Block coding is normally referred to as mB/nB coding; it replaces each m-bit group with an n-bit group

4B/5B
The four binary/five binary (4B/5B) coding scheme was designed to be used in com-
bination with NRZ-. In 4B/5B, the 5-bit output that replaces the 4-bit input has no more than one leading
zero (left bit) and no more than two trailing zeros (right bits).
8B/10B
The eight binary/ten binary (8B/10B) encoding is similar to 4B/5B encoding except that a group of 8 bits of data is now
substituted by a 10-bit code. It provides greater error detection capability than 4B/5B. The 8B/10B block coding is actually
a combination of 5B/6B and 3B/4B encoding. The five most significant bits of a 10-bit block are fed into the 5B/6B
encoder; the three least significant bits are fed into a 3B/4B encoder. The coding has 210 − 28 = 768 redundant groups that
can be used for disparity checking and error detection.

B8ZS: is commonly used in North America. In this technique, eight consecutive zero-level voltages are replaced by the
sequence 000VB0V substitutes
1. If the number of nonzero pulses after the last substitution is odd, the substitution pattern will be 000V, which makes the
total number of nonzero pulses even.
2. If the number of nonzero pulses after the last substitution is even, the substitution pattern will be B00V, which makes the
total number of nonzero pulses even
HDB3 substitutes four consecutive zeros with 000V or B00V depending on the number of nonzero pulses after the last
substitution.
HDB3 is commonly used outside of North America. In this technique, which is more conservative than B8ZS, four
consecutive zero-level voltages are replaced with a sequence of 000V or B00V.

2.2-2 Analog-to-digital Conversion

PCM
PCM consists of three steps to digitize an analog signal:
1. Sampling: Nyquist theorem, the sampling rate must be at least 2 times the highest frequency contained in the signal
2. Quantization: Sampling results in a series of pulses of varying amplitude values ranging between two limits: a min and a
max. The amplitude values are infinite between the two limits. We need to map the infinite amplitude values onto a finite
set of known values. This is achieved by dividing the distance between min and max into L zones, each of height =
(Vmax - Vmin)/L
3. Binary encoding

2.2-3 Transmission Modes

The transmission of binary data across a link can be accomplished in either parallel or serial mode. In
parallel mode, multiple bits are sent with each clock tick. In serial mode, 1 bit is sent with each clock tick. While there is
only one way to send parallel data, there are three subclasses of serial transmission: asynchronous, synchronous, and
isochronous.

● In asynchronous transmission, we send 1 start bit (0) at the beginning and 1 or more stop bits (1s) at the end of
each byte. There may be a gap between each byte.
● Asynchronous here means “asynchronous at the byte level,” but the bits are still synchronized; their durations are
the same.
● In synchronous transmission, we send bits one after another without start or stop bits or gaps. It is the
responsibility of the receiver to group the bits.

Chapter 2.3 Analog Transmission

2.3-1 Digital-to-analog Conversion
Digital data needs to be carried on an analog signal. A carrier signal (frequency fc) performs the function of transporting
the digital
data in an analog waveform. The analog carrier signal is manipulated to uniquely identify the digital data being carried.

Data Rate Versus Signal Rate

S=N x 1/r band
Bit rate is the number of bits per second. Baud rate isthe number of signalelements per second.In the analog transmission
of digital data, the baudrate is less thanor equal to the bit rate.S=Nx1/r baudsWhere r is the number of data bits per signal
element

Amplitude Shift Keying

In amplitude shift keying, the amplitude of the carrier signal is varied to create signalelements. Both frequency and phase
remain constant while the amplitude changes
Binary ASK (BASK)
ASK is normally implemented using only two levels. The peak amplitude of one signal level is 0; the other is the same as
the amplitude of the carrier frequency.

Bandwidth for BFSK B =(1+d) x S (S is the signal rate and the B is the bandwidth.) the required bandwidth has a
minimum value of S and maximum value of 2S.

Frequency Shift Keying

The digital data stream changes the frequency of the carrier signal, fc
For example, a “1” could be represented by f1=fc +∆f; and a “0” could be represented by f2=fc-∆f.
Both peak amplitude and phase remain constant for all signal elements.
Binary FSK (BFSK)
Binary FSK consider two carrier frequencies. Normally the carrier frequencies are very high, and the difference between
them is very small.
Bandwidth for BFSK: If the difference between the two frequencies (f1 and f2) is 2∆f, then the required BW B will be: B =
(1+d)xS +2∆f
Coherent and Non Coherent
In a non-coherent FSK scheme, when we change from one frequency to the other, we do not adhere to the current phase of
the signal. In coherent FSK, the switch from one frequency signal to the other only occurs at the same phase in the signal.
Multi level FSK
Similarly to ASK, FSK can use multiple bits per signal element. That means we need to provision for multiple frequencies,
each one to represent a group of data bits. The bandwidth for FSK can be higher B = (1+d)xS + (L-1)/2∆f = LxS
Phase Shift Keyeing
We vary the phase shift of the carrier signal to represent digital data.
The bandwidth requirement, B is: B = (1+d)xS
PSK is much more robust than ASK as it is not that vulnerable to noise

