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Module 2 CN 18EC71

This document contains lecture notes on data link layer protocols for a computer networks course. It discusses the data link layer services of framing, flow control, error control, and introduces the concepts of nodes and links in a network. It describes how data is encapsulated and decapsulated as it travels from the source to destination through intermediate nodes. The notes also cover media access control protocols for random access, and introduce Ethernet and wireless LAN technologies.

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0% found this document useful (0 votes)
52 views61 pages

Module 2 CN 18EC71

This document contains lecture notes on data link layer protocols for a computer networks course. It discusses the data link layer services of framing, flow control, error control, and introduces the concepts of nodes and links in a network. It describes how data is encapsulated and decapsulated as it travels from the source to destination through intermediate nodes. The notes also cover media access control protocols for random access, and introduce Ethernet and wireless LAN technologies.

Uploaded by

Chethan Bhat B S
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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CANARA

ENGINEERING COLLEGE
Benjanapadavu, Mangalore-574219

NOTES
Computer Networks
For 7TH Semester (CBCS)
Course Code: 18EC71
Prepared By :Rajitha A A

Module- 2
DATA LINK LAYER, MAC,

WIRED and WIRELESS LAN’s

DEPARTMENT OF
ELECTRONICS & COMMUNICATION
ENGINEERING
MODULE 2
SYLLABUS:
Data-link Layer:

Introduction: Nodes and Links, Services, Categories of link, Sublayers, Link Layer addressing: Types
of addresses, ARP. Data Link Control (DLC) services: Framing, Flow and Error Control, Data Link
Layer Protocols: Simple Protocol, Stop and Wait protocol, Piggybacking. (9.1, 9.2(9.2.1, 9.2.2), 11.1,
11.2 of Text)

Media Access Control:

Random Access: ALOHA, CSMA, CSMA/CD, CSMA/ CA.(12.1 of Text)

Wired and Wireless LANs:

Ethernet Protocol, Standard Ethernet. Introduction to wireless LAN: Architectural Comparison,


Characteristics, Access Control. (13.1, 13.2(13.2.1 to 13.2.5), 15.1 of Text) L1,L2, L3

Text Books:

Forouzan, Data Communications and Networking, 5th Edition, McGraw Hill, 2013, ISBN: 1-
25906475-3.
MODULE-2

DATA LINK LAYER, MAC, WIRED AND WIRELESS LAN’s


INTRODUCTION

➢ The Internet is a combination of networks glued together by connecting devices (routers or


switches). If a packet is to travel from a host to another host, it needs to pass through these
networks.
➢ Fig.1 shows communication at the data-link layer. Communication at the data-link layer is
made up of five separate logical connections between the data-link layers in the path.

Fig. 1: Communication at the data-link layer

➢ The data-link layer at Alice’s computer communicates with the data-link layer at router R2.
The data-link layer at router R2 communicates with the data-link layer at router R4, and so
on. Finally, the data-link layer at router R7 communicates with the data-link layer at Bob’s
computer. Only one data-link layer is involved at the source or the destination, but two data-
link layers are involved at each router.
➢ The reason is that Alice’s and Bob’s computers are each connected to a single network, but
each router takes input from one network and sends output to another network. Note that

although switches are also involved in the data-link-layer communication, for simplicity we
have not shown them in the Fig. 1.

NODES AND LINKS

➢ Communication at the data-link layer is node-to-node. A data unit from one point in the
Internet needs to pass through many networks (LANs and WANs) to reach another point.
Theses LANs and WANs are connected by routers.
➢ It is customary to refer to the two end hosts and the routers as nodes and the networks in
between as links. Fig. 2 is a simple representation of links and nodes when the path of the
data unit is only six nodes.

Fig. 2: Nodes and Links

➢ The first node is the source host; the last node is the destination host. The other four nodes
are four routers. The first, the third, and the fifth links represent the three LANs; the second
and the fourth links represent the two WANs.

SERVICES

➢ The data-link layer is located between the physical and the network layers. The datalink layer
provides services to the network layer; it receives services from the physical layer.

Let us discuss services provided by the data-link layer.


➢ The duty scope of the data-link layer is node-to-node. When a packet is travelling in the
Internet, the data-link layer of a node (host or router) is responsible for delivering a datagram
to the next node in the path. For this purpose, the data-link layer of the sending node needs to
encapsulate the datagram received from the network in a frame and the data-link layer of the
receiving node needs to decapsulate the datagram from the frame. In other words, the data-
link layer of the source host needs only to encapsulate, the data-link layer of the destination
host needs to decapsulate, but each intermediate node needs to both encapsulate and decapsulate.
➢ The reason for encapsulation and decapsulation at each intermediate node is that each link
may be using a different protocol with a different frame format. Even if one link and the next
are using the same protocol, encapsulation and decapsulation are needed because the link-
layer addresses are normally different. An analogy may help in this case.
➢ Assume a person needs to travel from her home to her friend’s home in another city. The
traveler can use three transportation tools. She can take a taxi to go to the train station in her
own city, then travel on the train from her own city to the city where her friend lives, and
finally reach her friend’s home using another taxi. Here we have a source node, a destination
node, and two intermediate nodes. The traveler needs to get into the taxi at the source node,
get out of the taxi and get into the train at the first intermediate node (train station in the city
where she lives), get out of the train and get into another taxi at the second intermediate node
(train station in the city where her friend lives), and finally get out of the taxi when she
arrives at her destination. A kind of encapsulation occurs at the source node, encapsulation
and decapsulation occur at the intermediate nodes, and decapsulation occurs at the
destination node. Our traveler is the same, but she uses three transporting tools to reach the
destination. Fig. 3 shows the encapsulation and decapsulation at the data-link layer.
Fig. 3: A communication with only three nodes
➢ For simplicity, we have assumed that we have only one router between the source and
destination. The datagram received by the data-link layer of the source host is encapsulated
in a frame. The frame is logically transported from the source host to the router. The frame is
decapsulated at the data-link layer of the router and encapsulated at another frame. The new
frame is logically transported from the router to the destination host. Note that, although we
have shown only two data-link layers at the router, the router actually has three data-link
layers because it is connected to three physical links.

SERVICES PROVIDED BY DATA LINK LAYER

➢ Framing
• The first service provided by the data-link layer is framing. The data-link layer at each
node needs to encapsulate the datagram (packet received from the network layer) in a
frame before sending it to the next node.
• The node also needs to decapsulate the datagram from the frame received on the logical
channel. Although we have shown only a header for a frame, but frame may have both a
header and a trailer. Different data-link layers have different formats for framing.

➢ Flow Control

• Whenever we have a producer and a consumer, we need to think about flow control. If
the producer produces items that cannot be consumed, accumulation of items occurs.
• The sending data-link layer at the end of a link is a producer of frames; the receiving
data-link layer at the other end of a link is a consumer. If the rate of produced frames is
higher than the rate of consumed frames, frames at the receiving end need to be buffered
while waiting to be consumed (processed).
• We cannot have an unlimited buffer size at the receiving side. We have two choices. The
first choice is to let the receiving data-link layer drop the frames if its buffer is full. The
second choice is to let the receiving data-link layer send feedback to the sending data-link
layer to ask it to stop or slow down.
• Different data-link-layer protocols use different strategies for flow control. Since flow
control also occurs at the transport layer, with a higher degree of importance.

➢ Error Control

• At the sending node, a frame in a data-link layer needs to be changed to bits, transformed
to electromagnetic signals, and transmitted through the transmission media.
• At the receiving node, electromagnetic signals are received, transformed to bits, and put
together to create a frame.
• Since electromagnetic signals are susceptible to error, a frame is susceptible to error. The
error needs first to be detected. After detection, it needs to be either corrected at the
receiver node or discarded and retransmitted by the sending node. And also, error
detection and correction are an issue in every layer (node-to-node or host-to-host).

➢ Congestion Control

• Although a link may be congested with frames, which may result in frame loss, most
data-link-layer protocols do not directly use a congestion control to alleviate congestion,
although some wide-area networks do.
• In general, congestion control is considered an issue in the network layer or the transport
layer because of its end-to-end nature.

TWO CATEGORIES OF LINKS

➢ Although two nodes are physically connected by a transmission medium such as cable or air,
we need to remember that the data-link layer controls how the medium is used.
➢ We can have a data-link layer that uses the whole capacity of the medium; we can also have a
data-link layer that uses only part of the capacity of the link. In other words, we can have a
point-to-point link or a broadcast link.
➢ In a point-to-point link, the link is dedicated to the two devices; in a broadcast link, the link is
shared between several pairs of devices. For example, when two friends use the traditional
home phones to chat, they are using a point-to-point link; when the same two friends use
their cellular phones, they are using a broadcast link (the air is shared among many cell
phone users).

TWO SUBLAYERS

➢ To better understand the functionality of and the services provided by the link layer, we can
divide the data-link layer into two sublayers: data link control (DLC) and media access
control (MAC). This is not unusual because LAN protocols use the same strategy.

➢ The data link control sublayer deals with all issues common to both point-to-point and
broadcast links; the media access control sublayer deals only with issues specific to broadcast
links. In other words, we separate these two types of links at the data-link layer, as shown in
Fig. 4.

Fig. 4: Dividing the data-link layer into two sublayers

LINK-LAYER ADDRESSING

➢ The data-link layer is the link-layer addresses where IP addresses used as the identifiers at
the network layer that define the exact points in the Internet where the source and destination
hosts are connected. However, in a connectionless internetwork such as the Internet we
cannot make a datagram reach its destination using only IP addresses.
➢ The reason is that each datagram in the Internet, from the same source host to the same
destination host, may take a different path. The source and destination IP addresses define the
two ends but cannot define which links the datagram should pass through.
➢ Here, IP addresses in a datagram should not be changed. If the destination IP address in a
datagram change, the packet never reaches its destination; if the source IP address in a
datagram change, the destination host or a router can never communicate with the source if a response
needs to be sent back or an error needs to be reported back to the source. The above discussion shows
that we need another addressing mechanism in a connectionless internetwork: the link-layer addresses
of the two nodes.

