Câu 1: Explain the difference between connectionless unacknowledged service and connectionless

acknowledged service. How do the protocols that provide these services differ?

Connectionless unacknowledged service is a type of network service in which data packets are sent
without any guarantee of delivery. If a packet is lost, the sender will not be notified. This type of service is
often used for applications where reliability is not as important as speed, such as streaming media or
online gaming.
Connectionless acknowledged service is a type of network service in which data packets are sent with
an acknowledgement from the receiver. This means that the sender knows if the packet was received
correctly. If a packet is lost, the sender will resend it. This type of service is often used for applications
where reliability is important, such as file transfer or email.
The protocols that provide these services differ in the way they handle packet loss. In connectionless
unacknowledged service, the sender does not know if a packet was received correctly. If a packet is lost,
the sender will simply send it again. This can lead to duplicate packets being received by the receiver. In
connectionless acknowledged service, the sender knows if a packet was received correctly because the
receiver sends an acknowledgement. This means that the sender does not need to resend packets that
were lost.
Here is a table that summarizes the key differences between connectionless unacknowledged service
and connectionless acknowledged service:
Feature Connectionless unacknowledged service Connectionless acknowledged service
Guarantee of delivery No Yes
Handling of packet loss Packets are simply resent Packets are not resent
Use cases Streaming media, online gaming 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?

Connection-oriented acknowledged service is a type of network service in which a connection is

established between the sender and receiver before any data is sent. This connection provides a guarantee
of delivery for all data packets. If a packet is lost, the sender will resend it until it is received correctly. This
type of service is often used for applications where reliability is critical, such as file transfer or video
conferencing
Connectionless acknowledged service is a type of network service in which data packets are sent
without establishing a connection between the sender and receiver. Each packet is acknowledged
individually, so if a packet is lost, the sender will only know that the packet was lost if the receiver does not
send an acknowledgement. This type of service is often used for applications where reliability is not as
critical as speed, such as email or streaming media.
The protocols that provide these services differ in the way they handle packet loss. In connection-
oriented acknowledged service, the sender knows if a packet was received correctly because the receiver
sends an acknowledgement. If a packet is lost, the sender will resend it until it is received correctly. This
ensures that all data packets are delivered reliably. In connectionless acknowledged service, the sender
does not know if a packet was received correctly unless the receiver sends an acknowledgement. If a
packet is lost, the sender will not know that the packet was lost unless the receiver does not send an
acknowledgement. This means that there is a small chance that some data packets may be lost. Here is a
table that summarizes the key differences between connection-oriented acknowledged service and
connectionless acknowledged service:
Feature Connection-oriented acknowledged service Connectionless acknowledged service Guarantee
of delivery Yes No
Handling of packet loss Packets are resent until they are received correctly Packets are not resent unless
the receiver does not send an acknowledgement
Use cases File transfer, video conferencing Email, streaming media

Câu 3: Explain the differences between PPP and HDLC.

• Protocol type: PPP is a byte-oriented protocol, while HDLC is a bit-oriented protocol. This means that PPP
deals with data in units of bytes, while HDLC deals with data in units of bits.
- Frame structure: The frame structure of PPP is more flexible than the frame structure of HDLC. PPP can
support a variety of network layer. protocols, while HDLC is typically used with only one network layer
protocol.
- Error detection: PPP uses a CRC (cyclic redundancy check) for error detection, while HDLC uses a FCS
(frame check sequence), CRC is a more reliable error detection mechanism than FCS.
- Support for multiple links: PPP can support multiple links, while HDLC can only support a single link. This
makes PPP more suitable for applications that require multiple connections, such as dial-up networking.
- Security: PPP supports a variety of security mechanisms, while HDLC does not. This makes PPP more
secure than HDLC.
Here is a table that summarizes the key differences between PPP and HDLC: Feature PPP HDLC Protocol
type Byte-oriented Bit-oriented Frame structure Flexible Rigid Error detection CRC FCS Support for multiple
links Yes No Does not support security mechanisms Security Supports a variety 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.The following adaptation functions are relevant to meeting the requirements of this transfer:

Handling of arbitrary message size: The voice signal is a continuous signal, so it needs to be divided into
small packets. The packets can be of different sizes, so the network needs to be able to handle arbitrary
message sizes.

Reliability and sequencing: The voice signal is a real-time signal, so it is important that the packets are
delivered reliably and in the correct order.

Pacing and flow control: The network needs to be able to pace the delivery of the packets so that the voice
signal does not become too delayed.

