Module 1️⃣

A periodic signal is decomposed into five sign waves with frequencies of 100, 300, 500, 700
and 900 Hz. What is the bandwidth of the signal? Draw the spectrum assuming all
components have a maximum amplitude of 10 volts. L

The bandwidth of a signal is the range of frequencies that it occupies. In the case of a
periodic signal, it is the difference between the highest frequency component and the
lowest frequency component.

In this case, the highest frequency component is 900 Hz and the lowest frequency
component is 100 Hz. Therefore, the bandwidth of the signal is 900 Hz - 100 Hz = 800
Hz.

What is the need for Analog to Analog modulation? Describe all the possible modulation
techniques?L

This modulation is generally needed when a bandpass channel is required. Bandpass

is a range of frequencies which are transmitted through a bandpass filter which is a
filter allowing specific frequencies to pass, preventing signals at unwanted
frequencies.

What is the role of Network layer and Data Link layer in OSI Reference Model? Why IP in
Network layer is purposely made connection less?L

The network layer has two main functions. One is breaking up segments into network
packets, and reassembling the packets on the receiving end. The other is routing
packets by discovering the best path across a physical network. The network layer
uses network addresses (typically Internet Protocol addresses) to route packets to a
destination node.

The data link layer establishes and terminates a connection between two
physically-connected nodes on a network. It breaks up packets into frames and sends
them from source to destination. This layer is composed of two parts—Logical Link
Control (LLC), which identifies network protocols, performs error checking and
synchronizes frames, and Media Access Control (MAC) which uses MAC addresses to
connect devices and define permissions to transmit and receive data.

IP is connectionless because it does not require a connection between hosts. It does

not sequence, acknowledge, or control the flow of data between hosts. IP treats each
datagram as a separate entity; it merely addresses the datagram and sends it out,
hoping it reaches the destination.

A low-pass signal is sampled with a bandwidth of 400 kHz using 1024 levels of quantization.
a. Calculate the bit rate of the digitized signal.
Bit rate = sampling rate × number of bits per sample
Low pass signal: frequency between 0 – 200 KHz BW = 200 KHz
Sampling rate >= 2 x fhighest = 2 x 200 KHz >= 400,000 samples /s
nb= log2 1024 = 10 bits/sample;
Bit rate = 400,000 x 10 = 4 Mbps
b. Calculate the SNRdB for this signal.
SNRdB = 6.02 nb + 1.76 = 6.02 x 10 + 1.76 = 61.96 dB
c. Calculate the PCM bandwidth of this signal.L
SNRdB = 6.02 nb + 1.76 = 6.02 x 10 + 1.76 = 61.96 dB

Match the following to one or more layers of the OSI model:

a. Reliable process-to-process message delivery Transport Layer
b. Route selection Network Layer
c. Defines frames Data link Layer
d. Provides user services such as e-mail and file transfer Application Layer
e. Transmission of bit stream across the physical medium Physical Layer

Show the Differential Manchester encoding for the bit pattern given below.
(i) 01001100
(ii) 10110011 M
(ans kiye ho to dm Nehal Bhuyan / WhatsApp)

In a noiseless channel with a bandwidth of 9000 hz transmitting a signal with two signal
levels. Calculate the bit rate? S

Consider a noiseless channel with a bandwidth of 9000 Hz transmitting a signal with

two signal levels. The maximum bit rate can be calculated as Bit Rate =2 x 9000 x log2
2 = 18000 bps. (Bit Rate = 2 x bandwidth x l0g2 L)

If the data link layer can detect errors between hops, why do you think we need another
checking mechanism at the transport layer?

The errors between the nodes can be detected by the data link layer control, but the
error at the node (between input port and output port) of the node cannot be detected
by the data link layer.
Suppose a computer sends a packet at the transport layer to another computer somewhere
on the Internet. There is no process with the destination port address running at the
destination computer. What will happen? S

Most protocols issue a special error message that is sent back to the source in this
case.

Define a DC component and its effect on digital transmission. S

When the voltage level in a digital signal is constant for a while, the spectrum creates
very low frequencies, called DC components, which present problems for a system
that cannot pass low frequencies.

A file contains 2 million bytes. How long does it take to download this file using a 56-Kbps
channel? M

With a 56-Kbps channel, it takes 16,000,000/56,000 = 289 s ≈ 5 minutes. (Convert

bytes to bits)

What is the need for Analog to Analog modulation? Describe all the possible modulation
techniques? L

Analog to Analog Conversion (Modulation) - GeeksforGeeks (Link)

For 6 devices in a network, what is the number of cable links required for a mesh, ring and
star topology? S

● n(n-1)/2 cable links are required for mesh 6(6-1)/2 = 15

● n for ring 6
● n-1 cable link for bus 5
● n cable link for star topology. 6

A signal travels from point A to point B. At point A, the signal power is 200 W. At point B, the
power is 170 W. What is the attenuation in decibels? M

For, 10 log10 (200/170) = 0.7058085

Similar qsn:
Encode the given data word 101110101 using RZ, NRZ-L, NRZ-I, Manchester and
differential Manchester schemes. L
(ans kiye ho to dm Nehal Bhuyan / WhatsApp)

Show the Manchester encoding for the bit pattern given below.
(i) 01001100
(ii) 10110011 M
(ans kiye ho to dm Nehal Bhuyan / WhatsApp)

Draw the OSI Model and explain the functions of different layers. L

The Open Systems Interconnection (OSI) model describes seven layers that computer
systems use to communicate over a network.