➢ A link-layer address is sometimes called a link address, sometimes a physical address, and
sometimes a MAC address. We use these terms interchangeably in this book. Since a link is
controlled at the data-link layer, the addresses need to belong to the data-link layer. When a
datagram passes from the network layer to the data-link layer, the datagram will be
encapsulated in a frame and two data-link addresses are added to the frame header. These
two addresses are changed every time the frame moves from one link to another. Fig. 5
demonstrates the concept in a small internet.

Fig. 5: IP addresses and link-layer addresses in a small internet

➢ In the internet in Fig. 5, we have three links and two routers. We also have shown only two
hosts: Alice (source) and Bob (destination). For each host, we have shown two addresses, the
IP addresses (N) and the link-layer addresses (L). Note that a router has as many pairs of
addresses as the number of links the router is connected to. We have shown three frames, one
in each link. Each frame carries the same datagram with the same source and destination
addresses (N1 and N8), but the link-layer addresses of the frame change from link to link. In
link 1, the link-layer addresses are L1 and L2. In link 2, they are L4 and L5. In link 3, they are L7 and
L8.
➢ Note that the IP addresses and the link-layer addresses are not in the same order. For IP
addresses, the source address comes before the destination address; for link-layer addresses,
the destination address comes before the source. The datagrams and frames are designed in
this way, and we follow the design.

➢ We may raise several questions:


• If the IP address of a router does not appear in any datagram sent from a source to a
destination, why do we need to assign IP addresses to routers? The answer is that in some
protocols a router may act as a sender or receiver of a datagram. For example, in routing
protocols, a router is a sender or a receiver of a message. The communications in these
protocols are between routers.
• Why do we need more than one IP address in a router, one for each interface? The answer
is that an interface is a connection of a router to a link. We will see that an IP address
defines a point in the Internet at which a device is connected. A router with n interfaces is
connected to the Internet at n points. This is the situation of a house at the corner of a
street with two gates; each gate has the address related to the corresponding street.
• How are the source and destination IP addresses in a packet determined? The answer is
that the host should know its own IP address, which becomes the source IP address in the
packet. As we know, the application layer uses the services of DNS to find the
destination address of the packet and passes it to the network layer to be inserted in the
packet.
• How are the source and destination link-layer addresses determined for each link? Again,
each hop (router or host) should know its own link-layer address and the destination link-
layer address is determined by using the Address Resolution Protocol.
• What is the size of link-layer addresses? The answer is that it depends on the protocol
used by the link. Although we have only one IP protocol for the whole Internet, we may
be using different data-link protocols in different links. This means that we can define the
size of the address when we discuss different link-layer protocols.

THREE TYPES OF ADDRESSES

Some link-layer protocols define three types of addresses: unicast, multicast, and broadcast.

➢ Unicast Address
Each host or each interface of a router is assigned a unicast address. Unicasting means one-
to-one communication. A frame with a unicast address destination is destined only for one
entity in the link.

Example:
The unicast link-layer addresses in the most common LAN, Ethernet, are 48 bits (six bytes)
that are presented as 12 hexadecimal digits separated by colons; for example, the following is
a link-layer address of a computer.
A3:34:45:11:92:F1
➢ Multicast Address
Some link-layer protocols define multicast addresses. Multicasting means one-to-many
communication. However, the jurisdiction is local (inside the link).

Example:
The multicast link-layer addresses in the most common LAN, Ethernet, are 48 bits (six bytes)
that are presented as 12 hexadecimal digits separated by colons.

The second digit, however, needs to be an even number in hexadecimal. The following
shows a multicast address:
A2:34:45:11:92:F1

➢ Broadcast Address
Some link-layer protocols define a broadcast address. Broadcasting means one-to-all
communication. A frame with a destination broadcast address is sent to all entities in the link.

Example:
The broadcast link-layer addresses in the most common LAN, Ethernet, are 48 bits, all 1s,
that are presented as 12 hexadecimal digits separated by colons. The following shows a
broadcast address:
FF:FF:FF:FF:FF:FF

ADDRESS RESOLUTION PROTOCOL (ARP)


➢ Anytime a node has an IP datagram to send to another node in a link, it has the IP address of
the receiving node. The source host knows the IP address of the default router. Each router
except the last one in the path gets the IP address of the next router by using its forwarding
table. The last router knows the IP address of the destination host. However, the IP address of
the next node is not helpful in moving a frame through a link; we need the link-layer address
of the next node. This is the time when the Address Resolution Protocol (ARP) becomes
helpful.
➢ The ARP protocol is one of the auxiliary protocols defined in the network layer, as shown in
Figure 9.6. It belongs to the network layer, but we discuss it in this chapter because it maps
an IP address to a logical-link address. ARP accepts an IP address from the IP protocol, maps
the address to the corresponding link-layer address, and passes it to the data-link layer.

Fig. 6: Position of ARP in TCP/IP protocol suite

➢ Anytime a host or a router needs to find the link-layer address of another host or router in its
network, it sends an ARP request packet. The packet includes the link-layer and IP addresses
of the sender and the IP address of the receiver. Because the sender does not know the link-
layer address of the receiver, the query is broadcast over the link using the link-layer
broadcast address (see Fig. 7).
Fig. 7: ARP operation
➢ Every host or router on the network receives and processes the ARP request packet, but only
the intended recipient recognizes its IP address and sends back an ARP response packet. The
response packet contains the recipient’s IP and link-layer addresses. The packet is unicast
directly to the node that sent the request packet.
➢ In Fig. 7a, the system on the left (A) has a packet that needs to be delivered to another system
(B) with IP address N2. System A needs to pass the packet to its data-link layer for the actual
delivery, but it does not know the physical address of the recipient. It uses the services of
ARP by asking the ARP protocol to send a broadcast ARP request packet to ask for the
physical address of a system with an IP address of N2.
➢ This packet is received by every system on the physical network, but only system B will
answer it, as shown in Fig. 7b. System B sends an ARP reply packet that includes its physical

address. Now system A can send all the packets it has for this destination using the physical
address it received.
CACHING
➢ A question that is often asked is this: If system A can broadcast a frame to find the link layer
address of system B, why can’t system A send the datagram for system B using a broadcast
frame? In other words, instead of sending one broadcast frame (ARP request), one unicast
frame (ARP response), and another unicast frame (for sending the datagram), system A can
encapsulate the datagram and send it to the network. System B receives it and keep it; other
systems discard it.
➢ To answer the question, we need to think about the efficiency. It is probable that system A
has more than one datagram to send to system B in a short period of time. For example, if
system B is supposed to receive a long e-mail or a long file, the data do not fit in one
datagram.
➢ Let us assume that there are 20 systems connected to the network (link): system A, system B,
and 18 other systems. We also assume that system A has 10 datagrams to send to system B in
one second.
• Without using ARP, system A needs to send 10 broadcast frames. Each of the 18 other
systems need to receive the frames, decapsulate the frames, remove the datagram and
pass it to their network-layer to find out the datagrams do not belong to them. This means
processing and discarding 180 broadcast frames.
• Using ARP, system A needs to send only one broadcast frame. Each of the 18 other
systems need to receive the frames, decapsulate the frames, remove the ARP message
and pass the message to their ARP protocol to find that the frame must be discarded. This
means processing and discarding only 18 (instead of 180) broadcast frames. After system
B responds with its own data-link address, system A can store the link-layer address in its
cache memory. The rest of the nine frames are only unicast. Since processing broadcast
frames is expensive (time consuming), the first method is preferable.

PACKET FORMAT
➢ Fig. 8 shows the format of an ARP packet. The names of the fields are self-explanatory. The
hardware type field defines the type of the link-layer protocol; Ethernet is given the type 1.
➢ The protocol type field defines the network-layer protocol: IPv4 protocol is (0800)16. The
source hardware and source protocol addresses are variable-length fields defining the link-
layer and network-layer addresses of the sender. The destination hardware address and
destination protocol address fields define the receiver link-layer and network-layer addresses.
➢ An ARP packet is encapsulated directly into a data-link frame. The frame needs to have a
field to show that the payload belongs to the ARP and not to the network-layer datagram.
Fig. 8: ARP packet
Example:
A host with IP address N1 and MAC address L1 has a packet to send to another host with IP
address N2 and physical address L2 (which is unknown to the first host). The two hosts are on
the same network. Fig. 9 shows the ARP request and response messages.

Fig. 9: Example for ARP Packet

DLC SERVICES
The data link control (DLC) deals with procedures for communication between two adjacent
nodes—node-to-node communication—no matter whether the link is dedicated or broadcast.
Data link control functions include framing and flow and error control.

FRAMING
➢ Data transmission in the physical layer means moving bits in the form of a signal from the
source to the destination. The physical layer provides bit synchronization to ensure that the
sender and receiver use the same bit durations and timing.
➢ The data-link layer, on the other hand, needs to pack bits into frames, so that each frame is
distinguishable from another.
➢ Our postal system practices a type of framing. The simple act of inserting a letter into an
envelope separates one piece of information from another; the envelope serves as the
delimiter. In addition, each envelope defines the sender and receiver addresses, which is
necessary since the postal system is a many to many carriers facility.
➢ Framing in the data-link layer separates a message from one source to a destination by
adding a sender address and a destination address. The destination address defines where the
packet is to go; the sender address helps the recipient acknowledge the receipt.
➢ Although the whole message could be packed in one frame, that is not normally done. One
reason is that a frame can be very large, making flow and error control very inefficient.
➢ When a message is carried in one very large frame, even a single-bit error would require the
retransmission of the whole frame. When a message is divided into smaller frames, a single-
bit error affects only that small frame.