Quadrature PSK
To increase the bit rate, we can code 2 or more bits onto one signal element. In QPSK, we parallelize the bit stream so that
every two incoming bits are split up and PSK a carrier frequency. One carrier frequency is phase shifted 90o from the other
- in quadrature. The two PSKed signals are then added to produce one of 4 signal elements. L = 4 here.
Constellation Diagrams

Quadrature Amplitude Modulation(QAM)=ASK+PSK

Bandwidth for QAM
The minimum bandwidth required for QAM transmission is the same as that required
for ASK and PSK transmission. QAM has the same advantages as PSK over ASK
2.3-2 Analog To Analog Conversion.
Analog-to-analog conversion is the representation of analog information by an analog signal.
Amplitude Modulation
A carrier signal is modulated only in amplitude value. The modulating signal is the envelope of the carrier
Bandwidth is 2B, where B is the bandwidth of the modulating signal
Since on both sides of the carrier fc , the spectrum is identical, we can discard one half, thus requiring a smaller bandwidth
for transmission.

Frequency Modulation
The modulating signal changes the f c of the carrier signal
The total bandwidth required for FM can be determined from the bandwidth of the audio signal:
B_FM = 2(1 + β)B. Where is usually 4.
Phase Modulation (PM)
The modulating signal only changes the phase of the carrier signal. The phase change manifests itself as a frequency
change but the instantaneous frequency change is proportional to the derivative of the amplitude. The bandwidth is higher
than for AM.

The total bandwidth required for PM can be determined from the bandwidth and maximum amplitude of the modulating
signal: B_PM = 2(1 + β)B. Where β = 2 most often.

Chapter 2.4 Bandwidth Utilization: Multiplexing and Spreading

Bandwidth utilization is the wise use of available bandwidth to achieve specific goals. Efficiency can be achieved by
multiplexing; sharing of the bandwidth between multiple users.
2.4-1 Multiplexing
Ghép kênh (Multiplexing) là truyền nhiều tín hiệu trên một kênh truyền.
Frequency-Division Multiplexing(FDM)
FDM is an analog multiplexing technique that combines analog signals.

Example: Five channels, each with a 100-kHz bandwidth, are to be multiplexed together. What is the minimum bandwidth
of the link if there is a need for a guard band of 10 kHz between the channels to prevent interference?
Solution For five channels, we need at least four guard bands. This means that the required bandwidth is at least 5 × 100 +
4 × 10 = 540 kHz
Example 6.3
Four data channels (digital), each transmitting at 1 Mbps, use a satellite channel of 1 MHz. Design an appropriate
configuration, using FDM.
Wavelength-division multiplexing (WDM)
WDM is an analog multiplexing technique to combine optical signals. WDM technology is very complex, the basic idea is
very simple.
Time-division multiplexing (TDM)
TDM is a digital process that allows several connections to share the high bandwidth of a link. Instead of sharing a portion
of the bandwidth as in FDM, time is shared. Each connection occupies a portion of time in the link.
Note that the same link is used as in FDM;but, the link is shown sectioned by time rather than by frequency.We can divide
TDM into two different schemes: synchronous or statistical.
In synchronous TDM, each input connection has an allotment in the output even if it is not sending data. In statistical TDM,
slots are dynamically allocated to improve bandwidth efficiency.

TDM is a digital multiplexing technique for combining several low-rate digital channels into one high-rate one.
Synchronous TDM
The data rate of the link is n times faster, and the unit duration is n times shorter.
Ex: The data rate for each input connection is 1 kbps. If 1 bit at a time is multiplexed (a unit is 1 bit), what is the duration of
1. each input slot,
2. each output slot, and
3. each frame?
Solution
1. The data rate of each input connection is 1 kbps. This means that the bit duration is 1/1000 sor 1 ms. The duration of the
input time slot is 1 ms (same as bit duration).
2. The duration of each output time slot is one-third of the input time slot. This means that the duration of the output time
slot is 1/3 ms.
3. Each frame carries three output time slots. So the duration of a frame is 3 × 1/3 ms, or 1 ms. The duration of a frame is
the same as the duration of an input unit.
Interleaving
The process of taking a group of bits from each input line for multiplexing is called interleaving. We interleave bits (1 - n)
from each input onto one output.
Data Rate Management
Not all input links maybe have the same data rate. Some links maybe slower. There maybe several different input link
speeds There are three strategies that can be used to overcome the data rate mismatch: multilevel, multislot and pulse
stuffing
Data rate matching
Multilevel: used when the data rate of theinput links are multiples of each other.
Multislot: used when there is a GCD betweenthe data rates. The higher bit rate channelsare allocated more slots per frame,
and the output frame rate is a multiple of each input link.
Pulse Stuffing: used when there is no GCD between the links. The slowest speed link will be brought up to the speed of the
other links by bit insertion, this is called pulse stuffing.
Synchronization
To ensure that the receiver correctly reads the incoming bits,, knows the incoming bit boundaries to interpret a “1” and a
“0”, a known bit pattern is used between the frames. The receiver looks for the anticipated bit and starts counting bits till
the end of the frame. Then it starts over again with the reception of another known bit. These bits (or bit patterns) are
called synchronization bit(s). They are part of the overhead of transmission.

