Câu 1: Explain the difference between connectionless unacknowledged service

and connectionless acknowledged service. How do the protocols that provide

these services differ?
Connectionless service is a network communication service that does not
establish a dedicated communication path between the sender and receiver. In
connectionless unacknowledged service, also known as “fire and forget”, the
sender sends a message without ensuring that it has been received by the
receiver. The receiver may or may not receive the message, and there is no
confirmation sent back to the sender.
In contrast, in connectionless acknowledged service, the sender transmits a
message and waits for a confirmation of receipt from the receiver. If the
receiver receives the message, it sends an acknowledgement back to the
sender. If the sender does not receive an acknowledgement, it can assume that
the message was not received.
The protocols that provide these services differ in their handling of packet loss
and protocol overhead. Connectionless unacknowledged service protocols,
such as UDP (User Datagram Protocol), are lightweight and have minimal
overhead but provide no guarantee of message delivery. Connectionless
acknowledged service protocols, such as ICMP (Internet Control Message
Protocol) and ARP (Address Resolution Protocol), have more overhead but
provide greater reliability by sending acknowledgements of successful message
delivery.
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 network communication
service that establishes a dedicated connection between the sender and
receiver before any data transmission takes place. In this service, the receiver
sends an acknowledgement to the sender after receiving each packet of data.
In contrast, connectionless acknowledged service does not establish a
dedicated connection before data transmission. The sender simply sends
packets to the receiver, and the receiver sends acknowledgements back to the
sender for each received packet.
The protocols that provide these services differ in their handling of packet loss
and
protocol overhead. Connection-oriented acknowledged service protocols, such
as TCP (Transmission Control Protocol), use a three-way handshake to establish
a reliable connection between the sender and receiver. This protocol also
implements flow control and congestion control mechanisms to ensure
efficient data transfer and minimize packet loss. Once the data transmission is
complete, the connection is terminated.
Connectionless acknowledged service protocols, such as UDP (User Datagram
Protocol), do not establish a dedicated connection with the receiver. Instead,
they simply send the packets to the receiver without guaranteeing their
delivery or order. These protocols have lower overhead and are useful in
applications where speed is more important than accuracy, such as online
gaming and live streaming.
In summary, connection-oriented acknowledged service is reliable but has
higher overhead and may be slower, while connectionless acknowledged
service is faster but less reliable and does not guarantee the order of packet
delivery.
Câu 3. Explain the differences between PPP and HDLC.
PPP HDLC
Full form Point-to-Point Protocol High-level Data Link
Control
Flexibility PPP is a more flexible protocol than HDLC primarily designed
HDLC. It can be used to carry for carrying only one
multiple network layer protocols, protocol.
including IP, IPX, and AppleTalk.
Error PPP has a better error detection HDLC uses a frame
detection mechanism. PPP uses a cyclic check sequence (FCS).
redundancy check (CRC) for
detecting errors. The CRC is
considered more effective in
detecting errors in data
transmission.
Configuration PPP is easier to config. PPP uses a HDLC requires manual
configuration protocol called LCP configuration of
(Link Control Protocol), which parameters, such as the
 automates the configuration address field and control
process. field.
Authentication PPP supports authentication HDLC does not support
mechanisms such as PAP (Password any authentication
Authentication Protocol) and CHAP mechanism.
(Challenge Handshake
Authentication Protocol), which
authenticate the identity of the
sender
Sliding PPP does not support any silding HDLC uses a sliding
Window window protocol. window protocol for
Protocol flow control. The sliding
window protocol is used
to optimize the flow of
data between the
sender and receiver,
ensuring that the
receiver is not
overwhelmed with too
much data at once.

In summary, PPP is a more flexible, easier to configure and better error

detection mechanism than HDLC. Additionally, PPP offers mechanisms like
authentication, which HDLC lacks. However, HDLC uses a sliding window
protocol for flow control, which PPP does not provide.

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 × 108 meters/second.

