver acknowledges
the received frames and indicates the next expected frame it is ready to receive. The sender adjusts the
size of the window based on acknowledgment information, allowing it to send new frames within the
updated window.

The Sliding Window Protocol provides better efficiency compared to the Stop-and-Wait Protocol by
allowing pipelining of frames, increasing throughput. It also supports selective repeat or go-back-n
mechanisms for handling frame retransmissions in case of errors.

These elementary data link protocols serve as building blocks for more complex protocols used in
modern communication networks. They provide reliable data transmission, flow control, and error
handling capabilities, albeit with some limitations. Advanced protocols like High-Level Data Link

5|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Control (HDLC), Point-to-Point Protocol (PPP), and Ethernet are based on these elementary protocols
but offer enhanced features and efficiencies to meet the requirements of various network environments.

In a Utopian Simplex Protocol, which assumes an error-free channel, a simple and straightforward
approach for data transmission is used. The protocol operates in a simplex mode, which means that data
is transmitted in only one direction, and there is no need for acknowledgments or retransmissions.

Here's how the Utopian Simplex Protocol works:

1. Sender Perspective:

 The sender divides the data into fixed-size frames.

 It sequentially transmits each frame to the receiver without any error checking or
acknowledgment.

 After transmitting each frame, the sender waits for a fixed amount of time to ensure that the
receiver has received the frame.

 The sender then proceeds to transmit the next frame in sequence.

2. Receiver Perspective:

 The receiver listens for incoming frames.

 Upon receiving a frame, the receiver assumes that it is error-free since the protocol assumes an
error-free channel.

 The receiver processes the received frame and utilizes the data as required.

 Since the protocol assumes an error-free channel, there is no need for acknowledgment or
retransmission.

It is important to note that the Utopian Simplex Protocol is a highly simplified model that assumes
perfect transmission without any errors or the need for error detection and correction mechanisms. In
reality, communication channels are prone to errors, noise, and disruptions. Therefore, more advanced
protocols incorporating error detection, retransmission, and error correction mechanisms are used to
ensure reliable data transmission in practical scenarios.

The Utopian Simplex Protocol serves as a basic conceptual model to understand the principles of data
transmission but does not address the complexities and challenges associated with real-world
communication channels.

A Simplex Stop and Wait Protocol for a Noisy Channel:

A Simplex Stop and Wait Protocol for a noisy channel is a basic data link protocol that accounts for
potential errors in the communication channel. Unlike the Utopian Simplex Protocol, this protocol

6|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
includes error detection and retransmission mechanisms to ensure reliable data transmission over a noisy
channel. Here's how the Simplex Stop and Wait Protocol for a Noisy Channel works:

1. Sender Perspective:

 The sender divides the data into fixed-size frames.

 It sequentially transmits each frame to the receiver.

 After transmitting each frame, the sender starts a timer and waits for an acknowledgment (ACK)
from the receiver.

 If the sender receives an ACK within the timeout period, it assumes successful transmission and
proceeds to send the next frame.

 If the sender does not receive an ACK within the timeout period, it assumes that the frame was
lost or damaged and retransmits the same frame.

2. Receiver Perspective:

 The receiver listens for incoming frames.

 Upon receiving a frame, it checks for errors using error detection techniques such as checksum or
cyclic redundancy check (CRC).

 If the received frame is error-free, the receiver sends an ACK to the sender indicating successful
receipt.

 If the received frame contains errors, the receiver discards the frame without sending an ACK,
triggering the sender to retransmit the frame.

3. Retransmission:

 If the sender does not receive an ACK within the timeout period, it assumes that the frame was
lost and retransmits the same frame.

 The sender keeps retransmitting the frame until it receives a corresponding ACK from the
receiver or reaches a maximum number of retransmission attempts.

 The receiver discards duplicate frames to avoid processing the same frame multiple times.

This Simplex Stop and Wait Protocol for a Noisy Channel provides a basic mechanism to handle errors in
the communication channel. It ensures that each frame is acknowledged before proceeding to the next
frame and includes retransmission of lost or damaged frames. However, it does not include advanced
error correction techniques, such as forward error correction (FEC), which can recover from errors
without retransmission.

7|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Real-world protocols like HDLC, PPP, or Ethernet incorporate more sophisticated error detection and
correction mechanisms to deal with noisy channels, providing higher reliability and efficiency in data
transmission.

Sliding Window Protocols:

Sliding Window Protocols are widely used data link layer protocols that allow for efficient and reliable
data transmission over a communication channel. These protocols utilize a sliding window mechanism to
control the flow of data between the sender and receiver. They provide advantages such as increased
throughput, improved efficiency, and the ability to transmit multiple frames before receiving
acknowledgments. Here are two commonly used sliding window protocols:

1. Go-Back-N (GBN) Protocol: The Go-Back-N Protocol is a sliding window protocol that offers a
simple and efficient approach to reliable data transmission. In this protocol, the sender is allowed
to transmit multiple frames without waiting for individual acknowledgments. The receiver
acknowledges the successful receipt of frames by sending cumulative acknowledgments. Key
features of the Go-Back-N Protocol include:

 Sender Perspective: The sender maintains a window of unacknowledged frames and continues
sending new frames as long as the window is not full. The sender starts a timer upon sending the
first frame in the window. If an acknowledgment is received within the timer's duration, the
sender advances the window and continues sending new frames. If the timer expires before
receiving an acknowledgment, the sender assumes that one or more frames were lost and
retransmits all the frames in the window.

 Receiver Perspective: The receiver receives frames and acknowledges them by sending
cumulative acknowledgments. The receiver discards out-of-order frames and requests
retransmission for any missing or corrupted frames. The receiver only accepts frames that fall
within the receiver's window, which defines the acceptable sequence numbers.

2. Selective Repeat (SR) Protocol: The Selective Repeat Protocol is another sliding window
protocol that provides improved efficiency compared to the Go-Back-N Protocol. In this protocol,
the sender is allowed to transmit multiple frames without waiting for individual
acknowledgments, similar to the Go-Back-N Protocol. However, the receiver can selectively
acknowledge individual frames rather than just cumulative acknowledgments. Key features of the
Selective Repeat Protocol include:

 Sender Perspective: The sender maintains a window of unacknowledged frames and continues
sending new frames as long as the window is not full. Each frame in the window is individually
timed. If a timeout occurs for a specific frame, only that frame is retransmitted while the other
frames in the window remain unaffected.

 Receiver Perspective: The receiver receives frames and individually acknowledges them. The
receiver discards out-of-order frames but buffers them for later delivery. When a missing frame is

8|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
detected, the receiver requests retransmission of that specific frame rather than all frames in the
window.

Both the Go-Back-N and Selective Repeat Protocols provide reliability and efficient data transmission by
utilizing sliding window mechanisms. However, the Selective Repeat Protocol offers better efficiency by
eliminating unnecessary retransmissions of frames that were successfully received. The choice of
protocol depends on factors such as the desired level of reliability, network conditions, and available
resources.

A One Bit Sliding Window Protocol:

A One Bit Sliding Window Protocol, also known as a One-Bit Go-Back-N Protocol, is a simplified
version of the sliding window protocol that uses a window size of one. This protocol allows the sender to
transmit one frame at a time before waiting for an acknowledgment from the receiver. It is commonly
used for communication over error-prone channels with a limited sequence number space. Here's how the
One Bit Sliding Window Protocol works:

1. Sender Perspective:

 The sender divides the data into fixed-size frames and assigns a sequence number to each frame,
typically using a single bit to represent the sequence number.

 The sender transmits one frame at a time and waits for an acknowledgment (ACK) from the
receiver.

 If the sender receives an ACK within a specified timeout period, it assumes successful
transmission and proceeds to send the next frame.

 If the sender does not receive an ACK within the timeout period, it assumes that the frame was
lost or damaged and retransmits the same frame.

2. Receiver Perspective:

 The receiver listens for incoming frames and checks the sequence number of each received frame.

 If the received frame's sequence number matches the expected sequence number, the receiver
sends an ACK to the sender, indicating successful receipt.

 If the received frame's sequence number does not match the expected sequence number, the
receiver discards the frame without sending an ACK, indicating that the frame was either lost or
duplicated. The receiver does not advance the expected sequence number until the correct frame
is received.

3. Retransmission:

 If the sender does not receive an ACK within the timeout period, it assumes that the frame was
lost and retransmits the same frame with the same sequence number.

9|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 The sender keeps retransmitting the frame until it receives the corresponding ACK or reaches a
maximum number of retransmission attempts.

The One Bit Sliding Window Protocol provides a simple mechanism for error recovery in a limited
sequence number space. It ensures reliable data transmission by retransmitting frames that are not
acknowledged within the timeout period. However, it is a relatively inefficient protocol as it waits for the
acknowledgment of each frame before transmitting the next one.

This protocol is typically used in scenarios where the sequence number space is small, such as low-speed
communication links or simple communication systems with limited resources. In more complex and
high-speed networks, protocols like Go-Back-N or Selective Repeat with larger window sizes are
preferred for improved efficiency and throughput.

A Protocol Using Go-Back-N-

A protocol using the Go-Back-N (GBN) algorithm is a sliding window protocol that provides reliable
data transmission over a communication channel. It allows the sender to transmit multiple frames without
waiting for individual acknowledgments and includes mechanisms for error detection and retransmission
of lost or corrupted frames. Here's how a protocol using the Go-Back-N algorithm works:

1. Sender Perspective:

 The sender divides the data into fixed-size frames and assigns a unique sequence number to each
frame.

 The sender maintains a window of unacknowledged frames, indicating the maximum number of
frames it can transmit without receiving acknowledgments.

 The sender starts a timer when the first frame in the window is sent.

 The sender continuously transmits frames within the window until it reaches the end of the
window or the entire data is transmitted.

 When a frame is sent, the sender stores a copy of the frame and waits for an acknowledgment
(ACK) from the receiver.

2. Receiver Perspective:

 The receiver listens for incoming frames and checks the sequence number of each frame.

 If the received frame's sequence number matches the expected sequence number, the receiver
accepts the frame and sends an ACK back to the sender.

 If the received frame's sequence number does not match the expected sequence number, the
receiver discards the frame and does not send an ACK. This indicates that the frame was either
lost or duplicated.

10 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 The receiver maintains a variable to track the expected sequence number of the next frame it
should receive.

3. Retransmission:

 If the sender does not receive an ACK within a specified timeout period, it assumes that one or
more frames were lost.

 The sender retransmits all the frames in the current window, starting from the oldest
unacknowledged frame.

 The sender resets the timer after retransmitting the frames.

4. Receiver Window:

 The receiver maintains a window that specifies the acceptable range of sequence numbers it can
receive. Any frames outside this window are ignored.

 The receiver acknowledges the highest in-order frame it has received, indicating that all frames
with a lower sequence number have been received successfully.

The Go-Back-N protocol ensures reliable transmission by retransmitting a window of frames when an
acknowledgment is not received within the timeout period. The receiver's window helps the sender to
know which frames need to be retransmitted.

It is important to note that the size of the sender and receiver windows, as well as the timeout period, can
impact the protocol's performance and efficiency. Selecting appropriate values for these parameters
depends on factors such as the channel characteristics, transmission delays, and network conditions.

The Go-Back-N protocol is commonly used in scenarios where there is a possibility of frame loss or
errors, such as in wireless or unreliable communication channels.

A Protocol Using Selective Repeat:

A protocol using the Selective Repeat algorithm is a sliding window protocol that provides reliable data
transmission over a communication channel. It allows the sender to transmit multiple frames without
waiting for individual acknowledgments and includes mechanisms for error detection, selective
retransmission of lost frames, and out-of-order frame handling. Here's how a protocol using the Selective
Repeat algorithm works:

1. Sender Perspective:

 The sender divides the data into fixed-size frames and assigns a unique sequence number to each
frame.

 The sender maintains a window of unacknowledged frames, indicating the maximum number of
frames it can transmit without receiving acknowledgments.

11 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 The sender continuously transmits frames within the window until it reaches the end of the
window or the entire data is transmitted.

 When a frame is sent, the sender stores a copy of the frame and waits for an acknowledgment
(ACK) from the receiver.

2. Receiver Perspective:

 The receiver listens for incoming frames and checks the sequence number of each frame.

 If the received frame's sequence number matches the expected sequence number, the receiver
accepts the frame, sends an ACK back to the sender, and delivers the frame to the upper layers.

 If the received frame's sequence number does not match the expected sequence number, the
receiver checks if it falls within the receiver window.

 If the frame's sequence number is within the receiver window, the receiver stores the frame in a
buffer, updates the expected sequence number, and sends an ACK back to the sender.

 If the frame's sequence number is outside the receiver window, the receiver discards the frame
without sending an ACK. This indicates that the frame is either out-of-order or a duplicate.

3. Retransmission:

 If the sender does not receive an ACK within a specified timeout period for a particular frame, it
assumes that the frame was lost or damaged.

 The sender retransmits only the specific frame that did not receive an ACK, rather than
retransmitting the entire window.

 The sender continues transmitting subsequent frames within the window.

4. Receiver Window:

 The receiver maintains a window that specifies the acceptable range of sequence numbers it can
receive. Any frames outside this window are ignored.

 The receiver acknowledges individual frames, indicating the successful receipt of each frame,
even if they are out-of-order.

The Selective Repeat protocol improves efficiency by eliminating unnecessary retransmissions of frames
that were successfully received. It allows the receiver to handle out-of-order frames by buffering them
until they can be delivered in the correct order.

Similar to the Go-Back-N protocol, the size of the sender and receiver windows, as well as the timeout
period, can impact the protocol's performance and efficiency. Proper configuration of these parameters is
essential for optimal operation based on the channel characteristics and network conditions.

12 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
The Selective Repeat protocol is commonly used in scenarios where errors, losses, or out-of-order frame
delivery can occur, such as in high-speed networks or satellite communications.

13 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

Unit-3
COMPUTER NETWORKS:
The Medium Access Control Sub layer-The Channel Allocation Problem-Static Channel Allocation-
Assumptions for Dynamic Channel Allocation, Multiple Access Protocols-Aloha-Pure aloha- slotted
aloha-Carrier Sense Multiple Access Protocols- Collision-Free Protocols-Limited Contention Protocols.
Wireless LAN Protocols- Ethernet-Classic Ethernet Physical Layer-Classic Ethernet MAC Sub-layer
Protocol-Ethernet Performance-Fast Ethernet- Wireless LANs-The 802.11 Architecture and Protocol
Stack-The 802.11 Physical Layer-The802.11 MAC Sub-layer Protocol- The 805.11 Frame Structure-
Services.

The Medium Access Control Sub layer:

The Medium Access Control (MAC) sublayer is a sublayer of the data link layer in the OSI model. It is
responsible for managing access to the shared communication medium, such as a network channel or a
wireless spectrum, when multiple devices or nodes are competing for access. The MAC sublayer ensures
that data frames are transmitted efficiently and without collisions in a multi-access network environment.
Here are some key aspects of the MAC sublayer:

1. Medium Access Control Methods:

 The MAC sublayer implements various medium access control methods, which determine how
devices access and share the communication medium. Some common MAC methods include:

 Carrier Sense Multiple Access (CSMA): Devices listen for carrier signals on the medium
and transmit only when the medium is idle.

 CSMA/CD (Carrier Sense Multiple Access with Collision Detection): Used in Ethernet
networks, devices listen for carrier signals and check for collisions. If a collision is
detected, devices follow a backoff algorithm and retransmit later.

 CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance): Used in wireless
networks, devices avoid collisions by using techniques like Request to Send (RTS) and
Clear to Send (CTS) handshakes.

 Token Passing: A token is passed among devices, allowing only the device holding the
token to transmit data.

2. Addressing and Frame Control:

 The MAC sublayer adds addressing information to the data frame, including source and
destination MAC addresses. These addresses uniquely identify devices within a local network.

 Frame control information includes control flags, frame type, and other control parameters that
govern the behavior of the data frame.

1|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Frame Synchronization:

 The MAC sublayer ensures that data frames are properly synchronized between the sender and
receiver. This synchronization allows the receiver to identify the start and end of each frame.

4. Flow Control:

 The MAC sublayer may incorporate flow control mechanisms to regulate the flow of data
between the sender and receiver. Flow control prevents data overload and avoids congestion in
the network.

5. Error Detection:

 The MAC sublayer typically includes error detection mechanisms, such as cyclic redundancy
check (CRC), to detect transmission errors and ensure the integrity of data frames.

The MAC sublayer plays a crucial role in managing access to the shared communication medium and
coordinating data transmission between devices. Different network technologies and protocols may
implement specific MAC sublayer functions tailored to their requirements. For example, Ethernet
networks utilize CSMA/CD as the MAC method, while Wi-Fi networks employ CSMA/CA. The MAC
sublayer works in conjunction with the physical layer to enable reliable and efficient communication in a
network.

The Channel Allocation Problem-Static Channel Allocation:

The Channel Allocation Problem refers to the challenge of efficiently allocating communication channels
to multiple users or devices in a shared medium or network. It is a fundamental problem in the design and
management of communication systems, particularly in scenarios where multiple users compete for
limited resources. The goal is to maximize the utilization of channels, minimize interference, and ensure
fair access to resources. The Channel Allocation Problem can be categorized into two main types: Static
Channel Allocation and Dynamic Channel Allocation.

1. Static Channel Allocation: Static Channel Allocation involves pre-assigning channels to users or
devices based on a predetermined plan or assignment scheme. The channel assignments remain
fixed over time and do not change dynamically. Static allocation methods include techniques like
Frequency Division Multiple Access (FDMA), Time Division Multiple Access (TDMA), or Code
Division Multiple Access (CDMA). Static Channel Allocation is suitable for scenarios with stable
traffic patterns, predictable communication requirements, and a fixed number of users.

2. Dynamic Channel Allocation: Dynamic Channel Allocation involves dynamically allocating

channels to users or devices based on their current demands, network conditions, and availability
of resources. Dynamic allocation methods aim to adaptively assign channels to optimize resource
utilization, minimize interference, and accommodate varying communication needs. Dynamic
allocation techniques include Carrier Sense Multiple Access (CSMA), Carrier Sense Multiple
Access with Collision Avoidance (CSMA/CA), or Reservation-based protocols like Dynamic
Reservation Multiple Access (DRMA) or Dynamic Time Division Multiple Access (DTDMA).

2|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Dynamic Channel Allocation is more suitable for dynamic network environments, where traffic
load, user demands, and network conditions vary over time.

The Channel Allocation Problem can be further complicated by factors such as interference, varying
channel conditions, quality of service requirements, fairness considerations, and the presence of multiple
access points or base stations. Various algorithms, protocols, and optimization techniques have been
developed to address the Channel Allocation Problem, aiming to improve channel utilization, mitigate
interference, and provide efficient and fair access to the shared medium.

The choice of channel allocation method depends on the specific requirements, characteristics, and
constraints of the communication system, including the number of users, available channels, traffic
patterns, mobility, interference levels, and QoS considerations. It is an active area of research and
optimization in wireless networks, cellular networks, satellite communication systems, and other multi-
user communication scenarios.

Static Channel Allocation:

Static Channel Allocation is a method of allocating communication channels to users or devices in

advance, where the channel assignments remain fixed over time. It involves dividing the available
communication channels among the users or devices based on a predetermined plan or assignment
scheme. Here are some key points about Static Channel Allocation:

1. Predefined Channel Assignment: In Static Channel Allocation, the channel assignments are
determined before the actual communication starts. Each user or device is assigned a specific
channel or a group of channels for their communication.

2. Fixed Channel Usage: Once the channels are allocated, they remain dedicated to the assigned
users or devices throughout the communication session. The channel assignments do not change
dynamically.

3. Deterministic Access: Static Channel Allocation ensures deterministic access to the channels, as
each user or device knows in advance which channel(s) they are allowed to use. This helps avoid
contention and collisions between users.

4. Allocation Schemes: Various allocation schemes can be used in Static Channel Allocation,
depending on the characteristics and requirements of the communication system. Common
schemes include:

 Frequency Division Multiple Access (FDMA): Channels are divided into non-overlapping
frequency bands, and each user or device is assigned a specific frequency band for
communication.

 Time Division Multiple Access (TDMA): Channels are divided into time slots, and each
user or device is allocated specific time slots for their communication. Users take turns
transmitting during their allocated time slots.

3|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 Code Division Multiple Access (CDMA): Users or devices share the same frequency
band, but each user is assigned a unique code that allows their transmissions to be
distinguished from others.

5. Advantages of Static Channel Allocation:

 Simplicity: Static Channel Allocation is relatively simple to implement, as the channel

assignments are predetermined and do not require dynamic coordination.

 Deterministic Performance: Since channel assignments are fixed, users or devices can
expect consistent performance without contention or randomness.

 Efficient Resource Utilization: With dedicated channels, the allocated resources are fully
utilized by the assigned users or devices.