7. Application Layer

The application layer is used by end-user software such as web browsers and email
clients. It provides protocols that allow software to send and receive information and
present meaningful data to users. A few examples of application layer protocols are
the Hypertext Transfer Protocol (HTTP), File Transfer Protocol (FTP), Post Office
Protocol (POP), Simple Mail Transfer Protocol (SMTP), and Domain Name System
(DNS).
6. Presentation Layer

The presentation layer prepares data for the application layer. It defines how two
devices should encode, encrypt, and compress data so it is received correctly on the
other end. The presentation layer takes any data transmitted by the application layer
and prepares it for transmission over the session layer.

5. Session Layer

The session layer creates communication channels, called sessions, between

devices. It is responsible for opening sessions, ensuring they remain open and
functional while data is being transferred, and closing them when communication
ends. The session layer can also set checkpoints during a data transfer—if the
session is interrupted, devices can resume data transfer from the last checkpoint.

4. Transport Layer

The transport layer takes data transferred in the session layer and breaks it into
“segments” on the transmitting end. It is responsible for reassembling the segments
on the receiving end, turning it back into data that can be used by the session layer.
The transport layer carries out flow control, sending data at a rate that matches the
connection speed of the receiving device, and error control, checking if data was
received incorrectly and if not, requesting it again.

3. Network Layer

The network layer has two main functions. One is breaking up segments into network
packets, and reassembling the packets on the receiving end. The other is routing
packets by discovering the best path across a physical network. The network layer
uses network addresses (typically Internet Protocol addresses) to route packets to a
destination node.

2. Data Link Layer

The data link layer establishes and terminates a connection between two
physically-connected nodes on a network. It breaks up packets into frames and sends
them from source to destination. This layer is composed of two parts—Logical Link
Control (LLC), which identifies network protocols, performs error checking and
synchronizes frames, and Media Access Control (MAC) which uses MAC addresses to
connect devices and define permissions to transmit and receive data.
1. Physical Layer

The physical layer is responsible for the physical cable or wireless connection
between network nodes. It defines the connector, the electrical cable or wireless
technology connecting the devices, and is responsible for transmission of the raw
data, which is simply a series of 0s and 1s, while taking care of bit rate control.

State FDM with a neat diagram. M

In FDM, the total bandwidth is divided to a set of frequency bands that do not
overlap. Each of these bands is a carrier of a different signal that is generated and
modulated by one of the sending devices. The frequency bands are separated from
one another by strips of unused frequencies called the guard bands, to prevent
overlapping of signals.
The modulated signals are combined together using a multiplexer (MUX) in the
sending end. The combined signal is transmitted over the communication channel,
thus allowing multiple independent data streams to be transmitted simultaneously.
At the receiving end, the individual signals are extracted from the combined signal
by the process of demultiplexing (DEMUX).

Draw the encoded signal for the NRZ-L and NRZ-I Scheme of the following bits;
001101101 M

(ans kiye ho to dm Nehal Bhuyan / WhatsApp)

What are the key elements of a protocol? S

The key elements of a protocol are syntax, semantics and timing.

Define distortion. S

Distortion means changes in the form or shape of the signal. This is generally seen in
composite signals made up with different frequencies.
In a noiseless channel with a bandwidth of 6000 hz transmitting a signal with two signal
levels. Calculate the bit rate? S

Consider a noiseless channel with a bandwidth of 6000 Hz transmitting a signal with

two signal levels. The maximum bit rate can be calculated as Bit Rate =2 x 6000 x log2
2 = 12000 bps. (Bit Rate = 2 x bandwidth x l0g2 L)

What is the difference between half-duplex and full-duplex transmission modes? S

A half-duplex transmission could be considered a one-way street between sender and

receiver. Full-duplex, on the other hand, enables two-way traffic at the same time.

Explain different types of topology with their advantages and disadvantages. L

Advantages and disadvantages of topologies (Link)

Differentiate between TDM and FDM. M

 Module 2️⃣

Given an account of the frame format of I-frame and S-frame in HDLC protocol, describing
the function of each field. L
I-frame (Information frame) in HDLC has the following fields:

Flag: marks the beginning and end of the frame

Address: contains the device address
Control: contains the frame type and control information
Information: contains the data being sent
Frame Check Sequence (FCS): used for error detection
Flag: marks the end of the frame
S-frame (Supervisory frame) in HDLC has the following fields:

Flag: marks the beginning and end of the frame

Address: contains the device address
Control: contains the frame type and control information specific to S-frame like
sequence numbers, Receiver ready/not ready etc
Information field is absent
Frame Check Sequence (FCS): used for error detection
Flag: marks the end of the frame

Difference between CSMA and CSMA/CD. M

What is bit stuffing? Bit stuff the following data:

0001111101111110011 S

Bit stuffing is a technique that adds extra bits to data to prevent control characters
confusion. To bit stuff the data 0001111101111110011, add a 0 after every 5
consecutive 1s, resulting in 000111110111110010011.
Draw the phase transition diagram for PPP. Distinguish between PAP and CHAP. L
What do you mean by multiple accesses? Briefly discuss the operations of CSMA/CD and
CSMA/CA random access methods. L

Multiple access refers to the ability of multiple devices to share a communication

channel and access it simultaneously. In a multiple access network, a communication
channel is shared by multiple devices, and the devices need a method to decide when
to transmit and when to listen.

CSMA/CD (Carrier Sense Multiple Access with Collision Detection) and CSMA/CA
(Carrier Sense Multiple Access with Collision Avoidance) are two different random
access methods used in LANs.