To determine the smallest possible frame size that allows continuous

transmission, we need to calculate the round-trip time (RTT) for a signal to
travel from Earth to the Moon and back.

2 × 𝑑𝑖𝑠𝑡𝑎𝑛𝑐𝑒
𝑅𝑇𝑇 =
𝑠𝑝𝑒𝑒𝑑 𝑜𝑓 𝑙𝑖𝑔ℎ𝑡

Converting the distance between Earth and the Moon to meters:

375,000 km × 1000 m/km = 375,000,000 meters

Plugging in the values:

2 × 375,000,000
𝑅𝑇𝑇 = = 5 (𝑠𝑒𝑐𝑜𝑛𝑑)
3 × 108
To achieve continuous transmission on a 1.5 Mbps link with HDLC, we need to
calculate the minimum frame size that can be transmitted within this RTT.

𝑙𝑖𝑛𝑘 𝑠𝑝𝑒𝑒𝑑 × 𝑅𝑇𝑇

𝑀𝑖𝑛𝑖𝑚𝑢𝑚 𝐹𝑟𝑎𝑚𝑒 𝑆𝑖𝑧𝑒 =
𝑒𝑓𝑓𝑖𝑐𝑖𝑒𝑛𝑐𝑦

The efficiency factor takes into account the protocol overhead, such as header
and trailer bits.

Assuming an efficiency factor of 80% (efficiency = 0.8), we get:

1.5 Mbps × 5 sec
Minimum frame size = = 11.25 𝑚𝑒𝑔𝑎𝑏𝑖𝑡𝑠
0.8

To convert to bytes:

11.25 megabits
= 1.41 megabytes
8 𝑏𝑖𝑡𝑠/𝑏𝑦𝑡𝑒

Therefore, the smallest possible frame size that allows continuous

transmission on a 1.5 Mbps HDLC link to the Moon is approximately 1.41
megabytes.

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?

To determine the maximum rate at which information can be transmitted over

a 1.5 Mbps geostationary satellite link using HDLC with 250-byte frames, we
need to take into account the overhead of the HDLC protocol.

The HDLC frame consists of several fields, including flag characters, address,
control, data, and CRC (Cyclic Redundancy Check). The flag characters mark the
beginning and end of the frame and are each one byte long. The address and
control fields are each one byte long. The CRC field is two bytes long.

Therefore, the total size of an HDLC frame is:

 𝐹𝑟𝑎𝑚𝑒 𝑆𝑖𝑧𝑒 = 𝐷𝑎𝑡𝑎 + 𝐴𝑑𝑑𝑟𝑒𝑠𝑠 (1 𝑏𝑦𝑡𝑒) + 𝐶𝑜𝑛𝑡𝑟𝑜𝑙 (1 𝑏𝑦𝑡𝑒)
+ 𝐶𝑅𝐶 (2 𝑏𝑦𝑡𝑒𝑠) + 𝐹𝑙𝑎𝑔 (1 𝑏𝑦𝑡𝑒) = 𝐷𝑎𝑡𝑎 + 6 𝑏𝑦𝑡𝑒𝑠

For 250-byte frames, the total frame size is:

𝐹𝑟𝑎𝑚𝑒 𝑆𝑖𝑧𝑒 = 250 + 6 = 256 𝑏𝑦𝑡𝑒𝑠

To calculate the maximum rate of information transmission, we need to divide

the link speed by the time it takes to transmit one frame, including the
overhead.