Frame Size
➢ Frames can be of fixed or variable size.
➢ In fixed-size framing, there is no need for defining the boundaries of the frames; the size
itself can be used as a delimiter. An example of this type of framing is the ATM WAN,
which uses frames of fixed size called cells.
➢ In variable-size framing, prevalent in local-area networks which need a way to define the end
of one frame and the beginning of the next.

➢ Here, two approaches were used for this purpose: a character-oriented approach and a bit-
oriented approach.
Character-Oriented Framing
➢ In character-oriented (or byte-oriented) framing, data to be carried are 8-bit characters from a
coding system such as ASCII. The header, which normally carries the source and destination
addresses and other control information and the trailer, which carries error detection
redundant bits, are also multiples of 8 bits.
➢ To separate one frame from the next, an 8-bit (1-byte) flag is added at the beginning and the
end of a frame. The flag, composed of protocol-dependent special characters, signals the start
or end of a frame. Figure 10 shows the format of a frame in a character-oriented protocol.

Fig. 10: A frame in a character-oriented protocol

➢ Character-oriented framing was popular when only text was exchanged by the data-link layers.
The flag could be selected to be any character not used for text communication. Now, however,
we send other types of information such as graphs, audio, and video; any character used for the
flag could also be part of the information. If this happens, the receiver, when it encounters this
pattern in the middle of the data, thinks it has reached the end of the frame.
➢ To fix this problem, a byte-stuffing strategy was added to character-oriented framing. In byte
stuffing (or character stuffing), a special byte is added to the data section of the frame when there
is a character with the same pattern as the flag. The data section is stuffed with an extra byte.
This byte is usually called the escape character (ESC) and has a predefined bit pattern.
➢ Whenever the receiver encounters the ESC character, it removes it from the data section and
treats the next character as data, not as a delimiting flag. Fig. 11 shows the situation.

Fig. 11: Byte stuffing and unstuffing

➢ Byte stuffing by the escape character allows the presence of the flag in the data section of the
frame, but it creates another problem. What happens if the text contains one or more escape
characters followed by a byte with the same pattern as the flag? The receiver removes the
escape character, but keeps the next byte, which is incorrectly interpreted as the end of the
frame.
➢ To solve this problem, the escape characters that are part of the text must also be marked by
another escape character. In other words, if the escape character is part of the text, an extra
one is added to show that the second one is part of the text.

➢ Character-oriented protocols present another problem in data communications. The universal


coding systems in use today, such as Unicode, have 16-bit and 32-bit characters that conflict
with 8-bit characters.

Bit-Oriented Framing
➢ In bit-oriented framing, the data section of a frame is a sequence of bits to be interpreted by
the upper layer as text, graphic, audio, video, and so on. However, in addition to headers (and
possible trailers), we still need a delimiter to separate one frame from the other.
➢ Most protocols use a special 8-bit pattern flag, 01111110, as the delimiter to define the
beginning and the end of the frame, as shown in Fig. 12.

Fig. 12: A frame in a bit-oriented protocol

➢ This flag can create the same type of problem we saw in the character-oriented protocols. That is,
if the flag pattern appears in the data, we need to somehow inform the receiver that this is not the
end of the frame. We do this by stuffing 1 single bit (instead of 1 byte) to prevent the pattern
from looking like a flag. The strategy is called bit stuffing.
➢ In bit stuffing, if 0 and five consecutive 1 bits are encountered, an extra 0 is added. This extra
stuffed bit is eventually removed from the data by the receiver. Note that the extra bit is added
after one 0 followed by five 1s regardless of the value of the next bit. This guarantees that the
flag field sequence does not inadvertently appear in the frame.
➢ Fig. 13 shows bit stuffing at the sender and bit removal at the receiver. Note that even if we have
a 0 after five 1s, we still stuff a 0. The 0 will be removed by the receiver. This means that if the
flaglike pattern 01111110 appears in the data, it will change to 011111010 (stuffed) and is not
mistaken for a flag by the receiver. The real flag 01111110 is not stuffed by the sender and is
recognized by the receiver.

Fig. 13: Bit stuffing and unstuffing

FLOW AND ERROR CONTROL


Here we elaborate on these two issues here. One of the responsibilities of the data-link control
sublayer is flow and error control at the data-link layer.

Flow Control
➢ Whenever an entity produces items and another entity consumes them, there should be a
balance between production and consumption rates.
➢ If the items are produced faster than they can be consumed, the consumer can be
overwhelmed and may need to discard some items.
➢ If the items are produced more slowly than they can be consumed, the consumer must wait,
and the system becomes less efficient.
➢ Flow control is related to the first issue. We need to prevent losing the data items at the
consumer site.
➢ In communication at the data-link layer, we are dealing with four entities: network and data-
link layers at the sending node and network and data-link layers at the receiving node.
Although we can have a complex relationship with more than one producer and consumer,
we ignore the relationships between networks and data-link layers and concentrate on the
relationship between two data-link layers, as shown in Fig. 14.
Fig. 14: Flow control at the data-link layer

➢ The Fig. 14 shows that the data-link layer at the sending node tries to push frames toward the
data-link layer at the receiving node. If the receiving node cannot process and deliver the
packet to its network at the same rate that the frames arrive, it becomes overwhelmed with
frames. Flow control in this case can be feedback from the receiving node to the sending
node to stop or slow down pushing frames.
Buffers
➢ Although flow control can be implemented in several ways, one of the solutions is normally
to use two buffers; one at the sending data-link layer and the other at the receiving data-link
layer.
➢ A buffer is a set of memory locations that can hold packets at the sender and receiver. The
flow control communication can occur by sending signals from the consumer to the producer.
➢ When the buffer of the receiving data-link layer is full, it informs the sending data-link layer
to stop pushing frames.

Example
➢ The above discussion requires that the consumers communicate with the producers on two
occasions: when the buffer is full and when there are vacancies. If the two parties use a
buffer with only one slot, the communication can be easier.
➢ Assume that each data-link layer uses one single memory slot to hold a frame. When this
single slot in the receiving data-link layer is empty, it sends a note to the network layer to
send the next frame.
Error Control
➢ Since the underlying technology at the physical layer is not fully reliable, we need to
implement error control at the data-link layer to prevent the receiving node from delivering
corrupted packets to its network layer.
➢ Error control at the data-link layer is normally very simple and implemented using one of the
following two methods. In both methods, a CRC is added to the frame header by the sender
and checked by the receiver.
• In the first method, if the frame is corrupted, it is silently discarded; if it is not corrupted,
the packet is delivered to the network layer. This method is used mostly in wired LANs
such as Ethernet.
• In the second method, if the frame is corrupted, it is silently discarded; if it is not
corrupted, an acknowledgment is sent (for the purpose of both flow and error control) to
the sender.

Combination of Flow and Error Control


➢ Flow and error control can be combined. In a simple situation, the acknowledgment that is
sent for flow control can also be used for error control to tell the sender the packet has
arrived uncorrupted.
➢ The lack of acknowledgment means that there is a problem in the sent frame. A frame that
carries an acknowledgment is normally called an ACK to distinguish it from the data frame.

CONNECTIONLESS AND CONNECTION-ORIENTED PROTOCOL


A DLC protocol can be either connectionless or connection oriented.

Connectionless Protocol
➢ In a connectionless protocol, frames are sent from one node to the next without any
relationship between the frames; each frame is independent.
➢ Here, the term connectionless here does not mean that there is no physical connection
(transmission medium) between the nodes; it means that there is no connection between
frames.
➢ The frames are not numbered and there is no sense of ordering. Most of the data-link
protocols for LANs are connectionless protocols.
Connection-Oriented Protocol
➢ In a connection-oriented protocol, a logical connection should first be established between
the two nodes (setup phase).
➢ After all frames that are somehow related to each other are transmitted (transfer phase), the
logical connection is terminated (teardown phase). In this type of communication, the frames
are numbered and sent in order.
➢ If they are not received in order, the receiver needs to wait until all frames belonging to the
same set are received and then deliver them in order to the network layer.
➢ Connection oriented protocols are rare in wired LANs, but we can see them in some point-to-
point protocols, some wireless LANs, and some WANs.

DATA-LINK LAYER PROTOCOLS

➢ Traditionally four protocols have been defined for the data-link layer to deal with flow and
error control: Simple, Stop-and-Wait, Go-Back-N, and Selective-Repeat. Although the first
two protocols still are used at the data-link layer, the last two have disappeared.
➢ The behavior of a data-link-layer protocol can be better shown as a finite state machine
(FSM). An FSM is thought of as a machine with a finite number of states. The machine is
always in one of the states until an event occurs.
➢ Each event is associated with two reactions: defining the list (possibly empty) of actions to
be performed and determining the next state (which can be the same as the current state).
➢ One of the states must be defined as the initial state, the state in which the machine starts
when it turns on. In Fig. 15, we show an example of a machine using FSM.
➢ We have used rounded-corner rectangles to show states, colored text to show events, and
regular black text to show actions. A horizontal line is used to separate the event from the
actions, although later we replace the horizontal line with a slash. The arrow shows the
movement to the next state.
➢ The Fig. 15 shows a machine with three states. There are only three possible events and three
possible actions. The machine starts in state I. If event 1 occurs, the machine performs
actions 1 and 2 and moves to state II. When the machine is in state II, two events may occur.
If event 1 occurs, the machine performs action 3 and remains in the same state, state II. If
event 3 occurs, the machine performs no action, but move to state I.