6. Limitations of Static Channel Allocation:

 Inflexibility: Static Channel Allocation is not adaptable to changes in the communication

environment, such as varying traffic loads or the arrival of new users or devices.

 Limited Scalability: The number of channels is fixed, which can limit the number of users
or devices that can be accommodated in the system.

 Inefficient Utilization: Channels may be underutilized if users or devices do not require

their allocated channels at all times, leading to wasted resources.

Static Channel Allocation is commonly used in scenarios where the number of users or devices is known
in advance, and the communication requirements are relatively stable. It is often employed in traditional
circuit-switched networks or dedicated communication systems where predictable channel access is
required.

However, in dynamic environments with varying traffic patterns or a large number of users, dynamic
channel allocation methods, such as Dynamic Channel Allocation (DCA), may be more suitable for
efficient channel utilization and adaptability to changing communication demands.

Assumptions for Dynamic Channel Allocation:

When considering Dynamic Channel Allocation, several assumptions are made to facilitate the effective
allocation of communication channels to users or devices. These assumptions serve as a foundation for
the design and implementation of dynamic channel allocation methods. Here are some common
assumptions:

1. Dynamic Channel Availability: It is assumed that there are a sufficient number of channels
available for allocation. The availability of channels may vary based on factors such as spectrum
availability, regulatory restrictions, or the presence of other users or systems sharing the same
spectrum.

4|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Dynamic Traffic Patterns: Dynamic Channel Allocation assumes that the communication traffic
patterns of users or devices can vary over time. Users may have different communication needs,
and the demand for channels may fluctuate based on factors like user behavior, application
requirements, or network conditions.

3. Channel Sensing and Access: Users or devices are assumed to have the capability to sense the
availability or occupancy of channels. This sensing can be done using techniques like Carrier
Sense Multiple Access (CSMA) or Clear Channel Assessment (CCA) in wireless networks. Users
can then attempt to access available channels based on sensing results.

4. Dynamic Channel Assignment: Dynamic Channel Allocation assumes that channels can be
dynamically assigned to users or devices based on their current demands and channel availability.
The allocation can be done using centralized control mechanisms, such as a base station or access
point, or through distributed algorithms where users negotiate or contend for available channels.

5. Interference Mitigation: Dynamic Channel Allocation aims to mitigate interference among users
or devices sharing the same channels. Techniques such as power control, frequency hopping, or
adaptive modulation can be employed to minimize interference and optimize channel allocation.

6. Quality of Service (QoS) Considerations: Dynamic Channel Allocation takes into account the
QoS requirements of different users or applications. Channels may be allocated based on
parameters such as bandwidth, latency, reliability, or priority to meet the specific QoS needs of
different users or applications.

7. Scalability: Dynamic Channel Allocation assumes the ability to handle a large number of users or
devices in the system. Efficient algorithms and protocols are designed to ensure that channel
allocation scales well with increasing user population and communication demands.

8. Feedback and Adaptation: Dynamic Channel Allocation assumes the availability of feedback
mechanisms to monitor the performance of allocated channels and adapt the allocation strategy
based on network conditions or user requirements. Feedback can be obtained through
acknowledgments, channel quality indicators, or measurements of channel utilization.

These assumptions enable the design and implementation of dynamic channel allocation methods that can
effectively manage and optimize channel usage in wireless communication systems. By considering the
dynamic nature of traffic patterns, channel availability, and interference conditions, dynamic channel
allocation can adaptively allocate channels to users or devices, ensuring efficient resource utilization and
meeting the QoS requirements of different applications.

Multiple Access Protocols:

Multiple Access Protocols are used in communication systems to allow multiple users or devices to
access a shared communication channel or medium. These protocols determine how the channel is shared

5|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
and provide mechanisms for coordinating access to avoid collisions and ensure efficient utilization of the
available resources. Here are some commonly used Multiple Access Protocols:

1. Aloha Protocol:

 Pure Aloha: Users can transmit data at any time, leading to potential collisions. If a
collision occurs, a random backoff time is used before retransmitting the data.

 Slotted Aloha: Time is divided into fixed slots, and users are allowed to transmit data only
at the beginning of a slot. Collisions can still occur, but the probability is reduced
compared to pure Aloha.

2. Carrier Sense Multiple Access (CSMA):

 CSMA: Users sense the medium before transmitting and only transmit when the channel is
idle. If a collision occurs, random backoff times are used for retransmission.

 CSMA/CD (Collision Detection): Similar to CSMA, but users continuously monitor the
channel during transmission. If a collision is detected, transmission is aborted, and a
backoff mechanism is used for retransmission. Used in Ethernet networks.

 CSMA/CA (Collision Avoidance): Used in wireless networks, CSMA/CA incorporates a

request-to-send (RTS) and clear-to-send (CTS) mechanism to avoid collisions and manage
access to the channel.

3. Time Division Multiple Access (TDMA):

 Users are allocated specific time slots, and they take turns transmitting during their
assigned slots. TDMA is commonly used in cellular networks and satellite communication
systems.

4. Frequency Division Multiple Access (FDMA):

 Users are allocated different frequency bands, and each user can transmit within their
assigned frequency band. FDMA is used in analog cellular networks and some satellite
communication systems.

5. Code Division Multiple Access (CDMA):

 Users share the same frequency band but use different codes to differentiate their
transmissions. CDMA allows multiple users to transmit simultaneously using different
codes, with the ability to separate and decode their signals at the receiver. It is commonly
used in modern cellular networks.

6. Orthogonal Frequency Division Multiple Access (OFDMA):

6|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 OFDMA divides the channel into multiple orthogonal subcarriers, allowing multiple users
to transmit simultaneously by assigning them different subcarriers. OFDMA is used in
modern wireless communication systems such as Wi-Fi and 4G/5G cellular networks.

These are just a few examples of Multiple Access Protocols used in different communication systems.
The choice of protocol depends on factors such as the nature of the communication medium, the number
of users, the required data rate, and the quality of service requirements. Each protocol has its advantages
and trade-offs in terms of throughput, latency, fairness, and collision avoidance. The selection of an
appropriate Multiple Access Protocol is crucial to ensure efficient and reliable communication in shared
channel environments.

Aloha-Pure aloha:

Aloha is a multiple access protocol used in shared communication networks to allow multiple users to
transmit data over a common communication channel. Pure Aloha is one variant of the Aloha protocol.
Here are some key features of Pure Aloha:

1. Random Access: In Pure Aloha, users can transmit data at any time, without any central
coordination or synchronization. Each user decides when to transmit based on its own data
availability.

2. Collision Detection: After transmitting a data frame, a user listens to the channel to detect if a
collision has occurred. If a collision is detected, it means that another user has simultaneously
transmitted, and both transmissions may have been corrupted.

3. Retransmission Mechanism: In the case of a collision, the user waits for a random period of
time and then retransmits the data frame. The random backoff time is chosen to minimize the
chances of collisions occurring again.

4. Continuous Operation: Pure Aloha operates continuously, with users attempting to transmit data
whenever they have it. There is no predetermined time slot or synchronization requirement.

5. Efficiency: Pure Aloha allows users to transmit data at any time, resulting in a high potential for
collisions. As a result, the protocol may not be highly efficient in terms of channel utilization
since collisions can lead to retransmissions and decreased overall throughput.

6. Simple Implementation: Pure Aloha is relatively simple to implement because it does not
require sophisticated coordination mechanisms or centralized control. Users only need to sense
the channel, transmit their data, and handle collisions during the collision detection phase.

Pure Aloha was first developed for wireless packet radio networks in the early 1970s. While it served as
the foundation for subsequent multiple access protocols, it has limitations in terms of efficiency,
particularly in high-traffic scenarios where collisions are more likely. Slotted Aloha, a variant of Pure
Aloha, was later introduced to improve efficiency by dividing time into fixed slots and aligning user
transmissions to those slots.

7|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Overall, Pure Aloha is an early example of a random access multiple access protocol that allowed users to
share a common communication channel without centralized coordination. While it has its limitations, it
laid the groundwork for further advancements in multiple access protocols used in modern
communication systems.

slotted aloha:
Slotted Aloha is a variant of the Aloha multiple access protocol that was introduced to improve the
efficiency of data transmissions in shared communication networks. It addresses some of the limitations
of the original Pure Aloha protocol. Here are the key features of Slotted Aloha:

1. Time Division: Slotted Aloha divides time into fixed slots, with each slot having a specific
duration. The duration of each slot is equal to the time required to transmit a single data frame.

2. Synchronized Transmissions: Users are required to transmit their data frames only at the
beginning of a time slot. This synchronization ensures that all transmissions start at the same time
and eliminates the possibility of collisions within a slot.

3. Collision Detection: After transmitting a data frame, a user listens to the channel to detect if a
collision has occurred during the slot. If a collision is detected, the user waits until the beginning
of the next slot to retransmit the data frame.

4. Efficiency Improvement: By dividing time into slots and synchronizing transmissions, Slotted
Aloha reduces the chances of collisions compared to Pure Aloha. Collisions can only occur
between slots, rather than within a slot, thereby improving the overall channel utilization and
throughput.

5. Implementation Complexity: Slotted Aloha requires a mechanism to synchronize user

transmissions with the beginning of slots. This synchronization can be achieved through various
means, such as a centralized clock or a distributed signaling scheme.

6. Backoff Mechanism: In the event of a collision, users in Slotted Aloha do not utilize a random
backoff mechanism like in Pure Aloha. Instead, they simply wait until the beginning of the next
slot to retransmit, which provides a deterministic retransmission strategy.

Slotted Aloha was introduced as a refinement of the original Aloha protocol to achieve higher efficiency
in shared communication networks. By dividing time into slots and synchronizing transmissions, it
reduces collisions and improves the overall throughput. Slotted Aloha has been widely used in various
communication systems, such as Ethernet-based LANs and wireless networks, as a fundamental
component of multiple access protocols.

Carrier Sense Multiple Access Protocols:

8|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Carrier Sense Multiple Access (CSMA) protocols are a class of multiple access protocols used in shared
communication networks. These protocols allow multiple users or devices to share a common
communication medium by sensing the medium before transmitting data. The key characteristic of
CSMA protocols is that users first sense the channel to check if it is idle (i.e., no other user is
transmitting) before initiating their own transmission. Here are some commonly used CSMA protocols:

1. CSMA:

 In basic CSMA, a user listens to the channel before transmitting. If the channel is sensed
as busy (i.e., another user is transmitting), the user defers its transmission and waits for the
channel to become idle.

 However, if multiple users defer their transmission and attempt to transmit simultaneously
after the channel becomes idle, collisions can occur. To handle collisions, CSMA employs
collision detection mechanisms and retransmission strategies.

2. CSMA/CD (Carrier Sense Multiple Access with Collision Detection):

 CSMA/CD is used in Ethernet-based LANs. It extends basic CSMA with the capability to
detect collisions.

 In CSMA/CD, users continuously monitor the channel while transmitting to detect

collisions. If a collision is detected, the transmitting user immediately aborts the
transmission, sends a jam signal to notify other users of the collision, and waits for a
random backoff period before retransmitting.

3. CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance):

 CSMA/CA is used in wireless networks and addresses the issue of hidden terminal
problem and collisions in a wireless environment.

 In CSMA/CA, users employ a virtual carrier sensing mechanism where they listen for
signals from other nearby devices before transmitting. This helps avoid collisions caused
by hidden terminals.

 CSMA/CA also incorporates a request-to-send (RTS) and clear-to-send (CTS)

mechanism, where users send control frames to reserve the channel and inform other users
about their intent to transmit.

The main objective of CSMA protocols is to ensure fair access to the shared channel and minimize
collisions. By sensing the channel before transmitting, users can avoid or mitigate collisions and improve
the efficiency of the communication medium. CSMA protocols are widely used in Ethernet-based LANs,
Wi-Fi networks, and other shared medium communication systems.

Collision-Free Protocols:

9|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Collision-free protocols are a class of multiple access protocols designed to completely eliminate
collisions among users or devices sharing a common communication medium. These protocols ensure
that only one user transmits at a time, thereby maximizing channel utilization and throughput. Here are
some examples of collision-free protocols:

1. Time Division Multiple Access (TDMA):

 TDMA divides time into fixed time slots, with each slot allocated to a specific user or
device. Users take turns transmitting during their assigned time slots, and collisions are
completely avoided.

 TDMA requires precise synchronization among users to ensure accurate slot timing. It is
commonly used in cellular networks and satellite communication systems.

2. Frequency Division Multiple Access (FDMA):

 FDMA allocates different frequency bands to individual users or devices. Each user is
assigned a unique frequency band, and simultaneous transmissions are possible without
collision.

 FDMA is used in analog cellular networks and some satellite communication systems.

3. Code Division Multiple Access (CDMA):

 CDMA allows multiple users to transmit simultaneously by assigning unique codes to

each user. The orthogonal nature of the codes enables the receiver to separate and decode
individual transmissions without collisions.

 CDMA is commonly used in modern cellular networks, such as 3G and 4G systems.

4. Space Division Multiple Access (SDMA):

 SDMA allocates different physical spatial regions, such as sectors or beams, to individual
users. Each user is assigned a specific spatial region, and simultaneous transmissions can
occur without interference or collisions.

 SDMA is employed in advanced wireless communication systems, such as massive

MIMO (Multiple Input Multiple Output) systems.

These collision-free protocols eliminate the possibility of collisions by employing various techniques,
such as time division, frequency division, code division, or spatial division. By ensuring that only one
user transmits at a given time or within a specific frequency or spatial region, collision-free protocols
optimize channel utilization and improve overall system performance.

It's important to note that while these protocols eliminate collisions within their designated domains (e.g.,
time slots, frequency bands, codes, or spatial regions), collisions can still occur when multiple users
attempt to access the shared medium concurrently. Therefore, collision-free protocols are typically used

10 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
in conjunction with other medium access control techniques, such as carrier sensing or contention
resolution mechanisms, to manage access in shared communication networks effectively.

Limited Contention Protocols:

Limited contention protocols are a class of multiple access protocols that aim to strike a balance between
collision avoidance and efficient utilization of the communication medium. These protocols allow a
limited number of users or devices to contend for access to the shared channel, reducing the chances of
collisions compared to pure contention-based protocols. Here are a few examples of limited contention
protocols:

1. Reservation-Based Protocols:

 In reservation-based protocols, users contend for access to the channel by making a

reservation in advance. This reservation process reduces the contention window and
collision probability.

 One example is the Reservation ALOHA protocol, where users reserve specific time slots
to transmit their data. The reservations are typically made using a control channel or a
dedicated reservation mechanism.

2. Demand-Assigned Multiple Access (DAMA):

 DAMA protocols allocate communication resources to users on-demand, based on their

request or demand for service.

 In DAMA, users request access to the channel when they have data to transmit, and the
system dynamically assigns resources to fulfill the requests. This reduces the contention
and improves the efficiency of resource utilization.

 DAMA protocols are commonly used in satellite communication systems to allocate

bandwidth to users as needed.

3. Hybrid Protocols:

 Hybrid protocols combine elements of contention-based access and reservation-based

access to strike a balance between efficiency and collision avoidance.

 One example is the Carrier Sense Multiple Access with Reservation (CSMA/CR)
protocol, which combines carrier sensing with a reservation mechanism. Users sense the
channel before transmitting and, if the channel is idle, can reserve the channel for their
exclusive use for a certain period, reducing the contention and potential collisions.

Limited contention protocols aim to reduce the contention and collision probability in shared
communication networks while maintaining a reasonable level of efficiency. By introducing reservation-
based mechanisms, demand-based resource allocation, or hybrid approaches, these protocols can improve
overall system performance and fairness among users. However, the effectiveness of limited contention

11 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
protocols depends on the specific network environment, traffic patterns, and the number of users
contending for the channel.

Wireless LAN Protocols:

Wireless LAN (WLAN) protocols are a set of standards and protocols that govern wireless
communication in local area networks. These protocols define how devices communicate with each other
over the airwaves, ensuring reliable and efficient wireless connectivity. Here are some widely used
WLAN protocols:

1. IEEE 802.11b:

 IEEE 802.11b operates in the 2.4 GHz frequency band and supports data rates up to 11
Mbps.

 It uses Direct Sequence Spread Spectrum (DSSS) modulation and provides backward
compatibility with older devices.

 While 802.11b has slower data rates compared to newer protocols, it has good signal
range and can penetrate obstacles relatively well.

2. IEEE 802.11a:

 IEEE 802.11a operates in the 5 GHz frequency band and supports data rates up to 54
Mbps.

 It uses Orthogonal Frequency Division Multiplexing (OFDM) modulation and provides

faster data rates compared to 802.11b.

 However, 802.11a has a shorter signal range and is more susceptible to obstacles and
interference.

3. IEEE 802.11g:

 IEEE 802.11g operates in the 2.4 GHz frequency band and supports data rates up to 54
Mbps.

 It combines the compatibility of 802.11b with the higher data rates of OFDM modulation
used in 802.11a.

 802.11g is backward-compatible with 802.11b devices and has a good signal range and
compatibility with existing networks.

4. IEEE 802.11n:

 IEEE 802.11n operates in both the 2.4 GHz and 5 GHz frequency bands and supports data
rates up to 600 Mbps (with Multiple-Input Multiple-Output, or MIMO, technology).

12 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 It uses OFDM modulation and introduces features like channel bonding and spatial
multiplexing to improve throughput and coverage.

 802.11n provides better signal range and resistance to interference compared to previous
protocols.

5. IEEE 802.11ac (Wi-Fi 5):

 IEEE 802.11ac operates in the 5 GHz frequency band and supports data rates up to several
Gbps (with MIMO and other enhancements).

 It uses improved OFDM modulation and introduces technologies like Multi-User MIMO
(MU-MIMO) and beamforming for better performance in crowded environments.

 802.11ac offers higher throughput and capacity compared to previous protocols and is
suitable for bandwidth-intensive applications.

6. IEEE 802.11ax (Wi-Fi 6):

 IEEE 802.11ax is the latest WLAN protocol designed to enhance network efficiency and
performance.

 Wi-Fi 6 operates in both the 2.4 GHz and 5 GHz bands and introduces technologies like
Orthogonal Frequency Division Multiple Access (OFDMA), Target Wake Time (TWT),
and improved MU-MIMO.

 802.11ax provides higher data rates, reduced latency, and improved network capacity
compared to previous protocols, making it suitable for dense deployments and demanding
applications.

These WLAN protocols define the communication standards and capabilities for wireless devices,
enabling seamless connectivity and data transfer in local area networks. The choice of protocol depends
on factors such as data rate requirements, range, interference considerations, and compatibility with
existing network infrastructure. It's important to ensure that devices and access points support the same
protocol to establish a reliable and efficient wireless network.

Ethernet:
Ethernet is a widely used technology for local area networks (LANs) that allows devices to communicate
with each other over a physical network. It is based on a set of standards defined by the Institute of
Electrical and Electronics Engineers (IEEE) under the IEEE 802.3 standard. Ethernet provides a reliable
and efficient method for transmitting data packets between devices in a LAN environment.

Key features of Ethernet include:

1. Physical Medium: Ethernet supports various types of physical media for transmitting data,
including twisted-pair copper cables, coaxial cables, and fiber optic cables. The choice of the

13 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
physical medium depends on factors such as distance requirements, data rate, and environmental
conditions.

2. Carrier Sense Multiple Access with Collision Detection (CSMA/CD): Ethernet uses a
contention-based access method, where devices share the same communication medium. Before
transmitting data, devices listen to the network to check if it is idle. If multiple devices attempt to
transmit simultaneously, collisions may occur. CSMA/CD allows devices to detect collisions and
retransmit data if necessary.

3. Ethernet Frame: Data is transmitted in Ethernet networks using frames. An Ethernet frame
consists of a header, payload, and trailer. The header includes source and destination MAC
addresses, while the trailer contains a cyclic redundancy check (CRC) for error detection.

4. Ethernet Switching: Ethernet networks often employ switches to improve performance and
manage network traffic. Switches analyze the destination MAC address of incoming frames and
forward them only to the appropriate port, reducing unnecessary traffic and collisions.

Ethernet supports different data rates, with the most common variations being Fast Ethernet (IEEE
802.3u) with a data rate of 100 Mbps, Gigabit Ethernet (IEEE 802.3ab) with a data rate of 1 Gbps, and 10
Gigabit Ethernet (IEEE 802.3ae) with a data rate of 10 Gbps. Higher-speed Ethernet standards, such as
40 Gigabit Ethernet and 100 Gigabit Ethernet, have also been developed to meet the increasing demands
of high-bandwidth applications.

Ethernet has evolved over the years to meet the growing needs of LANs, and it remains a widely adopted
technology for wired networking. It provides a flexible and scalable solution for connecting devices in
homes, offices, data centers, and other network environments.