CSMA/CD: In this method, devices listen to the channel before transmitting. If the
channel is busy, the device waits for a random amount of time and then listens again.
If the channel is still busy, the device waits again. If the channel is idle, the device can
transmit. If a collision occurs, the devices involved in the collision sense the collision
and stop transmitting, and the colliding devices wait for a random amount of time
before trying to transmit again.

CSMA/CA: In this method, devices listen to the channel before transmitting. If the
channel is busy, the device waits for a random amount of time and then listens again.
If the channel is still busy, the device waits again. If the channel is idle, the device can
transmit. If a collision occurs, the devices involved in the collision sense the collision
and stop transmitting. Instead of waiting for a random amount of time before trying to
transmit again like CSMA/CD, devices in CSMA/CA use a random back-off algorithm
to decide when to try to transmit again.

In summary, both CSMA/CD and CSMA/CA are multiple access methods where
devices listen to a channel before transmitting, but CSMA/CD uses collision detection
while CSMA/CA uses collision avoidance to handle the collision.

Define HDLC. Discuss about the different formats of HDLC frame with neat diagrams. L

High-Level Data Link Control (HDLC) generally provides flexibility to simply support
all options that are possible in various data transfer modes and configurations. To
provide flexibility, HDLC basically uses and explains three different types of frames.
Type of frame is basically determined by the control field of the frame. Each type of
frame generally serves as an envelope for transmission of various types of messages.
These three different classes of frames used in HDLC are given below.

1. I-frame : I-frame stands for Information frames. This frame is generally used for
transporting user data from the network layer. These frames actually carry
actual data or information of the upper layer and some control information.
This frame carries data along with both a send sequence number and an
acknowledgment number. It can also be used to piggyback acknowledgement
 information in case of ABM (Asynchronous Balanced Mode). The first bit of
this frame of the control field is 0.

2. S-frame : S-frame stands for Supervisory frames. These frames are basically
required and essential for error control and flow control. They also provide
control information. It contains or includes only an Acknowledgment number.
First two bit of this frame of the control field is 10. S-frame does not have any
information fields. This frame contains send and receive sequence numbers.

3. U-frame : U-frame stands for Unnumbered frames. These frames are also
required in various functions like link setup and disconnections. These frames
basically support control purposes and are not sequenced. First, two-bit of this
frame of the control field are 11. Some U-frame contain an information field
depending on the type. These frames are also used for different miscellaneous
purposes along with link management. U-frame is required for managing the
link itself. This frame does not include any type of acknowledgment
information i.e. in turn it includes or is contained in sequence number. These
frames are generally reserved for system management.

How TDM differs from FDM. What are the types of TDM? Explain each of them in detail. L

TDM (Time Division Multiplexing) and FDM (Frequency Division Multiplexing) are both
multiplexing techniques that allow multiple signals to share a single communication
channel. The main difference between them is the way they divide the channel into
smaller units. TDM divides the channel into time slots, while FDM divides the channel
into frequency bands. TDM assigns each device a specific time slot, while FDM
assigns each device a specific frequency band. TDM is efficient and predictable but
less flexible than FDM. FDM is more flexible but less efficient than TDM.
Synchronous TDM: In this type of TDM, the time slots are fixed and of equal length.
Each device is assigned a specific time slot, and it can only transmit during its
assigned time slot. The devices are synchronized with a common clock signal, and
the time slots are repeated periodically. This type of TDM is typically used in digital
networks such as T1 and E1.

Asynchronous TDM: In this type of TDM, the time slots are not fixed and may vary in
length. Each device is assigned a time slot based on its data rate, and it can transmit
whenever it has data to send. The devices are not synchronized with a common clock
signal, and the time slots are not repeated periodically. This type of TDM is typically
used in analog networks such as voice over telephone lines.

In variable-size framing, what are the ways there are to define the end of one frame and the
beginning of the next frame? S

In variable-size framing, there are several ways to define the end of one frame and the
beginning of the next frame, such as:

Using a special character or sequence of characters as a delimiter between frames.

Using a fixed number of idle bits or a predefined pattern of bits between frames.
Using a specific bit or field in the header of the frame to indicate the end of the frame.
Using the length field in the header of the frame to indicate the end of the frame.

What is the purpose of Virtual Circuit Identifier? S

The Virtual Circuit Identifier (VCI) is a unique identifier assigned to a virtual circuit that
is used to identify it among other virtual circuits in a network. It is used by a switch or
router to identify the specific virtual circuit to which a data packet belongs to and
forward it to the correct destination.

Explain the mechanism of Stop and Wait ARQ for different operations and how selective
Repeat ARQ differs from it. L

Stop-and-Wait ARQ is a type of Automatic Repeat Request (ARQ) error control

method used to ensure reliable data transfer over a noisy communication channel.
The mechanism of Stop-and-Wait ARQ is as follows:

● Data transmission: The sender sends a single frame of data at a time and waits
for an acknowledgement (ACK) from the receiver.
● Acknowledgement: The receiver receives the frame and sends an ACK to the
sender.
● Timeout: If the sender does not receive an ACK within a certain time period
(timeout), it retransmits the same frame.
● Frame acceptance: If the receiver receives the same frame multiple times, it
accepts only the first copy and discards the rest.
 ● Frame error: If the receiver detects an error in the received frame, it sends a
negative acknowledgement (NAK) to the sender, and the sender retransmits
the same frame.

The main difference between Stop-and-Wait ARQ and Selective Repeat ARQ is that in
the later, the receiver can selectively retransmit lost or corrupted frames, and the
sender retransmits only the lost or corrupted frames, not all the frames. This makes
Selective Repeat ARQ more efficient than Stop-and-Wait ARQ.