Fig. 15: Connectionless and connection-oriented service represented as FSMs

SIMPLE PROTOCOL
Our first protocol is a simple protocol with neither flow nor error control. We assume that
the receiver can immediately handle any frame it receives. In other words, the receiver can
never be overwhelmed with incoming frames. Fig. 16 shows the layout for this protocol.

Fig. 16: Simple protocol

➢ The data-link layer at the sender gets a packet from its network layer, makes a frame out of it,
and sends the frame. The data-link layer at the receiver receives a frame from the link,
extracts the packet from the frame, and delivers the packet to its network layer. The data-link
layers of the sender and receiver provide transmission services for their network layers.

FSMs
➢ The sender site should not send a frame until its network layer has a message to send. The
receiver site cannot deliver a message to its network layer until a frame arrives. We can show
these requirements using two FSMs.
➢ Each FSM has only one state, the ready state. The sending machine remains in the ready state
until a request comes from the process in the network layer. When this event occurs, the
sending machine encapsulates the message in a frame and sends it to the receiving machine.
➢ The receiving machine remains in the ready state until a frame arrives from the sending
machine. When this event occurs, the receiving machine decapsulates the message out of the
frame and delivers it to the process at the network layer. Fig. 17 shows the FSMs for the
simple protocol.

Fig. 17: FSMs for the simple protocol

Fig. 18 shows an example of communication using this protocol. It is very simple. The
sender sends frames one after another without even thinking about the receiver.

Fig. 18: An Example for Flow diagram

STOP-AND-WAIT PROTOCOL
➢ Our second protocol is called the Stop-and-Wait protocol, which uses both flow and error
control. We show a primitive version of this protocol here.
➢ In this protocol, the sender sends one frame at a time and waits for an acknowledgment before
sending the next one. To detect corrupted frames, we need to add a CRC to each data frame.
When a frame arrives at the receiver site, it is checked. If its CRC is incorrect, the frame is
corrupted and silently discarded.
➢ The silence of the receiver is a signal for the sender that a frame was either corrupted or lost.
Every time the sender sends a frame, it starts a timer. If an acknowledgment arrives before the
timer expires, the timer is stopped and the sender sends the next frame (if it has one to send).
➢ If the timer expires, the sender resends the previous frame, assuming that the frame was either
lost or corrupted. This means that the sender needs to keep a copy of the frame until its
acknowledgment arrives. When the corresponding acknowledgment arrives, the sender discards
the copy and sends the next frame if it is ready.
➢ Fig. 19 shows the outline for the Stop-and-Wait protocol. Note that only one frame and one
acknowledgment can be in the channels at any time.

Fig. 19: Stop-and-Wait protocol


FSMs
Fig. 20 shows the FSMs for our primitive Stop-and-Wait protocol. We describe the sender and
receiver states below.
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.

Fig. 20: FSM for the Stop-and-Wait protocol

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.

Example
Fig. 21 shows an example. The first frame is sent and acknowledged. The second frame is sent,
but lost. After time-out, it is resent. The third frame is sent and acknowledged, but the
acknowledgment is lost. The frame is resent. However, there is a problem with this scheme. The
network layer at the receiver site receives two copies of the third packet, which is not right. In
the next section, we will see how we can correct this problem using sequence numbers and
acknowledgment numbers.
Fig. 21: Flow diagram for Example

Sequence and Acknowledgment Numbers


➢ We saw a problem from previous Example that needs to be addressed and corrected.
Duplicate packets, as much as corrupted packets, need to be avoided. As an example, assume
we are ordering some item online. If each packet defines the specification of an item to be
ordered, duplicate packets mean ordering an item more than once.
➢ To correct the problem in Example 11.3, we need to add sequence numbers to the data
frames and acknowledgment numbers to the ACK frames. However, numbering in this case
is very simple. Sequence numbers are 0, 1, 0, 1, 0, 1, . . . ; the acknowledgment numbers can
also be 1, 0, 1, 0, 1, 0, …
➢ In other words, the sequence numbers start with 0, the acknowledgment numbers start with 1.
An acknowledgment number always defines the sequence number of the next frame to
receive.
Example
Fig. 22 shows how adding sequence numbers and acknowledgment numbers can prevent
duplicates. The first frame is sent and acknowledged. The second frame is sent, but lost. After
time-out, it is resent. The third frame is sent and acknowledged, but the acknowledgment is lost.
The frame is resent.

Fig. 22: Flow diagram for Example

FSMs with Sequence and Acknowledgment Numbers


We can change the FSM in Fig. 20 to include the sequence and acknowledgment numbers.

Piggybacking
➢ The two protocols we discussed are designed for unidirectional communication, in which
data is flowing only in one direction although the acknowledgment may travel in the other
direction. Protocols have been designed in the past to allow data to flow in both directions.
➢ However, to make the communication more efficient, the data in one direction is
piggybacked with the acknowledgment in the other direction. In other words, when node A is
sending data to node B, Node A also acknowledges the data received from node B. Because
piggybacking makes communication at the datalink layer more complicated, it is not a
common practice.

MEDIA ACCESS CONTROL (MAC)


➢ When nodes or stations are connected and use a common link called a multipoint or
broadcast link, we need a multiple-access protocol to coordinate access to the link. The
problem of controlling the access to the medium is similar to the rules of speaking in an
assembly.
➢ The procedures guarantee that the right to speak is upheld and ensure that two people do not
speak at the same time, do not interrupt each other, do not monopolize the discussion, and so
on.
➢ Many protocols have been devised to handle access to a shared link. All of these protocols
belong to a sublayer in the data-link layer called media access control (MAC). We categorize
them into three groups, as shown in Fig. 23.

Fig. 23: Taxonomy of multiple-access protocols

RANDOM ACCESS
➢ In random-access or contention methods, no station is superior to another station and none is
assigned control over another. 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. This
decision depends on the state of the medium (idle or busy). In other words, each station can
transmit when it desires on the condition that it follows the predefined procedure, including
testing the state of the medium.
➢ Two features give this method its name. First, there is no scheduled time for a station to
transmit. Transmission is random among the stations. That is why these methods are called
random access. Second, no rules specify which station should send next. Stations compete
with one another to access the medium. That is why these methods are also called contention
methods.
➢ In a random-access method, each station has the right to the medium without being
controlled by any other station. However, if more than one station tries to send, there is an
access conflict—collision—and the frames will be either destroyed or modified. To avoid
access conflict or to resolve it when it happens, each station follows a procedure that answers
the following questions:

• When can the station access the medium?

• What can the station do if the medium is busy?


• How can the station determine the success or failure of the transmission?
• What can the station do if there is an access conflict?
➢ The random-access methods have evolved from a very interesting protocol known as
ALOHA, which used a very simple procedure called multiple access (MA). The method was
improved with the addition of a procedure that forces the station to sense the medium before
transmitting. This was called carrier sense multiple access (CSMA). This method later
evolved into two parallel methods: carrier sense multiple access with collision detection
(CSMA/CD), which tells the station what to do when a collision is detected, and carrier sense
multiple access with collision avoidance (CSMA/CA), which tries to avoid the collision.

ALOHA
➢ ALOHA, the earliest random-access method, was developed at the University of Hawaii in
early 1970. It was designed for a radio (wireless) LAN, but it can be used on any shared
medium.
➢ It is obvious that there are potential collisions in this arrangement. The medium is shared
between the stations. When a station sends data, another station may attempt to do so at the
same time. The data from the two stations collide and become garbled.

Pure ALOHA
➢ The original ALOHA protocol is called pure ALOHA. This is a simple but elegant protocol.
The idea is that each station sends a frame whenever it has a frame to send (multiple access).
However, since there is only one channel to share, there is the possibility of collision
between frames from different stations. Fig. 24 shows an example of frame collisions in pure
ALOHA.

Fig. 24: Frames in a pure ALOHA network

➢ There are four stations (unrealistic assumption) that contend with one another for access to
the shared channel. The Fig. 24 shows that each station sends two frames; there are a total of

eight frames on the shared medium. Some of these frames collide because multiple frames
are in contention for the shared channel. Fig. 24 shows that only two frames survive: one
frame from station 1 and one frame from station 3. We need to mention that even if one bit of
a frame coexists on the channel with one bit from another frame, there is a collision and both
will be destroyed. It is obvious that we need to resend the frames that have been destroyed
during transmission.
➢ The pure ALOHA protocol relies on acknowledgments from the receiver. When a station
sends a frame, it expects the receiver to send an acknowledgment. If the acknowledgment
does not arrive after a time-out period, the station assumes that the frame (or the
acknowledgment) has been destroyed and resends the frame. A collision involves two or
more stations. If all these stations try to resend their frames after the time-out, the frames will
collide again.
➢ Pure ALOHA dictates that when the time-out period passes, each station waits a random
amount of time before resending its frame. The randomness will help avoid more collisions.
We call this time the backoff time TB.
➢ Pure ALOHA has a second method to prevent congesting the channel with retransmitted
frames. After a maximum number of retransmissions attempts Kmax, a station must give up
and try later. Fig. 25 shows the procedure for pure ALOHA based on the above strategy.

Fig. 25: Procedure for pure ALOHA protocol

➢ The time-out period is equal to the maximum possible round-trip propagation delay, which is
twice the amount of time required to send a frame between the two most widely separated
stations (2 × Tp). The backoff time TB is a random value that normally depends on K (the

number of attempted unsuccessful transmissions). The formula for TB depends on the


mplementation. One common formula is the binary exponential backoff.
➢ In this method, for each retransmission, a multiplier R = 0 to 2K − 1 is randomly chosen and
multiplied by Tp (maximum propagation time) or Tfr (the average time required to send out a
frame) to find TB. Note that in this procedure, the range of the random numbers increases
after each collision. The value of Kmax is usually chosen as 15.

