Câu 1: Explain the difference between connectionless unacknowledged service and

connectionless acknowledged service. How do the protocols that provide these services
differ?
-Data packets are sent using a type of network service called a connectionless
unacknowledged service with no assurance that they will arrive. The sender won't be
informed if a packet is misplaced. This type of service is frequently used for applications
like streaming media or online gaming where reliability is less crucial than speed.
-A type of network service known as connectionless acknowledged service sends
data packets with a recipient acknowledgement. This implies that the sender is aware of
whether the packet was correctly received. The sender will resend a lost packet.
Applications like file transfer and email, which depend on reliability, frequently use this
kind of service.
-The protocol that offers these services handles packet loss differently. The sender
has no way of knowing if a packet was correctly received when using connectionless
unacknowledged service. The sender will just send the packet again if it is lost. This may
result in the receiver receiving duplicate packets. The sender can tell if a packet was
correctly received in a connectionless acknowledged service because the recipient sends
an acknowledgement. This implies that packets that were lost do not need to be resent by
the sender.
Here is a table that summarizes the key different between connectionless
unacknowledged service and connectionless acknowledged service :

Feature Connectinoless Connectionless

Guarantee of delivery
Handing of packet loss
Use cases
unacknowledged service
No
Packets are simply resent
Streaming media, online
gaming
acknowledged service
Yes
Packets are not resent
File transfer, email

Câu 2. Explain the difference between connection-oriented acknowledged service and

connectionless acknowledged service. How do the protocols that provide these services
differ?
Câu 3: Explain the differences between PPP and HDLC.

-Type of protocol: HDLC is a bit-oriented protocol, whereas PPP is a byte-

oriented protocol. Thus, HDLC deals with data in units of bits, whereas PPP deals with
data in units of bytes.
-Frame structure: The PPP frame structure is more adaptable than the HDLC
frame structure. PPP is capable of supporting a wide range of network layer protocols,
whereas HDLC is typically used with just one network layer protocol.
-For error detection, PPP employs a CRC (cyclic redundancy check), while HDLC
employs a FCS (frame check sequence). Compared to FCS, CRS is a more trustworthy
method of error detection.
-Multiple links supported: PPP can support multiple links, whereas HDLC can
only support a single link. PPP is thus better suited for multi-connection applications like
dial-up networking.
-Security: PPP is more secure than HDLC and supports a wider range of security
mechanisms. Due to this, PPP is more secure than HDLC.

Feature PPP HDLC

Protocol type Byte- oriented Bit - oriendted

Frame structure Flexible

Rigid
Error deletion CRR FCS

Support for multiple links Yes No

Security Supports a variety Does not support security mechanisms.

of security
mechanisms

Câu 4:
A 1.5 Mbps communications link is to use HDLC to transmit information to the moon.
What is the smallest possible frame size that allows continuous transmission? The
distance between earth and the moon is approximately 375,000 km, and the speed of light
is 3 x 108 meters/second.
The smallest possible frame size that allows continuous transmission is the size of
the round-trip propagation delay. The round-trip propagation delay is the time it
takes for a signal to travel from Earth to the Moon and back.

The distance between Earth and the Moon is 375,000 km, so the round-trip
propagation delay is 2 * 375,000 km / 3 x 10^8 meters/second = 250 milliseconds.

The data rate of the communications link is 1.5 Mbps, so the smallest possible frame
size is 1.5 Mbps * 250 milliseconds = 375,000 bits.

In bytes, the smallest possible frame size is 375,000 bits / 8 bits/byte = 46,875 bytes.

Therefore, the smallest possible frame size that allows continuous transmission is
46,875 bytes.

Câu 5: Suppose HDLC is used over a 1.5 Mbps geostationary satellite link. Suppose that
250-byte frames are used in the data link control. What is the maximum rate at which
information can be transmitted over the link?

The maximum rate at which information can be transmitted over the link is 299,800 bits
per second.