𝑓𝑟𝑎𝑚𝑒 𝑠𝑖𝑧𝑒 256 𝑏𝑦𝑡𝑒𝑠

𝑇𝑖𝑚𝑒 𝑡𝑜 𝑡𝑟𝑎𝑛𝑠𝑚𝑖𝑡 𝑜𝑛𝑒 𝑓𝑟𝑎𝑚𝑒 = =
𝑙𝑖𝑛𝑘 𝑠𝑝𝑒𝑒𝑑 1.5 𝑀𝑏𝑝𝑠
= 0,001707 𝑠𝑒𝑐𝑜𝑛𝑑𝑠

Therefore, the maximum rate of information transmission is:

𝑑𝑎𝑡𝑎 𝑟𝑎𝑡𝑒
𝑀𝑎𝑥𝑖𝑚𝑢𝑛 𝑟𝑎𝑡𝑒 =
𝑒𝑓𝑓𝑖𝑐𝑖𝑒𝑛𝑐𝑦

The efficiency factor takes into account the protocol overhead, such as the
header and trailer bits.

Assuming an efficiency factor of 80% (efficiency = 0.8), we get:

(250 bytes × 8 bits/byte)/0.001707 seconds
𝑀𝑎𝑥𝑖𝑚𝑢𝑛 𝑟𝑎𝑡𝑒 = =
0.8
25,154 bits per second

Therefore, the maximum rate at which information can be transmitted over a

1.5 Mbps geostationary satellite link using HDLC with 250-byte frames is
approximately 23,154 bits 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.

Given:

N = 60 data sources

Probability of having a packet in a given T-second period, p = 0.1

The multiplexer can transmit eight packets every T seconds

Any packets that cannot be transmitted in the first line are directed to the
second line

To find:

Average number of packets that are transmitted on the first line

Average number of packets that are transmitted in the second line

We can model this scenario as a binomial distribution problem, where each

data source has a probability p of transmitting a packet and there are N such
sources.

The probability of k sources transmitting a packet is given by the binomial

distribution formula:

𝑁
𝑃(𝑘) = ( ) 𝑝𝑘 (1 − 𝑝)𝑁−𝑘
𝑘

where (𝑁𝑘) is the binomial coefficient, given by:

𝑁 𝑁!
( )=
𝑘 𝑘! (𝑁 − 𝑘)!

We want to find the average number of packets that are transmitted on the
first line and the second line. Let X be the total number of packets generated in
a T-second period. Then, we can divide X into two parts: Y, the number of
packets transmitted on the first line, and Z, the number of packets transmitted
on the second line.

Since we can transmit up to eight packets on the first line, we have:

 𝑌 = min (𝑋, 8)

For the remaining packets, we send them on the second line, so we have:

𝑍 = max (0, 𝑋 − 8)

The expected value of Y is given by:

8

𝐸(𝑌) = ∑ 𝑘𝑃(𝑘)
𝑘=0

where P(k) is the probability of k sources transmitting a packet, as computed

using the binomial distribution formula.

Similarly, the expected value of Z is given by:

𝑁

𝐸(𝑍) = ∑ 𝑘𝑃(𝑘)
𝑘=9

where P(k) is again the probability of k sources transmitting a packet.

Let's compute these values using the given parameters:

N = 60

p = 0.1

T = 1 second (since we are considering a T-second period)

M = 8 (maximum packets that can be transmitted on the first line)

First, let's calculate the probabilities of k sources transmitting a packet:

𝑁
𝑃(𝑘) = ( ) 𝑝𝑘 (1 − 𝑝)𝑁−𝑘
𝑘
For k = 0 to 8:

P(0) = (60 choose 0) * 0.1^0 * 0.9^60 = 0.026

 60
𝑃(0) = ( ) 0.10 (1 − 0.1)60 = 0.026
0
60
𝑃(1) = ( ) 0.11 (1 − 0.1)59 = 0.157
1

60
𝑃(2) = ( ) 0.12 (1 − 0.1)58 = 0.318
2
60
𝑃(3) = ( ) 0.13 (1 − 0.1)57 = 0.306
3
60
𝑃(4) = ( ) 0.14 (1 − 0.1)56 = 0.185
4
60
𝑃(5) = ( ) 0.15 (1 − 0.1)55 = 0.08
5
60
𝑃(6) = ( ) 0.16 (1 − 0.1)54 = 0.027
6
60
𝑃(7) = ( ) 0.17 (1 − 0.1)53 = 0.007
7
60
𝑃(8) = ( ) 0.18 (1 − 0.1)52 = 0.001
8