2.4-2 Spread Spectrum

In spread spectrum (SS), we combine signals from different sources to fit into a larger bandwidth, but our goals are to
prevent eavesdropping and jamming. To achieve these goals, spread spectrum techniques add redundancy.
Spread spectrum is designed to be used in wireless applications in which stations must be able to share the medium without
interception by an eavesdrop- per and without being subject to jamming from a malicious intruder. The frequency hopping
spread spectrum (FHSS) technique uses M different carrier frequencies that are modulated by the source signal. At one
moment, the signal modulates one carrier frequency; at the next moment, the signal modulates another carrier frequency.
The direct sequence spread spectrum (DSSS) technique expands the bandwidth of a signal by replacing each data bit with n
bits using a spreading code. In other words, each bit is assigned a code of n bits, called chips.

Chapter 3.1 Error Detection and Correction

Overview of Data Link layer
Role: transform the physical layer (a raw transmission facility) to a link responsible for node-to-node
communications.
Responsibilities:
+ Framing: divides the bit stream received from the network layer to manageable data units (frames).
+ Addressing: adds a header to a frame to define the address of sender and receiver.
+ Flow control: to avoid overwhelming (choáng ngợp) receiver when receiving rate is smaller than sending rate.
+ Error control: adds mechanisms to detect and retransmit damaged, duplicate and lost frames. Media access
control: determines which devices get access to a shared link at a given time the when more than 2 devices are
connected with the same link
Data can be corrupted during transmission. Some applications require that errors be detected and corrected.
Types of Errors
Whenever bits flow from one point to another, they are subject to unpredictable changes because of interference. This
interference can change the shape of the signal. The term single-bit error means that only 1 bit of a given data unit (such as
a byte, character, or packet) is changed from 1 to 0 or from 0 to 1. The term burst error means
that 2 or more bits in the data unit have changed from 1 to 0 or from 0 to 1.
To detect or correct errors, we need to send extra (redundant) bits with data.The central concept in detecting or correcting
errors is redundancy. To be able to detect or correct errors, we need to send some extra bits with our
data. These redundant bits are added by the sender and removed by the receiver.
Detection Versus Correction: The correction of errors is more difficult than the detection.
Coding
o Redundancy is achieved through various coding schemes. The sender adds redundant bits through a process that creates a
relationship between the redundant bits and the actual data bits.
o The receiver checks the relationships between the two sets of bits to detect errors. The ratio of redundant bits to data bits
and the robustness of the process are important factors in any coding scheme.
Block Coding
In block coding, we divide our message into blocks, each of k bits, called datawords. We add r redundant
bits to each block to make the length n = k + r. The resulting n-bit blocks are called codewords.
The 4B/5B block coding is example of this type of coding. In this coding scheme, k = 4 and n = 5. As we saw, we have 2^k
= 16 datawords and 2^n = 32 codewords. We saw that 16 out of 32 codewords are used for message transfer and the rest are
either used for other purposes or unused.
Error Detection
If The following two conditions are met, the receiver can detect a change in the original codeword.
1. The receiver has a list of valid codewords.
2. The original codeword has changed to an invalid one
● Enough redundancy is added to detect an error.
● The receiver knows an error occurred but does not know which bit(s) is(are) in error.
● Has less overhead than error correction.
An error-detecting code can detect only the types of errors for which it is designed; other types of errors may remain
undetected.