Demonstrate the mechanism used in Go-Back –N ARQ for normal and damaged/lost frame
operations. L

Go-Back-N ARQ is a type of Automatic Repeat Request (ARQ) error control method
that is similar to selective repeat ARQ, but it allows the sender to transmit multiple
frames before waiting for an acknowledgement (ACK) from the receiver. The
mechanism of Go-Back-N ARQ is as follows:

1. Data transmission: The sender sends a certain number of frames (N) before
waiting for an acknowledgement from the receiver. These frames are
sequentially numbered and the sender keeps track of the highest
acknowledged number, which is called the Next Expected Acknowledge (NEA).
2. Acknowledgement: The receiver receives the frames and sends an ACK for
each successfully received frame. The ACK contains the number of the next
expected frame.
3. Frame acceptance: If the receiver receives a duplicate frame, it discards it and
sends an ACK for the previously acknowledged frame.
4. Frame error: If the receiver detects an error in a received frame, it discards it
and sends a negative acknowledgement (NAK) for the frame number that
contains the error.
5. Lost frame: If the receiver doesn't receive a frame, it sends a NAK for the next
expected frame.
6. Timeout: If the sender doesn't receive an ACK or NAK within a certain time
period (timeout), it retransmits all the frames starting from the frame number
specified in the NAK or the NEA.

In case of lost frame or damaged frame, the sender retransmits the lost frame or the
damaged frame after receiving a NAK for that frame. The sender also retransmits all
the frames after the lost or damaged frame until the Next Expected Acknowledge
(NEA) or the highest acknowledged number.

In summary, Go-Back-N ARQ allows the sender to transmit multiple frames before
waiting for an acknowledgement, this makes it more efficient than Stop-and-Wait
ARQ, but less efficient than Selective Repeat ARQ as it retransmits all frames after the
lost or damaged frame.
List down different techniques of error detection. S

Different techniques of error detection include:

● Parity check
● Cyclic Redundancy Check (CRC)
● Checksum
● Hash functions
● Forward Error Correction (FEC)
● Hamming Code
● Error-Correcting Code (ECC)

What are the two main methods of error correction? S

The two main methods of error correction are:

● Forward Error Correction (FEC): It is a method where the sender adds
redundant data to the original data, before transmitting it, so that the receiver
can use this redundant data to detect and correct errors.
● Automatic Repeat Request (ARQ): It is a method where the receiver sends an
acknowledgement (ACK) to the sender after receiving a frame of data, if the
frame is received without errors. If the receiver detects an error in the received
frame, it sends a negative acknowledgement (NAK) to the sender, so that the
sender can retransmit the frame.

Differentiate between error detection and error correction. S

Error detection and error correction are both methods used to ensure the integrity of
data during transmission over a noisy communication channel.
● Error detection: is a method used to detect errors that occur during data
transmission. It does this by adding extra bits, called redundancy bits, to the
data, which can be used to check whether errors have occurred or not.
● Error correction: is a method used to correct errors that occur during data
transmission. It does this by using redundant information added to the data,
called error-correcting code, to detect and correct errors in the data.

What are the responsibilities of the data link layer? S

The data link layer is responsible for providing reliable data transfer between devices
on a single link. It provides error detection and correction, flow control, and media
access control. In short, the data link layer ensures that data is transmitted correctly
and reliably between devices on a local network.
Differentiate between circuit switching and packet switching. M

What is the operation of CRC? Given a 10-bit sequence 1011001011 and a divisor of 1101,
find the CRC. Verify your answer.

CRC (Cyclic Redundancy Check) is an error detection method that uses a

mathematical algorithm to generate a checksum, called a CRC value, based on the
data being transmitted. The receiver uses the same algorithm to calculate the CRC
value of the received data and compares it to the transmitted CRC value. If the values
match, the data is assumed to be error-free; otherwise, an error is detected.

Construct the Hamming code for the bit sequence 110011001. L

The Hamming Code is a type of error-correcting code that can detect and correct
errors in a bit sequence. To construct the Hamming code for the bit sequence
110011001, the following steps are performed:
1. Find the number of redundant bits, r, needed to make the total number of bits,
n, equal to 2^r - 1.
In this case, we need to add 3 redundant bits to the 9 data bits.
2. Place the redundant bits in positions that correspond to powers of 2. This is
called parity bit positioning. In this case, we will place the redundant bits in
positions 1, 2 and 4.
3. Calculate the value of the redundant bits by applying the parity bit formula.
4.
The redundant bits are calculated as follows:

Parity bit 1: (1, 3, 5, 7, 9) = 1 ⊕ 0 ⊕ 0 ⊕ 1 ⊕ 1 = 1

Parity bit 2: (2, 3, 6, 7) = 1 ⊕ 0 ⊕ 0 ⊕ 1 = 0
Parity bit 4: (4, 5, 6, 7) = 0 ⊕ 1 ⊕ 0 ⊕ 1 = 0
The final Hamming code for the bit sequence 110011001 is 110110001, where the first,
second and fourth positions are the redundant bits.

The receiver can check if a transmission error has occurred by comparing the
received Hamming code and the calculated Hamming code. If they don't match, it can
use the redundant bits to correct the error.

How fiber optic cable differs from coaxial cable? M

Differentiate among UTP and STP cable. M
 Module 3️⃣

What is dotted decimal notation in IPv4? How many numbers of bytes are required to
represent an address in dotted decimal notation? S

Dotted decimal notation is a human-readable representation of IPv4 addresses that

separates the four octets of the IP address with a period. Each octet is represented by
a decimal number between 0 and 255, and the IP address is written as four decimal
numbers separated by periods.
Each number of bytes required to represent an address in dotted decimal notation is 4
bytes.