Vulnerable time
➢ Let us find the vulnerable time, the length of time in which there is a possibility of collision.
We assume that the stations send fixed-length frames with each frame taking Tfr seconds to
send. Fig. 26 shows the vulnerable time for station B.
Fig. 26: Vulnerable time for pure ALOHA protocol

➢ Station B starts to send a frame at time t. Now imagine station A has started to send its frame
after t − Tfr. This leads to a collision between the frames from station B and station A. On the
other hand, suppose that station C starts to send a frame before time t + Tfr. Here, there is also
a collision between frames from station B and station C. Looking at Fig. 26, we see that the
vulnerable time during which a collision may occur in pure ALOHA is 2 times the frame
transmission time.

Throughput
➢ Let us call G the average number of frames generated by the system during one frame
transmission time. Then it can be proven that the average number of successfully transmitted
frames for pure ALOHA is S = G × e−2G. The maximum throughput Smax is 0.184, for G =
1/2. (We can find it by setting the derivative of S with respect to G to 0).

➢ In other words, if one-half a frame is generated during one frame transmission time (one
frame during two frame transmission times), then 18.4 percent of these frames reach their
destination successfully. We expect G = 1/2 to produce the maximum throughput because
the vulnerable time is 2 times the frame transmission time. Therefore, if a station generates
only one frame in this vulnerable time (and no other stations generate a frame during this
time), the frame will reach its destination successfully.

Slotted ALOHA
➢ Pure ALOHA has a vulnerable time of 2 × Tfr. This is so because there is no rule that defines
when the station can send. A station may send soon after another station has started or just
before another station has finished. Slotted ALOHA was invented to improve the efficiency
of pure ALOHA.
➢ In slotted ALOHA we divide the time into slots of Tfr seconds and force the station to send
only at the beginning of the time slot. Fig. 27 shows an example of frame collisions in slotted
ALOHA.

Fig. 27: Frames in a slotted ALOHA network

➢ Because a station is allowed to send only at the beginning of the synchronized time slot, if a
station misses this moment, it must wait until the beginning of the next time slot. This means
that the station which started at the beginning of this slot has already finished sending its
frame. Of course, there is still the possibility of collision if two stations try to send at the
beginning of the same time slot. However, the vulnerable time is now reduced to one-half,
equal to Tfr. Fig. 28 shows the situation.

Fig. 28: Vulnerable time for slotted ALOHA protocol


Throughput
➢ It can be proven that the average number of successful transmissions for slotted ALOHA is S
= G × e−G. The maximum throughput Smax is 0.368, when G = 1. In other words, if one frame
is generated during one frame transmission time, then 36.8 percent of these frames reach
their destination successfully.
➢ We expect G = 1 to produce maximum throughput because the vulnerable time is equal to the
frame transmission time. Therefore, if a station generates only one frame in this vulnerable
time (and no other station generates a frame during this time), the frame will reach its
destination successfully.

CSMA
➢ To minimize the chance of collision and, therefore, increase the performance, the CSMA
method was developed. The chance of collision can be reduced if a station senses the
medium before trying to use it. Carrier sense multiple access (CSMA) requires that each
station first listen to the medium (or check the state of the medium) before sending. In other
words, CSMA is based on the principle “sense before transmit” or “listen before talk.”
➢ CSMA can reduce the possibility of collision, but it cannot eliminate it. The reason for this is
shown in Fig. 28, a space and time model of a CSMA network. Stations are connected to a
shared channel (usually a dedicated medium).
➢ The possibility of collision still exists because of propagation delay; when a station sends a
frame, it still takes time (although very short) for the first bit to reach every station and for
every station to sense it. In other words, a station may sense the medium and find it idle, only
because the first bit sent by another station has not yet been received.

Fig. 28: Space/time model of a collision in CSMA


➢ At time t1, station B senses the medium and finds it idle, so it sends a frame. At time t2 (t2 >
t1), station C senses the medium and finds it idle because, at this time, the first bits from
station B have not reached station C. Station C also sends a frame. The two signals collide
and both frames are destroyed.

Vulnerable Time
➢ The vulnerable time for CSMA is the propagation time Tp. This is the time needed for a
signal to propagate from one end of the medium to the other. When a station sends a frame
and any other station tries to send a frame during this time, a collision will result.
➢ But if the first bit of the frame reaches the end of the medium, every station will already have
heard the bit and will refrain from sending. Fig. 29 shows the worst case. The leftmost
station, A, sends a frame at time t1, which reaches the rightmost station D, at time t1 + Tp.
The gray area shows the vulnerable area in time and space.

Fig. 29: Vulnerable time in CSMA


Persistence Methods
➢ What should a station do if the channel is busy? What should a station do if the channel is
idle? Three methods have been devised to answer these questions: the 1-persistent method,
the nonpersistent method, and the p-persistent method. Fig. 30 shows the behavior of three
persistence methods when a station finds a channel busy. Fig. 31 shows the flow diagrams
for these methods.
➢ 1-Persistent: The 1-persistent method is simple and straightforward. In this method, after the
station finds the line idle, it sends its frame immediately (with probability 1). This method
has the highest chance of collision because two or more stations may find the line idle and
send their frames immediately. We will see later that Ethernet uses this method.

➢ Nonpersistent: In the nonpersistent method, a station that has a frame to send senses the line.
If the line is idle, it sends immediately. If the line is not idle, it waits a random amount of
time and then senses the line again. The nonpersistent approach reduces the chance of
collision because it is unlikely that two or more stations will wait the same amount of time
and retry to send simultaneously. However, this method reduces the efficiency of the network
because the medium remains idle when there may be stations with frames to send.

Fig. 30: Behavior of three persistence methods

➢ p-Persistent: The p-persistent method is used if the channel has time slots with a slot
duration equal to or greater than the maximum propagation time. The p-persistent approach
combines the advantages of the other two strategies. It reduces the chance of collision and
improves efficiency. In this method, after the station finds the line idle it follows these steps:
• With probability p, the station sends its frame.
• With probability q = 1 − p, the station waits for the beginning of the next time slot and
checks the line again.
o If the line is idle, it goes to step 1.
o If the line is busy, it acts as though a collision has occurred and uses the backoff
procedure.
Fig. 31: Flow diagram for three persistence methods

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.
➢ In this method, a station monitors the medium after it sends a frame to see if the transmission
was successful. If so, the station is finished. If, however, there is a collision, the frame is sent
again.
➢ To better understand CSMA/CD, let us look at the first bits transmitted by the two stations
involved in the collision. Although each station continues to send bits in the frame until it
detects the collision, we show what happens as the first bits collide. In Fig. 32, stations A and
C are involved in the collision.
➢ At time t1, station A has executed its persistence procedure and starts sending the bits of its
frame. At time t2, station C has not yet sensed the first bit sent by A. Station C executes its
persistence procedure and starts sending the bits in its frame, which propagate both to the left
and to the right. The collision occurs sometime after time t2.
Fig. 32: Collision of the first bits in CSMA/CD

➢ Station C detects a collision at time t3 when it receives the first bit of A’s frame. Station C
immediately (or after a short time, but we assume immediately) aborts transmission. Station
A detects collision at time t4 when it receives the first bit of C’s frame; it also immediately
aborts transmission. Looking at

the Fig. 32, we see that A transmits for the duration t4 − t1; C transmits for the duration t3 −
t2.
➢ Now that we know the time durations for the two transmissions, we can show a more
complete graph in Fig. 33.

Fig. 33: Collision and abortion in CSMA/CD

Minimum Frame Size


➢ For CSMA/CD to work, we need a restriction on the frame size. Before sending the last bit of
the frame, the sending station must detect a collision, if any, and abort the transmission. This
is so because the station, once the entire frame is sent, does not keep a copy of the frame and
does not monitor the line for collision detection. Therefore, the frame transmission time Tfr
must be at least two times the maximum propagation time Tp.
➢ To understand the reason, let us think about the worst-case scenario. If the two stations
involved in a collision are the maximum distance apart, the signal from the first takes time
Tp to reach the second, and the effect of the collision takes another time TP to reach the first.
So, the requirement is that the first station must still be transmitting after 2Tp.

Procedure
➢ Now let us look at the flow diagram for CSMA/CD in Fig. 34. It is similar to the one for the
ALOHA protocol, but there are differences. The first difference is the addition of the
persistence process. We need to sense the channel before we start sending the frame by using
one of the persistence processes, we discussed previously (nonpersistent, 1-persistent, or p-
persistent). The corresponding box can be replaced by one of the persistence processes
shown in Fig. 31.

Fig. 34: Flow diagram for the CSMA/CD

➢ The second difference is the frame transmission. In ALOHA, we first transmit the entire
frame and then wait for an acknowledgment. In CSMA/CD, transmission and collision
detection are continuous processes. We do not send the entire frame and then look for a
collision. The station transmits and receives continuously and simultaneously (using two
different ports or a bidirectional port). We use a loop to show that transmission is a
continuous process. We constantly monitor in order to detect one of two conditions: either
transmission is finished or a collision is detected. Either event stops transmission. When we
come out of the loop, if a collision has not been detected, it means that transmission is
complete; the entire frame is transmitted. Otherwise, a collision has occurred.
➢ The third difference is the sending of a short jamming signal to make sure that all other
stations become aware of the collision.

Energy Level
➢ We can say that the level of energy in a channel can have three values: zero, normal, and
abnormal. At the zero level, the channel is idle. At the normal level, a station has successfully
captured the channel and is sending its frame. At the abnormal level, there is a collision and
the level of the energy is twice the normal level. A station that has a frame to send or is
sending a frame needs to monitor the energy level to determine if the channel is idle, busy, or
in collision mode. Fig. 35 shows the situation.

