LECTURE NOTES

ON
Data Communication and Networks

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UNIT II
Data Link Layer - Design issues - Channel allocation problem - Multiple access
protocols - Ethernet - Wireless LAN - 802.11 architecture

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 UNIT II

DATAGRAM NETWORKS
Data Link Layer: Error detection and correction Data Link Control:
CYCLIC REDUNDANCY CHECK (CRC):
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. For example, if 1011000
is a codeword and we cyclically left-shift, then 0110001 is also a codeword. In this case, if we
call the bits in the first word ao to a6' and the bits in the second word bo to b6.

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In the encoder, the dataword has k bits (4 here); the codeword has n bits (7 here). The size of the
dataword is augmented by adding n - k (3 here) Os to the right-hand side of the word. The n-bit result
is fed into the generator. The generator uses a divisor of size n - k + I (4 here), predefined and agreed
upon. The generator divides the augmented dataword by the divisor (modulo-2 division). The
quotient of the division is discarded; the remainder (r2r1r0) is appended to the dataword to create the
codeword. The decoder receives the possibly corrupted codeword. A copy of all n bits is fed to the
checker which is a replica of the generator. The remainder produced by the checker is a syndrome of
n - k (3 here) bits, which is fed to the decision logic analyzer. The analyzer has a simple function. If
the syndrome bits are all as, the 4 leftmost bits of the codeword are accepted as the dataword
(interpreted as no error); otherwise, the 4 bits are discarded (error). Encoder
Let us take a closer look at the encoder. The encoder takes the dataword and augments it with n -
k number of 0's. It then divides the augmented dataword by the divisor, as shown in Figure
10.15.

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 As in decimal division, the process is done step by step. In each step, a copy of the divisor is
XORed with the 4 bits of the dividend. The result of the XOR operation (remainder) is 3 bits (in
this case), which is used for the next step after 1 extra bit is pulled down to make it 4 bits long.
There is one important point we need to remember in this type of division. If the leftmost bit of
the dividend (or the part used in each step) is 0, the step cannot use the regular divisor; we need
to use an all-Os divisor. When there are no bits left to pull down, we have a result. The 3-bit
remainder forms the check bits (r2' r1 and r0). They are appended to the dataword to create the
codeword.
Framing, Flow and Error Control Protocols
Decoder
The codeword can change during transmission. The decoder does the same division process as
the encoder. The remainder of the division is the syndrome. If the syndrome is all Os, there is no
error; the dataword is separated from the received codeword and accepted. Otherwise, everything
is discarded. Figure 10.16 shows two cases: The left hand figure shows the value of syndrome
when no error has occurred; the syndrome is 000. The right-hand part of the figure shows the
case in which there is one single error. The syndrome is not all 0s .

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 Divisor
Let us first consider the divisor. We need to note the following points:
1. The divisor is repeatedly XORed with part of the dividend.
2. The divisor has n - k + 1 bits which either are predefined or are all Os. In other words, the bits
do not change from one dataword to another. In our previous example, the divisor bits were
either 1011 or 0000. The choice was based on the leftmost bit of the part of the augmented data
bits that are active in the XOR operation.
3. A close look shows that only n - k bits of the divisor is needed in the XOR operation. The
leftmost bit is not needed because the result of the operation is always 0, no matter what the
value of this bit. The reason is that the inputs to this XOR operation are either both Os or both 1s.
In our previous example, only 3 bits, not 4, is actually used in the XOR operation.
Polynomials
A better way to understand cyclic codes and how they can be analyzed is to represent them as
polynomials.
A pattern of Os and 1s can be represented as a polynomial with coefficients of 0 and 1. The
power of each term shows the position of the bit; the coefficient shows the value of the bit.
Figure 10.21 shows a binary pattern and its polynomial representation. In Figure 10.21a we show
how to translate a binary pattern to a polynomial; in Figure 1O.21b we show how the polynomial
can be shortened by removing all terms with zero coefficients.

Figure 10.21 shows one immediate benefit; a 7-bit pattern can be replaced by three terms. The
benefit is even more conspicuous when we have a polynomial such as x23 + X3 + 1. Here the bit
pattern is 24 bits in length (three Is and twenty-one Os) while the polynomial is just three terms.

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 PARITY CHECK CODE
The most familiar error-detecting code is the simple parity-check 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 Is in the codeword even. Although some implementations
specify an odd number of 1's. The minimum Hamming distance for this category is dmin =2,
which means that the code is a single-bit error-detecting code; it cannot correct any error.
Our first code (Table 10.1) is a parity-check code with k -= 4 and n =5.

Figure 10.10 shows a possible structure of an encoder (at the sender) and a decoder
(at the receiver). The encoder uses a generator that takes a copy of a 4-bit dataword (ao, aI' a2'
and a3) and generates a parity bit roo The dataword bits and the parity bit create the 5-bit
codeword. The parity bit that is added makes the number of Is in the codeword even.

