CN Notes by Mohan

7/31/2019 CN Notes by Mohan

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1) ISO / OSI MODEL:

ISO refers International Standards Organization was

established in 1947, it is a multinational body

dedicated to worldwide agreement on international

standards. ISO refers International StandardsOrganization was established in 1947, it is a OSI

refers to Open System Interconnection that covers

all aspects of network communication. It is a

standard of ISO. Here open system is a model that

allows any two different systems to communicate

regardless of their underlying architecture. Mainly, it

is not a protocol it is just a model.

OSI MODEL

The open system interconnection model is a layered

framework. It has seven separate but interrelated

layers. Each layer having unique responsibilities.

ARCHITECTURE

The architecture of OSI model is a layered

architecture. The seven layers are,

1. Physical layer 2. Data link layer 3. Network layer 4.

Transport layer 5. Session layer 6. Presentation layer

7. Application layer

The figure shown below shows the layers involved

when a message sent from A to B pass through

some intermediate devices. Both the devices A and

B are formed by the framed architecture. And the

intermediate nodes only having the layers are

physical, Data link and network. In every device each

layer gets the services from the layer just below to

it. When the device is connected to some other

device the layer of one device communicates with

the corresponding layer of another device. This is

known as peer to peer process. Each layer in the

sender adds its own information to the message.

This information is known is header and trailers.

When the information added at the beginning of

the data is known as header. Whereas added at the

end then it called as trailer. Headers added at layers

2, 3, 4, 5, 6. Trailer added at layer 2. Each layer is

connected with the next layer by using interfaces.

Each interface defines what information and

services a layer must provide for the layer above it.

ORGANIZATION OF LAYERS

The seven layers are arranged by three sub groups.

1. Network Support Layers 2. User Support Layers

3. Intermediate Layer

Network Support Layers:

Physical, Datalink and Network layers come under

the group. They deal with the physical aspects of

the data such as electrical specifications, physical

connections, physical addressing, and transport

timing and reliability.

User Support Layers:

Session, Presentation and Application layers comes

under the group. They deal with the interoperability

between the software systems. Intermediate Layer

The transport layer is the intermediate layer

between the network support and the user supportlayers.

FUNCTIONS OF THE LAYERS

PHYSICAL LAYER

The physical layer coordinates the functions

required to transmit a bit stream over a physical

medium. It deals with the mechanical and electrical

specifications of the interface and the transmission

medium.

The functions are,

1. Physical Characteristics of Interfaces and Media:

It defines the electrical and mechanical

characteristics of the interface and the media.

It defines the types of transmission medium

2. Representation of Bits

To transmit the stream of bits they must beencoded into signal. It defines the type of encoding

weather electrical or optical.

3. Data Rate

It defines the transmission rate i.e. the number of

bits sent per second.

4. Synchronization of Bits

The sender and receiver must be synchronized at bit

level.


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The transport layer creates a connection between

the two end ports.

It involves three steps. They are,

1. Connection establishment

2. Data transmission

3. Connection discard

4. Flow Control

Flow control is performed at end to end level

5. Error Control

Error control is performed at end to end level.

SESSION LAYER

It acts as a dialog controller. It establishes, maintains

and synchronizes the interaction between the

communication devices.The responsibilities are,

1. Dialog Control

2. Synchronization

It adds a synchronization points into a stream of

bits.The session layer allows two systems to enterinto a dialog. It allows the communication between

the devices.

PRESENTATION LAYER

The presentation layer is responsible for the

semantics and the syntax of the information

exchanged.The responsibilities are,

1. Translation

Different systems use different encoding systems.The presentation layer is responsible for

interoperability between different systems. The

presentation layer t the sender side translates the

information from the sender dependent format to a

common format. Likewise, at the receiver side

presentation layer translate the information from

common format to receiver dependent format.

2. Encryption

To ensure security encryption/decryption is used

.Encryption means transforms the original

information to another form

Decryption means retrieve the original information

from the encrypted data

3. Compression

It used to reduce the number of bits to be

transmitted.

APPLICATION LAYER

The application layer enables the user to access the

network. It provides interfaces between the users to

the network.The responsibilities are,

1. Network Virtual Terminal

It is a software version of a physical terminal and

allows a user to log onto a remote host.

2. File Transfer, Access, and Management

It allows a user to access files in a remote computer,

retrieve files, and manage or control files in a

remote computer.