For k = 9 to 60:

𝑁
𝑃(𝑘) = ( ) 𝑝𝑘 (1 − 𝑝)𝑁−𝑘
𝑘
For simplicity, we can use the complement rule and subtract the sum of
probabilities from 0 to 8 from 1:

𝑃(𝑘) = 1 − 𝑃(0 ≤ 𝑋 ≤ 8)
=1
− (𝑃(0) − 𝑃(1) − 𝑃(2) − 𝑃(3) − 𝑃(4) − 𝑃(5) − 𝑃(6) − 𝑃(7)
− 𝑃(8)) = 0.996
Now, let's calculate the expected values of Y and Z:
8

𝐸(𝑌) = ∑ 𝑘𝑃(𝑘) = 00.026 + 10.157 + 20.318 + 30.306 + 40.185

𝑘=0
+ 50.080 + 6 ∗ 0.027 + 7

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 meet the requirement of transferring a single real-time telephone voice

signal across a packet network with a maximum delay of 20 ms, the following
adaptation functions are relevant:

Timing: The timing adaptation function is critical in ensuring that each voice
sample is delivered within the required deadline. The network must be able to
synchronize its clock with the sender and receiver to maintain the required
time intervals between packets.

Reliability and sequencing: To ensure that each voice sample is delivered

without loss or misordering, the reliability and sequencing adaptation function
is necessary. This requires the use of error detection and correction
mechanisms, as well as sequencing and resequencing of packets at the receiver
end.

Pacing and flow control: To prevent packet loss due to congestion, pacing and
flow control mechanisms are necessary. These mechanisms regulate the rate
at which packets are transmitted and received to match the capacity of the
network.
Addressing: Addressing is necessary to identify the source and destination of
each voice sample. It also enables routing of packets through the network.

b. There are two approaches for meeting the requirements of a real-time

telephone voice signal over a packet network: the hop-by-hop approach and
the end-to-end approach.

The hop-by-hop approach involves implementing the required adaptation

functions at each intermediate node in the packet network. Each node
processes the packets it receives before forwarding them to the next node.
This approach can introduce additional delays and overhead due to processing
at each node. Furthermore, if a node fails, the entire communication may
become compromised.

The end-to-end approach involves implementing the required adaptation

functions only at the endpoints of the communication path, i.e., the sender
and receiver of the voice signal. The packets are transmitted through the
network without modification, and any required processing is performed at the
endpoints. This approach minimizes delays and overhead, but it may not be
suitable for networks with high packet loss rates or variable delays.

In general, the end-to-end approach is preferred for real-time voice

communications over packet networks because it minimizes delays and
overhead. However, the hop-by-hop approach may be necessary in some
situations, such as when the network has high delay or loss rates, or when
additional processing is necessary at intermediate nodes.

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 modified Stop-and-Wait protocol still operates correctly because it

ensures that every frame is received correctly before the next one is sent. If a
frame is found in error, the sender immediately resends the last transmitted
frame, which guarantees that the receiver will receive a correct copy of the
frame.

b. The state transition diagram would need to be modified to reflect the new
operation. Specifically, a new transition would need to be added from the
"Frame Received, ACK/NAK Lost" state back to the "Frame Sent" state,
indicating that the sender should immediately resend the last transmitted
frame in response to the error.

c. The main effect of introducing the immediate-retransmission feature is to

improve the protocol's error recovery capabilities. With this feature, errors can
be quickly corrected by resending the last transmitted frame. This reduces the
time required for error recovery and increases the overall efficiency of the
protocol. However, it also introduces additional network traffic, which could
potentially increase congestion and delay.