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 This is normally done by adding the 4 bits of the dataword (modulo-2); the result is the
parity bit. In other words,

If the number of 1s is even, the result is 0; if the number of 1s is odd, the result is 1. In
both cases, the total number of 1s in the codeword is even. The sender sends the codeword
which may be corrupted during transmission. The receiver receives a 5-bit word. The
checker at the receiver does the same thing as the generator in the sender with one
exception: The addition is done over all 5 bits. The result, which is called the syndrome, is
just 1 bit. The syndrome is 0 when the number of Is in the received codeword is even;
otherwise, it is 1.

The syndrome is passed to the decision logic analyzer. If the syndrome is 0, there is no
error in the received codeword; the data portion of the received codeword is accepted as
the dataword; if the syndrome is 1, the data portion of the received codeword is discarded.
The dataword is not created.

Noiseless Channel and Noisy Channel Protocol:

Noiseless Channel
An ideal channel in which no frames are lost, duplicated or corrupted is regarded as Noiseless
Channel.
Simplest Protocol
 In simplest protocol, there is no flow control and error control mechanism. It is a
unidirectional protocol in which data frames travel in only one direction (from sender
to receiver).
 Also, the receiver can immediately handle any received frame with a processing time
that is small enough to be negligible.

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 The protocol consists of two distinct procedures :a sender and receiver. The sender runs
in the data link layer of the source machine and the receiver runs in the data link layer of the
destination machine. No sequence number or acknowledgements are used here.
Stop and Wait Protocol
 The simplest retransmission protocol is stop-and-wait.
 Transmitter (Station A) sends a frame over the communication line and then waits for a
positive or negative acknowledgement from the receiver (station B).
 If no error occurs in the transmission, station B sends a positive acknowledgement
(ACK) to station A.
 Now, the transmitter starts to send the next frame. If frame is received at station B with
errors, then a negative acknowledgement(NAK) is sent to station A. In this case, station 'A' must
retransmit the old packet in a new frame.
 There is also a possibility that the information frames or ACKs may get lost.
 Then, the sender is equipped with a timer. If no recognizable acknowledgement is
received when the timer expires at the end of time out interval, the same frame is sent again.
 The sender which sends one frame and then waits for an acknowledgement
before process is known as stop and wait.

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 Noisy Channels
Consider the normal situation of a communication channel that makes errors. Frames may be
either damaged or lost completely.

1. Stop and Wait Automatic Repeat Request

 In a noisy communication channel, if a frame is damaged in transit, the receiver
hardware will detect this when it computes the checksum.
 If a damaged frame is received, it will be discarded and transmitter will retransmit the
same frame after receiving a proper acknowledgement.
 If the acknowledgement frame gets lost and data link layer on 'A' eventually times out.
Not having received an ACK, it assumes that its data frame was lost or damaged and sends the
frame containing packet 1 again. This duplicate frame also arrives at data link layer on 'B', thus
part of file will be duplicated and protocol is said to be failed.
 A typical approach to solve this problem is the provision of a sequence number in the
header of the message.
 The receiver can then check the sequence number determine if the message is a duplicate
since only message is transmitted at any time.
 The sending and receiving station needs only 1-bit alternating sequence of '0' or '1'
to maintain the relationship of the transmitted message and its ACK/ NAK.
 A modulo-2 numbering scheme is used where the frames are alternatively label with '0'
or '1' and positive acknowledgements are of the form ACK 0 and ACK 1.
2. Sequence numbers
 The protocol specifies that frames need to be numbered. This is done by using sequence
number. A field is added to the data frame to hold the sequence number of that frame.
 The sequence numbers are based on modulo-2 arithmetic.
 Stop-and-wait ARQ is the simplest mechanism for error and flow control.

HDLC
High-level Data Link Control (HDLC) is a bit-oriented protocol for communication over
point- to-point and multipoint links.

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 Configurations and Transfer Modes
HDLC provides two common transfer modes that can be used in different configurations:
normal response mode (NRM) and asynchronous balanced mode (ABM).
Normal Response Mode
In normal response mode (NRM), the station configuration is unbalanced. We have one
primary station and multiple secondary stations. A primary station can send commands; a
secondary station can only respond. The NRM is used for both point-to-point and
multiple-point links, as shown in Figure 11.25.
Asynchronous Balanced Mode
In asynchronous balanced mode (ABM), the configuration is balanced. The link is point-
to-point, and each station can function as a primary and a secondary (acting as peers), as
shown in Figure
11.26. This is the common mode today.