For a given IP address 192.168.1.0/26, L

i. Find the subnet mask.
ii. How many subnets will the subnet mask provide?
iii. What is the block size for a subnet mask?
iv. What are the valid subnets?
v. How many valid hosts are available per subnet?
vi. What is the broadcast address of each subnet?
vii. What is the network address of each subnet?

For the given IP address 192.168.1.0/26,

i. The subnet mask for this IP address is 255.255.255.192, which is derived from
the number of bits in the subnet mask (26) and the subnet mask notation (CIDR
notation)
ii. With a subnet mask of 255.255.255.192, it will provide 64 subnets.
iii. The block size for this subnet mask is 64.
iv. Valid subnets are
192.168.1.0
192.168.1.64
192.168.1.128
192.168.1.192
v. There are 62 valid hosts available per subnet
vi. The broadcast address of each subnet is the last IP of the subnet block. For
example, the broadcast address of subnet 192.168.1.0 is 192.168.1.63, the
broadcast address of subnet 192.168.1.64 is 192.168.1.127 and so on.
vii. The network address of each subnet is the first IP of the subnet block. For
example, the network address of subnet 192.168.1.0 is 192.168.1.0, the network
address of subnet 192.168.1.64 is 192.168.1.64 and so on.
Generate the shortest path tree for the given topology using the link-state routing protocol,
taking E as the root node. (incomplete question)

Link State Routing in computer networks || Link state routing algorithm || Computer Net…

3.6 Dijkstra Algorithm - Single Source Shortest Path - Greedy Method

Distinguish between multicast and broadcast. S

Broadcast sends to all devices on a network, Multicast sends to a specific group of

devices on a network.

Differentiate between Link state routing and Distance vector routing. L

Why does RIP use UDP instead of TCP?

RIP (Routing Information Protocol) is a distance-vector routing protocol that is used to

distribute routing information within a network. It uses UDP (User Datagram Protocol)
instead of TCP (Transmission Control Protocol) for several reasons:

1. Low overhead: UDP is a connectionless, simple protocol that does not require the
overhead of establishing and maintaining a connection, as is necessary with TCP.
This makes it more efficient for applications that need to send a large number of
small packets, such as routing protocols.
 2. Less delay: Because UDP does not require the overhead of establishing and
maintaining a connection, it incurs less delay than TCP. This is important for
real-time applications like routing protocols, where delay can cause problems.
3. No congestion control: UDP does not provide any mechanism for congestion
control, which makes it more suitable for routing protocols, where the goal is to
quickly propagate routing information regardless of network congestion.
4. Broadcasting: UDP allows for broadcasting packets to all devices on a network,
which makes it a good fit for routing protocols that need to send information to all
routers in a network.

In summary, using UDP for RIP allows for more efficient, less delayed and faster
communication, it also allows for broadcasting packets.

What is a mask in IPv4 addressing? What is a default mask in IPv4 addressing?

A mask in IPv4 addressing is a 32-bit value that is used to specify which portion of an
IP address represents the network address and which portion represents the host
address. The mask is typically represented in dotted decimal notation (e.g.
255.255.255.0) and is used in conjunction with an IP address to determine the network
and host addresses.

A default mask is the mask value that is used when no mask is explicitly specified.
The default mask is determined by the class of the IP address. The three classes of IP
addresses (A, B, and C) have different default masks.
● Class A IP addresses have a default mask of 255.0.0.0, the first octet is used
for the network address and the remaining three octets are used for the host
address.
● Class B IP addresses have a default mask of 255.255.0.0, the first two octets
are used for the network address and the remaining two octets are used for the
host address.
● Class C IP addresses have a default mask of 255.255.255.0, the first three
octets are used for the network address and the remaining octet is used for the
host address.
It's worth noting that the default mask method of IP addressing is now considered as
an obsolete method and replaced by CIDR (classless inter-domain routing) which
allows for more flexible and fine-grained IP address allocation.

A block of addresses is granted to a small organization. If one of the addresses is

205.16.37.39/28. Find the class, net-id, host-id, first address, last address and the number of
addresses in the block. L

● The IP address is 205.16.37.39/28, which means it is a CIDR notation.

● The class of this IP address is Class C, which means the first three octets of the
address (205.16.37) represent the network address, and the last octet (39)
represents the host address.
 ● The net-id of this IP address is 205.16.37.32, the net-id is the network address of
the IP block, and it is obtained by setting the host bits of the IP address to 0.
● The host-id of this IP address is 39, the host-id of the IP address is the last octet of
the IP address which is represented by the host bits of the IP address
● The first address of the block is 205.16.37.32, it is the same as the net-id
● The last address of the block is 205.16.37.47, it is obtained by setting the host bits
of the last IP address of the block to 1.
● The number of addresses in the block is 16, it is calculated by 2^(32-28) = 2^4 = 16
● The block of addresses is a Class C block, the net-id is 205.16.37.32, the host-id is
39, the first address of the block is 205.16.37.32, the last address of the block is
205.16.37.47, and the block contains 16 addresses.

In a block of addresses, we know the IP address of one host is 182.44.82.16/26. What is the
first address (network address) and the last address in this block? M

The IP address is 182.44.82.16/26, which means it is a CIDR notation.

The network address (also known as the first address) of this block is 182.44.82.0, it is
obtained by setting the host bits of the IP address to 0.