Fig. 35: Energy level during transmission, idleness, or collision

Throughput

The throughput of CSMA/CD is greater than that of pure or slotted ALOHA. The maximum
throughput occurs at a different value of G and is based on the persistence method and the value
of p in the p-persistent approach. For the 1-persistent method, the maximum throughput is
around 50 percent when G = 1. For the nonpersistent method, the maximum throughput can go
up to 90 percent when G is between 3 and 8.
Traditional Ethernet
One of the LAN protocols that used CSMA/CD is the traditional Ethernet with the data rate of 10
Mbps. Now, it is good to know that the traditional Ethernet was a broadcast LAN that used the 1-
persistence method to control access to the common media. Later versions of Ethernet try to
move from CSMA/CD access methods.
CSMA/CA
➢ Carrier sense multiple access with collision avoidance (CSMA/CA) was invented for
wireless networks. Collisions are avoided using CSMA/CA’s three strategies: the interframe
space, the contention window, and acknowledgments, as shown in Fig. 36. We discuss RTS
and CTS frames later.
➢ Interframe Space (IFS): First, collisions are avoided by deferring transmission even if the
channel is found idle. When an idle channel is found, the station does not send immediately.
It waits for a period of time called the interframe space or IFS. Even though the channel may
appear idle when it is sensed, a distant station may have already started transmitting. The
distant station’s signal has not yet reached this station. The IFS time allows the front of the
transmitted signal by the distant station to reach this station. After waiting an IFS time, if the
channel is still idle, the station can send, but it still needs to wait a time equal to the
contention window. The IFS variable can also be used to prioritize stations or frame types.
For example, a station that is assigned a shorter IFS has a higher priority.

Fig. 36: Flow diagram of CSMA/CA


➢ Contention Window: The contention window is an amount of time divided into slots. A
station that is ready to send chooses a random number of slots as its wait time. The number
of slots in the window changes according to the binary exponential backoff strategy. This
means that it is set to one slot the first time and then doubles each time the station cannot
detect an idle channel after the IFS time. This is very similar to the p-persistent method
except that a random outcome defines the number of slots taken by the waiting station. One
interesting point about the contention window is that the station needs to sense the channel
after each time slot. However, if the station finds the channel busy, it does not restart the
process; it just stops the timer and restarts it when the channel is sensed as idle. This gives
priority to the station with the longest waiting time as shown in Fig. 37.

Fig. 37: Contention window

➢ Acknowledgment: With all these precautions, there still may be a collision resulting in
destroyed data. In addition, the data may be corrupted during the transmission. The positive
acknowledgment and the time-out timer can help guarantee that the receiver has received the
frame.

Frame Exchange Time Line


Fig. 38 shows the exchange of data and control frames in time.
1. Before sending a frame, the source station senses the medium by checking the energy level at
the carrier frequency.
a. The channel uses a persistence strategy with backoff until the channel is idle.

b. After the station is found to be idle, the station waits for a period of time called the DCF
interframe space (DIFS); then the station sends a control frame called the request to send
(RTS).
2. After receiving the RTS and waiting a period of time called the short interframe sp.ace
(SIFS), the destination station sends a control frame, called the clear to send (CTS), to the
source station. This control frame indicates that the destination station is ready to receive

data.
Fig. 38: CSMA/CA and NAV

3. The source station sends data after waiting an amount of time equal to SIFS.
4. The destination station, after waiting an amount of time equal to SIFS, sends an
acknowledgment to show that the frame has been received. Acknowledgment is needed in
this protocol because the station does not have any means to check for the successful arrival
of its data at the destination. On the other hand, the lack of collision in CSMA/CD is a kind
of indication to the source that data have arrived.

Network Allocation Vector


➢ How do other stations defer sending their data if one station acquires access? In other words,
how is the collision avoidance aspect of this protocol accomplished? The key is a feature
called NAV.
➢ When a station sends an RTS frame, it includes the duration of time that it needs to occupy
the channel. The stations that are affected by this transmission create a timer called a network
allocation vector (NAV) that shows how much time must pass before these stations are
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allowed to check the channel for idleness. Each time a station accesses the system and sends an RTS
frame, other stations start their NAV. In other words, each station, before sensing the physical medium
to see if it is idle, first checks its NAV to see if it has expired. Fig. 38 shows the idea of NAV.

Collision During Handshaking


➢ What happens if there is a collision during the time when RTS or CTS control frames are in transition,
often called the handshaking period? Two or more stations may try to send RTS frames at the same
time. These control frames may collide.
➢ However, because there is no mechanism for collision detection, the sender assumes there has been a
collision if it has not received a CTS frame from the receiver. The backoff strategy is employed, and
the sender tries again.

Hidden-Station Problem
➢ The solution to the hidden station problem is the use of the handshake frames (RTS and CTS). Fig. 38
also shows that the RTS message from B reaches A, but not C.
➢ However, because both B and C are within the range of A, the CTS message, which contains the
duration of data transmission from B to A, reaches C. Station C knows that some hidden station is
using the channel and refrains from transmitting until that duration is over.

CSMA/CA and Wireless Networks


➢ CSMA/CA was mostly intended for use in wireless networks. The procedure described above,
however, is not sophisticated enough to handle some particular issues related to wireless networks,
such as hidden terminals or exposed terminals.
➢ We will see how these issues are solved by augmenting the above protocol with handshaking features.

WIRED LANs: ETHERNET


After discussing the general issues related to the data-link layer, it is time to discuss the wired LANs.
Although over a few decades many wired LAN protocols existed, only the Ethernet technology survives
today. This is the reason that we discuss only this technology here.

ETHERNET PROTOCOL
➢ We discussed about TCP/IP protocol suite does not define any protocol for the data-link or the physical
layer. In other words, TCP/IP accepts any protocol at these two layers that can provide services to the
network layer. The data-link layer and the physical layer are the territory of the local and wide area

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networks.

➢ This means that when we discuss these two layers, we are talking about networks that are using them.
We learned that a local area network (LAN) is a computer network that is designed for a limited
geographic area such as a building or a campus. Although a LAN can be used as an isolated network
to connect computers in an organization for the sole purpose of sharing resources, most LANs today are
also linked to a wide area network (WAN) or the Internet.
➢ In the 1980s and 1990s several different types of LANs were used. All these LANs used a media-
access method to solve the problem of sharing the media. The Ethernet used the CSMA/CD approach.
The Token Ring, Token Bus, and FDDI (Fiber Distribution Data Interface) used the token-passing
approach. During this period, another LAN technology, ATM LAN, which deployed the high-speed
WAN technology (ATM), appeared in the market.
➢ Almost every LAN except Ethernet has disappeared from the marketplace because Ethernet was able to
update itself to meet the needs of the time. Several reasons for this success have been mentioned in the
literature, but we believe that the Ethernet protocol was designed so that it could evolve with the
demand for higher transmission rates. It is natural that an organization that has used an Ethernet LAN
in the past and now needs a higher data rate would update to the new generation instead of switching to
another technology, which might cost more. This means that we confine our discussion of wired LANs
to the discussion of Ethernet.

IEEE Project 802


➢ Here, we need to briefly discuss the IEEE standard that we often encounter in text or real life. 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 model or TCP/IP protocol suite.
➢ Instead, it is a way of specifying functions of the physical layer and the data-link layer of major LAN
protocols. The relationship of the 802 Standard to the TCP/IP protocol suite is shown in Fig. 39.
➢ 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. Logical Link Control (LLC).
➢ We know 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 (LLC). Framing is handled in both the LLC sublayer and the MAC sublayer.
➢ The LLC provides a single link-layer control protocol for all IEEE LANs. This means LLC protocol

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can provide interconnectivity between different LANs because it makes the MAC sublayer transparent.

Media Access Control (MAC)


➢ Earlier we discussed multiple access methods including random access, controlled access, and
channelization. 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 defines the
token-passing method for Token Ring and Token Bus LANs. As we mentioned in the previous section,
part of the framing function is also handled by the MAC layer.

Fig. 39: IEEE standard for LANs

ETHERNET EVOLUTION
The Ethernet LAN was developed in the 1970s by Robert Metcalfe and David Boggs. Since then, it has
gone through four generations: Standard Ethernet (10 Mbps), Fast Ethernet (100 Mbps), Gigabit Ethernet
(1 Gbps), and 10 Gigabit Ethernet (10 Gbps), as shown in Fig. 40.

Fig. 40: Ethernet evolution through four generations STANDARD

ETHERNET
We refer to the original Ethernet technology with the data rate of 10 Mbps as the Standard Ethernet.
Although most implementations have moved to other technologies in the Ethernet evolution, there are

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some features of the Standard Ethernet that have not changed during the evolution.

CHARACTERISTICS

1. Connectionless and Unreliable Service


➢ Ethernet provides a connectionless service, which means each frame sent is independent of the
previous or next frame. Ethernet has no connection establishment or connection termination phases.

➢ The sender sends a frame whenever it has it; the receiver may or may not be ready for it. The sender
may overwhelm the receiver with frames, which may result in dropping frames.
➢ If a frame drops, the sender will not know about it. Since IP, which is using the service of Ethernet, is
also connectionless, it will not know about it either.
➢ If the transport layer is also a connectionless protocol, such as UDP, the frame is lost and salvation
may only come from the application layer. However, if the transport layer is TCP, the sender TCP does
not receive acknowledgment for its segment and sends it again.
➢ Ethernet is also unreliable like IP and UDP. If a frame is corrupted during transmission and the
receiver finds out about the corruption, which has a high level of probability of happening because of
the CRC-32, the receiver drops the frame silently. It is the duty of high-level protocols to find out
about it.