Hamming Distance
One of the central concepts in coding for error control is the idea of the Hamming distance. The Hamming distance
between two words (of the same size) is the number of differences between the corresponding bit
The Hamming distance between two words is the number of differences between corresponding bits.
Let us find the Hamming distance between two pairs of words.
1. The Hamming distance d(000, 011) is 2 because (000 ⊕ 011) is 011 (two 1s).
2. The Hamming distance d(10101, 11110) is 3 because (10101 ⊕ 11110) is 01011 (three 1s)The minimum Hamming
distance is the smallest Hamming distance between all possible pairs in a set of words.
To guarantee the detection of up to s errors in all cases, the minimum Hamming distance in a block code must be
d_min=s+1
To guarantee correction of up to t errors in all cases, the minimum Hamming distance in a block code must be d_min = 2t
+ 1.
Linear Block Codes
Almost all block codes used today belong to a subset called linear block codes. A linear block code is a code in which the
exclusive OR (addition modulo-2) of two valid codewords creates another valid codeword.
In a linear block code, the exclusive OR (XOR) of any two valid codewords creates another valid codeword.
A simple parity-check code is a single-bit error-detecting code in which n = k + 1 with dmin = 2. Even parity (ensures that
a codeword has an even number of 1’s) and odd parity (ensures that there are an odd number of 1’s in the codewor
Parity-Check Code
This code is a linear block code. In this code, a k-bit dataword is changed to an n-bit codeword where n = k + 1. The extra
bit, called the parity bit, is selected to make the total number of 1s in the codeword even.
A parity-check code can detect an odd number of errors.

All Hamming codes have dmin = 3 (2 bit error detection and single bit error
correction). A codeword consists of n bits of which k are data bits and r are check bits.
Let m = r, then we have: n = 2^m -1 and k = n-m
Cyclic codes are special linear block codes with one extra property. In a cyclic code, if a codeword is cyclically shifted
(rotated), the result is another codeword.

Check sum
Sender site:
1. The message is divided into 16-bit words.
2. The value of the checksum word is set to 0.
3. All words including the checksum are added using one’s complement addition.
4. The sum is complemented and becomes the checksum.
5. The checksum is sent with the data.
Receiver site:
1. The message (including checksum) is divided into 16-bit words.
2. All words are added using one’s complement addition.
3. The sum is complemented and becomes the new checksum.
4. If the value of checksum is 0, the message is accepted; otherwise, it is rejected.
Chapter 3.2 Data Link Control
Framing
•The data link layer needs to pack bits into frames.
•Each frame is distinguishable from another.
•Framing types:
• Fixed-Size Framing: Boundary between frames is
not necessary e.g. ATM networks.
• Variable-Size Framing: need methods to define the end and the beginning of frames. We will discuss Character-oriented
and Bit-oriented methods.

Flow And Error Control

The most important responsibilities of the data link layer are flow control and error control. These functions are known as
data link control. Error control in the data link layer is based on automatic repeat request, which is the retransmission of
data.
Flow control refers to a set of procedures used to restrict (hạn chế) the amount of data that the sender can send before
waiting for acknowledgment.
Protocols
- How the data link layer can combine framing, flow control, and error control to achieve the delivery of data from one
node to another?
- The protocols are normally implemented in software.

Noiseless Channels
Stop-and-Wait Protocol: uses both flow and error control.
Sender States
The sender is initially in the ready state, but it can move between the ready and blocking state.
❑Ready State. When the sender is in this state, it is only waiting for a packet from the network layer. If a packet comes
from the network layer, the sender creates a frame, saves a copy of the frame, starts the only timer and sends the frame. The
sender then moves to the blocking state.
❑ Blocking State. When the sender is in this state, three events can occur:
a. If a time-out occurs, the sender resends the saved copy of the frame and restarts the timer.
b. If a corrupted ACK arrives, it is discarded.
c. If an error-free ACK arrives, the sender stops the timer and discards the saved copy of the frame. It then moves to the
ready state.
Receiver
The receiver is always in the ready state. Two events may occur:
a. If an error-free frame arrives, the message in the frame is delivered to the network layer and an ACK is sent.
b. If a corrupted frame arrives, the frame is discarded.
Noisy Channels
Three protocols in this section that use error control.
● Stop-and-Wait Automatic Repeat Request
● Go-Back-N Automatic Repeat Request
● Selective Repeat Automatic Repeat Request
Stop-and-Wait ARQ
• Keep a copy of the sent frame and retransmit the frame when the timer expires.
• Use sequence numbers to number the frames.
• The acknowledgment number always announces the sequence number of the next frame expected.
• Sequence numbers are based on modulo-2 arithmetic

Link utilization
• The system can send 20,000 bits during the time it takes for the data to go from the sender to the receiver and then back
again.
• However, the system sends only 1000 bits i.e. the link utilization is only 1000/20,000, or 5%.
• For a link with a high bandwidth or long delay, the use of Stop-and-Wait ARQ wastes the capacity of the link.
Go-Back-N Automatic Repeat Request
Selective Repeat Automatic Repeat Request
• Send several frames before receiving acknowledgments
• Keep a copy of sent frames before acknowledgements arrive
• In the Go-Back-N Protocol, the sequence numbers are modulo 2m, where m is the size of the sequence number field in
bits.