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 Frames
To provide the flexibility necessary to support all the options possible in the modes and
configurations just described, HDLC defines three types of frames: information frames (I-
frames), supervisory frames (S-frames), and unnumbered frames (V-frames). Each type of
frame serves as an envelope for the transmission of a different type of message. I-frames
are used to transport user data and control information relating to user data
(piggybacking). S-frames are used only to transport control information. V-frames are
reserved for system management. Information carried by V-frames is intended for
managing the link itself.
Frame Format
Each frame in HDLC may contain up to six fields, as shown in Figure 11.27: a beginning
flag field, an address field, a control field, an information field, a frame check sequence
(FCS) field,and an ending flag field. In multiple-frame transmissions, the ending flag of
one frame can serve as the beginning flag of the next frame.

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 Fields
Let us now discuss the fields and their use in different frame types.
o Flag field. The flag field of an HDLC frame is an 8-bit sequence with the bit pattern
01111110 that identifies both the beginning and the end of a frame and serves as a
synchronization pattern for the receiver.
o Address field. The second field of an HDLC frame contains the address of the
secondary station. If a primary station created the frame, it contains a to address. If a
secondary creates the frame, it contains a from address. An address field can be 1 byte or
several bytes long, depending on the needs of the network. One byte can identify up to 128
stations (l bit is used for another purpose). Larger networks require multiple-byte address
fields. If the address field is only 1 byte, the last bit is always a 1. If the address is more
than 1 byte, all bytes but the last one will end with 0; only the last will end with 1. Ending
each intermediate byte with 0 indicates to the receiver that there are more address bytes to
come.
o Control field. The control field is a 1- or 2-byte segment of the frame used for flow and
error control. The interpretation of bits in this field depends on the frame type.
o Information field. The information field contains the user's data from the network layer
or management information. Its length can vary from one network to another.
o FCS field. The frame check sequence (FCS) is the HDLC error detection field.

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 Control Field
The control field determines the type of frame and defines its functionality. The format is
specific for the type of frame, as shown in Figure 11.28.

Control Field for I-Frames

I-frames are designed to carry user data from the network layer. In addition, they can
include flow and error control information (piggybacking). The subfields in the control
field are used to the control field is 0, this means the frame is an I-frame. The next 3 bits,
called N(S), define the sequence number of the frame. Note that with 3 bits, we can define
a sequence number between 0 and 7; but in the extension format, in which the control field
is 2 bytes, this field is larger. The last 3 bits, called N(R), correspond to the
acknowledgment number when piggybacking is used. The single bit between N(S) and
N(R) is called the PIF bit. The PIP field is a single bit with a dual purpose. It has meaning
only when it is set (bit = 1) and can mean poll or final. It means poll when the frame is
sent by a primary station to a secondary (when the address field contains the address of the
receiver). It means final when the frame is sent by a secondary to a primary.
Control Field for S-Frames
Supervisory frames are used for flow and error control whenever piggybacking is either
impossible or inappropriate. S-frames do not have information fields. If the first 2 bits of
the control field is 10, this means the frame is an S-frame. The last 3 bits, called N(R),
corresponds to the acknowledgment number (ACK) or negative acknowledgment number
(NAK) depending on the type of S-frame. The 2 bits called code is used to define the type

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 of S-frame itself. With 2 bits, we can have four types of S-frames, as described below:
o Receive ready (RR). If the value of the code subfield is 00, it is an RR S-frame. This
kind of frame acknowledges the receipt of a safe and sound frame or group of frames. In
this case, the value N(R) field defines the acknowledgment number.
o Receive not ready (RNR). If the value of the code subfield is 10, it is an RNR S-frame.
This kind of frame is an RR frame with additional functions. It acknowledges the receipt
of a frame or group of frames, and it announces that the receiver is busy and cannot
receive more frames. It acts as a kind of congestion control mechanism by asking the
sender to slow down. The value of NCR is the acknowledgment number.
o Reject (REJ). If the value of the code subfield is 01, it is a REJ S-frame. This is a NAK
frame, but not like the one used for Selective Repeat ARQ. It is a NAK that can be used in
Go-Back-N ARQ to improve the efficiency of the process by informing the sender, before
the sender time expires, that the last frame is lost or damaged. The value of NCR) is the
negative acknowledgment number.
o Selective reject (SREJ). If the value of the code subfield is 11, it is an SREJ S-frame.
This is a NAK frame used in Selective Repeat ARQ. Note that the HDLC Protocol uses
the term selective reject instead of selective repeat. The value of N(R) is the negative
acknowledgment number.
Control Field for V-Frames
Unnumbered frames are used to exchange session management and control information
between connected devices. Unlike S-frames, U-frames contain an information field, but
one used for system management information, not user data. As with S-frames, however,
much of the information carried by U-frames is contained in codes included in the control
field. U-frame codes are divided into two sections: a 2-bit prefix before the PtF bit and a 3-
bit suffix after the PtF bit. Together, these two segments (5 bits) can be used to create up
to 32 different types of U- frames.