The last address of this block is 182.44.82.63, it is obtained by setting the host bits of
the last IP address of the block to 1.

The total number of addresses in the block can be calculated by 2^(32-26) = 2^6 = 64

It's worth noting that the address 182.44.82.16/26 belongs to the block, and it's the
first host address in the block and it's not the network address, the network address
also known as the first address is 182.44.82.0.

Write the following masks in slash notation (/n).

a. 255.255.255.0
b. 255.0.0.0
c. 255.255.224.0 M

a. The mask 255.255.255.0 can be written in slash notation as /24

b. The mask 255.0.0.0 can be written in slash notation as /8
c. The mask 255.255.224.0 can be written in slash notation as /19

In the slash notation, the number after the slash represents the number of 1's in the
binary representation of the mask.

It's worth noting that, the mask 255.255.255.0 is a Class C mask, 255.0.0.0 is a Class A
mask, and 255.255.224.0 is a Class B mask, they are all used in CIDR (classless
inter-domain routing) which allows for more flexible and fine-grained IP address
allocation.
Find the range of addresses in the following blocks.
a. 123.56.77.32/29
b. 200.17.21.128/27
c. 17.34.16.0/23 M

a. The IP address 123.56.77.32/29 is a CIDR notation, The net-id of this IP address

is 123.56.77.32, and the range of addresses in this block is 123.56.77.32 -
123.56.77.39.
b. The IP address 200.17.21.128/27 is a CIDR notation, The net-id of this IP
address is 200.17.21.128, and the range of addresses in this block is
200.17.21.128 - 200.17.21.159
c. The IP address 17.34.16.0/23 is a CIDR notation, The net-id of this IP address is
17.34.16.0, and the range of addresses in this block is 17.34.16.0 - 17.34.17.255

It's worth noting that the net-id is the first IP address of the block, and the last IP
address is obtained by setting the host bits to 1. The total number of addresses in the
block can be calculated by 2^(32-n) where n is the number of bits in the mask.

Given a fragmented datagram (in IPv4) with an offset of 120, how can you determine the first
and last byte numbers? S

The first byte number is 960 (120 * 8) and the last byte number can be calculated as
66495 (65535 + first byte number) if it's the last fragment.

State protocol stack of PPP. M

PPP (Point-to-Point Protocol) is a protocol stack that is used to establish a

point-to-point link between two devices, such as a router and a remote host. The
protocol stack of PPP includes the following layers:

Physical Layer: This is the lowest layer of the PPP protocol stack and it is responsible
for establishing and maintaining the physical link between the two devices.

Data Link Layer: This layer is responsible for providing reliable data transfer across
the physical link. It includes the following sublayers:
a. Link Control Protocol (LCP): This sublayer is responsible for establishing,
configuring, and maintaining the PPP link.
b. Authentication Protocol (AP): This sublayer is responsible for authenticating
the devices at each end of the link.
c. Link Quality Report (LQR): This sublayer is responsible for monitoring the
link quality.

Network Layer: This layer is responsible for providing network-layer services to the
PPP link. It includes the following sublayers:
a. Internet Protocol Control Protocol (IPCP): This sublayer is responsible for
configuring the IP address and other parameters for the PPP link.
 b. IPv6 Control Protocol (IPv6CP): This sublayer is responsible for configuring
the IPv6 address and other parameters for the PPP link.

Higher-layer protocols: PPP provides a mechanism for encapsulating higher-layer

protocols such as IP, IPv6, IPX, AppleTalk, and others, to be transported over the PPP
link.

In short, PPP protocol stack includes Physical, Data Link, Network, and Higher-layer
protocols to establish and maintain a point-to-point link

What is the range of private IPs for class A and class B?

Private IPs for class A: 10.0.0.0 - 10.255.255.255

Private IPs for class B: 172.16.0.0 - 172.31.255.255

How can you find the first IP, last IP and number of nodes in a network, if the mask is given
as a 32 bit IP sequence? S

To find the first IP, last IP, and number of nodes in a network, you need to know the IP
address and mask, and perform bitwise AND operation between the IP address and
the mask.
To find the number of nodes, you have to subtract the number of host bits from 32,
then raise 2 to that power.

Find the class, net-id and host-id of the following IP address: 188.25.45.48 S

The IP address 188.25.45.48 is Class B, net-id is 188.25.0.0 and host-id is 0.0.45.48

 Module 4️⃣

Briefly explain the "Three way handshaking" process in a TCP connection. L

The "Three-way Handshaking" process, also known as the "TCP Three-way

handshake" is the process used to establish a reliable, connection-oriented session
between two systems using TCP (Transmission Control Protocol). The process
involves the exchange of three messages between the two systems, as follows:

1. The first message, called the "SYN" (Synchronize) message, is sent by the
initiating system (the client) to the receiving system (the server). The SYN
message contains a randomly generated initial sequence number, which will be
used to number the bytes of data sent during the session.
2. The second message, called the "SYN-ACK" (Synchronize-Acknowledgment)
message, is sent by the receiving system (the server) to the initiating system
(the client). The SYN-ACK message contains the server's own randomly
generated initial sequence number and an acknowledgment of the client's
initial sequence number.
3. The third message, called the "ACK" (Acknowledgment) message, is sent by
the initiating system (the client) to the receiving system (the server). The ACK
message contains the client's own acknowledgment of the server's initial
sequence number.

Once the three-way handshake is complete, a reliable, connection-oriented session is

established between the two systems, and data can be exchanged using the agreed
upon sequence numbers. The session will be closed with a similar process in which
the closing party sends a message called FIN, the other party ACK's it and then the
closing party ACKs the ACK.