The send window is an abstract concept defining an imaginary box of size 2^m − 1 with three variables: Sf, Sn, and S_size.
The send window can slide one or more slots when a valid acknowledgment arrives

The receive window is an abstract concept defining an imaginary box of size 1 with one single variable Rn. The window
slideswhen a correct frame has arrived; sliding occurs one slot at a time.

In Go-Back-N ARQ, the size of the send window must be less than 2^m;the size of the receiver window is always 1.
Stop-and-Wait ARQ is a special case of Go-Back-N ARQ in which the size of the send window is 1.
Send window for Selective Repeat ARQ
ARQ go-back-N đơn giản hóa phía thu, bộ thu chỉ cần 1 biến Rn, do đó không cần bộ đệm cho các khung không đúng thứ
tự, các khung này bị loại bỏ, nên không hiệu quả
• Chỉ gửi lại 1 khung bị lỗi thay vì phải gửi lại toàn bộ N khung tính từ khung bắt đầu bị lỗi-> gọi là lặp lại tự động có lựa
chọn. Kích thước cửa sổ gửi và nhận cùng bằng 2^(m-1)
Receive window for Selective Repeat ARQ
Giao thức này cho phép bên nhận nhận nhiều khung không đúng thứ tự và giữ cho đến khi có đủ các khung theo đến theo
đúng thứ tự
Design of Selective Repeat ARQ

Selective Repeat ARQ, window size

Trong Selective Repeat ARQ, kích thước của cửa sổ gửi và nhận nhiều nhất là 2^(m-1)
In Selective Repeat ARQ, the size of the sender and receiver window must be at most one-half of 2^m

HDLC
High-level Data Link Control (HDLC) is a bit-oriented protocol for communication over point-to-point and multipoint
links. It implements the ARQ mechanisms we discussed in this chapter.
Point-to-point Protocol
Although HDLC is a general protocol that can be used for both point-to-point and multipoint configurations, one of the
most common protocols for point-to-point access is the Point-to-Point Protocol (PPP). PPP is a byte-oriented protocol.
PPP frame format

PPP is a byte-oriented protocol using byte stuffing with the escape byte 01111101.

Chapter 3.3: Multiple Access

Data link layer divided into two functionality-oriented sublayers: Data-link control and multiple-access resolution
Taxonomy of multiple-access protocols discussed in this chapter
Random access protocol
No station is superior to another station and none is assigned the control over another. No station permits, or does not
permit, another station to send. At each instance, a station that has data to send uses a procedure defined by the protocol to
make a decision on whether or not to send.
ALOHA, the earliest random access method

The data from the two stations collide and become garbled. The idea is that each station sends a frame whenever it has a
frame to sendsince there is only one channel to share, there is the possibility of collision between frames from different
stations. Four stations (unrealistic assumption) that contend with one another for access to the shared channel.
Pure ALOHA vulnerable time = 2 x T_fr.
The throughput for pure ALOHA is S = G × e −2G .The maximum throughput Smax = 0.184 when G= (1/2).
G là số khung trung bình tạo ra bởi hệ thống trong thời gian truyền dẫn 1 khung.
The throughput for slotted ALOHA is S = G × e^ −G. The maximum throughput Smax = 0.368 when G = 1.
CSMA/CD
The CSMA method does not specify the procedure following a collision. Carrier sense multiple access with collision
detection (CSMA/CD) augments the algorithm to handle the collision.
CSMA/CA Carrier sense multiple access with collision avoidance (CSMA/CA) was invented for wireless networks.
Collisions are avoided through the use of CSMA/CA’s three strategies: the interframe space, the contention window, and
acknowledgments. In CSMA/CA, the IFS can also be used to define the priority of a station or a frame. In CSMA/CA, if
the station finds the channel busy, it does not restart the timer of the contention window; it stops the timer and restarts it
when the channel becomes idle.
Controlled Access
In controlled access, the stations consult (hỏi) one another to find which station has the right to send. A station cannot send
unless it has been authorized by other stations. We discuss three popular controlled-access methods.
Channelization
Channelization is a multiple-access method in which the available bandwidth of a link is shared in time, frequency, or
through code, between different stations. In this section, we discuss three channelization protocols.