Point-to-point protocol
A point-to-point connection provides a dedicated link between two devices. The entire
capacity of the link is reserved for transmission between those two devices. Most point-to-
point connections use an actual length of wire or cable to connect the two ends, but other

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 options, such as microwave or satellite links, are also possible. When you change
television channels by infrared remote control, you are establishing a point-to-point
connection between the remote control and the television's control system.

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 MEDIUM ACCESS PROTOCOL
ALOHA:
In the 1970s, Norman Abramson and his colleagues at the University of Hawaii devised a new
and elegant method to solve the channel allocation problem. Their work has been extended by
many researchers since then (Abramson, 1985).
Although Abramson's work, called the ALOHA system, used ground-based radio broadcasting,
the basic idea is applicable to any system in which uncoordinated users are competing for the use
of a single shared channel. There are two versions of ALOHA: pure and slotted. They differ with
respect to whether time is divided into discrete slots into which all frames must fit. Pure ALOHA
does not require global time synchronization; slotted ALOHA does.
Pure ALOHA:
The basic idea of an ALOHA system is simple: let users transmit whenever they have data to be
sent. There will be collisions, of course, and the colliding frames will be damaged. However, due
to the feedback property of broadcasting, a sender can always find out whether its frame was
destroyed by listening to the channel, the same way other users do. With a LAN, the feedback is
immediate; with a satellite, there is a delay of 270 msec before the sender knows if the
transmission was successful. If listening while transmitting is not possible for some reason,
acknowledgements are needed. If the frame was destroyed, the sender just waits a random
amount of time and sends it again. The waiting time must be random or the same frames will
collide over and over, in lockstep. Systems in which multiple users share a common channel in a
way that can lead to conflicts are widely known as contention systems.

Fig.1 In pure ALOHA, frames are transmitted at completely arbitrary times.

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 A sketch of frame generation in an ALOHA system is given in Fig.1. We have made the frames
all the same length because the throughput of ALOHA systems is maximized by having a
uniform frame size rather than by allowing variable length frames.
Whenever two frames try to occupy the channel at the same time, there will be a collision and
both will be garbled. If the first bit of a new frame overlaps with just the last bit of a frame
almost finished, both frames will be totally destroyed and both will have to be retransmitted
later. The checksum cannot (and should not) distinguish between a total loss and a near miss.
Let the ''frame time'' denote the amount of time needed to transmit the standard, fixed length
frame (i.e., the frame length divided by the bit rate). At this point we assume that the infinite
population of users generates new frames according to a Poisson distribution with mean N
frames per frame time. (The infinite-population assumption is needed to ensure that N does no
decrease as users become blocked.) If N > 1, the user community is generating frames at a higher
rate than the channel can handle, and nearly every frame will suffer a collision.
For reasonable throughput we would expect 0 < N < 1. In addition to the new frames, the
stations also generate retransmissions of frames that previously suffered collisions. Let us further
assume that the probability of k transmission attempts per frame time, old and new combined, is
also Poisson, with mean G per frame time. Clearly, G≥N. At low load (i.e., N 0), there will be
few collisions, hence few retransmissions, so G N. At high load there will be many collisions, so
G > N. Under all loads, the throughput, S, is just the offered load, G, times the probability, P0, of
a transmission succeeding—that is, S = GP0, where P0 is the probability that a frame does not
suffer a collision.
A frame will not suffer a collision if no other frames are sent within one frame time of its start,
as shown in Fig.2.

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 Under what conditions will the shaded frame arrive undamaged? Let t be the time required to
send a frame. If any other user has generated a frame between time t0 and t0+tr, the end of that
frame will collide with the beginning of the shaded one. In fact, the shaded frame's fate was
already sealed even before the first bit was sent, but since in pure ALOHA a station does not
listen to the channel before transmitting, it has no way of knowing that another frame was
already underway. Similarly, any other frame started between t0+t and t0+2t will bump into the
end of the shaded frame.
The probability that k frames are generated during a given frame time is given by the Poisson
distribution:

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 so the probability of zero frames is just e-G. In an interval two frame times long, the mean
number of frames generated is 2G. The probability of no other traffic being initiated during the
entire vulnerable period is thus given by P0 = e -2G. Using S = GP0, we get

The relation between the offered traffic and the throughput is shown in Fig. 4-3. The maximum
throughput occurs at G = 0.5, with S = 1/2e, which is about 0.184. In other words, the best we
can hope for is a channel utilization of 18 per cent. This result is not very encouraging, but with
everyone transmitting at will, we could hardly have expected a 100 per cent success rate.
Slotted ALOHA:
In 1972, Roberts published a method for doubling the capacity of an ALOHA system (Robert,
1972). His proposal was to divide time into discrete intervals, each interval corresponding to one
frame. This approach requires the users to agree on slot boundaries. One way to achieve
synchronization would be to have one special station emit a pip at the start of each interval, like a
clock.
In Roberts' method, which has come to be known as slotted ALOHA, in contrast to Abramson's
pure ALOHA, a computer is not permitted to send whenever a carriage return is typed. Instead, it
is required to wait for the beginning of the next slot. Thus, the continuous pure ALOHA is turned
into a discrete one. Since the vulnerable period is now halved, the probability of no other traffic
during the same slot as our test frame is e-G which leads to