The data rate of the link is 1.5 Mbps, which is equal to 1.5 * 10^6 bits per second.
However, the overhead of the HDLC protocol is 250 * 8 = 2000 bits per frame. This
means that the maximum rate at which information can be transmitted over the link is 1.5
* 10^6 - 2000 = 299,800 bits per second.

In bytes, the maximum rate at which information can be transmitted over the link is
299,800 / 8 = 37,475 bytes per second.

Câu 6:
Suppose that a multiplexer receives constant-length packet from N = 60 data sources.
Each data source has a probability p = 0.1 of having a packet in a given T-second period.
Suppose that the multiplexer has one line in which it can transmit eight packets every T
seconds. It also has a second line where it directs any packets that cannot be transmitted
in the first line in a T-second period. Find the average number of packets that are
transmitted on the first line and the average number of packets that are transmitted in the
second line.

The average number of packets that are transmitted on the first line is given by:
 E[x_1] = np = 60 * 0.1 = 6
where n is the number of data sources and p is the probability of a data source having a
packet in a given T-second period.

The average number of packets that are transmitted in the second line is given by:

E[x_2] = np(1 - p/m) = 6 * 0.1 * (1 - 0.1/8) = 0.133333

where m is the capacity of the first line.

Therefore, the average number of packets that are transmitted on the first line is 6 and
the average number of packets that are transmitted in the second line is 0.133333.

Câu 7:
Consider the transfer of a single real-time telephone voice signal across a packet network.
Suppose that each voice sample should not be delayed by more than 20 ms.
a. Discuss which of the following adaptation functions are relevant to meeting the
requirements of this transfer: handling of arbitrary message size; reliability and
sequencing; pacing and flow control; timing; addressing; and privacy, integrity
and authentication.
b. Compare a hop-by-hop approach to an end-to-end approach to meeting the
requirements of the voice signal.

a.

To fulfill the requirements of this transfer, the aforementioned adaptation functions are
pertinent:.

Handling of messages of any size: Because the voice signal is continuous, it needs to be
split up into small packets. The network must be able to handle arbitrary message sizes
because the packets can be of different sizes.
Sequencing and dependability: Since the voice signal is a real-time signal, it is critical that
the packets are delivered with dependability.
Pacing and flow control are important so that the voice signal is not overly delayed.
The network must be able to pace the packet delivery.

Timing: To prevent the voice signal from being played back out of order, the network must
be able to monitor the timing of the packets.
The network must be able to address packets in order to deliver them to the intended
recipient.
The network must be able to safeguard the voice signal from unauthorized access,
modification, and replay. Privacy, integrity, and authentication are also required.
b.

Each hop in the network would handle the adaptation functions on its own in a hop-by-hop
method of meeting the requirements of the voice signal. Although this strategy would be
easy to put into practice, it would not be very dependable. The other hops would not be able
to recover a packet that was lost or delayed at one hop.

The network would have to offer end-to-end guarantees for the adaptation functions in
order to fully satisfy the demands of the voice signal. Although more difficult to
implement, this strategy would be more trustworthy.

Depending on the particular network and the needs of the application, different strategies
may be used to meet voice signal requirements. A hop-by-hop approach might be sufficient
if the network is dependable and the application does not require extremely low latency. An
end-to-end strategy may be required, though, if the network is unreliable or the application
requires extremely low latency.
 Câu 8 :
Consider the Stop-and-Wait protocol as described. Suppose that the protocol is modified
so that each time a frame is found in error at either the sender or receiver, the last
transmitted frame is immediately resent.

a. Show that the protocol still operates correctly.

b. Does the state transition diagram need to be modified to describe the new
operation?
c. What is the main effect of introducing the immediate-retransmission feature?

a.

When using the Stop-and-Wait protocol, a frame is sent, an acknowledgement is

awaited, and then the next frame is sent. The frame is resent in the event that the
acknowledgement is not received.

The protocol is altered by the immediate-retransmission feature so that the frame is

sent again as soon as an error is discovered. As a result, the sender need not wait for
the acknowledgement before sending the frame again.