Classic Ethernet Physical Layer

Classic Ethernet, also known as 10BASE-T Ethernet, refers to the original implementation of Ethernet
technology using twisted-pair copper cables as the physical medium. It operates at a data rate of 10 Mbps
(megabits per second) and is based on the IEEE 802.3 standard.

The physical layer of classic Ethernet includes the following components:

1. Twisted-Pair Cables: Classic Ethernet uses Category 3 or Category 5 twisted-pair cables for data
transmission. These cables consist of pairs of insulated copper wires twisted together, which helps
reduce interference and crosstalk. The maximum length of a twisted-pair Ethernet segment is 100
meters.

2. Manchester Encoding: Classic Ethernet uses Manchester encoding to represent data as electrical
signals on the twisted-pair cables. In Manchester encoding, each bit is divided into two time slots,
and the transition of the signal (high to low or low to high) occurs in the middle of the time slot.
The presence or absence of a transition represents the value of the bit.

14 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Carrier Sense Multiple Access with Collision Detection (CSMA/CD): Classic Ethernet
employs CSMA/CD as the access method. Before transmitting data, a device listens to the
network to check if it is idle. If the network is busy, the device waits for a random period before
attempting to transmit again. If collisions occur, devices involved in the collision detect the
collision and abort their transmissions. They then wait for a random amount of time before
retrying.

4. Medium Attachment Unit (MAU): The Medium Attachment Unit is a device that connects the
Ethernet interface on a device (e.g., a computer) to the twisted-pair cable. It performs tasks such
as signal encoding/decoding, impedance matching, and amplification.

Classic Ethernet has laid the foundation for modern Ethernet technologies, and while its data rate of 10
Mbps may seem relatively slow by today's standards, it played a crucial role in the development and
popularization of Ethernet as a LAN technology. It provided a cost-effective and scalable solution for
connecting devices and formed the basis for subsequent Ethernet standards with higher data rates.

Classic Ethernet MAC Sub-layer Protocol:

The Classic Ethernet MAC (Media Access Control) sub-layer protocol defines the rules and procedures
for accessing and transmitting data on an Ethernet network. It operates at the data link layer of the OSI
model and is responsible for managing access to the shared communication medium and ensuring reliable
data transmission. Here are the key aspects of the Classic Ethernet MAC sub-layer protocol:

1. Framing: The MAC sub-layer encapsulates higher-layer data into Ethernet frames. Each frame
consists of a preamble, destination and source MAC addresses, EtherType field (indicating the
protocol type), data payload, and a frame check sequence (FCS) for error detection.

2. Addressing: Ethernet frames use MAC addresses to identify the source and destination devices.
A MAC address is a unique identifier assigned to the network interface card (NIC) of each
Ethernet device. The MAC sub-layer uses the destination MAC address to determine whether to
accept or forward a received frame.

3. CSMA/CD: Classic Ethernet employs the Carrier Sense Multiple Access with Collision
Detection (CSMA/CD) protocol for medium access. Before transmitting data, a device listens to
the network to check if it is idle. If the network is busy, the device waits until it becomes idle. If
collisions occur due to multiple devices attempting to transmit simultaneously, CSMA/CD detects
the collisions and initiates a collision recovery process.

4. Binary Exponential Backoff: When a collision occurs, the colliding devices employ a binary
exponential backoff algorithm to resolve the contention. Each device waits for a random amount
of time before retransmitting, with the waiting time increasing exponentially with each collision.
This helps prevent repeated collisions and allows for fair access to the medium.

15 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
5. Jam Signal: In case a collision is detected, the colliding devices transmit a jam signal to notify
other devices on the network about the collision. This ensures that all devices are aware of the
collision and can take appropriate action.

The Classic Ethernet MAC sub-layer protocol is a fundamental part of Ethernet networks, providing a
mechanism for devices to access the shared communication medium and transmit data reliably. While
modern Ethernet standards have introduced advancements and higher data rates, the underlying principles
of the MAC sub-layer protocol remain consistent across Ethernet implementations.

Ethernet Performance-Fast Ethernet:

Fast Ethernet is an Ethernet standard that operates at a data rate of 100 Mbps (megabits per second). It is
an improvement over the original Classic Ethernet, which operates at 10 Mbps. The increased data rate of
Fast Ethernet offers significant performance enhancements and allows for faster data transmission in
local area networks (LANs).

Here are some key aspects of Fast Ethernet performance:

1. Increased Data Rate: With a data rate of 100 Mbps, Fast Ethernet provides a tenfold increase in
bandwidth compared to Classic Ethernet. This higher data rate allows for faster file transfers,
quicker access to network resources, and improved overall network performance.

2. Backward Compatibility: Fast Ethernet maintains backward compatibility with Classic

Ethernet. This means that Fast Ethernet devices can interoperate with Classic Ethernet devices,
allowing for a smooth transition and integration of new equipment into existing networks.

3. Enhanced Throughput: The increased data rate of Fast Ethernet enables higher throughput,
which is particularly beneficial for bandwidth-intensive applications. It provides faster data
transfers and reduces network congestion, allowing for improved performance in environments
with high data demands.

4. Reduced Latency: Fast Ethernet's higher data rate helps reduce network latency, enabling
quicker response times and improved real-time application performance. This is especially
important for applications that require low latency, such as VoIP (Voice over IP) and video
conferencing.

5. Support for Full-Duplex Operation: Fast Ethernet supports full-duplex operation, which allows
for simultaneous data transmission and reception. In full-duplex mode, devices can send and
receive data simultaneously on separate pairs of wires, effectively doubling the available
bandwidth. This further enhances network performance and throughput.

6. Compatibility with Existing Infrastructure: Fast Ethernet uses the same frame format and
CSMA/CD access method as Classic Ethernet. This means that existing Ethernet switches, hubs,
and cabling infrastructure can be used with Fast Ethernet, minimizing the need for extensive
network upgrades.

16 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Overall, Fast Ethernet significantly improves network performance compared to Classic Ethernet. Its
higher data rate, backward compatibility, increased throughput, reduced latency, and support for full-
duplex operation make it well-suited for LANs that require faster and more efficient data transmission.
However, it is worth noting that Fast Ethernet has been superseded by even faster Ethernet standards,
such as Gigabit Ethernet and 10 Gigabit Ethernet, which offer even higher performance for modern
network requirements.

Wireless LANs:

Wireless LANs (Local Area Networks) are networks that allow devices to connect and communicate
wirelessly within a limited area. They provide the flexibility and convenience of wireless connectivity,
eliminating the need for physical cables and allowing devices to connect to the network from anywhere
within the coverage area. Wireless LANs are widely used in homes, offices, public spaces, and other
environments to enable wireless communication and internet access.

Here are some key aspects of wireless LANs:

1. Wireless Access Points (WAPs): Wireless LANs utilize wireless access points, also known as
base stations or routers, to provide connectivity to wireless devices. These access points transmit
and receive data wirelessly, acting as the central point for communication within the wireless
network. Multiple access points can be deployed to extend the coverage area and support a larger
number of devices.

2. Wireless Standards: Wireless LANs operate based on various wireless standards defined by the
IEEE 802.11 family. The most common standards include 802.11a, 802.11b, 802.11g, 802.11n,
802.11ac, and 802.11ax (Wi-Fi 6). Each standard supports different data rates, frequency bands,
and features. The standards are backward compatible, allowing devices using older standards to
connect to newer networks.

3. Frequency Bands: Wireless LANs operate in different frequency bands, including the 2.4 GHz
and 5 GHz bands. The 2.4 GHz band provides broader coverage but is more susceptible to
interference from other devices, such as microwave ovens and Bluetooth devices. The 5 GHz
band offers higher data rates and is less crowded, resulting in potentially better performance and
less interference.

4. Security: Wireless LANs require robust security measures to protect data transmitted over the
airwaves. Encryption protocols like WPA2 (Wi-Fi Protected Access 2) or the newer WPA3
provide encryption and authentication mechanisms to secure wireless communication. It is
important to implement strong passwords, use encryption, and regularly update firmware to
mitigate security risks.

5. Mobility and Roaming: Wireless LANs allow devices to move within the coverage area while
maintaining a connection. This enables mobile devices like laptops, smartphones, and tablets to
seamlessly roam between different access points without losing connectivity. Roaming is

17 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
particularly important in larger wireless networks, such as those found in office buildings or
public spaces.

6. Network Management: Wireless LANs require proper network management to ensure optimal
performance and security. This includes tasks such as configuring access points, managing
channel allocation, monitoring network traffic, and applying quality of service (QoS) settings to
prioritize specific types of traffic.

Wireless LANs have revolutionized the way we connect and communicate, providing flexible and
convenient network access. They have become an essential part of modern networking, enabling
mobility, collaboration, and access to online resources. As technology continues to advance, wireless
LANs are evolving to support higher data rates, improved coverage, and enhanced security features to
meet the increasing demands of wireless connectivity.

The 802.11 Architecture and Protocol Stack:

The 802.11 architecture and protocol stack refer to the set of standards and protocols defined by the IEEE
802.11 working group for wireless local area networks (WLANs). The 802.11 standards define the
specifications for wireless communication, including the physical layer (PHY) and the medium access
control (MAC) layer protocols. Here is an overview of the 802.11 architecture and protocol stack:

1. Physical Layer (PHY):

 The 802.11 PHY defines the physical characteristics of wireless communication, including
the modulation schemes, frequency bands, and data rates.

 The PHY layer specifications vary depending on the 802.11 standard being used, such as
802.11a, 802.11b, 802.11g, 802.11n, 802.11ac, and 802.11ax.

 Each PHY standard supports specific frequency bands (e.g., 2.4 GHz, 5 GHz) and
employs different modulation techniques to transmit data over the air.

2. Medium Access Control (MAC) Layer:

 The 802.11 MAC layer provides the protocols and mechanisms for accessing the wireless
medium and managing the transmission of data frames.

 It implements the Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA)
protocol to avoid collisions in the wireless environment.

 The MAC layer handles functions like frame fragmentation, interframe spacing,
acknowledgment, and error handling.

 The MAC layer is responsible for managing the distribution of available channels among
the wireless devices.

18 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Logical Link Control (LLC) Layer:

 The 802.11 LLC layer is responsible for providing a consistent interface to the higher
layers of the network protocol stack.

 It establishes connections between the MAC layer and the higher-layer protocols, such as
the Internet Protocol (IP).

 The LLC layer handles tasks like addressing, error control, and flow control for data
transmission.

4. Service Access Point (SAP):

 The SAP is an interface between the LLC layer and the higher-layer protocols, such as IP.

 It allows the LLC layer to communicate with the network layer protocols and provides
services like data encapsulation and transmission.

5. Security:

 The 802.11 standards also include provisions for wireless network security, including
encryption and authentication mechanisms.

 Common security protocols used in 802.11 networks include Wired Equivalent Privacy
(WEP), Wi-Fi Protected Access (WPA), and the more secure WPA2/WPA3.

The 802.11 architecture and protocol stack provide the foundation for wireless LAN communication.
These standards ensure interoperability and compatibility between different devices and networks,
enabling seamless wireless connectivity and communication.

The 802.11 Physical Layer:

The 802.11 Physical Layer (PHY) is a key component of the 802.11 wireless LAN (WLAN) standard. It
defines the physical characteristics of wireless communication, including the modulation schemes,
channel coding, transmission rates, and frequency bands used for data transmission. The 802.11 PHY
operates at the lowest layer of the protocol stack and is responsible for converting digital data into analog
signals for transmission over the wireless medium. Here are some important aspects of the 802.11
Physical Layer:

1. Frequency Bands:

 The 802.11 PHY operates in multiple frequency bands, including 2.4 GHz and 5 GHz.

 The 2.4 GHz band is more common and offers better coverage but is susceptible to
interference from other devices operating in the same band.

 The 5 GHz band provides higher data rates and is less congested, resulting in better
performance in environments with high wireless activity.

19 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Modulation Schemes:

 The 802.11 PHY uses various modulation schemes to encode digital data into analog
signals for transmission.

 Common modulation schemes include Binary Phase Shift Keying (BPSK), Quadrature
Phase Shift Keying (QPSK), Quadrature Amplitude Modulation (QAM), and Orthogonal
Frequency Division Multiplexing (OFDM).

 The choice of modulation scheme depends on factors such as the data rate, channel
conditions, and interference.

3. Channelization:

 The 802.11 PHY divides the available frequency spectrum into multiple channels to
support concurrent wireless communications.

 Channelization methods vary across different PHY standards. For example, 802.11b/g/n
uses a 20 MHz channel width, while 802.11a/ac/ax supports wider channel widths, such as
40 MHz or 80 MHz.

 Channel selection and allocation are crucial for optimizing network performance and
mitigating interference.

4. Data Rates:

 The 802.11 PHY supports different data rates depending on the modulation scheme,
channel bandwidth, and other factors.

 The available data rates vary across different PHY standards and can range from a few
megabits per second (Mbps) to several gigabits per second (Gbps).

 Higher data rates enable faster transmission of data and support bandwidth-intensive
applications.

5. Encoding and Error Correction:

 The 802.11 PHY employs various encoding and error correction techniques to enhance
data reliability and mitigate the impact of channel noise and interference.

 Forward Error Correction (FEC) codes, such as Convolutional Codes and Reed-Solomon
Codes, are used to detect and correct errors in the received signals.

 These techniques help improve the overall transmission quality and reduce the need for
retransmissions.

6. Multiple Antenna Techniques:

20 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 Some 802.11 PHY standards support multiple antenna techniques like Multiple-Input
Multiple-Output (MIMO) and beamforming.

 MIMO uses multiple antennas at both the transmitter and receiver to improve data
throughput, increase range, and enhance signal quality.

 Beamforming focuses the wireless signal towards the intended receiver, increasing signal
strength and improving coverage.

The 802.11 Physical Layer plays a crucial role in enabling wireless communication and providing the
foundation for higher-layer protocols and functionalities in WLANs. It ensures efficient and reliable
transmission of data over the wireless medium, adapting to varying channel conditions and optimizing
network performance.

The802.11 MAC Sub-layer Protocol:

The 802.11 MAC (Media Access Control) sub-layer protocol is a component of the 802.11 wireless LAN
(WLAN) standard that operates in the data link layer. It is responsible for managing access to the shared
wireless medium, coordinating transmissions between wireless devices, and handling issues such as
collision avoidance and medium contention. The MAC sub-layer protocol ensures efficient and fair
distribution of the wireless channel among multiple devices. Here are some key features and
functionalities of the 802.11 MAC sub-layer protocol:

1. Medium Access Mechanism:

 The MAC sub-layer uses the Carrier Sense Multiple Access with Collision Avoidance
(CSMA/CA) mechanism to control access to the wireless medium.

 CSMA/CA involves sensing the channel before transmitting data to avoid collisions with
other concurrent transmissions.

 It uses a virtual carrier sensing mechanism known as the Network Allocation Vector
(NAV) to detect ongoing transmissions and defer transmission until the channel is clear.

2. Distributed Coordination Function (DCF):

 The DCF is the fundamental access method of the 802.11 MAC sub-layer.

 It employs a contention-based approach, where devices contend for access to the wireless
medium.

 DCF uses a random backoff algorithm to resolve collisions that may occur due to multiple
devices attempting to transmit simultaneously.

 The backoff algorithm introduces a random delay to reduce the chances of repeated
collisions.

21 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Point Coordination Function (PCF):

 The PCF is an optional coordination method provided by the 802.11 MAC sub-layer.

 It operates in conjunction with the DCF and introduces a centralized coordination

mechanism.

 In PCF, a central device called the Point Coordinator (PC) controls access to the wireless
medium by allocating specific time slots to devices for transmission.

 The PCF can be used to provide time-bounded services and prioritize certain types of
traffic.

4. Fragmentation and Aggregation:

 The MAC sub-layer supports fragmentation and aggregation techniques to optimize data
transmission.

 Fragmentation allows a large data frame to be divided into smaller fragments that can be
transmitted more reliably over the wireless medium.

 Aggregation enables the grouping of multiple smaller frames into a single larger frame,
reducing overhead and improving efficiency.

5. Power Management:

 The MAC sub-layer includes power management mechanisms to conserve energy in

wireless devices.

 It supports power-saving modes, such as the Power Save Polling (PSP) mode, where
devices can enter a sleep state and wake up periodically to check for pending data.

 Power management mechanisms help prolong battery life in mobile devices and reduce
energy consumption.

6. Quality of Service (QoS) Support:

 The MAC sub-layer provides QoS support to prioritize different types of traffic based on
their requirements.

 It implements the Enhanced Distributed Channel Access (EDCA) mechanism, which

defines multiple access categories with different priorities.

 QoS parameters such as contention window sizes and access categories allow for
differentiated treatment of real-time and non-real-time traffic.

The 802.11 MAC sub-layer protocol plays a crucial role in managing access to the wireless medium and
ensuring efficient and fair communication between wireless devices. It incorporates mechanisms to

22 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
handle contention, control collisions, optimize transmission, conserve power, and provide quality of
service for different types of traffic.

The 805.11 Frame Structure-Services:

The 802.11 frame structure refers to the format and organization of data within frames used in the 802.11
wireless LAN (WLAN) standard. The frame structure defines how information is encapsulated and
transmitted over the wireless medium. It consists of various fields that convey important information and
provide services for reliable and efficient communication. Here are the key components of the 802.11
frame structure and the services they offer:

1. Frame Control Field:

 The Frame Control field contains several subfields that specify the frame type, duration,
and other control information.

 It includes the Protocol Version field, Type and Subtype fields, and Flags that indicate
various control functions.

 The Frame Control field helps devices identify the type of frame and interpret the
information contained within it.

2. Duration/ID Field:

 The Duration/ID field specifies the duration of the transmission for certain frame types or
serves as an identifier for other frame types.

 For example, in a Request to Send (RTS) frame, this field indicates the duration of the
subsequent transmission.

 In an Acknowledgment (ACK) frame, the Duration/ID field acknowledges the successful

reception of a frame.

3. Address Fields:

 The 802.11 frame structure includes several address fields that identify the source and
destination devices.

 The Address fields can include the Receiver Address (RA), Transmitter Address (TA),
and Destination Address (DA).

 The frame may also contain fields for Source Address (SA), BSSID (Basic Service Set
Identifier), and more, depending on the frame type.

4. Sequence Control Field:

 The Sequence Control field helps maintain frame ordering and ensures reliable delivery.

23 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 It includes a Fragment Number field and a Sequence Number field.

 The Fragment Number field is used to identify fragments of a fragmented frame, while the
Sequence Number field tracks the order of frames.

5. Frame Body:

 The Frame Body carries the actual data being transmitted.

 It can contain various types of information, such as payload data, management

information, control information, or encapsulated higher-layer protocol data.

6. Frame Check Sequence (FCS):

 The FCS field contains a checksum or cyclic redundancy check (CRC) value calculated
over the frame contents.

 The FCS enables the receiving device to verify the integrity of the received frame and
detect transmission errors.

The services provided by the 802.11 frame structure include:

1. Data Transmission and Delivery:

 The frame structure encapsulates and carries data, ensuring its delivery to the intended
recipient.

 Address fields identify the source and destination devices, allowing for proper routing and
delivery of frames.

2. Control and Coordination:

 The Frame Control field and other control information within the frame structure enable
devices to coordinate their activities and manage access to the wireless medium.

 Services like Request to Send (RTS), Clear to Send (CTS), and Acknowledgment (ACK)
help control transmissions and avoid collisions.

3. Error Detection:

 The FCS field provides a mechanism for error detection.

 The receiving device checks the FCS value to determine if the frame has been received
correctly or if errors occurred during transmission.

4. Fragmentation and Reassembly:

 The frame structure supports fragmentation and reassembly of large data frames.

24 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By : T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 Fragmentation breaks down a large frame into smaller fragments for reliable transmission,
and reassembly reconstructs the fragments at the receiving end.

The 802.11 frame structure ensures the reliable and efficient transmission of data over the wireless
medium. It provides necessary control, addressing, and error detection mechanisms to enable devices to
communicate effectively within a WLAN.

25 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MA NAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

Unit-4
COMPUTER NETWORKS:

The Network Layer Design Issues – Store and Forward Packet Switching-Services Provided to the
Transport layer- Implementation of Connectionless Service-Implementation of Connection Oriented
Service- Comparison of Virtual Circuit and Datagram Networks, Routing Algorithms-The Optimality
principle-Shortest path, Flooding, Distance vector, Link state, Hierarchical. Congestion Control
algorithms-General principles of congestion control, Congestion prevention polices, Approaches to
Congestion Control-Traffic Aware Routing- Admission Control-Traffic Throttling-Load Shedding. Internet
Working: How networks differ- How networks can be connected- Tunneling, internetwork routing-,
Fragmentation, network layer in the internet – IP protocols-IP Version 4 protocol-, IP addresses-,
Subnets-IP Version 6-The main IPV6 header- Internet control protocols- ICMP-ARP-DHCP.