As you can see from Fig.3, slotted ALOHA peaks at G = 1, with a throughput of S=1/e or about
0.368, twice that of pure ALOHA. If the system is operating at G = 1, the probability of an
empty slot is 0.368. The best we can hope for using slotted ALOHA is 37 percent of the slots
empty, 37 percent successes, and 26 percent collisions. Operating at higher values of G reduces
the number of empties but increases the number of collisions exponentially.
To see how this rapid growth of collisions with G comes about, consider the transmission of a
test frame. The probability that it will avoid a collision is e-G, the probability that all the other

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 users are silent in that slot. The probability of a collision is then just 1 – e-G. The probability of a
transmission requiring exactly k attempts, (i.e., k – 1 collisions followed by one success) is

CDMA
Code-division multiple access (CDMA) was conceived several decades ago. Recent
advances in electronic technology have finally made its implementation possible. CDMA differs
from FDMA in that only one channel occupies the entire bandwidth of the link. It differs from
TDMA in that all stations can send data simultaneously; there is no timesharing.
Let us first give an analogy. CDMA simply means communication with different codes.
For example, in a large room with many people, two people can talk privately in English if
nobody else understands English. Another two people can talk in Chinese if they are the only
ones who understand Chinese, and so on. In other words, the common channel, the space of the
room in this case, can easily allow communication between several couples, but in different
languages (codes).

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 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 Cj, 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 O.
2. If we multiply each code by itself, we get 4 (the number of stations).
With these two properties in mind, let us see how the above four stations can send data using the
same common channel, as shown in Figure.
Station 1 multiplies (a special kind of multiplication, as we will see) its data by its code to get
d,. Cj. Station 2 multiplies its data by its code to get d2 . <z- and so on. The data that go on the
channel are the sum of all these terms, as shown in the box. Any station that wants to receive
data from one of the other three multiplies the data on the channel by the code of the sender. For
example, suppose stations 1 and 2 are talking to each other. Station 2 wants to hear what station
1 is saying. It multiplies the data on the channel by C r- the code of station 1.
Because (cI . cI) is 4, but (c2 . cl), (c3 . cl)' and (c4 . cI) are all Os, station 2 divides the result by
4 to get the data from station 1.
Chips CDMA is based on coding theory. Each station is assigned a code, which is a sequence of
numbers called chips, as shown in below figure. The codes are for the previous example.
Later in this chapter we show how we chose these sequences. For now, we need to know that we
did not choose the sequences randomly; they were carefully selected. They are called orthogonal
sequences and have the following properties:

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 1. Each sequence is made of N elements, where N is the number of stations.
2. If we multiply a sequence by a number, every element in the sequence is multiplied by that
element. This is called multiplication of a sequence by a scalar. For example,
2. [+1 +1 -1 -1] = [+2 +2 -2 -2]
3. If we multiply two equal sequences, element by element, and add the results, we get N, where
N is the number of elements in each sequence. This is called the inner product of two equal
sequences. For example, [+1 +1-1-1]. [+1 +1-1-1] = 1 + 1 + 1 + 1 =4
4. If we multiply two different sequences, element by element, and add the results, we get O.
This is called the inner product of two different sequences. For
example, [+ 1 +1 -1 -1] • [+1 +1 + 1 + 1] = 1 + 1 - 1 - 1 = 0
5. Adding two sequences means adding the corresponding elements. The result is another
sequence. For example,
[+1+1-1-1) + [+1+1+1+1)= [+2+2 0 0]

CSMA (Carrier Sense Multiple Access Protocols):

With slotted ALOHA the best channel utilization that can be achieved is 1/e. This is
hardly surprising, since with stations transmitting at will, without paying attention to what the
other stations are doing, there are bound to be many collisions. In local area networks, however,
it is possible for stations to detect what other stations are doing, and adapt their behaviour
accordingly. These networks can achieve a much better utilization than 1/e. In this section we
will discuss some protocols for improving performance. Protocols in which stations listen for a
carrier (i.e., a transmission) and act accordingly are called carrier sense protocols. A number of
them have been proposed. Kleinrock and Tobagi (1975) have analysed several such protocols in
detail. Below we will mention several versions of the carrier sense protocols.
1.1-persistent CSMA:
The first carrier sense protocol that we will study here is called 1-persistent CSMA (Carrier
Sense Multiple Access). When a station has data to send, it first listens to the channel to see if
anyone else is transmitting at that moment. If the channel is busy, the station waits until it
becomes idle. When the station detects an idle channel, it transmits a frame. If a collision occurs,
the station waits a random amount of time and starts all over again. The protocol is called 1-
persistent because the station transmits with a probability of 1 when it finds the channel idle.