The immediate retransmission feature won't affect the protocol's ability to function
properly. The receiver will send a negative acknowledgement if a frame is received
incorrectly. The sender will then immediately send the frame again.
b.

The new operation can be described using the state transition diagram as is. The
sender will now enter the "Resend frame" state as soon as an error is discovered,
which is the only difference.
c.

Reducing the amount of lost frames is the main result of the immediate-
retransmission feature. This is due to the fact that the frame is sent again as soon as
an error is discovered, which reduces the amount of time the frame can be lost in the
network.

The immediate-retransmission feature enhances the protocol's throughput as well.

Because the sender can send more frames in a given amount of time without having
to wait for the acknowledgement, this is due to the fact that the acknowledgement is
not required every time a frame is sent.
Câu 9:
Suppose that two peer-to-peer processes provide a service that involves the transfer of
discrete messages. Suppose that the peer processes are allowed to exchange PDUs that
have a maximum size of M bytes including H bytes of header. Suppose that a PDU is not
allowed to carry information from more than one message.
a. Develop an approach that allows the peer processes to exchange messages of arbitrary
size.
b. What essential control information needs to be exchanged between the peer processes?
c. Now suppose that the message transfer service provided by the peer processes is shared
by several message source-destination pairs. Is additional control information required,
and if so, where should it be placed?

a.

We can employ a process known as fragmentation to enable the peer processes to

communicate messages of any size. Using this method, the message is divided into
fragments, or smaller-than-the-largest PDU-sized pieces, that are then combined into a
single message. The fragments are then transmitted as individual PDUs.
The original message is created by the receiver by piecing together the pieces. The peer
processes are in charge of fragmentation and reassembly.
b.

The essential control information that needs to be exchanged between the peer processes
includes:

-The size of the message.

-The number of fragments.
-The sequence number of each fragment.
c.

Additional control information is necessary if multiple message source-destination pairs

share the message transfer service offered by the peer processes. The extra control
information in question consists of:
-The message's sender and recipient.
-which message type.
-The message's importance.
To make sure that the messages are processed in the proper order and are routed to the
intended recipient, additional control information is required.
The PDU's header is a suitable location for the additional control information.
There is enough room in the PDU's header, which can be up to H bytes long, to include
the extra control information.
Câu 10:
A 1 Mbyte file is to be transmitted over a 1 Mbps communication line that has a bit error
rate of p = 10-6.
a. What is the probability that the entire file is transmitted without errors? Note for n
large and p very small, (1 − p)n ≈ e-np.
b. The file is broken up into N equal-sized blocks that are transmitted separately.
What is the probability that all the blocks arrive correctly without error? Does
dividing the file into blocks help?
c. Suppose the propagation delay is negligible, explain how Stop-and-Wait ARQ can
help deliver the file in error-free form. On the average how long does it take to
deliver the file if the ARQ transmits the entire file each time?
 .
A. The probability that entire file is transmitted without error is given by :
P = (1 - 10 ^ 6) ^ 1048576 ≈ 1
Therefore, the probability that the entire file is transmitted without errors is very
close to 1.

B.
The probability that all the blocks arrive correctly without error is given by :
P=(1-p)^N
The probability that all the blocks arrive correctly without error is :
P = (1 - 10 ^ 6 ) ^ N
The probability of a block transmitted error is the same as the probability of the
entire file being transmitted without error. So, the probability that all blocks arrive correctly
without error is also very close to 1.
Dividing the file into blocks does help to improve the reliability of the transmission.
This is because if one block is corrupted, the other blocks are sill likely to be transmitted
correctly.
C. Stop-and-Wait ARQ is a simple and effective way to deliver a file in error- free
form. The sender transmits the file one block at a time. The receiver acknowledges each
block that it receives correctly. If the receiver does not a block correctly, it sends a
negative acknowledgement. The sender then resends the block.
The Stop-and-Wait ARQ protocol ensures that the file is delivered in error-free form
by retransmiting any blocks that are corrupted. The protocol also ensure that the file is
delivered in the correct order by acknowledging each block that is received correctly.
On the average, it takes 2RTT to deliver a file using Stop-and-Wait ARQ, where
RTT is the round-trip time.This is because the sender has to wait for an acknowledgement
before it can transmit the next block.
A table summarizing the time it takes to deliver a file using Stop-and-Wait ARQ:

Event Time
Sender transmit a block RTT
Receiver receives the block correctly 0
Receiver send an acknowledgement RTT
Sender receives the acknowledgement 0

Câu 11:

In this activity, you are given the network address of 192.168.1.0/24 to subnet and
provide the IP addressing for the Packet Tracer network. Each LAN in the network
requires at least 25 addresses for end devices, the switch and the router. The
connection between R1 to R2 will require an IP address for each end of the link.

a. Based on the topology, how many subnets are needed?

b. How many bits must be borrowed to support the number of subnets in the topology
table?
c. How many subnets does this create?
d. How many usable hosts does this create per subnet?
 Note: If your answer is less than the 25 hosts required, then you borrowed too
many bits.

Câu 12:

Five stations (S1-S5) are connected to an extended LAN through transparent bridges (B1-
B2), as shown in the following figure. Initially, the forwarding tables are empty. Suppose
the following stations transmit frames: S1 transmits to S5, S3 transmit to S2, S4 transmits
to S3, S2 transmits to S1, and S5 transmits to S4. Fill in the forwarding tables with
appropriate entries after the frames have been completely transmitted.
Câu 13:
Consider the network in Figure.

a) Use the Dijkstra algorithm to find the set of shortest paths from node 4 to other
nodes.
Iteration N D1 D2 D3 D5 D6

Initial
b) Find the set of associated routing table entries (Destination, Next Hop, Cost)

Destination Cost Next Hop

D1 4 D2
D2 1 D2
D3 2 D3
D5 3 D5
D6 3 D3
14)
 You are a network technician assigned to install a new network for a customer. You
must create multiple subnets out of the 192.168.1.0/24 network address space to meet
the following requirements:
- The first subnet is the LAN-A network. You need a minimum of 50 host IP
addresses.
- The second subnet is the LAN-B network. You need a minimum of 40 host
IP addresses.
- You also need at least two additional unused subnets for future network
expansion.
Note: Variable length subnet masks will not be used. All of the device subnet masks
should be the same length.
Answer the following questions to help create a subnetting scheme that meets the stated
network requirements:

a. How many host addresses are needed in the largest required subnet?
b. What is the minimum number of subnets required?
c. The network that you are tasked to subnet is 192.168.1.0/24. What is the /24
subnet mask in binary?
d. The subnet mask is made up of two portions, the network portion, and the host
portion. This is represented in the binary by the ones and the zeros in the
subnet mask.

In the network mask, what do the ones and zeros represent?

e. When you have determined which subnet mask meets all of the stated network
requirements, derive each of the subnets. List the subnets from first to last in the
table. Remember that the first subnet is 192.168.0.0 with the chosen subnet
mask.
Câu 15:

Suppose that Selective Repeat ARQ is modified so that ACK messages contain a list of
the next m frames that it expects to receive.

Solutions follow questions:

a. How does the protocol need to be modified to accommodate this change?

b. What is the effect of the change on protocol performance?

Q.16. (2 marks)
Suppose the size of an uncompressed text file is 1 megabyte
Note: Explain your answer in details.
a. How long does it take to download the file over a 32 kilobit/second modem?

b. How long does it take to take to download the file over a 1 megabit/second
modem?
c. Suppose data compression is applied to the text file. How much do the transmission
times in parts (a) and (b) change?
Q17. (2 marks)
Let g(x)=x3+x+1. Consider the information sequence 1001. Find the codeword
corresponding to the preceding information sequence. Using polynomial arithmetic we obtain

Note: Explain your answer in details.