The Network Layer Design Issues

The network layer, also known as the Internet layer, is a crucial component of the TCP/IP protocol suite.
It is responsible for facilitating communication between different networks by routing packets from the
source to the destination across multiple intermediate networks. The design of the network layer
involves various key issues that need to be addressed to ensure efficient and reliable network
communication. Here are some of the major design issues in the network layer:

1. Addressing: The network layer requires a unique addressing scheme to identify each device on a
network. IP (Internet Protocol) addresses, specifically IPv4 and IPv6, are used for this purpose.
Design issues include the allocation and assignment of IP addresses, address space management,
and the transition from IPv4 to IPv6.

2. Routing: Routing is a critical function of the network layer, responsible for determining the
optimal path for packet forwarding. Design issues in routing involve the selection of routing
algorithms, establishing routing tables, maintaining routing information, and ensuring efficient
and accurate packet delivery.

3. Packet Fragmentation and Reassembly: Network layer protocols may need to fragment packets
into smaller units for transmission across networks with different maximum transmission unit
(MTU) sizes. Packet fragmentation and reassembly mechanisms need to be designed to handle
this process efficiently and ensure successful reassembly at the destination.

4. Quality of Service (QoS): QoS refers to the ability to prioritize and manage network resources to
meet specific service requirements. Designing QoS mechanisms in the network layer involves
determining policies for packet prioritization, traffic shaping, congestion control, and resource
reservation.

1|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
5. Security: Network layer security focuses on protecting the network infrastructure and data from
unauthorized access, attacks, and other security threats. Design issues include the
implementation of encryption, authentication, access control, and intrusion detection
mechanisms at the network layer.

6. Network Address Translation (NAT): NAT is used to translate private IP addresses to public IP
addresses, allowing devices on private networks to communicate with devices on the public
Internet. NAT design issues involve defining translation methods, managing NAT tables, and
addressing compatibility concerns.

7. Mobility Support: The network layer needs to address the challenges of supporting mobile
devices that can move between different networks while maintaining connectivity. Design issues
include mobile IP protocols, location management, and seamless handover mechanisms.

8. Interoperability: The network layer should support interoperability among different network
technologies and protocols. Design issues include standardization, protocol encapsulation, and
the integration of heterogeneous networks.

9. Scalability: Network layer design should consider the ability to scale the network infrastructure
to accommodate a growing number of devices and increasing network traffic. Issues include
hierarchical addressing, routing scalability, and efficient resource allocation.

10. Network Management: Network layer design should incorporate mechanisms for network
monitoring, fault detection, performance management, and configuration management. These
mechanisms enable network administrators to manage and troubleshoot network devices and
ensure optimal network operation.

These are some of the key design issues in the network layer. Addressing these issues effectively is
crucial for building robust, scalable, and secure network architectures.

Store and Forward Packet Switching-Services Provided to the Transport layer:

Store and Forward packet switching provides several services to the Transport layer of the TCP/IP
protocol stack. The Transport layer is responsible for end-to-end communication between applications
running on different hosts. Here are the services provided by Store and Forward packet switching to the
Transport layer:

1. Reliable Delivery: Store and Forward packet switching ensures reliable delivery of packets to the
Transport layer. Each packet is received and analyzed by the intermediate network nodes before
being forwarded. This allows for error detection and correction mechanisms, such as checksum
verification and retransmission, to be applied. The Transport layer can rely on the network's
error handling capabilities provided by Store and Forward packet switching to ensure that
packets are delivered correctly and in the correct order.

2|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Congestion Control: Store and Forward packet switching plays a role in managing network
congestion, which can impact the performance of the Transport layer. By buffering packets
when the outgoing link or the next hop is congested, Store and Forward packet switching
prevents packet loss and avoids overwhelming the network. This helps in maintaining the
stability and fairness of the network, ensuring that the Transport layer can continue its
communication without being adversely affected by congestion.

3. Flow Control: Store and Forward packet switching can assist with flow control, which regulates
the rate of data transmission between the source and destination. The intermediate network
nodes can buffer packets when necessary, allowing for the smooth flow of data from the
Transport layer. This enables the Transport layer to adjust its transmission rate based on the
feedback received from the network, avoiding overflow or underflow of data and maintaining
efficient data transfer.

4. Segmentation and Reassembly: Store and Forward packet switching allows for the segmentation
and reassembly of data at the network layer. The Transport layer typically sends data in larger
chunks, which are segmented into smaller packets at the network layer for transmission. The
receiving node in the Store and Forward packet switching network reassembles the packets back
into the original data units and delivers them to the Transport layer. This segmentation and
reassembly process enables the Transport layer to transmit data across networks with different
maximum transmission unit (MTU) sizes, ensuring compatibility and efficient data transfer.

5. Network-Level Addressing: Store and Forward packet switching relies on network-level

addressing, such as IP addresses, to identify the source and destination of packets. The
Transport layer uses these addresses to establish end-to-end communication between
applications running on different hosts. Store and Forward packet switching ensures that the
network-layer addresses are correctly maintained and utilized, allowing the Transport layer to
establish connections and exchange data with the desired destination.

These services provided by Store and Forward packet switching to the Transport layer contribute to
reliable, efficient, and orderly communication between applications across the network. The Transport
layer can rely on the underlying Store and Forward packet switching network to handle error detection,
congestion control, flow control, and segmentation/reassembly, ensuring the smooth operation of end-
to-end communication.

Implementation of Connectionless Service:

Implementing a connectionless service involves designing and implementing protocols and mechanisms
that enable communication between network entities without establishing a dedicated connection.
Here are some key aspects to consider when implementing a connectionless service:

3|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Protocol Design: Design a protocol that supports connectionless communication. This protocol
defines the format of the data packets, header fields, and the rules for packet transmission,
reception, and handling. For example, User Datagram Protocol (UDP) is a common
connectionless transport protocol used in the Internet Protocol (IP) suite.

2. Addressing: Develop a mechanism for addressing network entities in a connectionless

environment. This typically involves using network-layer addressing such as IP addresses. Each
packet should contain the source and destination addresses in the packet header, allowing
routers to forward the packets to the appropriate destination.

3. Packet Handling: Implement functions to handle packet transmission and reception. This
includes packaging the data into packets, adding appropriate headers, and sending them to the
network. On the receiving side, packets are received, and the headers are examined to extract
the necessary information.

4. Routing: In connectionless communication, each packet is independently routed through the

network based on its destination address. Implement routing algorithms and mechanisms t hat
determine the next hop for each packet. This involves maintaining routing tables, updating them
dynamically, and selecting the appropriate output interface for packet forwarding.

5. Error Detection and Handling: Connectionless communication does not guarantee reliable
delivery, so error detection mechanisms are crucial. Implement error detection techniques such
as checksums or cyclic redundancy checks (CRC) to verify packet integrity. If errors are detected,
appropriate actions, such as discarding or retransmitting packets, should be taken.

6. Application Layer Handling: The application layer needs to handle the specifics of the
connectionless service. This includes managing packet sequencing, duplicate detection, and any
required reliability mechanisms. Depending on the application requirements, additional
protocols or techniques may need to be implemented to provide reliability, congestion control,
or other desired services.

7. Testing and Validation: Thoroughly test the implementation to ensure that the connectionless
service functions correctly. Test various scenarios, such as packet loss, out-of-order delivery, and
network congestion, to validate the robustness and performance of the implementation.

It's important to note that the specifics of implementing a connectionless service may vary depending
on the underlying network architecture, protocols, and requirements. However, the general principles
mentioned above provide a framework for designing and implementing a connectionless service that
allows for efficient and flexible communication between network entities.

4|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Implementation of Connection Oriented Service:

Implementing a connection-oriented service involves designing and implementing protocols and

mechanisms that establish a dedicated connection between network entities before data transmission
occurs. Here are the key aspects to consider when implementing a connection-oriented service:

1. Protocol Design: Design a protocol that supports connection-oriented communication. This

protocol defines the format of control messages for connection establishment, maintenance,
and termination. It also specifies rules for data transmission and flow control. The Transmission
Control Protocol (TCP) is a commonly used connection-oriented transport protocol in the
Internet Protocol (IP) suite.

2. Connection Establishment: Implement mechanisms for initiating a connection between the

communicating entities. This typically involves a three-way handshake process, where the
entities exchange control messages to establish and synchronize the connection parameters,
such as sequence numbers and window sizes.

3. Connection Maintenance: Develop mechanisms to ensure the continuity and reliability of the
connection during data transmission. This includes managing acknowledgment messages,
retransmission of lost or corrupted packets, and updating connection parameters dynamically to
adapt to changing network conditions.

4. Flow Control: Implement flow control mechanisms to regulate the rate of data transmission
between the sender and receiver. This involves managing the sender's transmission rate based
on the receiver's capacity and feedback, using techniques such as sliding window protocols.

5. Error Detection and Handling: Connection-oriented services include error detection mechanisms
to ensure data integrity. Implement error detection techniques, such as checksums or cyclic
redundancy checks (CRC), to verify packet integrity. If errors are detected, appropriate actions,
such as requesting retransmission, should be taken.

6. Connection Termination: Implement mechanisms for orderly termination of the connection. This
involves exchanging control messages to close the connection gracefully, ensuring all data has
been transmitted and acknowledged. It may also involve dealing with any remaining packets in
transit or retransmission timers.

7. Application Layer Handling: The application layer needs to interact with the connection-oriented
service to initiate data transmission, handle sequencing, manage acknowledgments, and
interpret received data. Applications typically use APIs or libraries provided by the connection-
oriented protocol to establish and manage connections, send data, and receive data reliably.

8. Testing and Validation: Thoroughly test the implementation to ensure the connection-oriented
service functions correctly. Test various scenarios, such as connection establishment and

5|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
termination, data transmission, retransmission, and flow control, to validate the reliability,
performance, and robustness of the implementation.

It's important to note that the specifics of implementing a connection-oriented service may vary
depending on the underlying network architecture, protocols, and requirements. However, the general
principles mentioned above provide a framework for designing and implementing a connection-
oriented service that enables reliable and ordered communication between network entities.

Comparison of Virtual Circuit and Datagram Networks:

Virtual Circuit Networks and Datagram Networks are two different approaches to handling
communication in computer networks. Let's compare them based on key characteristics:

1. Connection Establishment:

 Virtual Circuit: Virtual Circuit networks require a connection establishment phase before
data transmission can occur. A dedicated path is established between the source and
destination nodes, typically using a setup phase involving a three-way handshake. Once
the connection is established, subsequent data packets follow the same path.

 Datagram: Datagram networks do not require a connection establishment phase. Each

data packet is treated independently and can take different paths to reach the
destination. There is no dedicated path established prior to data transmission.

2. Packet Delivery:

 Virtual Circuit: In Virtual Circuit networks, packets follow a predetermined path through
the network. Each packet carries a virtual circuit identifier, and intermediate network
nodes use this identifier to forward packets along the established path. This guarantees
in-order delivery of packets.

 Datagram: Datagram networks do not guarantee in-order delivery of packets. Each

packet is treated independently and can take different paths through the network. This
can lead to potential out-of-order delivery.

3. Reliability:

 Virtual Circuit: Virtual Circuit networks offer reliability guarantees since the connection is
established beforehand. Error detection and correction mechanisms are typically
implemented at the network layer to ensure reliable data transmission.

 Datagram: Datagram networks do not provide inherent reliability guarantees. Error

detection mechanisms, such as checksums, may be used, but error recovery and
retransmission are typically handled at the transport or application layer.

6|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
4. Overhead:

 Virtual Circuit: Virtual Circuit networks require additional overhead due to the need for
connection establishment and maintenance. Resources such as bandwidth and routing
tables are reserved for the duration of the connection, even if no data is being
transmitted.

 Datagram: Datagram networks have lower overhead since no connection setup or

maintenance is required. Resources are allocated on-demand for each packet, allowing
for more efficient use of network resources.

5. Routing:

 Virtual Circuit: In Virtual Circuit networks, routing decisions are made during the
connection establishment phase. Once the path is established, subsequent packets
follow the predetermined route.

 Datagram: Datagram networks determine routing for each packet independently.

Routers examine the destination address of each packet and make routing decisions
based on their routing tables.

6. Flexibility:

 Virtual Circuit: Virtual Circuit networks offer predictable and deterministic paths for data
transmission. This can be advantageous for real-time applications that require consistent
latency and ordered delivery.

 Datagram: Datagram networks offer more flexibility in terms of routing and adaptability
to changing network conditions. They can handle varying traffic patterns and are suitable
for applications that do not require strict ordering or deterministic paths.

In summary, Virtual Circuit networks provide ordered and reliable data transmission with a dedicated
path, but require connection setup and maintenance. Datagram networks offer more flexibility and
lower overhead but do not guarantee in-order delivery or inherent reliability. The choice between the
two depends on the specific requirements of the application and the desired trade-offs between
reliability, overhead, and flexibility.

Routing Algorithms:

Routing algorithms are a fundamental component of computer networks, responsible for determining
the optimal paths for data packets to travel from a source to a destination through a network. Various
routing algorithms have been developed, each with its own characteristics, advantages, and limitations.
Here are some common routing algorithms:

7|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Shortest Path Algorithms:

 Dijkstra's Algorithm: This algorithm calculates the shortest path between a source node
and all other nodes in a network by iteratively selecting the node with the minimum
distance.

 Bellman-Ford Algorithm: It determines the shortest path between a source node and all
other nodes in a network, considering the possibility of negative edge weights.

2. Distance Vector Routing:

 Routing Information Protocol (RIP): This protocol uses the distance vector algorithm,
where each router shares its routing table with its neighbors. Routers exchange
information to determine the best paths to destinations based on distance metrics.

 Interior Gateway Routing Protocol (IGRP): It is a proprietary distance vector routing

protocol developed by Cisco. IGRP incorporates additional factors like bandwidth and
delay in addition to hop count for calculating the best paths.

3. Link State Routing:

 Open Shortest Path First (OSPF): OSPF is a link state routing protocol that uses the
Shortest Path First (SPF) algorithm. It builds a complete map of the network by
exchanging link state information among routers, allowing them to calculate the shortest
paths based on the network topology.

 Intermediate System to Intermediate System (IS-IS): IS-IS is another link state routing
protocol similar to OSPF, commonly used in large-scale networks. It uses the SPF
algorithm and operates at the data-link layer.

4. Path Vector Routing:

 Border Gateway Protocol (BGP): BGP is an exterior gateway protocol used for routing
between different autonomous systems (ASes) in the Internet. BGP uses path vector
routing, where routing decisions are based on a combination of path attributes and
policies.

5. Hierarchical Routing:

 Enhanced Interior Gateway Routing Protocol (EIGRP): EIGRP is a hybrid routing protocol
that combines elements of both distance vector and link state algorithms. It supports
hierarchical routing, allowing networks to be divided into smaller areas for more efficient
routing.

8|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
6. Adaptive Routing:

 Adaptive algorithms, such as the Adaptive Shortest Path First (ASPF) algorithm,
continuously monitor network conditions and dynamically adjust routing decisions based
on real-time information, such as link congestion or failures.

7. Software-Defined Networking (SDN):

 SDN is an approach that separates the control plane from the data plane, allowing
network administrators to centrally control and manage network routing using software.
Routing decisions in an SDN environment are typically made by a centralized controller
based on network-wide knowledge.

These are just a few examples of routing algorithms used in computer networks. The selection of a
routing algorithm depends on factors such as network size, topology, scalability requirements,
reliability, and specific application needs. Routing algorithms play a critical role in ensuring efficient and
reliable data transmission across networks.

The Optimality principle-Shortest path:

The Optimality principle, also known as the Principle of Optimality or Bellman's principle of optimality, is
a fundamental concept in graph theory and routing algorithms. It states that an optimal solution to a
problem contains within it optimal solutions to its subproblems.

In the context of shortest path algorithms, the Optimality principle can be summarized as follows: If
there is a shortest path from node A to node B, then the subpath from A to any intermediate node C
along that path must also be a shortest path from A to C.

This principle allows us to break down the problem of finding the shortest path between two nodes into
smaller subproblems and solve them independently. By applying the Optimality principle recursively, we
can find the optimal solution for the entire problem.

Shortest path algorithms, such as Dijkstra's Algorithm and the Bellman-Ford Algorithm, leverage the
Optimality principle to efficiently find the shortest path in a graph. These algorithms iteratively calculate
the shortest path estimates for each node by considering the optimal paths to its neighboring nodes.

Here's a simplified example to illustrate the Optimality principle in action:

Suppose we want to find the shortest path from node A to node D in a graph with nodes A, B, C, and D.
Let's assume the shortest path is A → B → C → D. According to the Optimality principle:

 The shortest path from A to B is a direct path from A to B.

 The shortest path from A to C is the path A → B → C, which is a subpath of the overall shortest
path.

9|Page LOYO LA INST IT UT E O F T EC HNO LO GY A ND MA NAG EMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 The shortest path from A to D is the path A → B → C → D, which is the complete shortest path.

By applying the Optimality principle, we can determine that the shortest path from A to D includes the
shortest path from A to C, which in turn includes the shortest path from A to B. This principle allows us
to break down the problem into smaller subproblems and solve them independently while ensuring that
the overall solution is optimal.

The Optimality principle plays a crucial role in the efficiency and correctness of shortest path
algorithms. It enables us to make optimal decisions at each step of the algorithm based on the
principle's guarantee that subpaths within the optimal solution are also optimal.

Flooding:

Flooding is a simple and basic routing technique used in computer networks. It involves broadcasting
data packets or messages to all nodes in a network, regardless of their destination. Each node receiving
the packet then forwards it to all of its connected neighbors, except for the node from which it received
the packet. This process continues until all nodes in the network have received the packet.

Flooding is typically used in situations where there is no predefined or known routing information, such
as in small-scale or ad hoc networks, or when there is a need for robustness and redundancy. Here are
some key points about flooding:

1. Broadcast Communication: Flooding relies on the broadcast communication model, where a

packet is sent to all nodes within the network. This ensures that the packet reaches every node,
regardless of its location or destination.

2. No Prior Knowledge: Unlike other routing algorithms that require knowledge of the network
topology or routing tables, flooding does not rely on any prior information about the network. It
blindly floods the network with packets to ensure their delivery.

3. Redundant Paths: Flooding allows packets to follow multiple paths within the network, as each
node forwards the packet to all of its neighbors. This redundancy helps to ensure packet delivery
even in the presence of link failures or congestion.

4. Packet Replication: Since every node forwards the received packet to all of its neighbors,
flooding can result in packet replication and network congestion. This issue can be mitigated by
using techniques such as hop count limits or sequence numbers to control the propagation of
duplicate packets.

5. Broadcast Storms: If not properly controlled, flooding can lead to broadcast storms, where
packets endlessly circulate within the network, causing excessive network traffic and congestion.
Techniques like packet suppression or random delays can be used to address this issue.

10 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
6. Inefficient Use of Resources: Flooding can be resource-intensive, as it consumes network
bandwidth and processing power in transmitting and forwarding packets. In large networks,
where the number of nodes is significant, flooding can become inefficient due to the exponential
growth of packet replication and network overhead.

7. Limited Scalability: Flooding is not scalable for large networks due to the indiscriminate
broadcast nature and the associated resource consumption. As the number of nodes increases,
the network becomes overwhelmed with excessive traffic and packet duplication.

8. Application-Specific Use: Flooding is often used in specific scenarios, such as broadcasting

routing information updates, distributing network discovery packets, or disseminating time-
critical information in emergency situations.

While flooding is a straightforward and robust approach for packet dissemination, it has limitations in
terms of scalability, resource utilization, and network congestion. Therefore, it is typically used in small -
scale networks or as a component of more advanced routing protocols that employ flooding selectively
for specific purposes.

Distance vector:

Distance Vector is a routing algorithm used to determine the best path or route for data packets in a
computer network. It operates by exchanging information about distances (metrics) between routers,
enabling each router to build and maintain a routing table containing the best path to each destination.

The key characteristics of the Distance Vector algorithm are as follows:

1. Iterative Updates: Distance Vector algorithms use iterative updates, where routers exchange
routing information periodically with their neighboring routers. These updates contain
information about the distances to various destinations.

2. Distance Metrics: Distance Vector algorithms assign metrics to measure the desirability of a
particular path. Common metrics include hop count (the number of routers crossed) or link cost
(a value based on factors like bandwidth, delay, or reliability).

3. Routing Table: Each router maintains a routing table that stores information about the best path
to reach each destination based on the received distance vectors. The table includes the next
hop router and the metric associated with the path.

4. Distance Vector Exchange: Routers share their routing tables (distance vectors) with neighboring
routers during update exchanges. By comparing the received distance vectors, routers can
update their own routing tables to reflect more optimal paths.

11 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
5. Bellman-Ford Equation: The Distance Vector algorithm utilizes the Bellman-Ford equation to
calculate the best path. It iteratively updates the routing table by comparing the received
distance vectors and choosing the path with the minimum metric to each destination.