The three-way handshake process is important to ensure that both systems are ready
to communicate and that the connection is established in a reliable and secure
manner.

The following is the content of a TCP header in hexadecimal format. L

04720017 00000001 00001000 400207FF 0000000F

a. What is the source port number?

b. What is the destination port number?
c. What is the sequence number?
d. What is the acknowledgement number?
e. What is the length of the header?
f. What is the type of the segment?
g. What is the window size?

a. The source port number is located in the first 2 bytes of the TCP header (04 72
in hexadecimal), which corresponds to a decimal value of 1170.
 b. The destination port number is located in the next 2 bytes of the TCP header
(00 17 in hexadecimal), which corresponds to a decimal value of 23.
c. The sequence number is located in the next 4 bytes of the TCP header (00 00
00 01 in hexadecimal), which corresponds to a decimal value of 1.
d. The acknowledgment number is located in the next 4 bytes of the TCP header
(00 00 00 08 in hexadecimal), which corresponds to a decimal value of 8.
e. The length of the header can be calculated from the HLEN field (4 bits) in the
fourth byte of the TCP header (40 in hexadecimal), which corresponds to a
decimal value of 4. So, the length of the header is 4 x 4 bytes = 16 bytes.
f. The type of the segment cannot be determined from the given information. The
type of the segment is conveyed by the values of the flags and is determined
by the combination of the 6 control bits (URG, ACK, PSH, RST, SYN, FIN) found
in the flags field in the TCP header.
g. The window size is located in the next 2 bytes of the TCP header (07 FF in
hexadecimal), which corresponds to a decimal value of 2047.

What role does ICMP play for an IP packet? What are the various types of ICMP packets
that are sent back to the sender?

ICMP (Internet Control Message Protocol) is a network protocol that is used to send
error messages and operational information about network conditions. ICMP is
typically used by IP (Internet Protocol) to provide feedback about the success or
failure of an IP packet's journey through the network.

The role of ICMP for an IP packet is to provide a mechanism for the sender to receive
information about the state of the network, such as whether a destination host is
reachable or if a packet has been lost. ICMP messages are typically generated by
network devices, such as routers, in response to errors or exceptional conditions that
occur when processing an IP packet.

There are several types of ICMP packets that are sent back to the sender, including:

1. Echo Request/Reply (ping): This type of ICMP packet is used to test the
reachability of a host and measure the round-trip time for packets to travel
from the source host to the destination host and back.

2. Destination Unreachable: This type of ICMP packet is sent back to the sender
when a destination host or network is unreachable. This can occur due to a
variety of reasons, such as a network being down or a host being offline.

3. Time Exceeded: This type of ICMP packet is sent back to the sender when a
packet has been discarded because it exceeded the maximum time allowed for
it to traverse the network.
 4. Parameter Problem: This type of ICMP packet is sent back to the sender when
an error has been encountered in the header of an IP packet.

5. Redirect: This type of ICMP packet is sent back to the sender when a more
efficient route for the packet's destination is available.

6. Echo Request/Reply (ping) and Timestamp Request/Reply: These types of

ICMP packet are used to determine the round-trip time and to measure the
clock offset between the sender and the destination.

Compare the TCP header and the UDP header. List the fields in the TCP header that are
missing from the UDP header.

The UDP header typically includes the following fields:

● Source port: Identifies the source of the datagram
● Destination port: Identifies the destination of the datagram
● Length: Indicates the total length of the UDP header and data
● Checksum: Verifies the integrity of the header and data

The fields that are missing from the UDP header compared to the TCP header are:
● Sequence number
● Acknowledgment number
● Data offset
● Flags
● Window
● Urgent pointer
● Options
In short, the UDP header is simpler and shorter than the TCP header, it does not have
the features of reliability, flow control, and error correction that TCP provides.
Differentiate among UDP and TCP.

The following is a dump of a UDP header in hexadecimal format.

0632000DOO lCE217
a. What is the source port number?
b. What is the destination port number?
c. What is the total length of the user datagram?
d. What is the length of the data?

a. The source port number is located in the first 2 bytes of the UDP header (06 32
in hexadecimal), which corresponds to a decimal value of 1570.
b. The destination port number is located in the next 2 bytes of the UDP header
(00 0D in hexadecimal), which corresponds to a decimal value of 13.
c. The total length of the user datagram is located in the next 2 bytes of the UDP
header (OO in hexadecimal) which corresponds to a decimal value of OO.
However, the given hexadecimal dump is not complete and it seems to be a
typo, it doesn't have the correct format of a hexadecimal representation, thus
it's impossible to extract the value of the total length.
d. The length of the data can't be extracted from the provided hexadecimal dump
since we don't have the total length of the user datagram and we can't
calculate it by subtracting the length of the header from the total length of the
user datagram.
In TCP, if the value of HLEN is 0111, how many bytes of option is included in the segment?
What is the significance of the "Sequence number" and "Acknowledgement number" field in
a TCP segment?

In TCP (Transmission Control Protocol), the HLEN (Header Length) field is 4 bits long
and it specifies the length of the TCP header in 32-bit words.
If the value of HLEN is 0111, it corresponds to a decimal value of 7. This means that
the length of the TCP header is 7 32-bit words, or 7 x 4 = 28 bytes.

So, if the value of HLEN is 0111, the number of bytes of options included in the
segment is 28 bytes - 20 bytes (the fixed size of the TCP header without options) = 8
bytes.