3. Mail Services

4. Directory Service

It provides the basis for e-mail forwarding and

storage.It provides distributed database sources and

access for global information about various objects

and services.

2) DIFFERENCE BETWEEN TCP/IP AND OSI

Number of layers: TCP/IP defines only 5 layers

(although these are not specifically mentioned in

the standards)

Functions performed at a given layer:

In the OSI model each layer performs specific

functions.

In TCP/IP different protocols may be defined within

a layer, each performing different functions. What is

common about a set of protocols at the same layer

is that they share the same set of support protocols

at the next lower layer.

Interface between adjacent layers:


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In the OSI model, a protocol at a given layer may be

substituted by a new one without impacting on

adjacent layers.

In TCP/IP the strict use of all layers is not

mandated.

3) ERROR DETECTION AND CORRECTION

ERROR

Networks must be able to transfer data from one

device to another with complete accuracy. Some

part of a message will be altered in transit than that

the entire content will arrive intact. Many factors like

line noise can alter or wipe out one or more bits of

a given data unit. This is known as errors.

TYPES OF ERRORS

There are two types. They are,

1. Single Bit Error

It means that only one bit of a given data unit is

changed from 1 to

0 or from 0 to 1.

2. Burst Bit Error

It means that two or more bits in the data unit have

changed.

ERROR DETECTION

A burst bit does not necessarily means that the

errors occur in consecutive

Bits The length of the bust error is measured from

the first corrupted bit to the last corrupted bit.

Some bits in between may not be corrupted. For

reliable communication errors must be detected

and corrected. For error detection we are using

many mechanisms.

REDUNDANCY

One error detection mechanism is sending every

data unit twice. The receiving device then would be

able to do a bit for bit comparison between the two

versions of the data. Any discrepancy would indicatean error, and an appropriate correction mechanism

could be used. But instead of repeating the entire

data stream, a shorter group of bits may be

appended to the end of each unit. This technique is

called redundancy because extra bits are redundant

to the information. They are discarded as soon as

the accuracy of the transmission has beendetermined.

TYPES

Four types of redundancy checks are used in data

communications. They are,

1. vertical redundancy check (VRC)

2. longitudinal redundancy check (LRC)

3. cyclic redundancy check (CRC)

4. checksum

VERTICAL REDUNDANCY CHECK:

It is also known as parity check. In this technique a

redundant bit called a parity bit is appended to

every data unit so that the total number of 1s in the

unit including the parity bit becomes even for even

parity or odd for odd parity. In even parity, the data

unit is passed through the even parity generator. It

counts the number of 1s in the data unit. If odd

number of 1s, then it sets 1 in the parity bit to make

the number of 1s as even. If the data unit having

even number of 1s then it sets in the parity bit to

maintain the number of 1s as even. When it reaches

its destination, the receiver puts all bits through an

even parity checking function. If it counts even

number of 1s than there is no error. Otherwise there

is some error.

EXAMPLE:

The data is : 01010110

The VRC check : 010101100

In odd parity, the data unit is passed through the

odd parity generator. It counts the number of 1s in

the data unit. If even number of 1s, then it sets 1 in

the parity bit to make the number of 1s as odd. If

the data unit having odd number of 1s then it sets

in the parity bit to maintain the number of 1s as

odd. When it reaches its destination, the receiver

puts all bits through an odd parity checking


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function. If it counts odd number of 1s than there is

no error. Otherwise there is some error.

EXAMPLE

The data is: 01010110

The VRC check: 01010111

LONGITUDINAL REDUNDANCY CHECK

In this, a block of bits is organized in a table (rows

and columns). For example, instead of sending a

block of 32 bits, we organize them in a table made

of four roes and eight columns. We then calculate

the parity bit for each column and create a new row

of eight bits which are the parity bits for the whole

block

CYCLIC REDUNDANCY CHECK

CRC is based on binary division. In this a sequence

of redundant bits, called CRC remainder is

appended to the end of a data unit so that the

resulting data unit becomes exactly divisible by a

second predetermined binary number. At its

destination, the incoming data unit is divided by the

same number. If at this step there is no reminder,

the data unit is assumed to be intact and therefore

accepted. A remainder indicates that the data unit

has been changed in transit and therefore must be

rejected. Here, the remainder is the CRC. It must

have exactly one less bit than the divisor, and

appending it to the end of the data string must

make the resulting bit sequence exactly divisible by

the divisor.