6. Count-to-Infinity Problem: Distance Vector algorithms may encounter the count-to-infinity

problem, where incorrect or stale routing information causes routing loops or incorrect path
selection. Techniques such as split horizon, poison reverse, or triggered updates are employed
to mitigate this issue.

7. Convergence Time: Distance Vector algorithms require some time to converge and reach a
stable state where routers have accurate and consistent routing information. The convergence
time can be influenced by factors such as network size, topology changes, or the frequency of
update exchanges.

Examples of Distance Vector routing protocols include Routing Information Protocol (RIP) and Interior
Gateway Routing Protocol (IGRP). RIP uses hop count as the metric and broadcasts its entire routing
table to neighboring routers. IGRP, a Cisco proprietary protocol, incorporates additional factors like
bandwidth and delay in its metric calculation.

Distance Vector algorithms are relatively simple to implement and require less computational overhead
compared to other routing algorithms. However, they may suffer from slow convergence and
suboptimal paths due to their limited knowledge of the network beyond their immediate neighbors.
Consequently, Distance Vector algorithms are commonly used in smaller networks or as a part of larger,
more sophisticated routing protocols.

Link state:

Link State is a routing algorithm used to determine the best path or route for data packets in a
computer network. Unlike Distance Vector algorithms, which rely on exchanging routing information
with neighboring routers, Link State algorithms focus on building a complete and accurate map of the
network's topology.

The key characteristics of the Link State algorithm are as follows:

1. Link State Advertisements (LSAs): Each router in the network collects information about its
directly connected links, such as the link state and cost. This information is encapsulated in Link
State Advertisements (LSAs), which contain details about the router's interfaces, neighboring
routers, and link states.

2. Flooding LSAs: Routers using the Link State algorithm flood LSAs to all other routers in the
network. This ensures that every router has a consistent and up-to-date view of the network's
topology.

12 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Building a Network Map: Each router receives LSAs from its neighboring routers and builds a
map or database of the network's topology. This map includes information about all routers and
links in the network.

4. Dijkstra's Shortest Path First (SPF) Algorithm: Link State algorithms typically use Dijkstra's SPF
algorithm to calculate the shortest path from the source router to all other routers in the
network. The SPF algorithm uses the network map or database to determine the best path
based on link costs.

5. Routing Table Calculation: Using the SPF algorithm, each router calculates its routing table,
which contains the best path to each destination based on the shortest paths calculated by the
SPF algorithm. The routing table includes the next hop router and the total cost associated with
the path.

6. Event-Driven Updates: Link State algorithms are event-driven, meaning they respond to changes
in the network, such as new links, link failures, or topology changes. When a change occurs,
routers update their LSAs and flood the updated LSAs to inform other routers.

7. Rapid Convergence: Link State algorithms converge relatively quickly compared to Distance
Vector algorithms. Each router independently calculates its shortest paths using the SPF
algorithm, leading to faster convergence and more accurate routing decisions.

Examples of Link State routing protocols include Open Shortest Path First (OSPF) and Intermediate
System to Intermediate System (IS-IS). OSPF is widely used in IP networks, while IS-IS is commonly used
in large-scale networks, particularly in service provider environments.

Link State algorithms provide a more accurate and detailed understanding of the network's topology,
enabling routers to make better routing decisions. However, they require more memory and
computational resources to maintain and process the network map. Link State algorithms are suitable
for larger networks where scalability, faster convergence, and more efficient path selection are
important factors.

Hierarchical:

In the context of networking, "hierarchical" typically refers to a hierarchical network architecture or

design. It involves organizing the network infrastructure into multiple hierarchical layers or levels, each
serving a specific purpose and providing different services. The hierarchical approach helps in achieving
scalability, manageability, and efficient resource utilization. Here are the key aspects of a hierarchical
network design:

1. Core Layer:

 The core layer is the backbone of the network and handles high-speed, high-capacity
traffic.

13 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 It provides connectivity between different distribution layers or to external networks.

 The core layer focuses on speed and reliability and typically utilizes high-bandwidth links
and switches/routers with minimal packet processing.

2. Distribution Layer:

 The distribution layer aggregates network connections from various access layer devices.

 It provides routing, policy enforcement, and traffic filtering between different network
segments or subnets.

 The distribution layer may include switches, routers, and firewalls to facilitate network
segmentation, load balancing, and security policies.

3. Access Layer:

 The access layer connects end-user devices, such as computers, printers, and access
points, to the network.

 It provides local connectivity and handles the traffic between end devices and the
distribution layer.

 The access layer may include switches, wireless access points, and network interface cards
(NICs) for wired and wireless connections.

By adopting a hierarchical network design, organizations can benefit from the following advantages:

1. Scalability:

 Hierarchical networks can scale easily by adding more devices or expanding individual
layers without affecting the entire network.

 The separation of functions and traffic at different layers allows for more efficient
resource allocation and management.

2. Simplified Management:

 The hierarchical structure simplifies network administration and troubleshooting.

 Each layer can be managed independently, allowing for focused monitoring,

configuration, and maintenance tasks.

3. Improved Performance:

 The hierarchical design optimizes network performance by reducing network congestion

and improving traffic flow.

14 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 Traffic aggregation and segmentation at different layers prevent unnecessary broadcasts or
collisions, leading to improved efficiency.

4. Enhanced Security:

 Hierarchical networks enable better security enforcement and control by implementing

security policies at the distribution layer.

 Segmentation of network segments allows for the isolation and protection of sensitive
resources and data.

5. Cost-Effectiveness:

 The hierarchical approach can result in cost savings by utilizing appropriate hardware and
bandwidth resources at each layer.

 By distributing network resources efficiently, organizations can optimize their

infrastructure investment.

Overall, a hierarchical network design provides a structured and efficient framework for organizing and
managing complex networks, allowing for better scalability, performance, management, and security.

Congestion Control algorithms:

Congestion control algorithms are designed to manage and regulate the flow of data in a network to
prevent or mitigate congestion. These algorithms aim to ensure fair resource allocation, maintain network
stability, and minimize packet loss. Here are some widely used congestion control algorithms:

1. TCP Congestion Control:

 Transmission Control Protocol (TCP) implements various congestion control algorithms

to manage congestion in the internet.

 Notable TCP congestion control algorithms include:

 Reno: Uses additive increase and multiplicative decrease (AIMD) to adjust the
sending rate based on packet loss.

 NewReno: An enhancement of Reno that improves its performance during fast

recovery after packet loss.

 Cubic: Provides a TCP-friendly congestion control mechanism with a more

aggressive increase and slower decrease compared to Reno.

 BBR (Bottleneck Bandwidth and RTT): Uses a model-based approach to estimate

the available bandwidth and RTT to optimize congestion control.

2. UDP-Based Congestion Control:

15 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 User Datagram Protocol (UDP) does not have built-in congestion control mechanisms like
TCP.

 However, congestion control algorithms like LEDBAT (Low Extra Delay Background
Transport) can be used with UDP-based applications to achieve congestion control and
fairness.

3. Explicit Congestion Notification (ECN):

 ECN is a mechanism that allows routers to signal network congestion to endpoints without
dropping packets.

 Endpoints can detect ECN markings and adjust their transmission rate accordingly to
prevent congestion.

 TCP ECN and UDP ECN are congestion control extensions that leverage ECN to improve
congestion control in the network.

4. Random Early Detection (RED):

 RED is a queue management algorithm used in routers to detect and manage congestion.

 It randomly drops or marks packets before the queue becomes completely full, preventing
global synchronization of TCP flows.

 RED aims to maintain a low average queue size, thus improving network performance and
fairness.

5. Forward Explicit Congestion Notification (FECN) and Backward Explicit Congestion

Notification (BECN):

 FECN and BECN are congestion notification mechanisms used in Frame Relay networks.

 FECN is set by the source to indicate congestion in the forward direction, while BECN is
set by the network to indicate congestion in the backward direction.

 Based on these indications, the source adjusts its transmission rate to alleviate congestion.

6. Network-Assisted Congestion Control:

 Some congestion control mechanisms involve network-assisted approaches, where routers

actively participate in congestion control.

 Examples include XCP (Explicit Control Protocol) and RCP (Rate Control Protocol),
which involve collaborative control between routers and endpoints to manage congestion.

16 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
These are just a few examples of congestion control algorithms used in various network protocols and
technologies. The selection and implementation of specific algorithms depend on the network
environment, application requirements, and the underlying protocol being used.

General Principles Of Congestion Control:

The general principles of congestion control aim to manage and regulate network traffic to prevent or
mitigate congestion. These principles help ensure fair resource allocation, maintain network stability,
and minimize packet loss. Here are the key principles of congestion control:

1. Traffic Monitoring and Measurement:

 Congestion control starts with the monitoring and measurement of network traffic.

 Network devices and protocols collect data on various congestion indicators, such as
packet loss, queue length, and round-trip time (RTT).

 These measurements provide insights into network conditions and help identify
congestion.

2. Congestion Detection:

 Congestion detection involves analyzing the collected traffic measurements to determine

if congestion exists in the network.

 Various congestion detection algorithms and techniques are used to identify congestion
patterns and trends.

 Common indicators of congestion include increased packet loss, high queue occupancy,
and longer delays.

3. Congestion Notification:

 Once congestion is detected, network devices notify the source nodes about the
presence of congestion.

 Explicit congestion notification mechanisms, such as ECN (Explicit Congestion

Notification), can be used to inform the endpoints about congestion.

 Implicit congestion signals, such as increased delays or dropped packets, also serve as
notifications to the source nodes.

4. Congestion Avoidance:

 Congestion avoidance techniques aim to prevent congestion from occurring or

worsening.

17 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 These techniques regulate the traffic flow to ensure that the network operates within its
capacity limits.

 Congestion avoidance typically involves adjusting the transmission rate or window size of
data flows based on feedback received from congestion indicators.

5. Congestion Reaction:

 In cases where congestion is unavoidable or cannot be sufficiently prevented, congestion

control mechanisms react to mitigate its impact.

 This can involve adjusting the transmission rate, managing queue sizes, or prioritizing
certain types of traffic to alleviate congestion.

 Congestion reaction mechanisms aim to distribute the available network resources fairly
and efficiently among competing flows.

6. Fairness and Quality of Service (QoS):

 Congestion control principles emphasize fairness in resource allocation among different

network flows.

 Fairness ensures that no single flow or user monopolizes the available bandwidth,
allowing all flows to receive their fair share.

 QoS mechanisms may be employed to prioritize certain types of traffic, such as real-time
or latency-sensitive applications, over others during congestion.

7. Continuous Monitoring and Adaptation:

 Congestion control is an ongoing process that requires continuous monitoring and

adaptation to changing network conditions.

 Network devices and protocols dynamically adjust their behavior based on the feedback
received from congestion indicators and notifications.

 The goal is to maintain network stability, optimal resource utilization, and satisfactory
performance under varying traffic loads.

By following these general principles, congestion control mechanisms can effectively manage network
congestion, ensure fair resource allocation, and maintain the overall performance and stability of the
network.

18 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Congestion Prevention Polices:

Congestion prevention policies are proactive measures implemented to minimize the occurrence of
congestion in a network. These policies aim to manage network resources efficiently, optimize traffic
flow, and prevent congestion before it becomes a problem. Here are some commonly used congestion
prevention policies:

1. Traffic Engineering:

 Traffic engineering involves the optimization of network paths, routing, and resource
allocation to prevent congestion.

 By analyzing network traffic patterns and demands, traffic engineering policies can
allocate resources intelligently and dynamically.

 This can include load balancing, route optimization, and traffic prioritization techniques
to efficiently utilize available network capacity.

2. Quality of Service (QoS):

 QoS policies prioritize certain types of traffic over others based on their requirements
and importance.

 By assigning different traffic classes or service levels, QoS mechanisms ensure that critical
or delay-sensitive traffic receives preferential treatment.

 QoS techniques may include traffic classification, packet marking, and traffic shaping to
allocate resources according to predefined service-level agreements (SLAs).

3. Admission Control:

 Admission control policies determine whether to accept or reject new traffic flows based
on the availability of network resources.

 By considering the network's current capacity and the expected impact of new flows,
admission control prevents overloading of network resources.

 This helps ensure that the network can handle the requested traffic without
compromising the performance of existing flows.

4. Congestion-Aware Routing:

 Congestion-aware routing policies take into account network congestion levels when
determining the optimal paths for data transmission.

19 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 By avoiding congested links or selecting alternative routes, congestion-aware routing
policies help distribute traffic evenly across the network.

 This helps prevent congestion hotspots and improves overall network performance.

5. Traffic Policing and Shaping:

 Traffic policing and shaping policies regulate the rate of incoming traffic to align with
available network resources.

 Traffic policing enforces predetermined traffic rate limits and drops or marks packets
that exceed these limits.

 Traffic shaping techniques smooth out traffic bursts by buffering and delaying packets,
ensuring a more controlled and consistent flow of data.

6. Buffer Management:

 Effective buffer management policies play a crucial role in congestion prevention.

 By appropriately sizing and managing network buffers, these policies help avoid buffer
overflow and associated packet loss.

 Buffer management techniques may include drop-tail, random early detection (RED), or
weighted random early detection (WRED) algorithms.

7. Network Monitoring and Analysis:

 Continuous monitoring of network performance and traffic behavior allows for early
detection and identification of potential congestion issues.

 Network analysis tools can provide insights into traffic patterns, bottlenecks, and
resource utilization, enabling proactive congestion prevention actions.

Implementing a combination of these congestion prevention policies can help maintain network
efficiency, minimize congestion-related problems, and deliver a better user experience. The specific
policies adopted depend on the network architecture, traffic characteristics, and performance
objectives of the network.

Approaches To Congestion Control-Traffic Aware Routing

Traffic-aware routing is an approach to congestion control that considers the current traffic conditions
and congestion levels when making routing decisions. It aims to dynamically route traffic along paths
that have lower congestion, thus optimizing network resource utilization and avoiding congestion
hotspots. Here are the key aspects and benefits of traffic-aware routing:

20 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Real-time Traffic Monitoring: Traffic-aware routing relies on real-time monitoring of network
traffic. This involves collecting data on link utilization, packet loss, latency, and other congestion
indicators. Various monitoring techniques such as packet sniffing, flow monitoring, or network
telemetry can be used to gather this information.

2. Congestion Metrics: Traffic-aware routing utilizes congestion metrics or parameters to quantify

the congestion levels on different network links. These metrics can include link utilization, queue
length, available bandwidth, or any other measure that reflects the current congestion state.

3. Traffic Engineering: Based on the collected traffic and congestion information, traffic-aware
routing algorithms dynamically calculate the optimal paths for traffic transmission. These
algorithms consider both the shortest path and the path with the least congestion to make
routing decisions. By distributing traffic across less congested paths, the overall network
congestion can be reduced.

4. Load Balancing: Traffic-aware routing promotes load balancing by distributing traffic evenly
across multiple network paths. It ensures that no individual path becomes heavily congested
while other paths remain underutilized. Load balancing helps utilize the available network
capacity efficiently and prevents congestion from occurring.

5. Dynamic Path Selection: Unlike traditional routing protocols that use static path selection,
traffic-aware routing continuously updates and adjusts path selection based on real-time
congestion information. It allows the routing decisions to adapt to changing traffic patterns and
network conditions. This flexibility ensures that traffic is routed through less congested paths,
optimizing network performance.

6. Fault Tolerance: Traffic-aware routing can also improve network fault tolerance by rerouting
traffic in the event of link failures or congestion spikes. When congestion is detected on a
particular path, traffic can be quickly rerouted along alternative paths to avoid the congested
area and maintain network connectivity.

7. Quality of Service (QoS) Support: Traffic-aware routing can prioritize certain types of traffic
based on their QoS requirements. For example, real-time or delay-sensitive traffic can be routed
along paths with lower congestion to ensure timely delivery. This QoS support improves the
overall user experience and guarantees performance for critical applications.

By incorporating traffic-aware routing techniques, networks can achieve better congestion control,
improved network performance, and efficient utilization of network resources. These approaches are
particularly beneficial in large-scale networks with varying traffic patterns and dynamic congestion
conditions.

21 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Admission Control-Traffic Throttling-Load Shedding:

Admission Control, Traffic Throttling, and Load Shedding are three approaches commonly used in
congestion control to prevent network congestion and maintain overall network performance. Let's
explore each approach in more detail:

1. Admission Control:

 Admission Control is a mechanism that determines whether to accept or reject new

traffic flows based on the availability of network resources.

 When a new flow requests access to the network, admission control evaluates whether
there is sufficient capacity to accommodate the flow without degrading the performance
of existing flows.

 By rejecting flows when network resources are already heavily utilized, admission control
prevents overloading and congestion in the network.

 Admission control policies can consider factors such as available bandwidth, buffer
capacity, QoS requirements, and traffic priorities to make informed decisions.

2. Traffic Throttling:

 Traffic Throttling involves regulating the rate of incoming traffic to align it with the
available network resources.

 When congestion is detected or anticipated, traffic throttling mechanisms restrict the

transmission rate of certain flows to prevent congestion from worsening.

 Traffic throttling can be implemented at different levels, such as at the network edge,
within specific network domains, or even at the application layer.

 This approach can involve techniques like traffic shaping, where the traffic flow is
smoothed by buffering and delaying packets, ensuring a controlled and manageable flow
of data.

3. Load Shedding:

 Load Shedding is a technique used to selectively discard or drop certain packets or data
to reduce the overall network load during congestion.

 When congestion occurs, the network can prioritize critical or high-priority traffic while
shedding or dropping lower-priority traffic or non-essential data.

 Load shedding can be implemented based on pre-defined policies or by dynamically

adjusting the priority of different traffic flows based on their importance.
22 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN
 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
 This approach helps alleviate congestion by reducing the overall demand on network
resources, ensuring that critical traffic receives adequate bandwidth and reducing the
likelihood of packet loss.

Admission control, traffic throttling, and load shedding are complementary approaches that can be used
individually or in combination to manage congestion. The specific approach or combination depends on
the network environment, traffic characteristics, and the desired level of congestion control. By
implementing these techniques, network administrators can optimize resource utilization, ensure
quality of service, and maintain network stability even during periods of high demand or congestion.

Internet Working:

Internet working refers to the process of connecting multiple distinct networks to create a larger,
interconnected network, which is the foundation of the global Internet. It involves the integration of
different network technologies, protocols, and devices to enable seamless communication and data
exchange between networks. Here are key aspects of internet working:

1. Network Interconnectivity: Internet working enables the interconnectivity of diverse networks,

such as local area networks (LANs), wide area networks (WANs), and metropolitan area
networks (MANs). These networks can be based on different technologies, such as Ethernet, Wi-
Fi, fiber optics, or satellite links.

2. Internetworking Devices: Internetworking devices play a crucial role in connecting and routing
traffic between networks. Routers, switches, gateways, and network bridges are examples of
devices used to facilitate interconnection and data forwarding across networks.

3. Protocols and Standards: The Internet relies on a set of protocols and standards that ensure
compatibility and interoperability between networks. The TCP/IP protocol suite is the primary
protocol used for internet working, providing a common language for transmitting and routing
data across interconnected networks.

4. Routing: Routing is a fundamental function in internet working, where routers analyze and make
decisions on how to forward data packets across networks. Routing protocols, such as Border
Gateway Protocol (BGP) and Open Shortest Path First (OSPF), enable the exchange of routing
information and the selection of optimal paths for data transmission.

5. Addressing and Naming: Internet working requires unique addresses and names to identify
devices and networks. IP addresses are used for network and host identification, while domain
names provide human-readable names for network resources, such as websites. Addressing
schemes like IPv4 and IPv6 are used to assign unique addresses to devices.

6. Network Address Translation (NAT): NAT is a technique used in internet working to translate
private IP addresses used within local networks into public IP addresses used on the internet.

23 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
NAT enables multiple devices within a private network to share a single public IP address,
facilitating the connection between private networks and the internet.

7. Virtual Private Networks (VPNs): VPNs are used to create secure, private connections over public
networks, such as the internet. VPN technologies allow remote users or branch offices to
securely access resources within a corporate network, extending the reach of private networks
over public infrastructure.

8. Network Security: Internet working brings challenges in terms of network security. Firewalls,
intrusion detection systems (IDS), and virtual private networks (VPNs) are deployed to protect
networks and data from unauthorized access, attacks, and data breaches.

Internet working is vital for enabling global communication, data exchange, and access to resources on
a massive scale. It allows organizations, individuals, and devices from around the world to connect,
collaborate, and access information seamlessly, forming the foundation of the modern interconnected
world.

How networks differ- How networks can be connected- Tunneling:

Networks can differ in various ways, including their size, geographical coverage, topology, protocols,
and purpose. Here are some key factors that can differentiate networks:

1. Size: Networks can range from small-scale networks like home or office LANs to large-scale
networks like wide area networks (WANs) that span across cities, countries, or even continents.