Sequence number
This 32 bit field defines the number assigned to the first byte of data contained in this
segment.
As TCP is a stream transport protocol, to ensure connectivity, each byte to be
transmitted is numbered.
The sequence number tells the destination which byte in this sequence is the first
byte in the segment.
Acknowledgment number
This 32 bit field defines the byte number that the receiver of the segment is expecting
to receive from the other party.
If the receiver of the segment has successfully received byte number x from the other
party, it returns x+1 as the acknowledgment number. Acknowledgement and data can
be piggybacked together.

Define a Socket? What role does it play in data delivery across multiple networks?

A socket is an endpoint for sending or receiving data across a computer network. It is

a combination of an IP address and a port number, used to identify a specific process
running on a host.
The role of sockets in data delivery across multiple networks is to act as a bridge
between the application layer and the network layer. Sockets allow applications to
send and receive data using a standard interface, regardless of the underlying
network protocol. This means that an application can use the same socket code to
communicate over different networks, such as TCP/IP or UDP.
Sockets also enable communication between different computers on a network, by
providing a way to identify and address the endpoint of a connection. They enable the
use of multiple networks such as IP and other protocols to be used together, to form a
communication channel.
 Module 5️⃣

Explain the architecture of the Domain Name System(DNS) to map the domain name to IP
address using the recursive method. L

Mapping a name to an address or an address to a name is called name-address

resolution.
Recursive Resolution:
The resolver asks for a recursive answer from a DNS server.
The server must respond with the complete answer.
If it does not know the answer the server itself asks a parent server in the hierarchy.
If the parent does not know, the parent asks a higher level server in the hierarchy.
Eventually the resolver will be told the answer by the first DNS server the resolver
contacted.

With the help of a diagram, explain the working of HTTP request and response from a
Client-Server communication perspective. Why is HTTP a stateless protocol? L

HTTP (Hypertext Transfer Protocol) is a protocol that is used for communication

between a server and a client. The following is a step-by-step overview of the process
of communication between an HTTP client (such as a web browser) and an HTTP
server (such as a website):
 1. The client sends an HTTP request to the server. The request includes a method
(such as GET or POST) and a URL (Uniform Resource Locator) that specifies
the location of the resource it is requesting.
2. The server receives the request and processes it. It then sends an HTTP
response message to the client.
3. The response message includes a status code (such as 200 for success or 404
for not found) and any data that is requested, such as HTML, XML, JSON, or
plain text.
4. The client receives the response and processes the data. If the status code is
200, the client displays the requested information. If the status code is 404, the
client shows an error message to the user.
5. The client and server can continue to exchange messages to complete the
task.
6. Once the communication is complete, the client and server close their
connections.

HTTP is a stateless protocol as both the client and server know each other
only during the current request. Due to this nature of the protocol, both the
client and server do not retain the information between various requests of the
web pages.

Briefly explain the functions of FTP? What are the default port numbers used by FTP? M

It helps to transfer files from one computer to another by providing access to

directories or folders on remote computers and allows software, data, text files to be
transferred between different kinds of computers. The end-user in the connection is
known as localhost and the server which provides data is known as the remote host.

FTP utilizes two ports, a 'data' port and a 'command' port (also known as the control
port). These are port 20 for the data port and port 21 for the 'command' port.

What is the difference between an active document and a dynamic document? M

A dynamic document is the product of a program run by a server as requested by a

browser. An active document is the product of a program sent from the server to the
client and run at the client site.

What is a proxy server and how is it related to HTTP? M

A proxy server lessens network traffic by rejecting unwanted requests, forwarding

requests to balance and optimize server workload, and fulfilling requests by serving
data from cache rather than unnecessarily contacting the true destination server.
HTTP Server has proxy server capabilities built in.
What is an advantage of a hierarchical name space over a flat name space for a system the
size of the Internet? S

When the name space is large, searching a name in a hierarchical structure

(tree) is much faster than searching it in a flat structure (linear). The first can
use a binary search; the second needs to use a sequential search.

What are the three FTP transmission modes? S

In FTP, there are three types of Transmission modes stream, block, and compressed.

Consider different activities related to email:

m1: Send an email from a email client to a email server
m2: Download an email from mailbox server to a email client
m3: Checking email in a web browser
What are the application-layer protocols used in each activity? S

m1 : SMTP m2 : POP m3 : HTTP

Sending an email will be done through user agent and message transfer agent by SMTP,
downloading an email from mailbox is done through POP, checking email in a web
browser is done through HTTP.

Identify the correct order in which the following actions take place in an interaction between
a web browser and a web server.
1.The web browser requests a web page using HTTP.
2.The web browser establishes a TCP connection with the webserver.
3.The web server sends the requested web page using HTTP.

➡️ ➡️ ➡️
4.The web browser resolves the domain name using DNS.
4 2 1 3

Important Numericals PDF ➡️ cn numericals.pdf

 Mod 1, 2 MidSem Important Questions

1. Different topologies and their advantages and disadvantages. Number of links

required in each case if n number of devices are there.

2. Encode a given bit sequence using line coding techniques

3. like 1011110011 using NRZ-L,NRZ-I etc

4. OSI and TCP/IP reference model

5. Differentiate bit rate and baud rate.

6. Compare and contrast FDM,TDM.

7. STP,UTP,Coaxial cable,Fibre optic cable

8. Given a dataword and divisor find codewords using CRC.

9. List error detection methods.

10. Differentiate Packet switching and Circuit Switching

11. List the responsibilities of Data Link Layer

12. Define Virtual Circuit Identifier.

13. Describe all analog to analog conversion techniques.