In FDMA, the available bandwidth of the common channel is divided into bands that are separated by guard bands.
In TDMA, the bandwidth is just one channel that is timeshared between different stations.
In CDMA, one channel carries all transmissions simultaneously
Idea Let us assume we have four stations, 1, 2, 3, and 4, connected to the same channel. The data from station 1 are d1,
from station 2 are d2, and so on. The code assigned to the first station is c1, to the second is c2, and so on. We assume that
the assigned codes have two properties.
1. If we multiply each code by another, we get 0.
2. If we multiply each code by itself, we get 4 (the number of stations).
The number of sequences in a Walsh table needs to be N = 2^m
Chapter 4.1 Logical Addressing
Ipv4 Addresses
An IPv4 address is a 32-bit address that uniquely and universally defines the connection of a device to the Internet
The address space of IPv4 is 2^32 or 4,294,967,296. Each number needs to be less than or equal to 255.

Classful addressing, the address space is divided into five classes: A, B, C, D, and E.
In classful addressing, a large part of the available addresses were wasted.
Advs: Given an address, we can easily find the class of the address and, since the prefix length for each class is fixed, we
can find the prefix length immediately. In other words, the prefix length in classful addressing is inherent in the address; no
extra information is needed to extract the prefix and the suffix.

Classless Addressing

CIDR = Classless Inter-Domain Routing

Classful addressing, which is almost obsolete, is replaced with classless addressing
Ex1: A classless address is given as 167.199.170.82/27. We can find the above three pieces of infor-
mation as follows. The number of addresses in the network is 2^(32 − n )= 2^5 = 32 addresses.

Address Mask
Another way to find the first and last addresses in the block is to use the address mask. The address mask is a 32-bit
number in which the n leftmost bits are set to 1s and the rest of the bits (32 − n) are set to 0s. A computer can easily find
the address mask because it is the complement of (232 − n − 1). The reason for defining a mask in this way is that it can be
used by a computer program to extract the information in a block, using the three bit-wise operations NOT, AND, and OR.
1. The number of addresses in the block N = NOT (mask) + 1.
2. The first address in the block = (Any address in the block) AND (mask).
3. The last address in the block = (Any address in the block) OR [(NOT (mask)].

Ex2: Repeat Ex1 using the mask. The mask in dotted-decimal notation is 256.256.256.224. The AND, OR, and NOT
operations can be applied to individual bytes using calculators and applets at the book website.

Number of addresses in the block: N = NOT (mask) + 1= 0.0.0.31 + 1 = 32 addresses

First address: First = (address) AND (mask) = 167.199.170.82
Last address: Last = (address) OR (NOT mask) = 167.199.170.255

The first address in a block is normally not assigned to any device; it is used as the network address that represents the
organization to the rest of the world.
Each address in the block can be considered as a two-level hierarchical structure: the leftmost n bits (prefix) define the
network; the rightmost 32 − n bits define the host.
Subnetting
An organization (or an ISP) that is granted a range of addresses may divide the range into several subranges and assign
each subrange to a subnetwork (or subnet). Note that nothing stops the organization from creating more levels. A
subnetwork can be divided into several sub-subnetworks.
A sub-subnetwork can be divided into several sub-sub-subnetworks.
Designing Subnets
The subnetworks in a network should be carefully designed to enable the routing of packets. We assume the total number of
addresses granted to the organization is N, the prefix length is n, the assigned number of addresses to each subnetwork is
N_sub, and the prefix length for each subnetwork is nsub. Then the following steps need to be carefully followed to
guarantee the proper operation of the subnetworks.
❑ The number of addresses in each subnetwork should be a power of 2.
❑ The prefix length for each subnetwork should be found using the following formula: n_sub = 32 − log2N_sub
❑ The starting address in each subnetwork should be divisible by the number of addresses in that subnetwork. This can be
achieved if we first assign addresses to larger subnetworks.
.After designing the subnetworks, the information about each subnetwork, such as first and last address, can be found using
the process we described to find the information about each network in the Internet.
Example
Ex: An ISP is granted a block of addresses starting with 190.100.0.0/16 (65,536 addresses). The ISP needs to distribute
these addresses to three groups of customers as follows:
a. The first group has 64 customers; each needs 256 addresses.
b. The second group has 128 customers; each needs 128 addresses.
c. The third group has 128 customers; each needs 64 addresses.
Design the subblocks and find out how many addresses are still available after these allocations.

Chapter 4.2. - Internet Protocol

Internetworking: connecting networks together to make an internetwork or an Internet.
Communication at the network layer in the Internet is connectionless. Switching at the network layer in the Internet uses
the datagram approach to packet switching.

IPv4
The Internet Protocol version 4 (IPv4) is the delivery mechanism used by the TCP/IP protocols.

Ex1:: An IPv4 packet has arrived with the first 8 bits as shown: 01000010 The receiver discards the packet. Why?
=> There is an error in this packet. The 4 leftmost bits (0100) show the version, which is correct （4= IPv4, 6= IPv6). The
next 4 bits (0010) show an invalid header length (2 × 4 = 8). The minimum number of bytes in the header must be 20. The
packet has been corrupted in transmission.