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 for the exchange of messages of arbitrary size within the given
constraints, the peer processes can use a technique known as segmentation
and reassembly. This involves dividing a message into smaller segments, each
of which can fit within a single PDU, and then sending these segments over
multiple PDUs. The receiver can then reassemble the segments back into the
original message.

b. The essential control information that needs to be exchanged between the

peer processes includes:

Sequence numbers: These are used to ensure that all segments are received in
the correct order and that no segments are missing or duplicated.

Acknowledgment numbers: These are used to confirm that a segment has

been successfully received by the receiver.

Window sizes: These are used to allow the sender to adjust the number of
unacknowledged segments it can send at any given time based on how much
space is available in the receiver's buffer.

c. If the message transfer service is shared by several source-destination pairs,

additional control information may be required to differentiate between the
different messages being sent. This information could be placed in the header
of each PDU and could include the source and destination addresses, session
identifiers, or any other information needed to identify the specific message
being sent. Additionally, the control information used to manage the flow of
PDUs between the sender and receiver may also need to be adjusted to
account for multiple concurrent connections. For example, each connection
may require its own sequence

and acknowledgment numbers to ensure that segments are properly tracked

and acknowledged for each individual message.

Câu 10. A 1 Mbyte file is to be transmitted over a 1 Mbps communication line

that has a bit error rate of p = 106 .

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 1 Mbyte file is transmitted without errors can
be calculated as follows: 𝑃 = (1 − 𝑝)𝑛

where p is the bit error rate and n is the number of bits in the file.

Since the file size is 1 Mbyte, which is equal to 8 million bits, we can calculate
the probability as:

𝑃 = (1 − 10−6 )8000000

𝑃 ≈ 𝑒 −8

Therefore, the probability that the entire file is transmitted without errors is
approximately 𝑒 −8 , which is a very small probability.

b. If the file is broken up into N equal-sized blocks and transmitted separately,

the probability that all blocks arrive correctly without error can be calculated
as:

𝑃 = (1 − 𝑝)𝑛 × 𝑁

where p is the bit error rate, n is the number of bits in each block, and N is the
total number of blocks.

Dividing the file into blocks does help because if an error occurs in one block,
only that block needs to be retransmitted instead of the entire file.

Assuming each block is equally sized at 1/N Mbytes or 8/N million bits, the
probability can be calculated as:
8
𝑃 = (1 − 10−6 )𝑁×𝑁 = (1 − 10−6 )8 = 0.99992

Therefore, the probability that all blocks arrive correctly without error is
approximately 0.999992.

c. Stop-and-Wait ARQ (Automatic Repeat Request) can help deliver the file in
error-free form by ensuring that each block is successfully received before
transmitting the next block. In this protocol, the sender transmits one block at
a time and waits for an acknowledgment from the receiver before transmitting
the next block.

Assuming the propagation delay is negligible, the time required to deliver the
file using Stop-and-Wait ARQ can be calculated as follows:

Time required to transmit one block = n/p, where n is the number of bits in
each block and p is the bit rate of the communication line.

Time required to receive an acknowledgment for one block

= 2 × propagation delay

Total time required to transmit and receive one block

= n/p + 2 × propagation delay

Since there are N blocks to be transmitted, the total time required to deliver
the file would be N times the time required to transmit and receive one block:
𝑛
𝑇𝑜𝑡𝑎𝑙 𝑡𝑖𝑚𝑒 = 𝑁( + 2 × 𝑝𝑟𝑜𝑝𝑎𝑔𝑎𝑡𝑖𝑜𝑛 𝑑𝑒𝑙𝑎𝑦)
𝑝

Assuming a negligible propagation delay, the total time required to deliver the
entire 1 Mbyte file would be:

8000000
𝑇𝑜𝑡𝑎𝑙 𝑡𝑖𝑚𝑒 = + 2 × 0 = 8 𝑠𝑒𝑐𝑜𝑛𝑑𝑠
106
However, this assumes no errors occur during transmission. If errors occur,
additional time will be required for retransmission until all blocks are received
correctly.

8000000