2. Geographical Coverage: Networks can be localized within a specific physical location, such as a
building or campus (LAN), or they can have a broader coverage area, such as a metropolitan area
(MAN) or a global reach (WAN).

3. Topology: Networks can be organized in different topologies, such as bus, star, ring, mesh, or
hybrid. The topology determines how devices are connected and the structure of the network.

4. Protocols: Networks can use different communication protocols, such as Ethernet, TCP/IP, ATM,
or MPLS. The protocols define the rules and standards for data transmission and network
operation.

5. Purpose: Networks can serve various purposes, including data sharing, resource sharing,
communication, collaboration, internet access, or specialized applications like voice or video
conferencing.

Networks can be connected to each other using various methods, depending on the type of networks
and the connectivity requirements. Some common methods of network connection include:

24 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Local Interconnection: Local networks can be interconnected using network switches or routers
within the same physical location. This allows devices from different networks to communicate
and share resources.

2. Wide Area Network (WAN) Connection: WANs can be connected using leased lines, dedicated
connections, or virtual private networks (VPNs). These connections enable data transmission
and communication between geographically dispersed networks.

3. Internet Connection: Networks can be connected to the internet through internet service
providers (ISPs) using technologies such as broadband, DSL, fiber-optic, or satellite connections.
Internet connectivity allows access to the global network of networks.

4. Network Gateways: Gateways act as intermediaries between different networks, facilitating

communication and data exchange. They perform protocol translation and enable connectivity
between networks with different protocols or architectures.

Tunneling is a technique used to encapsulate one network protocol within another network protocol. It
allows the transmission of packets from one network over another network by encapsulating the
original packets within the payload of the outer protocol. The encapsulated packets are then
transmitted across the intermediate network and de-encapsulated at the receiving end to recover the
original packets. Tunneling is commonly used to create virtual private networks (VPNs) and enable
secure communication over public networks like the internet. Examples of tunneling protocols include
IPsec (Internet Protocol Security), L2TP (Layer 2 Tunneling Protocol), and GRE (Generic Routing
Encapsulation).

Internetwork Routing-, Fragmentation

Internetwork routing refers to the process of forwarding data packets between networks in an
internetwork, such as the Internet. It involves determining the optimal path for data transmission and
making routing decisions based on various factors like network topology, routing protocols, and
network congestion. Here are some key aspects of internetwork routing:

1. Routing Protocols: Internetwork routing relies on routing protocols that facilitate the exchange
of routing information among routers. Examples of routing protocols include OSPF (Open
Shortest Path First), BGP (Border Gateway Protocol), and RIP (Routing Information Protocol).
These protocols enable routers to discover network paths, exchange routing tables, and
calculate the best paths for data transmission.

2. Routing Tables: Each router in an internetwork maintains a routing table that contains
information about network destinations and the associated next-hop routers. The routing table
helps routers determine the best path for forwarding packets to their intended destinations.

25 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Interior Gateway Protocol (IGP): IGPs are routing protocols used within an autonomous system
(AS), which is a collection of networks under a single administrative domain. IGPs, like OSPF or
RIP, exchange routing information among routers within the AS to determine the best paths for
intra-domain routing.

4. Exterior Gateway Protocol (EGP): EGPs are routing protocols used for inter-domain routing
between different autonomous systems. BGP is the most widely used EGP and is responsible for
exchanging routing information and making routing decisions between autonomous systems.

5. Routing Metrics: Routing protocols use various metrics to calculate the optimal paths for data
transmission. Metrics can include factors such as hop count, link bandwidth, delay, load, or path
reliability. By considering these metrics, routing protocols select paths that meet specific criteria,
such as minimizing delay or maximizing available bandwidth.

6. Dynamic Routing: Dynamic routing protocols continuously adapt to changes in the network, such
as link failures or network congestion. They update routing tables dynamically, allowing routers
to discover new paths or reroute traffic in response to changes in network conditions.

Fragmentation, on the other hand, refers to the process of dividing large data packets into smaller
fragments to accommodate transmission over a network with a smaller maximum transmission unit
(MTU). When a packet is too large to fit within the MTU of a network segment or link, it needs to be
fragmented into smaller units that can be transmitted separately and reassembled at the receiving end.
Here are some key points about fragmentation:

1. Fragmentation Process: Fragmentation involves breaking the original packet into smaller
fragments, each with a specific size that fits within the MTU of the network. The fragmentation
process adds fragmentation headers to each fragment, indicating the offset and identification of
the original packet.

2. Reassembly: At the receiving end, the fragmented packets are reassembled by using the
information provided in the fragmentation headers. The receiver collects all the fragments and
arranges them in the correct order to reconstruct the original packet.

3. Fragmentation Overhead: Fragmentation introduces additional overhead in terms of increased

header size and processing overhead at both the sender and receiver. It can impact network
performance and increase the processing load on routers and hosts.

4. Path MTU Discovery: To avoid fragmentation, the Path MTU Discovery (PMTUD) technique is
used, where the sender determines the maximum MTU along the path to the destination. The
sender then adjusts the packet size to fit within the path's MTU, eliminating the need for
fragmentation.

26 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Fragmentation is typically handled by network devices, such as routers, based on the MTU of each
network segment. However, minimizing fragmentation is desirable to reduce overhead and improve
overall network performance. Path MTU Discovery and appropriate packet sizing can help optimize data
transmission and avoid unnecessary fragmentation.

NETWORK LAYER IN THE INTERNET:

The network layer in the Internet is responsible for the logical addressing, routing, and forwarding of
data packets across different networks. It plays a crucial role in enabling communication between hosts
(devices) on different networks by providing end-to-end connectivity. The main protocol used at the
network layer in the Internet is the Internet Protocol (IP).

Here are key aspects of the network layer in the Internet:

1. Internet Protocol (IP): IP is the primary protocol used at the network layer. It provides logical
addressing to devices connected to the Internet, assigning unique IP addresses to each device.
IPv4 (Internet Protocol version 4) and IPv6 (Internet Protocol version 6) are the two versions of
IP currently in use.

2. IP Packet: Data is transmitted in the form of IP packets at the network layer. An IP packet
includes a header and a payload. The header contains information such as source and
destination IP addresses, protocol information, and flags. The payload consists of the upper-
layer data being transmitted.

3. Logical Addressing: IP addresses are used for logical addressing at the network layer. An IP
address uniquely identifies a device on a network or the Internet. IPv4 addresses are 32-bit
numbers, while IPv6 addresses are 128-bit numbers.

4. Routing: The network layer performs routing, which involves determining the optimal path for
data packets to reach their destination across multiple networks. Routing protocols, such as
OSPF (Open Shortest Path First) and BGP (Border Gateway Protocol), are used to exchange
routing information and make routing decisions.

5. Forwarding: Once the optimal path is determined, the network layer is responsible for
forwarding data packets to the next hop on the route. Routers examine the destination IP
address in the packet header and use routing tables to determine the appropriate outgoing
interface for forwarding the packet.

6. Fragmentation: The network layer may perform packet fragmentation when the packet size
exceeds the maximum transmission unit (MTU) of a network segment. Fragmentation involves
breaking a packet into smaller fragments that can be transmitted and reassembled at the
receiving end.

27 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
7. Quality of Service (QoS): The network layer supports QoS mechanisms to prioritize certain types
of traffic over others. QoS ensures that critical or time-sensitive data, such as voice or video,
receives higher priority and better service in terms of bandwidth, delay, and reliability.

8. Network Address Translation (NAT): NAT is a technique used at the network layer to enable the
sharing of a single public IP address among multiple devices within a private network. NAT
translates private IP addresses to a public IP address when packets are forwarded to the
Internet.

The network layer in the Internet provides the necessary functions to establish end-to-end connectivity,
enable routing across networks, and ensure efficient and reliable data transmission between hosts. It
forms a vital part of the overall Internet architecture, allowing global communication and data
exchange.

IP protocols-IP Version 4 protocol:

IP Version 4 (IPv4) is the fourth iteration of the Internet Protocol, which is the network layer protocol
used in the majority of today's networks. Here are key aspects of the IPv4 protocol:

1. Addressing: IPv4 uses 32-bit addresses, typically represented as four decimal numbers separated
by dots (e.g., 192.168.0.1). Each address consists of a network portion and a host portion. The
network portion identifies the network to which a device belongs, while the host portion
identifies the specific device within that network.

2. Address Classes: IPv4 initially defined three address classes: Class A, Class B, and Class C. Each
class had a different range of address space and was designed for networks of varying sizes.
Class D was reserved for multicast addresses, and Class E was reserved for experimental
purposes.

3. Subnetting: To allocate address space more efficiently and accommodate networks of different
sizes, subnetting was introduced. Subnetting allows the division of a network into multiple
subnetworks, each with its own network portion and host portion. Subnetting enables more
efficient utilization of available IP addresses.

4. Address Resolution Protocol (ARP): ARP is used in IPv4 networks to map an IP address to its
corresponding MAC address. When a device wants to send data to another device on the same
network, it needs to resolve the MAC address associated with the destination IP address using
ARP.

5. Fragmentation and Reassembly: IPv4 supports fragmentation and reassembly of packets when
the packet size exceeds the Maximum Transmission Unit (MTU) of a network segment.
Fragmentation involves breaking a packet into smaller fragments, which can be transmitted

28 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
across networks with different MTUs. At the destination, the fragments are reassembled to
reconstruct the original packet.

6. Routing: IPv4 uses routing protocols, such as OSPF (Open Shortest Path First) and BGP (Border
Gateway Protocol), to exchange routing information between routers. Routing tables are
maintained by routers to determine the best path for forwarding packets to their destination
based on the destination IP address.

7. Internet Control Message Protocol (ICMP): ICMP is a companion protocol to IPv4 and is used for
diagnostic and error reporting purposes. It includes features like ping (echo request/reply),
traceroute, and error messages, such as "Destination Unreachable" or "Time Exceeded."

Despite the widespread use of IPv4, its address space is limited, and the exhaustion of available IPv4
addresses led to the development of IPv6. IPv6 provides a significantly larger address space and
additional features to meet the growing demands of the Internet. However, IPv4 continues to be used
extensively, and mechanisms like Network Address Translation (NAT) have been employed to extend its
address availability.

IP ADDRESSES:

IP addresses are unique numerical identifiers assigned to devices on a network that use the Internet
Protocol (IP). They are used to identify and communicate with devices connected to a network,
including computers, servers, routers, and other network devices. IP addresses play a crucial role in
enabling communication between devices over the Internet.

Here are some key points about IP addresses:

1. Format: IP addresses are typically written in a decimal format known as "dotted-decimal

notation." In IPv4, addresses consist of four sets of numbers, each ranging from 0 to 255,
separated by dots. For example, 192.168.0.1 is a typical IPv4 address. In IPv6, addresses are 128-
bit numbers and are represented in hexadecimal format.

2. Version: There are two versions of IP addresses in use: IPv4 and IPv6. IPv4 addresses, with their
32-bit length, are more common and have been widely used for many years. IPv6 addresses,
with their 128-bit length, were introduced to overcome the limitations of IPv4's address space.

3. Public and Private IP Addresses: IP addresses can be categorized as either public or private.
Public IP addresses are globally unique and assigned by Internet Service Providers (ISPs) to
devices connected directly to the Internet. Private IP addresses are used within private
networks, such as local area networks (LANs), and are not globally routable. Private IP address
ranges are reserved and should not be used on the public Internet.

4. Dynamic and Static IP Addresses: IP addresses can be assigned dynamically or statically. Dynamic
IP addresses are assigned to devices temporarily by a DHCP (Dynamic Host Configuration
29 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN
 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Protocol) server. Static IP addresses are manually assigned to devices and remain fixed unless
manually changed.

5. Subnetting: Subnetting allows the division of an IP address space into smaller subnetworks or
subnets. It enables more efficient utilization of IP addresses and helps in network management
and organization.

6. Network and Host Portions: In an IP address, there is a network portion and a host portion. The
network portion identifies the specific network to which a device belongs, while the host portion
identifies the individual device within that network.

7. IP Address Classes: IPv4 originally defined five address classes (Class A, B, C, D, and E) to
designate different address ranges and network sizes. However, the concept of classes is less
relevant now due to the adoption of Classless Inter-Domain Routing (CIDR), which allows more
flexible allocation of IP addresses.

8. Address Resolution: Address resolution is the process of mapping an IP address to its

corresponding Media Access Control (MAC) address. This mapping is essential for devices to
communicate on the same local network.

IP addresses are fundamental to the functioning of the Internet, as they provide a means to identify and
locate devices on a network. They allow data to be routed between different networks, enabling
communication and data exchange on a global scale.

Subnets-IP Version 6-The Main IPV6 Header:

Subnets: In IP networking, a subnet refers to a portion of a network that shares a common network
address prefix. Subnetting allows for the division of a larger network into smaller subnetworks, enabling
efficient allocation of IP addresses and improved network management. Subnetting is typically done by
borrowing bits from the host portion of an IP address to create a separate network portion. Each subnet
within a network can have its own unique network address.

IPv6: IPv6 (Internet Protocol version 6) is the successor to IPv4 and was developed to address the
limitations of IPv4's address space. IPv6 uses a 128-bit address format, allowing for a significantly larger
address space compared to IPv4's 32-bit addresses. This expansion accommodates the growing number
of devices connected to the Internet. IPv6 addresses are represented in hexadecimal format and consist
of eight groups of four hexadecimal digits, separated by colons (:).

Main IPv6 Header: The main IPv6 header is the first fixed-length part of an IPv6 packet and contains
essential information for the handling and routing of the packet. Here are the key fields in the main IPv6
header:

1. Version: This 4-bit field indicates the IP version and is set to '0110' for IPv6.

30 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Traffic Class: This 8-bit field is used for quality of service (QoS) and traffic classification purposes.
It enables the prioritization and differentiation of different types of IPv6 traffic.

3. Flow Label: This 20-bit field is used to identify and label packets belonging to the same flow,
such as a real-time multimedia stream. It helps ensure that packets in the same flow are handled
consistently by routers.

4. Payload Length: This 16-bit field specifies the length of the IPv6 packet's payload (excluding the
IPv6 header) in octets.

5. Next Header: This 8-bit field identifies the type of the next header following the IPv6 header. It
indicates the type of the extension header or upper-layer protocol, such as TCP (Transmission
Control Protocol) or UDP (User Datagram Protocol).

6. Hop Limit: This 8-bit field specifies the maximum number of hops (routers) a packet can traverse
before being discarded. It is similar to the Time-to-Live (TTL) field in IPv4.

7. Source Address: This 128-bit field contains the IPv6 address of the packet's source device.

8. Destination Address: This 128-bit field contains the IPv6 address of the packet's intended
destination.

The main IPv6 header provides the basic information required for routing and forwarding IPv6 packets.
It is followed by optional extension headers, which can include additional information and features such
as fragmentation, security, and routing options.

Internet Control Protocols:

Internet Control Protocols (ICPs) are a set of protocols that are used for various control and
management functions within the Internet. These protocols enable the exchange of control messages
and provide mechanisms for network troubleshooting, error reporting, and diagnostic purposes. Here
are some commonly used Internet Control Protocols:

1. Internet Control Message Protocol (ICMP): ICMP is a core protocol in the Internet Protocol Suite
and is used for sending control messages and error reporting. It includes features like ping (echo
request/reply), traceroute, and error messages such as "Destination Unreachable" or "Time
Exceeded."

2. Address Resolution Protocol (ARP): ARP is used to map an IP address to its corresponding MAC
address on a local network. It helps in the resolution of IP addresses to physical addresses within
the same network segment.

31 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Reverse Address Resolution Protocol (RARP): RARP is used to obtain an IP address from a known
MAC address. It is primarily used in older systems to determine the IP address of diskless
workstations or devices without permanent storage.

4. Simple Network Management Protocol (SNMP): SNMP is used for network management and
monitoring. It allows administrators to manage and monitor network devices such as routers,
switches, and servers. SNMP enables the collection of various network statistics, configuration
information, and the ability to remotely manage devices.

5. Border Gateway Protocol (BGP): BGP is an exterior gateway protocol used to exchange routing
information between different autonomous systems (ASes) on the Internet. It is the protocol
that allows different networks to communicate and route traffic across the Internet.

6. Internet Group Management Protocol (IGMP): IGMP is used for managing IP multicast group
memberships. It enables hosts to join or leave multicast groups and facilitates the efficient
delivery of multicast traffic within a network.

7. Domain Name System (DNS): Although not a dedicated control protocol, DNS plays a vital role in
the Internet by translating domain names into IP addresses. It allows users to access websites
and services using human-readable domain names rather than IP addresses.

These are just a few examples of Internet Control Protocols. There are other protocols and mechanisms
used for various control and management functions within the Internet, each serving specific purposes
to ensure the smooth operation and management of network resources.

ICMP-ARP-DHCP:

ICMP (Internet Control Message Protocol): ICMP is a network protocol used for sending control
messages and error reporting in IP networks. It operates at the network layer of the TCP/IP protocol
suite. ICMP is primarily used for diagnostic and troubleshooting purposes. Some common uses of ICMP
include:

1. Echo Request/Reply (Ping): ICMP Echo Request/Reply messages are used to determine whether
a particular IP host is reachable and to measure round-trip time.

2. Destination Unreachable: ICMP Destination Unreachable messages are sent by routers to

indicate that a particular destination network or host is unreachable.

3. Time Exceeded: ICMP Time Exceeded messages are sent by routers to indicate that a packet's
time-to-live (TTL) value has reached zero, and the packet was discarded.

ARP (Address Resolution Protocol): ARP is used to map an IP address to its corresponding MAC address
on a local network. It operates at the data link layer of the TCP/IP protocol suite. ARP allows devices to

32 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
determine the MAC address of a destination device on the same network segment. Some key points
about ARP include:

1. Address Resolution: ARP resolves an IP address to its corresponding MAC address by sending an
ARP request broadcast on the local network. The device with the matching IP address responds
with its MAC address, allowing communication to take place.

2. ARP Table: Devices maintain an ARP table that stores IP-MAC address mappings for efficient
address resolution. This table is used to quickly look up MAC addresses without having to send
ARP requests for every destination.

3. Proxy ARP: Proxy ARP is a technique in which a device responds to an ARP request on behalf of
another device. It allows devices to appear as if they are on the same network segment when
they are actually on different segments.

DHCP (Dynamic Host Configuration Protocol): DHCP is a network protocol used to dynamically assign IP
addresses and other network configuration parameters to devices on a network. It operates at the
application layer of the TCP/IP protocol suite. DHCP simplifies the process of IP address allocation and
configuration. Some key points about DHCP include:

1. IP Address Assignment: DHCP servers assign IP addresses dynamically to devices on a network.

Devices can request an IP address when they connect to the network, and the DHCP server
provides an available address from a predefined pool.

2. Configuration Parameters: DHCP can also provide other configuration parameters such as
subnet mask, default gateway, DNS server addresses, and lease duration.

3. DHCP Discover/Offer/Request/Acknowledge (DORA): The DHCP client and server communication

follows a four-step process known as DORA. The client sends a DHCP Discover message to find
available servers, and the server responds with a DHCP Offer message. The client then requests
the offered IP address, and the server acknowledges the request with a DHCP Acknowledge
message.

ICMP, ARP, and DHCP are important protocols in TCP/IP networks, each serving specific functions in
communication, address resolution, and dynamic IP address allocation.

33 | P a g e LOY O LA IN ST IT UT E OF T EC HNO LO G Y AN D MANA GEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD

Unit-5
COMPUTER NETWORKS:
UNIT V:
The Transport Layer: Transport layer protocols: Introduction-services- port number-User data gram
protocol-User datagram-UDP services-UDP applications-Transmission control protocol: TCP services-
TCP features- Segment- A TCP connection- windows in TCP- flow control-Error control. Application
Layer –- World Wide Web: HTTP , FTP-Two connections-control connection-Data connection-security
of FTP-Electronic mail-Architecture- web based mail- email security- TELENET-local versus remote
Logging. Domain Name System: Name Space, DNS in Internet, - Resolution-Caching- Resource
Records- DNS messages- Registrars-security of DNS Name Servers.

The Transport Layer:

The Transport Layer is a crucial layer in the TCP/IP protocol stack and is responsible for the reliable and
efficient delivery of data between network hosts. Its main functions include segmentation of data, error
detection and correction, flow control, and multiplexing/demultiplexing of data streams. The two
primary transport layer protocols are the Transmission Control Protocol (TCP) and the User Datagram
Protocol (UDP).

1. Transmission Control Protocol (TCP): TCP is a connection-oriented protocol that provides

reliable, ordered, and error-checked delivery of data packets over an IP network. It offers
features such as acknowledgment of received data, retransmission of lost packets, and
congestion control to ensure the integrity and efficiency of data transmission. Key features of
TCP include:

 Connection Establishment: TCP establishes a connection between the sender and

receiver before data transmission using a three-way handshake mechanism.