2. Frame Format
The Ethernet frame contains seven fields, as shown in Fig. 40.

Fig. 40: Ethernet frame

➢ Preamble: This field contains 7 bytes (56 bits) of alternating 0s and 1s that alert the receiving system
to the coming frame and enable it to synchronize its clock if it’s out of synchronization. The pattern
provides only an alert and a timing pulse. The 56-bit pattern allows the stations to miss some bits at the

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beginning of the frame. The preamble is actually added at the physical layer and is not (formally) part
of the frame.
➢ Start frame delimiter (SFD): This field (1 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 are
(11)2 and alert the receiver that the next field is the destination address. This field is actually a flag that
defines the beginning of the frame. We need to remember that an Ethernet frame is a variable-length
frame. It needs a flag to define the beginning of the frame. The SFD field is also added at the physical
layer.
➢ Destination address (DA): This field is six bytes (48 bits) and contains the link layer address of the
destination station or stations to receive the packet. When the receiver sees its own link-layer address,
or a multicast address for a group that the receiver is a member of, or

a broadcast address, it decapsulates the data from the frame and passes the data to the upper layer
protocol defined by the value of the type field.
➢ Source address (SA): This field is also six bytes and contains the link-layer address of the sender of
the packet.
➢ Type: This field defines the upper-layer protocol whose packet is encapsulated in the frame. This
protocol can be IP, ARP, OSPF, and so on. In other words, it serves the same purpose as the protocol
field in a datagram and the port number in a segment or user datagram. It is used for multiplexing and
demultiplexing.
➢ Data: This field carries data encapsulated from the upper-layer protocols. It is a minimum of 46 and a
maximum of 1500 bytes. We discuss the reason for these minimum and maximum values shortly. If the
data coming from the upper layer is more than 1500 bytes, it should be fragmented and encapsulated in
more than one frame. If it is less than 46 bytes, it needs to be padded with extra 0s. A padded data
frame is delivered to the upper-layer protocol as it is (without removing the padding), which means
that it is the responsibility of the upper layer to remove or, in the case of the sender, to add the padding.
The upper-layer protocol needs to know the length of its data. For example, a datagram has a field that
defines the length of the data.
➢ CRC: The last field contains error detection information, in this case a CRC-32. The CRC is calculated
over the addresses, types, and data field. If the receiver calculates the CRC and finds that it is not zero
(corruption in transmission), it discards the frame.

3. Frame Length
➢ Ethernet has imposed restrictions on both the minimum and maximum lengths of a frame. The

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minimum length restriction is required for the correct operation of CSMA/CD.


➢ 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.
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 link-layer address. The Ethernet
address is 6 bytes (48 bits), normally written in hexadecimal notation, with a colon between the bytes. For
example, the following shows an Ethernet MAC address:
4A:30:10:21:10:1A
Transmission of Address Bits
The way the addresses are sent out online is different from the way they are written in hexadecimal
notation. The transmission is left to right, byte by byte; however, for each byte, the least significant bit is
sent first and the most significant bit is sent last. This means that the bit that defines an address as unicast
or multicast arrives first at the receiver. This helps the receiver to immediately known if the packet is
unicast or multicast.

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. Fig. 41 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. Note that with the way the bits are transmitted, the unicast/multicast bit is the first bit
which is transmitted or received.
➢ The broadcast address is a special case of the multicast address: the recipients are all the stations on the

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LAN. A broadcast destination address is forty-eight 1s.

Fig. 41: Unicast and multicast addresses

Distinguish Between Unicast, Multicast, and Broadcast Transmission


➢ Standard Ethernet uses a coaxial cable (bus topology) or a set of twisted-pair cables with a hub (star
topology) as shown in Fig. 42. We need to know that transmission in the standard Ethernet is always
broadcast, no matter if the intention is unicast, multicast, or broadcast. In the bus topology, when
station
➢ A sends a frame to station B, all stations will receive it. In the star topology, when station A sends a
frame to station B, the hub will receive it. Since the hub is a passive element, it does not check the
destination address of the frame; it regenerates the bits (if they have been weakened) and sends them to
all stations except station A. In fact, it floods the network with the frame.

Fig. 42: Implementation of standard Ethernet

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➢ The question is, then, how the actual unicast, multicast, and broadcast transmissions are distinguished
from each other. The answer is in the way the frames are kept or dropped.
• In a unicast transmission, all stations will receive the frame, the intended recipient keeps and
handles the frame; the rest discard it.
• In a multicast transmission, all stations will receive the frame, the stations that are members of the
group keep and handle it; the rest discard it.
• In a broadcast transmission, all stations (except the sender) will receive the frame and all stations
(except the sender) keep and handle it.

Access Method
Since the network that uses the standard Ethernet protocol is a broadcast network, we need to use an access
method to control access to the sharing medium. The standard Ethernet chose CSMA/CD with 1-persistent
method. Let us use a scenario to see how this method works for the Ethernet protocol.
➢ Assume station A in Fig. 42 has a frame to send to station D. Station A first should check whether any
other station is sending (carrier sense). Station A measures the level of energy on the medium (for a
short period of time, normally less than 100 μs). If there is no signal energy on the medium, it means
that no station is sending (or the signal has not reached station A). Station A interprets this situation as
idle medium. It starts sending its frame. On the other hand, if the signal energy level is not zero, it
means that the medium is being used by another station. Station A continuously monitors the medium
until it becomes idle for 100 μs. It then starts sending the frame. However, station A needs to keep a
copy of the frame in its buffer until it is sure that there is no collision. When station A is sure of this is
the subject.
➢ The medium sensing does not stop after station A has started sending the frame Station A needs to send
and receive continuously. Two cases may occur:

Case – 1:
• Station A has sent 512 bits and no collision is sensed (the energy level did not go above the regular
energy level), the station then is sure that the frame will go through and stops sensing the medium.
Where does the number 512 bits come from?
• If we consider the transmission rate of the Ethernet as 10 Mbps, this means that it takes the station
512/(10 Mbps) = 51.2 μs to send out 512 bits. With the speed of propagation in a cable (2 × 108
meters), the first bit could have gone 10,240 meters (one way) or only 5120 meters (round trip), have
collided with a bit from the last station on the cable, and have gone back.

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• In other words, if a collision were to occur, it should occur by the time the sender has sent out 512 bits
(worst case) and the first bit has made a round trip of 5120 meters. We should know that if the collision
happens in the middle of the cable, not at the end, station A hears the collision earlier and aborts the
transmission. We also need to mention another issue.
• The above assumption is that the length of the cable is 5120 meters. The designer of the standard
Ethernet actually put a restriction of 2500 meters because we need to consider the delays encountered
throughout the journey. It means that they considered the worst case. The whole idea is that if station A
does not sense the collision before sending 512 bits, there must have been no collision, because during
this time, the first bit has reached the end of the line and all other stations know that a station is sending
and refrain from sending.
• In other words, the problem occurs when another station (for example, the last station) starts sending
before the first bit of station A has reached it. The other station mistakenly thinks that the line is free
because the first bit has not yet reached it.
• The reader should notice that the restriction of 512 bits actually helps the sending station: The sending
station is certain that no collision will occur if it is not heard during the first 512 bits, so it can discard
the copy of the frame in its buffer.

Case – 2:
• Station A has sensed a collision before sending 512 bits. This means that one of the previous bits has
collided with a bit sent by another station. In this case both stations should refrain from sending and
keep the frame in their buffer for resending when the line becomes available.
• However, to inform other stations that there is a collision in the network, the station sends a 48-bit jam
signal. The jam signal is to create enough signal (even if the collision happens after a few bits) to alert
other stations about the collision.
• After sending the jam signal, the stations need to increment the value of K (number of attempts). If
after increment K = 15, the experience has shown that the network is too busy, the station needs to
abort its effort and try again. If K < 15, the station can wait a backoff time (TB in Fig. 25) and restart
the process.
• As Fig. 25 shows, the station creates a random number between 0 and 2K − 1, which means each time
the collision occurs, the range of the random number increases exponentially. After the first collision
(K = 1) the random number is in the range (0, 1). After the second collision

(K = 2) it is in the range (0, 1, 2, 3). After the third collision (K = 3) it is in the range (0, 1, 2,

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3, 4, 5, 6, 7).
• So, after each collision, the probability increases that the backoff time becomes longer. This is due to
the fact that if the collision happens even after the third or fourth attempt, it means that the network is
really busy; a longer backoff time is needed.

EFFICIENCY OF STANDARD ETHERNET


➢ The efficiency of the Ethernet is defined as the ratio of the time used by a station to send data to the
time the medium is occupied by this station. The practical efficiency of standard Ethernet has been
measured to be in which the parameter “a” is the number of frames that can fit on the medium. It can
be calculated as a = (propagation delay)/(transmission delay) because the transmission delay is the time
it takes a frame of average size to be sent out and the propagation delay is the time it takes to reach the
end of the medium.
➢ Note that as the value of parameter a decrease, the efficiency increases. This means that if the length of
the media is shorter or the frame size longer, the efficiency increases. In the ideal case, a = 0 and the
efficiency is 1.

IMPLEMENTATION
➢ The Standard Ethernet defined several implementations, but only four of them became popular during
the 1980s. Table 1 shows a summary of Standard Ethernet implementations.
➢ In the nomenclature 10BaseX, the number defines the data rate (10 Mbps), the term Base means
baseband (digital) signal, and X approximately defines either the maximum size of the cable in 100
meters (for example 5 for 500 or 2 for 185 meters) or the type of cable, T for unshielded twisted pair
cable (UTP) and F for fiber-optic.
➢ The standard Ethernet uses a baseband signal, which means that the bits are changed to a digital signal
and directly sent on the line.
Table 1: Summary of Standard Ethernet

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Implementations 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. As we know, Manchester encoding is self- synchronous, providing a
transition at each bit interval. Fig. 43 shows the encoding scheme for Standard Ethernet.