Ex2 In an IPv4 packet, the value of HLEN is 1000 in binary.How many bytes of options are being carried by this packet?
=> The HLEN value is 8, which means the total number of bytes in the header is 8 × 4, or 32 bytes. The first 20 bytes are
the base header, the next 12 bytes are the options.

Ex3: In an IPv4 packet, the value of HLEN is 5, and the value of the total length field is 0x0028. How many bytes of data
are being carried by this packet?
The HLEN value is 5, which means the total number of bytes in the header is 5 × 4, or 20 bytes (no options). The total
length is 40 bytes, which means the packet is carrying 20 bytes of data (40 − 20)

Ex4: An IPv4 packet has arrived with the first few hexadecimal digits as shown. 0x45000028000100000102 . . .How many
hops can this packet travel before being dropped? The data belong to what upper-layer protocol?
=>To find the time-to-live field, we skip 8 bytes. The time-to- live field is the ninth byte, which is 01. This means the
packet can travel only one hop. The protocol field is the next byte (02), which means that the upper-layer protocol is IGMP
Phân mảnh gói tin:
o Trường nhận dạng: 16 bit, nhận dạng gói tin được tạo ra từ nguồn, là duy nhất được copy vào tất cả các mảnh, giúp đích
tổng hợp lại gói tin
o Cờ: 3 bit, Bit đầu tiên không sử dụng, Bit thứ 2 (D): không phân mảnh, nếu nó bằng 1 thì nút không phải phân mảnh gói
tin, nếu nó bằng 0 thì có thể phân mảnh nếu cần. Bit cuối cùng (M): 1 tức là không phải mảnh cuối, 0 có nghĩa là mảnh
cuối hoặc chỉ có 1 mảnh.
o Offset: 13 bit cho biết vị trí tương đối của mảnh so với toàn bộ gói tin
o Được đo theo đơn vị 8 byte

IPv6
Transition From Ipv4 To Ipv6
Because of the huge number of systems on the Internet, the transition from IPv4 to IPv6 cannot happen suddenly. It takes a
considerable amount of time before every system in the Internet can move from IPv4 to IPv6. The transition must be
smooth to prevent any problems between IPv4 and IPv6 systems
3 strategy: Dual Stack; Tunneling; Header Translation
Chapter 4.3: Routing-Phân phối, chuyển tiếp và định tuyến
Phân Phối

Chuyển Tiếp - Forwarding

• Chuyển tiếp có nghĩa là đạt gói tin trên đường đi để đến đích
• Chuyển tiếp yêu cầu một host hay một router có bảng định tuyến.
• Khi một host có một gói tin gửi đi hay khi một router nhận được một gói tin cần được chuyển tiếp, thì sẽ dựa vào bảng
định tuyến để tìm đường đi đến đích cuối cùng.

Ví dụ: Xây dựng một bảng định tuyến cho router R1, sử dụng định tuyến trong Hình 4.4.6.
Ví dụ 2: Biểu diễn quá trình chuyển tiếp nếu một gói tin đến tại R1 trong Hình 4.4.6 với địa chỉ đích là 180.70.65.140.
=> Router thực hiện các bước sau:
1. Mặt nạ đầu tiên (/26) được áp dụng cho địa chỉ đích. Kết quả là 180.70.65.128, không tương thích với địa chỉ mạng
tương ứng.
2. Mặt nạ thứ hai (/25) được áp dụng cho địa chỉ đích. Kết quả là 180.70.65.128, phù hợp với địa chỉ mạng tương ứng. Địa
chỉ hop tiếp theo và giao diện m0 được chuyển cho ARP để thực hiện quá trình tiếp theo.

Ví dụ 3: Biểu diễn quá trình chuyển tiếp nếu một gói tin đến tại R1 trong Hình 4.4.6 với địa chỉ đích là 201.4.22.35.
=> Router thực hiện các bước sau:
1. Mặt nạ đầu tiên (/26) được áp dụng cho địa chỉ đích. Kết quả là 201.4.22.0, không phù hợp với địa chỉ mạng tương ứng.
2. mặt nạ thứ hai (/25) được áp dụng cho địa chỉ đích. Kết quả là 201.4.22.0, không tương thích với địa chỉ mạng tương ứng
(row 2).
3. Mặt nạ thứ ba (/24) được áp dụng cho địa chỉ đích. Kết quả là 201.4.22.0, phù hợp với địa chỉ mạng tương ứng. Địa chỉ
đích của gói tin và giao diện m3 được chuyển tới ARP.