 Reliable Delivery: TCP ensures reliable delivery of data by using sequence numbers,
acknowledgments, and retransmissions to handle lost or corrupted packets.

 Flow Control: TCP implements flow control mechanisms to manage the rate of data
transmission between sender and receiver, preventing overwhelming of the receiver
with data.

 Congestion Control: TCP includes congestion control algorithms to monitor network

congestion and adjust the transmission rate to avoid network congestion and ensure fair
sharing of network resources.

2. User Datagram Protocol (UDP): UDP is a connectionless protocol that provides a simple,
lightweight method for transmitting data packets over an IP network. Unlike TCP, UDP does not
provide built-in reliability mechanisms such as acknowledgments or retransmissions. UDP is

1|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
commonly used for applications that require low overhead and real-time communication, such
as streaming media, online gaming, and DNS. Key features of UDP include:

 Connectionless Communication: UDP does not establish a connection before data

transmission. Each UDP datagram is independent and can be sent and received without
any prior setup.

 Low Overhead: UDP has minimal protocol overhead, making it more efficient in terms of
packet size and processing requirements compared to TCP.

 Unreliable Delivery: UDP does not provide reliability guarantees, so lost or corrupted
packets are not automatically retransmitted. It is up to the application layer to handle
any required error detection or recovery.

 Fast Transmission: UDP offers faster transmission speeds than TCP due to its simplicity
and lack of congestion control mechanisms.

The Transport Layer plays a critical role in ensuring reliable and efficient data transfer between network
hosts. TCP is commonly used for applications that require reliable, ordered delivery of data, while UDP
is preferred for applications that prioritize low overhead and real-time communication. The choice
between TCP and UDP depends on the specific requirements of the application and the trade-offs
between reliability and efficiency.

TRANSPORT LAYER PROTOCOLS:

The transport layer in the TCP/IP protocol suite provides end-to-end communication services between hosts. It
encapsulates the data received from the upper layers and divides it into segments or datagrams for transmission
over the network. Here are some commonly used transport layer protocols:

1. Transmission Control Protocol (TCP): TCP is a connection-oriented protocol that provides reliable,
ordered, and error-checked delivery of data packets. It establishes a connection between sender and
receiver before data transmission, and it ensures that data is delivered in the correct order and without
errors. TCP offers features like flow control, congestion control, and retransmission of lost packets,
making it suitable for applications that require guaranteed delivery of data, such as web browsing, file
transfer, and email.

2. User Datagram Protocol (UDP): UDP is a connectionless protocol that provides a lightweight and low-
overhead method for transmitting data packets. It does not establish a connection before data
transmission and does not guarantee reliable delivery or order of packets. UDP is commonly used for
real-time applications or scenarios where speed and minimal overhead are prioritized, such as streaming
media, voice and video communication, online gaming, and DNS.

3. Stream Control Transmission Protocol (SCTP): SCTP is a reliable, message-oriented protocol designed for
transporting telephony signaling messages over IP networks. It offers features like reliable delivery,

2|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
congestion control, and multihoming support (ability to use multiple network interfaces). SCTP is
commonly used in telecommunications and Voice over IP (VoIP) applications.

4. Datagram Congestion Control Protocol (DCCP): DCCP is a transport layer protocol that provides a
congestion-controlled, unreliable delivery mechanism for applications that require real-time or near real-
time communication. It supports congestion control and allows applications to choose between reliable
and unreliable delivery modes based on their specific requirements. DCCP is used in applications like
streaming media, online gaming, and voice and video conferencing.

These are some of the commonly used transport layer protocols in the TCP/IP protocol suite. Each protocol has
its own characteristics and is suitable for specific types of applications and network scenarios. The choice of
transport protocol depends on factors such as reliability requirements, delay tolerance, overhead considerations,
and the nature of the application being used.

Introduction-services- port number:

Introduction:

The transport layer in the TCP/IP protocol suite is responsible for providing end-to-end communication services
between hosts. It ensures reliable delivery of data and manages the flow and congestion control mechanisms.
One of the key aspects of the transport layer is the identification of specific communication endpoints, which are
known as ports.

Services:

The transport layer provides two primary services:

1. Connection-oriented service: This service is provided by protocols such as TCP. It establishes a reliable
and ordered connection between the sender and receiver before data transfer. It ensures that data is
delivered without errors and in the correct order.

2. Connectionless service: This service is provided by protocols such as UDP. It does not establish a
connection before data transfer. Each packet, known as a datagram, is independent and can be
transmitted without any prior setup. This service is suitable for applications that prioritize low latency
and minimal overhead.

Port Number:

In the TCP/IP protocol suite, ports are used to identify specific processes or services running on a host. A port is a
16-bit number that uniquely identifies a particular application or service. Port numbers are categorized into three
ranges:

1. Well-known ports (0-1023): These ports are reserved for well-known services such as HTTP (port 80), FTP
(port 21), and DNS (port 53). These ports are standardized and widely recognized across the Internet.

2. Registered ports (1024-49151): These ports are used by registered applications and services that are not
considered well-known. They are typically assigned by the Internet Assigned Numbers Authority (IANA) to
ensure unique identification.

3|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Dynamic or private ports (49152-65535): These ports are also known as ephemeral ports. They are used
by client applications to establish outgoing connections. When a client initiates a connection, it is
assigned a temporary port number from this range.

Port numbers, along with the IP address, create a unique socket that enables communication between
applications on different hosts. The combination of an IP address and port number allows data to be delivered to
the appropriate process or service running on a host.

In summary, the transport layer provides services like reliable data delivery and manages communication
endpoints through the use of port numbers. Port numbers help in identifying specific processes or services,
allowing effective communication between applications on different hosts.

User Data Gram Protocol-User Datagram-UDP Services:

User Datagram Protocol (UDP) is a connectionless transport layer protocol in the TCP/IP protocol suite. It
provides a simple and lightweight method for transmitting data between network hosts. UDP is often used in
applications that require low latency and do not necessarily require reliable delivery of data.

User Datagram:

A user datagram is the basic unit of data in UDP. It consists of a header and payload. The UDP header is 8 bytes
long and contains the source port number, destination port number, length of the UDP datagram (including
header and payload), and a checksum for error detection.

UDP Services:

UDP provides the following services:

1. Connectionless Communication: UDP is a connectionless protocol, which means that it does not establish
a dedicated connection between sender and receiver before data transmission. Each UDP datagram is
treated independently and can be sent and received without any prior setup. This lack of connection
establishment overhead makes UDP a faster and more efficient choice for applications that prioritize
speed and minimal overhead.

2. Low Overhead: UDP has a smaller header size compared to TCP, resulting in lower overhead in terms of
network bandwidth and processing requirements. This makes UDP suitable for applications that require
fast transmission and have less stringent reliability requirements.

3. Unreliable Delivery: Unlike TCP, UDP does not provide built-in mechanisms for reliable delivery of data. It
does not guarantee that the data will be delivered to the receiver or that it will be delivered in the
correct order. This lack of reliability makes UDP suitable for applications where occasional packet loss or
out-of-order delivery can be tolerated, such as real-time streaming, video conferencing, and online
gaming.

4. Datagram Multiplexing: UDP supports the multiplexing of multiple data streams within a single IP
address. This means that multiple applications or services can use UDP simultaneously, each identified by

4|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
its own port number. The combination of the destination IP address and the destination port number
allows the receiver to determine which application or service the UDP datagram is intended for.

UDP is commonly used in applications that require low latency, real-time communication, or where reliability is
handled at the application layer. It is widely used for multimedia streaming, Voice over IP (VoIP), Domain Name
System (DNS) queries, and other time-sensitive applications where speed and efficiency are prioritized over
reliability.

UDP Applications-Transmission Control Protocol

UDP Services:

UDP provides the following services:

1. Connectionless Communication: UDP is a connectionless protocol, which means it does not establish a
dedicated connection before data transmission. Each UDP datagram is treated independently and can be
sent and received without any prior setup.

2. Low Overhead: UDP has a smaller header size compared to TCP, resulting in lower overhead in terms of
network bandwidth and processing requirements.

3. Unreliable Delivery: UDP does not guarantee reliable delivery of data. It does not provide mechanisms for
acknowledgment of received data, retransmission of lost packets, or flow control. Therefore, UDP is
suitable for applications where occasional packet loss or out-of-order delivery can be tolerated.

4. Datagram Multiplexing: UDP supports the multiplexing of multiple data streams within a single IP
address. Multiple applications or services can use UDP simultaneously, each identified by its own port
number.

UDP Applications:

UDP is commonly used in various applications, including:

1. Real-time Streaming: UDP is widely used for real-time streaming applications, such as live video
streaming, online gaming, and multimedia content delivery. These applications prioritize low latency and
real-time responsiveness over reliability.

2. Voice over IP (VoIP): VoIP applications, which enable voice communication over IP networks, often use
UDP. Voice packets need to be delivered quickly to maintain real-time conversation, and the occasional
loss of a packet may not significantly impact the quality of the call.

3. Domain Name System (DNS): DNS queries and responses are typically carried out using UDP. DNS is
responsible for translating domain names into IP addresses, and UDP's low overhead and speed make it
suitable for quick resolution of DNS queries.

4. Network Monitoring and Diagnostics: UDP is used in network monitoring and diagnostics tools, such as
network performance measurement tools or network probes. These tools often send UDP packets to
measure network latency, packet loss, and other performance metrics.

5|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Transmission Control Protocol (TCP):

While UDP provides a connectionless and unreliable transport service, the Transmission Control Protocol (TCP) is
a connection-oriented and reliable transport protocol. TCP offers services such as acknowledgment of received
data, retransmission of lost packets, flow control, and congestion control. TCP is commonly used in applications
that require guaranteed delivery and ordered data transmission, such as web browsing, file transfer, email, and
other applications where data integrity and reliability are essential. TCP ensures that data is delivered without
errors and in the correct order, but it incurs additional overhead compared to UDP due to its reliability
mechanisms.

TCP Services- TCP Features- Segment:

TCP Services:

TCP (Transmission Control Protocol) provides the following services:

1. Connection-oriented Communication: TCP establishes a connection between the sender and

receiver before data transmission. This connection is maintained throughout the communication
session, ensuring reliable delivery of data.

2. Reliable Delivery: TCP guarantees reliable delivery of data by implementing acknowledgment

mechanisms. It requires the receiver to send acknowledgments for received data, and the
sender retransmits any lost or unacknowledged segments.

3. Flow Control: TCP manages the flow of data between sender and receiver to avoid
overwhelming the receiver with more data than it can handle. It uses a sliding window
mechanism to regulate the rate of data transmission based on the receiver's ability to process
the data.

4. Congestion Control: TCP monitors the network for signs of congestion and adjusts its
transmission rate accordingly. It dynamically adjusts the window size and transmission rate to
prevent network congestion and maintain optimal performance.

5. Ordered Delivery: TCP ensures that data is delivered in the same order it was sent. It uses
sequence numbers to track and reorder out-of-order segments at the receiver's end.

TCP Features:

TCP offers the following features:

1. Connection Establishment and Termination: TCP uses a three-way handshake process to

establish a connection between sender and receiver. It also provides a graceful connection
termination mechanism to ensure all data is successfully transmitted and received before closing
the connection.

6|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
2. Error Detection and Retransmission: TCP uses checksums to detect errors in the received data. If
errors are detected, TCP requests retransmission of the corrupted segments to ensure data
integrity.

3. Full-Duplex Communication: TCP supports simultaneous bidirectional communication, allowing

data to flow in both directions between sender and receiver.

4. Multiplexing and Demultiplexing: TCP supports multiplexing and demultiplexing of multiple

concurrent connections on the same host. It uses port numbers to identify different applications
or services running on the same IP address.

Segment:

In TCP, data is divided into smaller units called segments for transmission over the network. Each
segment consists of a TCP header and a payload. The TCP header contains control information such as
source and destination port numbers, sequence numbers, acknowledgment numbers, and flags for
various control purposes.

Segments are the basic units of data exchanged between TCP entities. They are encapsulated within IP
packets for transmission over the network. At the receiving end, the TCP layer reassembles the received
segments to reconstruct the original data.

Overall, TCP provides reliable, ordered, and connection-oriented communication services. It ensures the
reliable delivery of data, manages flow and congestion control, and maintains the integrity and order of
transmitted data. Segments are the units of data used by TCP for transmission and reassembly.

A TCP CONNECTION- WINDOWS IN TCP- FLOW CONTROL-ERROR CONTROL:

A TCP Connection:

A TCP connection is a logical communication channel established between two network hosts
(computers) for reliable data transfer. The TCP connection provides a full-duplex, byte-oriented, and
stream-oriented communication between the sender and receiver.

The TCP connection establishment process involves a three-way handshake. It consists of the following
steps:

1. SYN: The client sends a SYN (synchronize) segment to the server, indicating its intent to establish
a connection. The SYN segment contains an initial sequence number.

2. SYN-ACK: Upon receiving the SYN segment, the server responds with a SYN-ACK segment. The
SYN-ACK segment acknowledges the client's SYN and includes the server's own initial sequence
number.

7|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. ACK: Finally, the client acknowledges the server's SYN-ACK segment by sending an ACK segment.
This completes the three-way handshake, and both client and server are now synchronized and
ready to exchange data.

Once the TCP connection is established, data can be transmitted bidirectionally between the sender and
receiver using TCP segments.

Windows in TCP:

In TCP, a window represents the amount of data that a sender can transmit without receiving an
acknowledgment from the receiver. The window size is dynamic and can be adjusted based on the flow
control mechanism and network conditions.

TCP uses a sliding window protocol to manage the flow of data between sender and receiver. The
sender maintains a window that indicates the next sequence number it expects to receive an
acknowledgment for. As the sender receives acknowledgments for transmitted data, the window slides
forward, allowing the sender to transmit additional data.

Flow Control:

Flow control in TCP ensures that the sender does not overwhelm the receiver with more data than it
can handle. TCP uses a mechanism called the sliding window to implement flow control.

The receiver specifies a receive window size, indicating the amount of data it is currently able to accept.
The sender adjusts its transmission rate based on the receive window size reported by the receiver. If
the receive window becomes smaller than the amount of data the sender has available to transmit, the
sender must pause transmission until more window space becomes available.

This flow control mechanism prevents the receiver from being overwhelmed by a fast sender and avoids
potential buffer overflow or data loss.

Error Control:

TCP provides reliable data transfer by implementing various error control mechanisms. These
mechanisms include:

1. Checksum: TCP uses a checksum to detect errors in the received data. The checksum is
calculated over the TCP header and data, and the receiver checks if the received checksum
matches the calculated checksum. If the checksums do not match, it indicates a transmission
error, and the segment is discarded.

2. Sequence Numbers and Acknowledgments: TCP assigns a sequence number to each segment to
ensure ordered delivery. The receiver sends an acknowledgment (ACK) for each received

8|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
segment, indicating the next expected sequence number. If the sender does not receive an ACK
for a transmitted segment within a certain timeout period, it retransmits the segment.

3. Retransmission: If a segment is lost or not acknowledged by the receiver within a certain

timeout period, TCP automatically retransmits the segment to ensure reliable delivery. The
receiver recognizes duplicate segments and discards them to prevent duplicate processing.

These error control mechanisms help TCP provide reliable data transfer over an unreliable network,
ensuring that data is received correctly and in the correct order.

In summary, TCP connections are established through a three-way handshake. TCP uses windows for
flow control, adjusting the transmission rate based on the receiver's window size. TCP also implements
error control mechanisms such as checksums, sequence numbers, acknowledgments, and
retransmissions to ensure reliable data transfer.

APPLICATION LAYER:

The Application Layer is the topmost layer of the OSI (Open Systems Interconnection) model and the
TCP/IP protocol suite. It is responsible for providing network services and protocols that enable
applications to communicate with each other over a network. The Application Layer interacts directly
with the end-user and provides the interface for applications to access network resources.

Functions of the Application Layer:

1. Application Services: The Application Layer provides various services that applications can use to
establish communication, exchange data, and perform specific functions. Examples of
application services include email, file transfer, remote login, web browsing, and domain name
resolution.

2. Application Protocols: The Application Layer defines specific protocols that applications use to
exchange data and communicate with each other. These protocols include HTTP (Hypertext
Transfer Protocol) for web browsing, SMTP (Simple Mail Transfer Protocol) for email
transmission, FTP (File Transfer Protocol) for file transfer, and DNS (Domain Name System) for
domain name resolution.

3. Data Formatting and Presentation: The Application Layer is responsible for formatting and
presenting data to the application layer at the receiving end. It ensures that data is in the
expected format and structure for the receiving application to process correctly.

4. Application Layer Security: The Application Layer can include security mechanisms and protocols
to ensure secure communication between applications. Examples include SSL/TLS (Secure
Sockets Layer/Transport Layer Security) for encrypted communication and digital certificates for
authentication.

9|P ag e LOYOLA INSTIT UT E OF T ECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Examples of Application Layer Protocols:

1. HTTP (Hypertext Transfer Protocol): Used for web browsing and communication between web
browsers and web servers.

2. FTP (File Transfer Protocol): Used for transferring files between a client and a server.

3. SMTP (Simple Mail Transfer Protocol): Used for sending and receiving emails between mail
servers.

4. DNS (Domain Name System): Used for resolving domain names to IP addresses.

5. DHCP (Dynamic Host Configuration Protocol): Used for dynamically assigning IP addresses and
network configuration parameters to devices on a network.

6. SNMP (Simple Network Management Protocol): Used for managing and monitoring network
devices and systems.

7. SSH (Secure Shell): Used for secure remote login and executing commands on a remote server.

8. POP (Post Office Protocol) and IMAP (Internet Message Access Protocol): Used for retrieving
emails from a mail server.

The Application Layer protocols and services enable applications to interact and communicate over a
network, providing a wide range of functionalities for users and applications alike.

World Wide Web:

The World Wide Web (WWW) is a system of interconnected documents and resources that are
accessed over the internet. It is an information space where webpages, images, videos, and other
multimedia content are stored and made available to users worldwide. The WWW is one of the most
popular and widely used services on the internet, and it has revolutionized the way we access and share
information.

Key Components of the World Wide Web:

1. Webpages: Webpages are documents written in HTML (Hypertext Markup Language), which is
the standard language for creating web content. Webpages can include text, images, links, and
multimedia elements.

2. Hyperlinks: Hyperlinks are clickable elements on webpages that allow users to navigate between
different webpages. By clicking on a hyperlink, users can access related content, visit other
websites, or jump to a different section of the same webpage.

3. Web Browsers: Web browsers are software applications that allow users to access and view
webpages. Popular web browsers include Google Chrome, Mozilla Firefox, Microsoft Edge, and

10 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Safari. Web browsers interpret the HTML code of webpages and render them as visual content
for users to interact with.

4. Web Servers: Web servers are computers or systems that host and serve webpages and other
web content. When a user requests a webpage, the web browser sends a request to the
appropriate web server, which then retrieves the requested webpage and sends it back to the
user's browser for display.

5. Uniform Resource Locators (URLs): URLs are the addresses used to locate webpages and
resources on the internet. A URL typically consists of the protocol (e.g., "http://" or "https://"),
the domain name (e.g., "www.example.com"), and the specific path or file name that points to
the desired webpage or resource.

6. HTTP and HTTPS: HTTP (Hypertext Transfer Protocol) is the protocol used for transferring data
between web browsers and web servers. It defines the rules and standards for communication
on the World Wide Web. HTTPS (HTTP Secure) is an extension of HTTP that adds encryption and
security features to protect sensitive information transmitted over the web.

7. Search Engines: Search engines are specialized websites or services that index and organize vast
amounts of web content. Users can enter keywords or queries into a search engine, and it
returns a list of relevant webpages and resources. Examples of popular search engines include
Google, Bing, and Yahoo.

The World Wide Web has transformed the way we access information, communicate, and conduct
business. It has enabled global connectivity and made information readily available to users around the
world. The web has also facilitated the growth of e-commerce, online education, social networking, and
many other online services and activities.

HTTP:

HTTP (Hypertext Transfer Protocol) is a protocol used for communication between web browsers and
web servers. It is the foundation of data communication on the World Wide Web and enables the
retrieval and display of webpages, images, videos, and other resources.

Key Features of HTTP:

1. Client-Server Model: HTTP follows a client-server model, where the client (typically a web
browser) sends requests to the server, and the server responds with the requested content.

2. Stateless Protocol: HTTP is a stateless protocol, meaning that each request-response cycle is
independent of previous interactions. The server does not retain any information about past
requests, and each request is treated as a separate transaction.

11 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Request Methods: HTTP defines various request methods that indicate the desired action to be
performed by the server. The most commonly used methods are:

 GET: Retrieves a resource (webpage, image, etc.) from the server.