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 The propagation delay has an important effect on the performance of the protocol. There
is a small chance that just after a station begins sending, another station will become ready to
send and sense the channel. If the first station's signal has not yet reached the second one, the
latter will sense an idle channel and will also begin sending, resulting in a collision. The longer
the propagation delay, the more important this effect becomes, and the worse the performance of
the protocol. Even if the propagation delay is zero, there will still be collisions. If two stations
become ready in the middle of a third station's transmission, both will wait politely until
thetransmission ends and then both will begin transmitting exactly simultaneously, resulting in a
collision. If they were not so impatient, there would be fewer collisions. Even so, this protocol is
far better than pure ALOHA because both stations have the decency to desist from interfering
with the third station's frame. Intuitively, this approach will lead to a higher performance than
pure ALOHA. Exactly the same holds for slotted ALOHA.
2. Non-persistent CSMA:
A second carrier sense protocol is nonpersistent CSMA. In this protocol, a conscious attempt is
made to be less greedy than in the previous one. Before sending, a station senses the channel. If
no one else is sending, the station begins doing so itself. However, if the channel is already in
use, the station does not continually sense it for the purpose of seizing it immediately upon
detecting the end of the previous transmission. Instead, it waits a random period of time and then
repeats the algorithm. Consequently, this algorithm leads to better channel utilization but longer
delays than 1-persistent CSMA.
3. P-persistent CSMA:
The last protocol is p-persistent CSMA. It applies to slotted channels and works as
follows. When a station becomes ready to send, it senses the channel. If it is idle, it transmits
with a probability p. With a probability q = 1 - p, it defers until the next slot. If that slot is also
idle, It either transmits or defers again, with probabilities p and q. This process is repeated until
either the frame has been transmitted or another station has begun transmitting. In the latter case,
the unlucky station acts as if there had been a collision (i.e., it waits a random time and starts
again). If the station initially senses the channel busy, it waits until the next slot and applies the
above algorithm. Figure 4 shows the computed throughput versus offered traffic for all three
protocols, as well as for pure and slotted ALOHA.

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 CSMA/CD (CSMA with Collision Detection)
Persistent and nonpersistent CSMA protocols are clearly an improvement over ALOHA
because they ensure that no station begins to transmit when it senses the channel busy. Another
improvement is for stations to abort their transmissions as soon as they detect a collision. In
other words, if two stations sense the channel to be idle and begin transmitting simultaneously,
they will both detect the collision almost immediately. Rather than finish transmitting their
frames, which are irretrievably garbled anyway, they should abruptly stop transmitting as soon as
the collision is detected. Quickly terminating damaged frames saves time and bandwidth.
This protocol, known as CSMA/CD (CSMA with Collision Detection) is widely used on
LANs in the MAC sublayer. In particular, it is the basis of the popular Ethernet LAN, so it is
worth devoting some time to looking at it in detail. CSMA/CD, as well as many other LAN
protocols, uses the conceptual model of Fig.5. At the point marked t0, a station has finished
transmitting its frame. Any other station having a frame to send may now attempt to do so. If two
or more stations decide to transmit simultaneously, there will be a collision. Collisions can be
detected by looking at the power or pulse width of the received signal and comparing it to the
transmitted signal.

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 After a station detects a collision, it aborts its transmission, waits a random period of time, and
then tries again, assuming that no other station has started transmitting in the meantime.
Therefore, our model for CSMA/CD will consist of alternating contention and transmission
periods, with idle periods occurring when all stations are quiet (e.g., for lack of work). Now let
us look closely at the details of the contention algorithm. Suppose that two stations both begin
transmitting at exactly time t0. How long will it take them to realize that there has been a
collision? The answer to this question is vital to determining the length of the contention period
and hence what the delay and throughput will be. The minimum time to detect the collision is
then just the time it takes the signal to propagate from one station to the other.
Based on this reasoning, you might think that a station not hearing a collision for a time equal to
the full cable propagation time after starting its transmission could be sure it had seized the
cable. By ''seized,'' we mean that all other stations knew it was transmitting and would not
interfere. This conclusion is wrong. Consider the following worst-case scenario. Let the time for
a signal to propagate between the two farthest stations be . At t0, one station begins transmitting.
At , an instant before the signal arrives at the most distant station, that station also begins
transmitting. Of course, it detects the collision almost instantly and stops, but the little noise
burst caused by the collision does not get back to the original station until time . In other words,
in the worst case a station cannot be sure that it has seized the channel until it has transmitted for
without hearing a collision. For this reason we will model the contention interval as a slotted
ALOHA system with slot width . On a 1-km long coaxial cable, . For simplicity we will assume
that each slot contains just 1 bit. Once the channel has been seized, a station can transmit at any
rate it wants to, of course, not just at 1 bit per sec.