First, a string of n-1 0s is appended to the data unit.

The number of 0s is one less than the number of

bits in the divisor which is n bits. Then the newly

elongated data unit is divided by the divisor using a

process called binary division. The remainder is CRC.

The CRC is replaces the appended 0s at the end of

the data unit. The data unit arrives at the receiver

first, followed by the CRC. The receiver treats whole

string as the data unit and divides it by the same

divisor that was used to find the CRC remainder. If

the remainder is 0 then the data unit is error free.

Otherwise it having some error and it must be

discarded.

CHECKSUM

The error detection method used by the higher

layer protocols is called checksum. It consists of two

arts. They are,

1. checksum generator 2. checksum checker

Checksum Generator:

In the sender, the checksum generator subdivides

the data unit into equal segments of n bits. These

segments are added with each other by using ones

complement arithmetic in such a way that the total

is also n bits long. That total is then complemented

and appended to the end of the data unit.

Checksum Checker:

The receiver subdivides the data unit as above andadds all segments together and complements the

result. If the extended data unit is intact, the total

value found by adding the data segments and the

checksum field should be zero. Otherwise the

packet contains an error and the receiver rejects it.

EXAMPLE see external ref.

ERROR CORRECTION

Error correction is handled in two ways. In one,

when an error is discovered, the receiver can have

the sender retransmit the entire data unit. In the

other, a receiver can use an error correcting code,

which automatically corrects certain errors.

Types of error correction:

1. Single bit error correction 2. Burst bit error

correction

Single Bit Error Correction

To correct a single bit error in an ASCII character,

the error correction code mustdetermine which of

the seven bits has changed. In this case we have to

determine eight different states: no error, error in

position 1, error in position 2, error in position 3,

error in position 4, error in position 5, error in

position 6, error in position 7. It looks like a three bit

redundancy code should be adequate because

three bits can show eight different states. But what

if an error occurs in the redundancy bits? Seven bits

of data and three bits of redundancy bits equal 10

bits. So three bits are not adequate.


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To calculate the number of redundancy bits (r)

required to correct a given number of data bits (m)

we must find a relationship between m and r.If the

total number of bits in a transmittable unit is m+r

then r must be able to indicate at least m+r+1

different state. Of these, one state means no errorand m+r states indicate the location of an error in

each of the m+r positions. So m+r+1 state must be

discoverable by r bits. And r bits can indicate 2r

different states. Therefore, 2r must be equal to or

greater than m+r+1;

2r >=m+r+1

NUMBER OF DATA

NUMBER OF REDUNDANCY BITS (R)

TOTAL BITS (M+R)

BITS (M) see external ref for example

Hamming Code:

The hamming code can be applied to data units of

any length and uses the relationship between data

and redundancy bits. Positions of redundancy bits in

hamming code The combinations used to calculate

each of the four r values for a seven bit data

sequence are as follows:

r1 :1,3,5,7,9,11 r2 : 2,3,6,7,10,11 r3 : 4,5,6,7 r4 :

8,9,10,11

Here, r1 bit is calculated using all bit positions

whose binary representation includes a 1 in the

rightmost position (0001, 0011, 0101, 0111, 1001,

and 1011). The r2 bit is calculated using all bit

positions with a 1 in the second position (0010,

0011, 0110, 0111, 1010 and 1011), and for r3 1 at

third bit position (0100, 0101, 0110 and 0111) for r41 at fourth bit position (1000, 1001, 1010 and 1011).

Calculating the r Values:

In the first step, we place each bit of the original

character in its appropriate positions in the 11 bit

unit. Then, we calculate the even parities for the

various bit combinations. The parity value of each

combination is the value of the corresponding r bit.

For example r1 is calculated to provide even parity

for a combination of bits 3, 5, 7, 9, 11.

Error Detection and Correction:

Example: see external ref

Burst Bit Error Correction:

Once the bit is identified, the receiver can reverse its

value and correct the error.

A hamming code can be designed to correct burst

errors of certain length. The number of redundancy

bits required to make these corrections, however, is

dramatically higher than that required for single bit

errors. To correct double bit errors, for example,

we must take into consideration that the two bits

can be a combination of any two bits in the entire

sequence. Three bit correction means any three bits

in the entire sequence and so on.