Ví dụ 4: Biểu diễn quá trình chuyển tiếp nếu một gói tin đến tại R1 trong Hình 4.4.6 với địa chỉ đích 18.24.32.78.
=> tất cả các mặt nạ đều được lần lượt áp dụng cho địa chỉ đích, nhưng không tìm thấy địa chỉ mạng phù hợp. Khi kết thúc
bảng, khối mô-đun đia địa chỉ của hop tiếp theo 180.70.65.200 và giao diện m2 đến ARP. Điều này có thể là một gói tin đi
ra ngoài cần được gửi đi, thông qua một router mặc định, để đến một nơi nào đó trong Internet.

Example 1:
As an example of hierarchical routing, let us consider Figure 4.4.9. A regional ISP is granted 16,384 addresses starting
from 120.14.64.0. The regional ISP has decided to divide this block into four subblocks, each with 4096 addresses. Three
of these subblocks are assigned to three local ISPs; the second subblock is reserved for future use. Note that the mask for
each block is /20 because the original block with mask /18 is divided into 4 blocks.
The first local ISP has divided its assigned subblock into 8 smaller blocks and assigned each to a small ISP. Each small ISP
provides services to 128 households, each using four addresses.
The second local ISP has divided its block into 4 blocks and has assigned the addresses to four large organizations. There is
a sense of hierarchy in this configuration. All routers in the Internet send a packet with destination address 120.14.64.0 to
120.14.127.255 to the regional ISP.
The third local ISP has divided its block into 16 blocks and assigned each block to a small organization. Each small
organization has 256 addresses, and the mask is /24.
 Unicast Routing Protocols
•A routing table can be either static or dynamic.
•A static table: manual entries.
•A dynamic table: is updated automatically when there is a change somewhere in the Internet.
•A routing protocol is a combination of rules and procedures that lets routers in the Internet inform each other of changes.

Distance Vector Routing: In distance vector routing, each node shares its routing table with its immediate neighbors
periodically and when there is a change.
Link State Routing
Chapter 5.1: Protocols: UDP, TCP, SCTP

Process-to-process Delivery
•A process is an application program running on a host.
•The transport layer is responsible for process-to- process delivery—the delivery of a packet, part of a message, from one
process to another.
•Two processes communicate in a client/server relationship.
User Datagram Protocol (Udp)
•The User Datagram Protocol (UDP) is called a connectionless, unreliable transport protocol.
•It does not add anything to the services of IP except to provide process-to-process communication instead of host-to-host
communication.

TCP
TCP is a connection-oriented protocol. It creates a virtual connection between two TCPs to send data.
In addition, TCP uses flow and error control mechanisms at the transport level.
Chapter 6: Application Layer
Dịch vụ truyền File FTP (File Transfer Protocol) là một trong những dịch vụ sớm nhất ứng dụng giao thức TCP/ IP. FTP
cho phép người dùng thực hiện các chức năng: Sao chép, Đổi tên, Xóa file, Tạo thư mục …..ở một hệ thống ở xa.
Hệ thống FTP ở xa thường yêu cầu người dùng cung cấp định danh ID và mật khẩu trước khi truy nhập hệ thống. Các máy
chủ thường cung cấp hai dạng dịch vụ truy nhập.
* Truy nhập vào các file công cộng dùng chung qua tài khoản ẩn danh (Anonymous).
* Truy nhập vào các file riêng chỉ dành cho những người sử dụng với quyền truy nhập ở mức hệ thống.
Network File System (NFS)
Hệ tập tin mạng (Network File System-NFS) cung cấp việc truy xuất trực tuyến các tập tin dùng chung. Người sử dụng có
thể thực hiện một chưng trình ứng dụng bất kỳ và sử dụng bất kỳ một tập tin nào trong việc xuất nhập. Bn thân tên các tập
tin không cho biết chúng cục bộ hay ở xa. NFS là một RPC (Remote Procedure Call )
Domain Name Service (DNS)
Đối với những người truy nhập Internet, việc nhớ nhiều địa chỉ IP cùng một lúc là rất khó. Do đó, các nhà thiết kế tạo nên
những tên dễ nhớ. Người dùng muốn truy nhập đến địa chỉ nào thì chỉ việc gõ bàn phím những tên đó vào. Tuy nhiên, giao
thức lớp mạng IP chỉ có thể hiệu và làm việc được với địa chỉ IP. Do vậy cần có sự chuyển đổi qua lại giữa tên và địa chỉ
IP. Việc chuyển đổi tên thành địa chỉ được thực hiện qua hệ thống tên miền (Domain Name System – DNS). Hệ thống DNS
thực chất là những CSDL (DNS database) chứa tên và địa chỉ tưng ứng cùng với các thông tin khác đi kèm.
Dịch vụ Mail: là dịch vụ thư điện tử. Để dịch vụ Mail hoạt động được thì phải đảm bảo 2 thành phần: Mail Server, Mail
Client.