 Q.18. (2 marks)
A router has the following CIDR entries in its routing table:
Address/mask Next hop

135.46.56.0/22 Interface 0

135.46.60.0/22 Interface 1

192.53.40.0 /23 Router 1

default Router 2

(a) What does the router do if a packet with an IP address 135.46.63.10 arrives?

(b) What does the router do if a packet with an IP address 135.46.57.14 arrives?

a.

With an IP address of 135.46, the router will forward the packet. 1 to interface 63.10.
This is so because the destination IP address matches the first entry in the routing
table, which has a subnet mask of /22. The first 22 bits of the IP address must match
for the packet to be routed to the interface, as indicated by the subnet mask of /22.
135.46's first 22 bits on an IP address. The first 22 bits of the subnet mask 135.46
correspond to 63.10. Consequently, the packet will be forwarded to Interface 1 at 60.0.
b.

The packet with an IP address of 135.46 will be forwarded by the router. A 57.14 to
interface 0 ratio. This is because neither of the first two routing table entries
correspond to the destination IP address. The routing table entry that routes all
packets to Router 2 by default will then be applied.
Câu 19:
A Large number of consecutive IP address are available starting at 198.16.0.0.
Suppose four organizations, A, B, C, D request 4000, 2000, 4000, and 8000
addresses, respectively. For each of these organizations, give:
1. the first IP address assigned
2. the last IP address assigned
3. the mask in the w.x.y.z/s notation
The start address, the ending address, and the mask are as follows:

Here are the details for each organization:

Organization A

Start address: 198.16.0.0

Last IP address assigned: 198.16.39.255
Mask: 255.255.252.0
The mask of 255.255.252.0 means that the first 23 bits of the IP address must match for
the packet to be routed to organization A. The first 23 bits of the IP address 198.16.0.0
match the first 23 bits of the subnet mask 255.255.252.0, so all packets with an IP address
in the range 198.16.0.0 to 198.16.39.255 will be routed to organization A.

Organization B

Start address: 198.16.40.0

Last IP address assigned: 198.16.63.255
Mask: 255.255.254.0
The mask of 255.255.254.0 means that the first 22 bits of the IP address must match for
the packet to be routed to organization B. The first 22 bits of the IP address 198.16.40.0
match the first 22 bits of the subnet mask 255.255.254.0, so all packets with an IP address
in the range 198.16.40.0 to 198.16.63.255 will be routed to organization B.

Organization C

Start address: 198.16.64.0

Last IP address assigned: 198.16.95.255
Mask: 255.255.252.0
The mask of 255.255.252.0 means that the first 23 bits of the IP address must match for
the packet to be routed to organization C. The first 23 bits of the IP address 198.16.64.0
match the first 23 bits of the subnet mask 255.255.252.0, so all packets with an IP address
in the range 198.16.64.0 to 198.16.95.255 will be routed to organization C.

Organization D

Start address: 198.16.96.0

Last IP address assigned: 198.16.127.255
Mask: 255.255.255.0
The mask of 255.255.255.0 means that all 32 bits of the IP address must match for the
packet to be routed to organization D. The first 32 bits of the IP address 198.16.96.0 match
the first 32 bits of the subnet mask 255.255.255.0, so all packets with an IP address in the
range 198.16.96.0 to 198.16.127.255 will be routed to organization D.

Câu 20:
(2 marks) Suppose an application layer entity wants to send an L-byte message to its
peer process, using an existing TCP connection. The TCP segment consists of the
message plus 20 bytes of header. The segment is encapsulated into an IP packet that has
an additional 20 bytes of header. The IP packet in turn goes inside an Ethernet frame that
has 18 bytes of header and trailer. What percentage of the transmitted bits in the physical
layer correspond to message information, if L = 100 bytes, 500 bytes, 1000 bytes.
Câu 21:
(2 marks) Consider the three-way handshake in TCP connection setup.
(a) Suppose that an old SYN segment from station A arrives at station B, requesting a
TCP connection. Explain how the three-way handshake procedure ensures that the
connection is rejected.
(b) Now suppose that an old SYN segment from station A arrives at station B, followed a
bit later by an old ACK segment from A to a SYN segment from B. Is this connection

Câu 22:
(2 marks) Suppose a header consists of four 16-bit words: (11111111 11111111,
11111111 00000000, 11110000 11110000, 11000000 11000000). Find the Internet
checksum for this code.