FUNCTIONS OF DATA LINK LAYER:

The data link layer is responsible for the following

functions. They are,

1. Line discipline or Access control 2. Flow control 3.

Error control 4. Framing

LINE DISCIPLINE

Communications requires at least two devices, one

to send and one to receive. If both devices areready to send some information and put their

signals on the link then the two signals collides each

other and became nothing. To avoid such a

situation the data link layer use a mechanism called

line discipline. and when it can send. It answers then

question, who should send now?

Line discipline coordinates the link system. It

determines which device can send. Line discipline

can serve in two ways:

1. enquiry / acknowledgement (ENQ / ACK) 2. poll /

select (POLL / SELECT)

ENQ / ACK:

This method is used in peer to peer

communications. That is where there is a dedicated

link between two devices. The initiator first

transmits a frame called an enquiry (ENQ) asking I

the receiver Is available to receive data. The receiver

must answer either with an acknowledgement (ACK)

frame if it ready to accept or with a negative

acknowledgement (NAK) frame if it is not ready. If


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the response is positive, the initiator is free to send

its data. Otherwise it waits, and try again. Once all

its data have been transmitted, the sending system

finishes with an end of transmission (EOT) frame.

POLL / SELECT

designated as primary and the other devices are

secondary.

This method of line discipline works with topologies

where one device is Whenever a multi point link

consists of a primary device and multiple secondary

devices using a single transmission line, all

exchanges must be made through the primary

device even when the ultimate destination is asecondary. The primary device controls the link. The

secondary devices follow its instructions. The

primary device only determines which device is

allowed to use the channel at a given time. The

primary asks the secondary if they have anything to

send; this function is called polling. And the primary

tells the target secondary to get ready to receive;

this function is called selecting.

POLL:

This function is used by the primary device to solicit

transmission from the secondary devices. The

secondary are not allowed to transmit data unless

asked by the primary device. When the primary

ready to receive data, it must ask (poll) each device

in turn if it has anything to send. If the secondary

have data to transmit it sends otherwise sends a

negative acknowledgment (NAK). the data frame

The primary then polls the next secondary. When

the response is positive (a data frame), the primary

reads the frame and returns an acknowledgment(ACK). There are two possibilities to terminate the

transmission: either the secondary sends all data,

finishing with an EOT frame, or the primary says

timers up. Then the primary cal polls the

remaining devices.

SELECT:

This mode of function is used whenever the primary

device has something to send. It alerts the intended

secondary device get ready to receive data. Before

sending data it sends the select (SEL) frame. The

receiver returns an ACK frame. Then the primary

sends data.

FLOW CONTROL AND ERROR CONTROL

FLOW CONTROL

It refers to a set of procedures used to restrict the

amount of data flow between sending and receiving

stations. It tells the sender how much data it can

transmit before it must wait for an

acknowledgement from the receiver. There are two

methods are used. They are,

1. stop and wait 2. sliding window

STOP AND WAIT:

In this method the sender waits for

acknowledgment after every frame it sends. Only

after an acknowledgment has been received, then

the sender sends the next frame. The advantage is

simplicity. The disadvantage is inefficiency.

SLIDING WINDOW:

In this method, the sender can transmit several

frames before needing an acknowledgment. The

receiver acknowledges only some of the frames,

using a single ACK to confirm the receipt of multiple

data frames. The sliding window refers to imaginary

boxes at both the sender and receiver. This window

provides the upper limit on the number of frames

that can be transmitted before requiring an

acknowledgement. To identify each frame the

sliding window scheme introduces the sequence

number. The frames are numbered as 0 to n-1. And

the size of the window is n-1. Here the size of the

window is 7 and the frames are numbered as

0,1,2,3,4,5,6,7.

SENDER WINDOW:

At the beginning the senders window contains n-1

frames. As frames are sent out the left boundary of

the window moves inward, shrinking the size of the

window. Once an ACK receives the window expands

at the right side boundary to allow in a number of

new frames equal to number of frames

acknowledged by that ACK.

EXAMPLE:

ERROR CONTROL


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Error control is implemented in such a way that

every time an error is detected, a negative

acknowledgement is returned and the specified

frame is retransmitted. This process is called

automatic repeat request (ARQ).

The error control is implemented with the flow

control mechanism. So there are two types in error

control. They are,

1. stop and wait ARQ 2. sliding window ARQ

STOP AND WAIT ARQ:

It is a form of stop and wait flow control, extended

to include retransmission of data in case of lost or

damaged frames.