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 Wired LANs: IEEE STANDARDS
In 1985, the Computer Society of the IEEE started a project, called Project 802, to set
standards to enable intercommunication among equipment from a variety of manufacturers.

Project 802 does not seek to replace any part of the OSI or the Internet model. Instead, it is a way
of specifying functions of the physical layer and the data link layer of major LAN protocols.
The standard was adopted by the American National Standards Institute (ANSI). In 1987, the
International Organization for Standardization (ISO) also approved it as an international standard
under the designation ISO 8802. The relationship of the 802 Standard to the traditional OSI
model is shown in Figure 1.
The IEEE has subdivided the data link layer into two sublayers:
logical link control (LLC) and media access control (MAC).
IEEE has also created several physical layer standards for different LAN protocols.

Data Link Layer

As we mentioned before, the data link layer in the IEEE standard is divided into two sublayers:
LLC and MAC. Logical Link Control (LLC)
We said that data link control handles framing, flow control, and error control. In IEEE Project
802, flow control, error control,and part of the framing duties are collected into one sublayer
called the logical link control. Framing is handled in both the LLC sublayer and the MAC
sublayer.
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 The LLC provides one single data link control protocol for all IEEE LANs. In this way, the LLC
is different from the media access control sublayer, which provides different protocols for
different LANs. A single LLC protocol can provide interconnectivity between different LANs

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 because it makes the MAC sublayer transparent. Figure 1 shows one single LLC protocol serving
several MAC protocols. Framing LLC defines a protocol data unit (PDU) that is somewhat
similar to that of HDLC. The header contains a control field like the one in HDLC; this field is
used for flow and error control. The two other header fields define the upper-layer protocol at the
source and destination that uses LLC. These fields are called the destination service access point
(DSAP) and the source service access point (SSAP). The other fields defined in a typical data
link control protocol such as HDLC are moved to the MAC sublayer. In other words, a frame
defined in HDLC is divided into a PDU at the LLC sublayer and a frame at the MAC sublayer,
as shown in Figure 2.
Need for LLC The purpose of the LLC is to provide flow and error control for the upper-layer
protocols that actually demand these services. For example, if a LAN or several LANs are used
in an isolated system, LLC may be needed to provide flow and error control for the application
layer protocols. However, most upper-layer protocols such as IP, do not use the services of LLC.

Media Access Control (MAC)

IEEE Project 802 has created a sublayer called media access control that defines the specific
access method for each LAN. For example, it defines CSMA/CD as the media access method for
Ethernet LANs and the tokenpassing method for Token Ring and Token Bus LANs. As we
discussed in the previous section, part of the framing function is also handled by the MAC layer.
In contrast to the LLC sublayer, the MAC sublayer contains a number of distinct modules; each
defines the access method and the framing format specific to the corresponding LAN protocol.

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 Physical Layer
The physical layer is dependent on the implementation and type of physical media used. IEEE
defines detailed specifications for each LAN implementation. For example, although there is
only one MAC sublayer for Standard Ethernet, there is a different physical layer specifications
for each Ethernet implementations.

STANDARD ETHERNET

MAC Sublayer
In Standard Ethernet, the MAC sublayer governs the operation of the access method. It also
frames data received from the upper layer and passes them to the physical layer.
Frame Format
The Ethernet frame contains seven fields: preamble, SFD, DA, SA, length or type of protocol
data unit (PDU), upper-layer data, and the CRe. Ethernet does not provide any mechanism for
acknowledging received frames, making it what is known as an unreliable medium.
Acknowledgments must be implemented at the higher layers. The format ofthe MAC frame is
shown in Figure 4.

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 D Preamble. The first field of the 802.3 frame contains 7 bytes (56 bits) of alternating Os and Is
that alerts the receiving system to the coming frame and enables it to synchronize its input
timing. The pattern provides only an alert and a timing pulse. The 56-bit pattern allows the
stations to miss some bits at the beginning of the frame. The preamble is actually added at the
physical layer and is not (formally) part of the frame.
D Start frame delimiter (SFD). The second field (l byte: 10101011) signals the beginning of the
frame. The SFD warns the station or stations that this is the last chance for synchronization. The
last 2 bits is 11 and alerts the receiver that the next field is the destination address.
Destination address (DA). The DA field is 6 bytes and contains the physical address of the
destination station or stations to receive the packet.
Source address (SA). The SA field is also 6 bytes and contains the physical address of the sender
of the packet. We will discuss addressing shortly.
Length or type. This field is defined as a type field or length field. The original Ethernet used this
field as the type field to define the upper-layer protocol using the MAC frame. The IEEE
standard used it as the length field to define the number of bytes in the data field. Both uses are
common today.
Data. This field carries data encapsulated from the upper-layer protocols. It is a minimum of 46
and a maximum of 1500 bytes.
CRC. The last field contains error detection information, in this case a CRC-32
Frame Length
Ethernet has imposed restrictions on both the minimum and maximum lengths of a frame,

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 as shown in Figure 5.