 POST: Submits data to be processed by the server (e.g., form submissions).

 PUT: Uploads a resource to the server.

 DELETE: Deletes a resource on the server.

 HEAD: Retrieves only the header information of a resource without the content.

4. Status Codes: HTTP uses status codes to indicate the outcome of a request. Some common
status codes include:

 200 OK: The request was successful, and the server returned the requested content.

 404 Not Found: The requested resource could not be found on the server.

 500 Internal Server Error: The server encountered an error while processing the request.

5. Headers: HTTP headers contain additional information about the request or response. Headers
can include details such as the content type, caching directives, authentication credentials, and
more.

6. Cookies: HTTP supports the use of cookies, which are small pieces of data sent from the server
and stored on the client's browser. Cookies are commonly used for session management, user
authentication, and tracking user preferences.

7. HTTPS: HTTPS (HTTP Secure) is an extension of HTTP that adds encryption and security features.
It uses SSL/TLS protocols to encrypt the data exchanged between the client and server, ensuring
privacy and integrity.

HTTP is the primary protocol for web communication and is widely supported by web browsers, servers,
and web-based applications. It enables the retrieval and delivery of web content and forms the basis of
modern web browsing and web-based services.

FTP-Two Connections-Control Connection:

FTP (File Transfer Protocol) is a standard network protocol used for transferring files between a client
and a server over a TCP/IP network. FTP operates using two separate connections: a control connection
and a data connection.

1. Control Connection: The control connection is established between the FTP client and the FTP
server. It handles the control and coordination of the file transfer session. The control

12 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
connection is used to send commands from the client to the server and receive responses from
the server.

During the control connection, the client sends FTP commands such as login credentials, directory
navigation commands (e.g., change directory, list directory), and file transfer commands (e.g., upload,
download). The server processes these commands and sends back appropriate responses indicating the
success or failure of the requested operation.

2. Data Connection: The data connection is used for actual file transfer between the client and the
server. Once the control connection has been established and authenticated, the client sends a
specific command to initiate the data connection. The data connection can be established in two
modes: active mode and passive mode.

 Active Mode: In active mode, the FTP server initiates a connection to the client's IP address and
port specified by the client. The client listens for incoming data connections and provides the
server with its IP address and port to establish the data connection. This mode requires the
client to allow incoming connections on its firewall.

 Passive Mode: In passive mode, the client initiates the data connection to the server. The server
listens for incoming data connections and provides the client with the IP address and port to
establish the data connection. This mode is typically used when the client is behind a firewall or
network address translation (NAT) device.

Once the data connection is established, actual file transfer occurs over this connection. The client and
server exchange data packets containing the file content, and the control connection remains open for
further commands and coordination.

By separating the control connection from the data connection, FTP allows for efficient and flexible file
transfers. The control connection handles the commands and responses, while the data connection
handles the actual file transfer. This two-connection approach enables simultaneous control and data
exchange and allows for reliable and efficient file transfers over the FTP protocol.

Data Connection-Security Of FTP-Electronic Mail-Architecture- Web Based Mail- Email Security:

The data connection in FTP is not inherently secure. FTP was designed with minimal security features,
and data transferred over the FTP data connection is transmitted in plain text, making it susceptible to
eavesdropping and unauthorized access. This lack of security is a significant drawback of traditional FTP.

To enhance the security of FTP, several secure alternatives have been developed, such as FTPS (FTP over
SSL/TLS) and SFTP (SSH File Transfer Protocol). FTPS adds SSL/TLS encryption to the FTP protocol,
securing both the control connection and the data connection. SFTP, on the other hand, is a separate
protocol that uses SSH (Secure Shell) for secure file transfers. Both FTPS and SFTP provide encryption

13 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
and authentication, protecting the confidentiality and integrity of data transmitted over the FTP
connection.

Moving on to electronic mail (email), it is a widely used communication method for exchanging
messages and files between individuals or organizations over computer networks. Email follows a client-
server architecture, where an email client (such as Microsoft Outlook or Gmail) connects to an email
server to send and receive messages.

In the case of web-based mail, also known as webmail, the email client is accessed through a web
browser rather than a dedicated desktop application. Webmail services, such as Gmail, Yahoo Mail, or
Outlook.com, provide a user-friendly interface for managing emails, contacts, and other features
directly from a web browser.

Email security is crucial due to the sensitive and confidential nature of email communication. The
following are some measures taken to enhance email security:

1. Transport Layer Security (TLS): TLS is used to encrypt the communication between email clients
and servers, ensuring that messages are transmitted securely. It prevents eavesdropping and
protects the confidentiality of email content.

2. Secure Authentication: Email servers typically use authentication methods, such as

username/password combinations, to verify the identity of users. Strong and unique passwords,
two-factor authentication, and secure login protocols (e.g., OAuth) are employed to enhance
security.

3. Spam and Malware Protection: Email servers and clients incorporate spam filters and antivirus
scanners to detect and filter out spam emails and malicious attachments or links. These
measures help protect users from phishing attempts and malware infections.

4. Digital Signatures and Encryption: Email encryption technologies, such as Pretty Good Privacy
(PGP) and Secure/Multipurpose Internet Mail Extensions (S/MIME), enable the use of digital
signatures and end-to-end encryption. Digital signatures verify the authenticity of the sender,
while encryption ensures that only the intended recipient can decrypt and read the email
content.

Email security is an ongoing challenge, and individuals and organizations need to stay vigilant by
implementing robust security practices, keeping software up to date, and being cautious of suspicious
emails and attachments.

14 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
TELENET-local versus remote Logging:

TELNET is a network protocol that provides a way to remotely log into a computer over a network,
allowing users to access and control remote systems as if they were directly connected to them. When
using TELNET, there are two types of logging: local logging and remote logging.

1. Local Logging: Local logging refers to logging into the local system or computer itself. In this
case, the user is physically present at the computer and interacts with it directly, using the
keyboard and display connected to the system. Local logging does not involve any network
communication or remote access.

For example, when you log into your personal computer or workstation by entering your username and
password on the local login screen, you are engaging in local logging. This type of logging is useful for
accessing and using the computer's resources and applications locally.

2. Remote Logging: Remote logging, on the other hand, involves logging into a remote system or
computer over a network. TELNET facilitates remote logging by establishing a connection
between the client (the computer from which the user initiates the TELNET session) and the
server (the remote computer being accessed).

With remote logging, the user can interact with the remote system as if they were sitting in front of it,
even though they are physically located elsewhere. The user can execute commands, access files, and
perform various operations on the remote system, all through the TELNET connection.

Remote logging via TELNET allows users to access and manage systems remotely, which is particularly
useful for system administration, troubleshooting, and remote access to servers or computers located
in different physical locations.

It's important to note that TELNET is an unencrypted protocol, meaning that the data transmitted
between the client and server is sent in plain text. This lack of encryption makes TELNET insecure for
transmitting sensitive information, such as passwords, over untrusted networks. As a result, TELNET is
often replaced by more secure protocols like SSH (Secure Shell) for remote access and management of
systems. SSH provides encryption and secure communication, ensuring the confidentiality and integrity
of data exchanged during remote sessions.

Domain Name System:

The Domain Name System (DNS) is a distributed hierarchical naming system that translates human-
readable domain names into IP addresses. It serves as a vital component of the internet infrastructure,
enabling users to access websites and other internet resources using easy-to-remember domain names
instead of numeric IP addresses.

Here's how the DNS works:

15 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
1. Domain Names: Domain names are hierarchical, consisting of multiple levels separated by dots.
For example, "www.example.com" is a domain name where "com" is the top-level domain
(TLD), "example" is the second-level domain (SLD), and "www" is a subdomain.

2. DNS Resolution: When a user enters a domain name in a web browser or any other network
application, the DNS resolution process begins to translate the domain name into an IP address.

3. DNS Query: The client (user's device) sends a DNS query to a DNS resolver (typically provided
by the internet service provider or configured manually). The query includes the domain name
that needs to be resolved.

4. Recursive Resolution: The DNS resolver receives the query and begins the recursive resolution
process. If the resolver has the requested domain name's IP address in its cache, it can provide the
response immediately. Otherwise, it proceeds with iterative queries to find the IP address.

5. DNS Hierarchy: The resolver contacts the root DNS servers, which are the top-level servers
responsible for handling queries for the TLDs (e.g., .com, .org, .net). The root DNS servers
respond with a referral to the authoritative DNS servers responsible for the requested TLD.

6. Authoritative DNS Servers: The resolver then contacts the authoritative DNS servers for the TLD
mentioned in the referral. For example, if the requested domain is "example.com," the resolver
contacts the authoritative DNS servers for the .com TLD.

7. Further Resolution: The authoritative DNS servers respond with a referral to the DNS servers
responsible for the requested SLD. The resolver continues this process until it reaches the DNS
servers responsible for the specific domain name being resolved.

8. IP Address Resolution: The authoritative DNS servers for the domain name respond with the IP
address associated with the domain name. The resolver caches this information for future use and
returns the IP address to the client.

9. Communication: The client receives the IP address from the DNS resolver and uses it to establish
a connection with the desired web server or network resource.

The DNS system is distributed, meaning that there are numerous DNS servers worldwide, each
responsible for specific domains or zones. This distributed nature ensures scalability, redundancy, and
improved performance of DNS resolution.

In addition to translating domain names into IP addresses, DNS also supports other record types, such as
MX (Mail Exchanger) records for email routing, CNAME (Canonical Name) records for aliases, and
TXT records for various purposes like domain verification or SPF (Sender Policy Framework)
information.

Overall, the DNS plays a critical role in making the internet accessible by translating user-friendly
domain names into the numeric IP addresses required for communication between devices and servers.

16 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
NAME SPACE:

Name space refers to the logical organization and allocation of names within a system. In the context of
computing and networking, name space commonly refers to the naming system used to identify and
distinguish various entities such as files, directories, network resources, domain names, and more.

Name spaces are designed to provide a structured and hierarchical approach to naming, allowing
unique and meaningful identifiers to be assigned to different entities. By organizing names in a name
space, it becomes easier to manage and locate resources within a system.

Here are a few examples of name spaces:

1. File System Name Space: In an operating system, the file system organizes files and directories
in a hierarchical structure. Each file or directory is assigned a unique name within the file
system's name space. This allows users and applications to reference specific files and
directories using their assigned names.

2. Network Name Space: In networking, the Domain Name System (DNS) provides a hierarchical
name space for mapping domain names to IP addresses. The DNS name space is organized into a
tree-like structure, with top-level domains (TLDs) at the root and subdomains branching out
from there. This enables users to access websites and other network resources using human-
readable domain names.

3. Object-Oriented Name Space: In object-oriented programming, name spaces are used to

prevent naming conflicts between different classes, functions, and variables. Name spaces allow
developers to group related code elements together and provide a unique identifier to each
element within the name space. This helps maintain code organization and avoid naming
collisions.

4. Communication Protocol Name Space: Communication protocols, such as HTTP, FTP, SMTP, and
others, define their own name spaces for different aspects of the protocol. For example, HTTP
defines a name space for HTTP methods (GET, POST, PUT, DELETE), HTTP headers, and status
codes. Each name within the protocol's name space has a specific meaning and is used to convey
information or control the behavior of the protocol.

Name spaces are crucial for maintaining order, uniqueness, and organization within a system. They
provide a structured approach to naming and help ensure that resources can be accessed and managed
efficiently.

DNS IN INTERNET:

In the context of the internet, the Domain Name System (DNS) plays a critical role in translating human-
readable domain names into IP addresses, allowing users to access websites, send emails, and use
various internet services.

17 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Here's how DNS operates in the internet:

1. DNS Hierarchy: The DNS system is organized hierarchically. At the top level, there are root DNS
servers that store information about the top-level domains (TLDs) such as .com, .org, .net, and
country-code TLDs like .uk, .de, etc. Below the root level, there are authoritative DNS servers
responsible for specific domain names and subdomains.

2. DNS Resolution Process: When a user enters a domain name (e.g., www.example.com) into a
web browser or other application, the DNS resolution process begins. The user's device sends a
DNS query to a DNS resolver, which can be provided by the internet service provider or
configured by the user.

3. Recursive Resolution: The DNS resolver receives the query and starts the recursive resolution
process. It first checks its cache to see if it has the IP address for the requested domain name. If
not, the resolver acts as a client and queries the root DNS servers to find the authoritative DNS
servers responsible for the TLD (.com in this case).

4. Authoritative DNS Servers: The root DNS servers respond to the resolver with the IP addresses of
the authoritative DNS servers for the requested TLD. The resolver then sends a query to the
authoritative DNS servers responsible for the TLD (.com) to obtain the IP addresses of the
authoritative DNS servers for the second-level domain (example.com).

5. Further Resolution: The authoritative DNS servers for the second-level domain respond with the
IP address of the authoritative DNS server for the subdomain (www.example.com). The resolver
continues this process until it reaches the authoritative DNS server for the specific domain
name.

6. IP Address Resolution: The authoritative DNS server for the requested domain name responds to
the resolver with the corresponding IP address. The resolver caches this information for future
use.

7. Communication: The resolver provides the IP address to the user's device, which can then
establish a connection with the web server associated with the domain name. This enables the
user to access the desired website or service.

DNS plays a crucial role in the functioning of the internet by providing a decentralized and distributed
naming system. It allows users to access websites and other internet resources using easy-to-remember
domain names, rather than relying on numeric IP addresses. The DNS system ensures efficient and
reliable translation of domain names into IP addresses, enabling seamless communication and access
across the internet.

18 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
RESOLUTION-CACHING:

Resolution and caching are two important aspects of the Domain Name System (DNS) that help improve
the efficiency and performance of DNS queries. Here's a brief explanation of resolution and caching in
DNS:

Resolution: Resolution in DNS refers to the process of translating a domain name into its corresponding
IP address. When a user enters a domain name in a web browser or any other network application, the
DNS resolution process is initiated to find the IP address associated with that domain name. The
resolution process involves querying DNS servers and following a hierarchical lookup until the IP address
is obtained.

Caching: Caching is a mechanism used in DNS to store the results of previous DNS queries for a certain
period of time. When a DNS resolver receives a response for a DNS query, it can store the obtained IP
address or other DNS information in its cache. The next time the same domain name is queried, the
resolver can retrieve the information from its cache instead of performing a new DNS lookup.

Caching provides several benefits:

1. Improved Performance: By caching DNS information, subsequent DNS queries for the same
domain name can be resolved faster since the resolver doesn't need to go through the entire
resolution process again. The cached information can be quickly retrieved and used.

2. Reduced Network Traffic: Caching reduces the number of DNS queries that need to traverse the
network. When a domain name is resolved from a local cache, it eliminates the need for
additional requests to authoritative DNS servers, reducing the network load and improving
network efficiency.

3. Load Distribution: Caching also helps distribute the load on DNS servers. When a resolver caches
DNS information, it can serve subsequent requests for the same domain name locally without
burdening the authoritative DNS servers.

4. Redundancy and Resilience: Caching provides a level of redundancy and resilience in DNS. If an
authoritative DNS server becomes unavailable, cached DNS records can still be used to resolve
domain names until the authoritative server becomes accessible again.

However, it's important to note that caching introduces a potential drawback: the possibility of
accessing outdated or expired DNS information. To mitigate this, DNS records are associated with a
time-to-live (TTL) value, which specifies how long the information should be considered valid. Resolvers
typically adhere to the TTL value and discard cached records once the TTL expires, ensuring that they
fetch fresh information from authoritative DNS servers.

Overall, resolution and caching are integral parts of DNS that help optimize performance, reduce
network traffic, and improve the efficiency of domain name resolution.
19 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN
 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
Resource Records- DNS Messages:

Resource Records: In the Domain Name System (DNS), resource records (RRs) are the fundamental
building blocks that store information about a domain name. Each resource record represents a specific
type of data associated with a domain name and provides information such as the IP address(es)
corresponding to the domain, mail server information, name server information, and more.

The structure of a resource record consists of several fields:

1. Name: The fully qualified domain name (FQDN) to which the resource record pertains.

2. Type: Indicates the type of data stored in the resource record. Some common types include A
(IPv4 address), AAAA (IPv6 address), MX (mail exchange), NS (name server), and CNAME
(canonical name).

3. Class: Specifies the class of the data, commonly IN (Internet).

4. Time to Live (TTL): Indicates the amount of time, in seconds, for which the resource record
should be considered valid before it needs to be refreshed.

5. Data: The actual data associated with the resource record, such as IP addresses, domain names,
or other relevant information.

DNS Messages: DNS messages are used for communication between DNS clients (resolvers) and DNS
servers. These messages contain queries and responses that are exchanged during the domain name
resolution process. DNS messages adhere to a specific format and are typically carried over User
Datagram Protocol (UDP) or Transmission Control Protocol (TCP).

A DNS message consists of a header section and a question/answer section:

1. Header Section: The header section contains information about the message, including flags,
identification number, query type (standard query, response, or other types), and other control
fields.

2. Question/Answer Section: This section contains the actual queries or responses. In a query, the
question section includes the domain name being queried and the desired resource record type.
In a response, the answer section includes the resource records that provide the requested
information.

DNS messages follow a client-server model, where the DNS client sends a query message to a DNS
server, and the server responds with the requested information or an error message if the information
is not available.

The DNS protocol supports recursive and iterative queries. In a recursive query, the DNS resolver sends
the query to a DNS server and expects a complete answer. If the server doesn't have the answer, it will

20 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
recursively contact other servers on behalf of the resolver until it obtains the answer. In an iterative
query, the DNS server provides the best information it has or refers the resolver to another DNS server
for further resolution.

By exchanging DNS messages, DNS clients and servers facilitate the resolution of domain names to IP
addresses and other relevant information, enabling users to access websites, send emails, and use
various internet services based on domain names.

Registrars-Security Of DNS Name Servers:

Registrars: Registrars are organizations or entities responsible for managing and registering domain
names on behalf of individuals or businesses. They are accredited by domain name registries and
provide domain registration services to customers. Registrars interact with the Domain Name System
(DNS) to allocate and maintain domain names in the global namespace.

The role of registrars includes:

1. Domain Registration: Registrars allow individuals or organizations to register and acquire domain
names. They handle the process of verifying domain name availability, collecting registration
information, and updating the DNS records with the registered domain name.

2. DNS Management: Registrars provide tools and interfaces for managing DNS settings associated
with registered domain names. This includes managing name server (NS) records, DNS zone
configuration, and other DNS-related settings.

3. WHOIS Database Management: Registrars maintain and update the WHOIS database, which
contains registration information for domain names. This information includes details such as
the domain owner's contact information, registration dates, and name server information.

Security of DNS Name Servers: DNS name servers play a critical role in resolving domain names to their
corresponding IP addresses and providing other DNS-related information. Ensuring the security of DNS
name servers is crucial to maintaining the integrity and availability of the DNS infrastructure. Here are
some important considerations for the security of DNS name servers:

1. Access Control: DNS name servers should have strict access controls in place to prevent
unauthorized access. This includes implementing strong authentication mechanisms, access
restrictions based on IP addresses or network segments, and proper user privilege management.

2. Patch Management: Keeping DNS servers up to date with the latest security patches is essential
to protect against known vulnerabilities. Regularly monitoring and applying patches helps
mitigate potential security risks.

21 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN

 Prepared By :T.V.GOPALA KRISHNA ,ASSOC PROF & HOD
3. Configuration Hardening: DNS servers should be configured securely following best practices.
This includes disabling unnecessary services, enforcing secure communication protocols (such as
DNS over TLS or DNS over HTTPS), and implementing secure DNS server software configurations.

4. DNSSEC (DNS Security Extensions): DNSSEC is a security extension to the DNS protocol that adds
digital signatures to DNS records, ensuring their authenticity and integrity. Deploying DNSSEC
helps protect against DNS spoofing and data manipulation attacks.

5. Monitoring and Logging: Implementing robust monitoring and logging mechanisms allows for
the detection of unusual or suspicious activities. Monitoring DNS traffic, analyzing logs, and
implementing intrusion detection systems help identify potential security incidents and enable
timely response.

6. DDoS Mitigation: DNS servers are frequently targeted by Distributed Denial of Service (DDoS)
attacks. Implementing DDoS mitigation strategies, such as using traffic filtering, rate limiting, and
employing DNS server redundancy, helps protect against these attacks.

7. Regular Audits and Assessments: Conducting regular security audits and assessments of DNS
infrastructure helps identify vulnerabilities, ensure compliance with security standards, and
implement necessary security improvements.

Ensuring the security of DNS name servers is a continuous effort that requires proactive measures,
regular monitoring, and timely response to security incidents. By implementing robust security
practices, registrars and DNS operators can help maintain the reliability and trustworthiness of the DNS
ecosystem.

22 | P a g e LOYOLA INST IT UT E OF TECHNOLOGY AND MANAGEMENT CN