Timing: The network needs to be able to keep track of the timing of the packets so that the voice signal is
not played back out of order.

Addressing: The network needs to be able to address the packets so that they can be delivered to the
correct destination.

Privacy, integrity and authentication: The network needs to be able to protect the voice signal from
unauthorized access, modification, and replay.

b.A hop-by-hop approach to meeting the requirements of the voice signal would involve each hop in the
network handling the adaptation functions independently. This approach would be simple to implement,
but it would not be very reliable. If a packet is lost or delayed at one hop, the other hops would not be able
to recover it.

An end-to-end approach to meeting the requirements of the voice signal would involve the network
providing end-to-end guarantees for the adaptation functions. This approach would be more reliable, but it
would be more complex to implement.

The best approach to meeting the requirements of the voice signal would depend on the specific network
and the requirements of the application. If the network is reliable and the application does not require very
low latency, then a hop-by-hop approach may be sufficient. However, if the network is not reliable or the
application requires very low latency, then an end-to-end approach may be necessary.

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.

The Stop-and-Wait protocol works by sending a frame, waiting for an acknowledgement, and then sending
the next frame. If the acknowledgement is not received, the frame is resent.

The immediate-retransmission feature modifies the protocol so that the frame is resent as soon as an error
is detected. This means that the sender does not have to wait for the acknowledgement before resending
the frame.

The protocol will still operate correctly with the immediate-retransmission feature. If a frame is received in
error, the receiver will send a negative acknowledgement. The sender will then immediately resent the
frame.

b.

The state transition diagram does not need to be modified to describe the new operation. The only
difference is that the sender will now enter the "Resend frame" state as soon as an error is detected.

c.
 The main effect of introducing the immediate-retransmission feature is to reduce the number of frames
that are lost. This is because the frame is resent as soon as an error is detected, so there is less time for the
frame to be lost in the network.

The immediate-retransmission feature also improves the throughput of the protocol. This is because the
sender does not have to wait for the acknowledgement before resending the frame, so the sender can
send more frames in a given period of time.

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.

To allow the peer processes to exchange messages of arbitrary size, we can use a technique called
fragmentation. This technique breaks the message into smaller pieces, called fragments, that are each
smaller than the maximum PDU size. The fragments are then sent as separate PDUs.

The receiver reassembles the fragments into the original message. The fragmentation and reassembly
process is handled by the peer processes.

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.
If the message transfer service provided by the peer processes is shared by several message source-
destination pairs, then additional control information is required. This additional control information
includes:

The source and destination of the message.

The type of message.

The priority of the message.

This additional control information is needed to ensure that the messages are routed to the correct
destination and that the messages are processed in the correct order.

The additional control information can be placed in the header of the PDU. The header of the PDU can be
up to H bytes long, so there is enough space to include the additional 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 the entire file is transmitted without errors is given by: P = (1 - p) ^ n where n is the
number of bits in the file and p is the bit error rate. In this case, n = 1048576 bits (1 Mbyte) and p = 10 ^ - 6
So, the probability that the entire file is transmitted without errors is: P = (1 - 10 ^ - 6) ^ 1048576 Therefore,
the probability that the entire file is transmitted without errors is very close to 1.

b. The file is broken up into N equal-sized blocks that are transmitted separately. The probability that all the
blocks arrive correctly without error is given by: P = (1 - p) "N where N is the number of blocks. In this case,
N is the number of bits in the file divided by the number of bits in each block. So, the probability that all
the blocks arrive correctly without error is: P (1-10^-6)^N The probability of a block being transmitted
without error is the same as the probability of the entire file being transmitted without error. So, the
probability that all the 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 still 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 receive 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
retransmitting any blocks that are corrupted. The protocol also ensures 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. Here is a table summarizing the time it
takes to deliver a file using Stop-and-Wait ARQ

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?
Based on the topology, there are 4 LANS in the network. So, 4 subnets are needed. b. The number of
bits that must be borrowed to support the number of subnets in the topology table is 2. This is because
2^2 = 4. Borrowing 2 bits from the host portion of the IP address creates 10 subnets. This is because
2^2-2=10. d. Each subnet has 16 usable hosts. This is because 2^4-2=14. Here is a table summarizing
the subnetting results:
Subnet Number of bits borrowed Number of subnets Usable hosts per subnet
192.168.1.0/26 2 10 16

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:
a) 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.
Consider the network in Figure.

b) 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