The minimum length restriction is required for the correct operation of CSMAlCD. An Ethernet
frame needs to have a minimum length of 512 bits or 64 bytes. Part of this length is the header
and the trailer. If we count 18 bytes of header and trailer (6 bytes of source address, 6 bytes of
destination address, 2 bytes of length or type, and 4 bytes of CRC), then the minimum length of
data from the upper layer is 64 - 18 = 46 bytes. If the upper-layer packet is less than 46 bytes,
padding is added to make up the difference.
The standard defines the maximum length of a frame (without preamble and SFD field) as 1518
bytes. If we subtract the 18 bytes of header and trailer, the maximum length of the payload is
1500 bytes. The maximum length restriction has two historical reasons. First, memory was very
expensive when Ethernet was designed: a maximum length restriction helped to reduce the size
of the buffer. Second, the maximum length restriction prevents one station from monopolizing
the shared medium, blocking other stations that have data to send.
Frame length:
Minimum: 64 bytes (512 bits) Maximum: 1518 bytes (12,144 bits)
Addressing
Each station on an Ethernet network (such as a PC, workstation, or printer) has its own network
interface card (NIC). The NIC fits inside the station and provides the station with a 6-byte
physical address. As shown in Figure 6, the Ethernet address is 6 bytes (48 bits), nonnally
written in hexadecimal notation, with a colon between the bytes.
06:01 :02:01:2C:4B

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 6 bytes =12 hex digits =48 bits
Figure 6 Example ofan Ethernet address in hexadecimal notation
Unicast, Multicast, and Broadcast Addresses A source address is always a unicast address-the
frame comes from only one station. The destination address, however, can be unicast, multicast,
or broadcast. Figure 7 shows how to distinguish a unicast address from a multicast address. If the
least significant bit of the first byte in a destination address is 0, the address is unicast; otherwise,
it is multicast.

A unicast destination address defines only one recipient; the relationship between the sender and
the receiver is one-to-one. A multicast destination address defines a group of addresses; the
relationship between the sender and the receivers is one-to-many. The broadcast address is a
special case of the multicast address; the recipients are all the stations on the LAN. A broadcast
destination address is forty-eight.
6.2.2 Physical Layer
The Standard Ethernet defines several physical layer implementations; four of the most common,
are shown in Figure 8.
Encoding and Decoding
All standard implementations use digital signaling (baseband) at 10 Mbps. At the sender, data are
converted to a digital signal using the Manchester scheme; at the receiver, the received signal is
interpreted as Manchester and decoded into data. Manchester encoding is self-synchronous,
providing a transition at each bit interval. Figure 9 shows the encoding scheme for Standard
Ethernet.

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 Fast Ethernet
Fast Ethernet was designed to compete with LAN protocols such as FDDI or Fiber Channel (or
Fibre Channel, as it is sometimes spelled). IEEE created Fast Ethernet under the name 802.3u.
Fast Ethernet is backward-compatible with Standard Ethernet, but it can transmit data 10 times
faster at a rate of 100 Mbps.
The goals of Fast Ethernet can besummarized as follows:
1. Upgrade the data rate to 100 Mbps.
2. Make it compatible with Standard Ethernet.
3. Keep the same 48-bit address.
4. Keep the same frame format.
5. Keep the same minimum and maximum frame lengths.

Gigabit Ethernet
The technology is based on fiber optic cable. Multi-mode fiber is able to transmit a gigabit rate to
at least 580 meters and with single-mode runs exceeding 3km.
Wireless LAN IEEE 802.11
Architecture
The standard defines two kinds of services: the basic service set (BSS) and the extended service
set (ESS).
Basic Service Set

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 IEEE 802.11 defines the basic service set (BSS) as the building block of a wireless LAN. A basic
service set is made of stationary or mobile wireless stations and an optional central base station,
known as the access point (AP). Figure 9 shows two sets in this standard.
The BSS without anAP is a stand-alone network and cannot send data to other BSSs. It is called
an ad hoc architecture. In this architecture, stations can form a network without the need of an
AP; they can locate one another and agree to be part of a BSS. A BSS with an AP is sometimes
referred to as an infrastructure network.

Extended Service Set

An extended service set (ESS) is made up of two or more BSSs with APs. In this case, the BSSs
are connected through a distribution system, which is usually a wired LAN. The distribution
system connects the APs in the BSSs. IEEE 802.11 does not restrict the distribution system; it
can be any IEEE LAN such as an Ethernet. Note that the extended service set uses two types of
stations: mobile and stationary. The mobile stations are normal stations inside a BSS. The
stationary stations are AP stations that are part of a wired LAN. Figure 10 shows an ESS.

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NOTES OF DCN PREPARED BY M. BALASUBRAMANIAN AP/BCA SSS COLLEGE ARCOT