Fig. 43: Encoding in a Standard Ethernet implementation

10Base5: Thick Ethernet


➢ The first implementation is called 10Base5, thick Ethernet, or Thick net. The nickname derives from
the size of the cable, which is roughly the size of a garden hose and too stiff to bend with your hands.
10Base5 was the first Ethernet specification to use a bus topology with an external transceiver
(transmitter/receiver) connected via a tap to a thick coaxial cable. Fig.
44 shows a schematic diagram of a 10Base5 implementation.

Fig. 44: 10Base5 implementation

➢ The transceiver is responsible for transmitting, receiving, and detecting collisions. The transceiver is
connected to the station via a transceiver cable that provides separate paths for sending and receiving.
This means that collision can only happen in the coaxial cable.
➢ The maximum length of the coaxial cable must not exceed 500 m, otherwise, there is excessive
degradation of the signal. If a length of more than 500 m is needed, up to five segments, each a
maximum of 500 meters, can be connected using repeaters.

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10Base2: Thin Ethernet


➢ The second implementation is called 10Base2, thin Ethernet, or Cheapernet. 10Base2 also uses a bus
topology, but the cable is much thinner and more flexible. The cable can be bent to pass very close to
the stations. In this case, the transceiver is normally part of the network interface card (NIC), which is
installed inside the station. Fig. 45 shows the schematic diagram of a 10Base2 implementation.

Fig. 45: 10Base2 implementation

➢ Note that the collision here occurs in the thin coaxial cable. This implementation is more cost effective
than 10Base5 because thin coaxial cable is less expensive than thick coaxial and the tee connections
are much cheaper than taps.
➢ Installation is simpler because the thin coaxial cable is very flexible. However, the length of each
segment cannot exceed 185 m (close to 200 m) due to the high level of attenuation in thin coaxial
cable.
10Base-T: Twisted-Pair Ethernet
➢ The third implementation is called 10Base-T or twisted-pair Ethernet. 10Base-T uses a physical star
topology. The stations are connected to a hub via two pairs of twisted cable, as shown in Fig. 46.

Fig. 46: 10Base-T implementation

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➢ Note that two pairs of twisted cable create two paths (one for sending and one for receiving) between
the station and the hub. Any collision here happens in the hub.

➢ Compared to 10Base5 or 10Base2, we can see that the hub actually replaces the coaxial cable as far as
a collision is concerned. The maximum length of the twisted cable here is defined as 100 m, to
minimize the effect of attenuation in the twisted cable.

10Base-F: Fiber Ethernet


➢ Although there are several types of optical fiber 10-Mbps Ethernet, the most common is called 10Base-
F. 10Base-F uses a star topology to connect stations to a hub.
➢ The stations are connected to the hub using two fiber-optic cables, as shown in Fig. 47.

Fig. 47: 10Base-F implementation

WIRELESS LANs
Wireless communication is one of the fastest-growing technologies. The demand for connecting devices
without the use of cables is increasing everywhere. Wireless LANs can be found on college campuses, in
office buildings, and in many public areas.

Architectural Comparison
Let us first compare the architecture of wired and wireless LANs to give some idea of what we need
to look for when we study wireless LANs.

Medium
➢ The first difference we can see between a wired and a wireless LAN is the medium. In a wired LAN,
we use wires to connect hosts. Earlier, we saw that we moved from multiple access to point-to-point
access through the generation of the Ethernet.

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➢ In a switched LAN, with a link-layer switch, the communication between the hosts is point to- point
and full-duplex (bidirectional).
➢ In a wireless LAN, the medium is air, the signal is generally broadcast. When hosts in a wireless LAN
communicate with each other, they are sharing the same medium (multiple access).
➢ In a very rare situation, we may be able to create a point-to-point communication between two wireless
hosts by using a very limited bandwidth and two-directional antennas. Here it is to know about the
multiple-access medium, which means we need to use MAC protocols.

Hosts
➢ In a wired LAN, a host is always connected to its network at a point with a fixed link layer address
related to its network interface card (NIC). Of course, a host can move from one point in the Internet to
another point.
➢ In this case, its link-layer address remains the same, but its network-layer address will change.
However, before the host can use the services of the Internet, it needs to be physically connected to the
Internet.
➢ In a wireless LAN, a host is not physically connected to the network; it can move freely (as we’ll see)
and can use the services provided by the network. Therefore, mobility in a wired network and wireless
network are totally different issues.

Isolated LANs
➢ The concept of a wired isolated LAN also differs from that of a wireless isolated LAN. A wired
isolated LAN is a set of hosts connected via a link-layer switch (in the recent generation of Ethernet).
➢ A wireless isolated LAN, called an ad hoc network in wireless LAN terminology, is a set of hosts that
communicate freely with each other. The concept of a link-layer switch does not exist in wireless
LANs. Fig. 48 shows two isolated LANs, one wired and one wireless.

Fig. 48: Isolated LANs: wired versus wireless

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Connection to Other Networks


➢ A wired LAN can be connected to another network or an internetwork such as the Internet using a
router. A wireless LAN may be connected to a wired infrastructure network, to a wireless
infrastructure network, or to another wireless LAN.
➢ The first situation is the one that we discuss in this section: connection of a wireless LAN to a wired
infrastructure network. Figure 15.2 shows the two environments.
➢ In this case, the wireless LAN is referred to as an infrastructure network, and the connection to the
wired infrastructure, such as the Internet, is done via a device called an access point (AP). Note that the
role of the access point is completely different from the role of a link- layer switch in the wired
environment.

➢ An access point is gluing two different environments together: one wired and one wireless.
Communication between the AP and the wireless host occurs in a wireless environment;
communication between the AP and the infrastructure occurs in a wired environment.

Fig. 50: Connection of a wired LAN and a wireless LAN to other networks

Moving between Environments


➢ A wired LAN or a wireless LAN operates only in the lower two layers of the TCP/IP protocol suite.
This means that if we have a wired LAN in a building that is connected via a router or a modem to the
Internet, all we need in order to move from the wired environment to a wireless environment is to
change the network interface cards designed for wired environments to the ones designed for wireless
environments and replace the link-layer switch with an access point.
➢ In this change, the link-layer addresses will change (because of changing NICs), but the network-layer
addresses (IP addresses) will remain the same; we are moving from wired links to wireless links.

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CHARACTERISTICS
There are several characteristics of wireless LANs that either do not apply to wired LANs or the existence
of which is negligible and can be ignored. Following are some of these characteristics here to pave the way
for discussing wireless LAN protocols.
➢ Attenuation

The strength of electromagnetic signals decreases rapidly because the signal disperses in all directions;
only a small portion of it reaches the receiver. The situation becomes worse with mobile senders that
operate on batteries and normally have small power supplies.
➢ Interference

Another issue is that a receiver may receive signals not only from the intended sender, but also from other
senders if they are using the same frequency band.
➢ Multipath Propagation

A receiver may receive more than one signal from the same sender because electromagnetic waves can be
reflected back from obstacles such as walls, the ground, or objects. The result is that the receiver receives
some signals at different phases (because they travel different paths). This makes the signal less
recognizable.

➢ Error
With the above characteristics of a wireless network, we can expect that errors and error detection are
more serious issues in a wireless network than in a wired network. If we think about the error level as the
measurement of signal-to-noise ratio (SNR), we can better understand why error detection and error
correction and retransmission are more important in a wireless network. SNR is enough to say that it
measures the ratio of good stuff to bad stuff (signal to noise). If SNR is high, it means that the signal is
stronger than the noise (unwanted signal), so we may be able to convert the signal to actual data. On the
other hand, when SNR is low, it means that the signal is corrupted by the noise and the data cannot be
recovered.

ACCESS CONTROL
➢ Maybe the most important issue we need to discuss in a wireless LAN is access control— how a
wireless host can get access to the shared medium (air). We know that the Standard Ethernet uses the
CSMA/CD algorithm. In this method, each host contends to access the medium and sends its frame if it
finds the medium idle.
➢ If a collision occurs, it is detected and the frame is sent again. Collision detection in CSMA/CD serves

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MODULE-2 VII SEM Computer Networks[18EC71]

two purposes. If a collision is detected, it means that the frame has not been received and needs to be
resent. If a collision is not detected, it is a kind of acknowledgment that the frame was received.
➢ The CSMA/CD algorithm does not work in wireless LANs for three reasons:
1. To detect a collision, a host needs to send and receive at the same time (sending the frame and
receiving the collision signal), which means the host needs to work in a duplex mode. Wireless
hosts do not have enough power to do so (the power is supplied by batteries). They can only send
or receive at one time.
2. Because of the hidden station problem, in which a station may not be aware of another station’s
transmission due to some obstacles or range problems, collision may occur but not be detected. Fig.
51 shows an example of the hidden station problem. Station B has a transmission range shown by
the left oval (sphere in space); every station in this range can hear any signal transmitted by station
B. Station C has a transmission range shown by the right oval (sphere in space); every station
located in this range can hear any signal transmitted by C. Station C is outside the transmission
range of B; likewise, station B is outside the transmission range of C. Station A, however, is in the
area covered by both B and C; it can hear any signal transmitted by B or C. The figure also shows
that the hidden station problem may also occur due to an obstacle. Assume that station B is sending
data to station A. In the middle of this transmission, station C also has data to send to station
A. However, station C is out of B’s range and transmissions from B cannot reach C. Therefore, C
thinks the medium is free. Station C sends its data to A, which results in a collision at A because
this station is receiving data from both B and C. In this case, we say that stations B and C are
hidden from each other with respect to A. Hidden stations can reduce the capacity of the network
because of the possibility of collision.
3. The distance between stations can be great. Signal fading could prevent a station at one end from
hearing a collision at the other end.

➢ To overcome the above three problems, Carrier Sense Multiple Access with Collision Avoidance
(CSMA/CA) was invented for wireless LANs.

Fig. 51: Hidden station problem

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