The Internet checksum is a 16-bit checksum that is used to verify the integrity of IP
packets. The checksum is calculated by adding the 16-bit words in the IP header, and then
taking the one's complement of the sum.

The header consists of four 16-bit words:

0xFFFF
0x0000
0xAAFF
0xCCCC
The checksum is calculated as follows:

0xFFFF + 0x0000 + 0xAAFF + 0xCCCC = 0x155D

The one's complement of 0x155D is 0xEAAB.

Therefore, the Internet checksum for this code is 0xEAAB.

Note: SV có thể làm cách khác nhưng kết quả đúng vẫn được tính điểm
Câu 23:

(2 marks)
Consider the 7-bit generator, G=10011, , and suppose that D has the value 1001010101.
What is the value of R? Show your all steps to have result.
Note: Explain your answer in details

The value of R is 1001010010.

Here are the steps:

The generator polynomial is G=10011. This means that the polynomial that is used to
generate the codewords is x^3 + x + 1.
The dataword is D=1001010101. This means that the dataword is a binary sequence with 7
bits.
The remainder R is calculated by dividing the dataword D by the generator polynomial G.
R = D - G * Quotient
The quotient is the number of times that the generator polynomial G divides evenly into the
dataword D. The remainder is the remainder that is left after the division.

In this case, the quotient is 1. This means that the generator polynomial G divides evenly
into the dataword D once. The remainder is therefore the last 3 bits of the dataword D,
which is 1010.

Therefore, the value of R is 1001010010.

Câu 24:

(2 marks)

Suppose two hosts, A and B, are separated by 20,000 kilometers and are connected by a
direct link of R = 2 Mbps. Suppose the propagation speed over the link is 2.5 x 108
meters/sec.
a. Calculate the bandwidth-delay product, R _ dprop.
b. Consider sending a file of 800,000 bits from Host A to Host B. Suppose the file is sent
continuously as one large message. What is the maximum number of bits that will be in
the link at any given time?
Note: Explain your answer in details

a. Calculate the bandwidth-delay product, R _ dprop.

The bandwidth-delay product is calculated as the product of the link capacity (R) and
the propagation delay (dprop). In this case, we have:

Bandwidth-delay product = R * dprop = 2 Mbps * (20,000 km / 2.5 x 10^8 m/s) =

500,000,000 bits
b. Consider sending a file of 800,000 bits from Host A to Host B. Suppose the file is
sent continuously as one large message. What is the maximum number of bits that will
be in the link at any given time?

The maximum number of bits that will be in the link at any given time is the sum of the
file size (800,000 bits) and the bandwidth-delay product (500,000,000 bits). This is
because the file is being sent continuously as one large message, so there will be a delay
between the time the first bit is sent and the time the last bit is received. During this
delay, the link will be filled with the file data.

Maximum number of bits in link = 800,000 bits + 500,000,000 bits = 500,800,000 bits
In other words, at any given time, there will be at most 500,800,000 bits in the link. This
is the maximum amount of data that can be transmitted by the sender before waiting for
acknowledgment.

Note: The bandwidth-delay product is an important concept in networking because it

limits the maximum throughput of a link. If the file size is larger than the bandwidth-
delay product, then the sender will have to stop and wait for acknowledgments before
sending more data. This can lead to a decrease in throughput and an increase in
latency.

Note:
Students have to follow the steps and complete the tasks in details in order to
have the results. If the students only write the result, that is, that result is not
marked or recorded.
- Students do examination on word file and answer by English