DAMAGED FRAME:

When a frame is discovered by the receiver to

contain an error, it returns a NAK frame and the

sender retransmits the last frame.

LOST DATA FRAME:

The sender is equipped with a timer that starts

every time a data frame is transmitted. If the frame

lost in transmission the receiver can never

acknowledge it. The sending device waits for an

ACK or NAK frame until its timer goes off, then it

tries again. It retransmits the last data frame.

LOST ACKNOWLEDGEMENT:

The data frame was received by the receiver but the

acknowledgement was lost in transmission. The

sender waits until the timer goes off, then it

retransmits the data frame. The receiver gets a

duplicated copy of the data frame. So it knows the

acknowledgement was lost so it discards the secondcopy.

SLIDING WINDOW ARQ

It is used to send multiple frames per time. The

number of frame is according to the window size.

The sliding window is an imaginary box which is

reside on both sender and receiver side. It has two

types. They are,

1. go-back-n ARQ 2. selective reject ARQ

GO-BACK-N ARQ:

In this method, if one frame is lost or damaged, all

frames sent since the last frame acknowledged or

retransmitted.

DAMAGED FRAME:

LOST FRAME:

LOST ACK:

SELECTIVE REPEAT ARQ

Selective repeat ARQ re transmits only the damaged

or lost frames instead of sending multiple frames.

The selective transmission increases the efficiency of

transmission and is more suitable for noisy link. The

receiver should have sorting mechanism.


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DAMAGED FRAME:

LOST FRAME

LOST ACK

15)CONGESTION CONTROL

Congestion in a network may occur if the load on

the network-the number of packets sent to the

network-is greater than the capacity of the network-

the number of packets a network can handle.

Congestion control refers to the mechanisms and

techniques to control the congestion and keep the

load below the capacity. Congestion in a network or

internetwork occurs because routers and switches

have queues-buffers that hold the packets before

and after processing. Congestion control refers totechniques and mechanisms that can either prevent

congestion, before it happens, or remove

congestion, after it has happened.

Congestion control categories:

1) Open loop congestion control :

In open-loop congestion control, policies are

applied to prevent congestion before it happens. In

these mechanisms, congestion control is handled by

either the source or the destination.

Retransmission Policy:

If the sender feels that a sent packet is lost or

corrupted, the packet needs to be retransmitted.

Retransmission in general may increase congestion

in the network. However, a good retransmission

policy can prevent congestion. The retransmission

policy and the retransmission timers must be

designed to optimize efficiency and at the same

time prevent congestion. For example, the

retransmission policy used by TCP (explained later)

is designed to prevent or alleviate congestion. The

type of window at the sender may also affect

congestion. The Selective Repeat window is better

than the Go-Back-N window for congestion control.

In the Go-Back-N window, when the timer for a

packet times out, several packets may be resent,

although some may have arrived safe and sound at

the receiver. This duplication may make the

congestion worse. The Selective Repeat window, on

the other hand, tries to send the specific packets

that have been lost or corrupted.


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Acknowledgment Policy

The acknowledgment policy imposed by the

receiver may also affect congestion. If the receiver

does not acknowledge every packet it receives, it

may slow down the sender and help preventcongestion. A receiver may send an

acknowledgment only if it has a packet to be sent

or a special timer expires. A receiver may decide to

acknowledge only N packets at a time. A good

discarding policy by the routers may prevent

congestion and at the same time may not harm the

integrity of the transmission. For example, in audio

transmission, if the policy is to discard less sensitive

packets when congestion is likely to happen, the

quality of sound is still preserved and congestion is

prevented or alleviated. An admission policy, whichis a quality-of-service mechanism, can also prevent

congestion in virtual-circuit networks. Switches in a

flow first check the resource requirement of a flow

before admitting it to the network. A router can

deny establishing a virtual circuit connection if there

is congestion in the network or if there is a

possibility of future congestion.

2) Closed-Loop Congestion Control

Closed-loop congestion control mechanisms try to

alleviate congestion after it happens.

Back Pressure:

The technique of backpressure refers to a

congestion control mechanism in which a

congested node stops receiving data from the

immediate upstream node or nodes. This may cause

the upstream node or nodes to become congested,

and they, in turn, reject data from their upstream

nodes or nodes. And so on. Backpressure is a node-

to-node congestion control that starts with a node

and propagates, in the opposite direction of data

flow, to the source. The backpressure technique can

be applied only to virtual circuit networks, in which

each node knows the upstream node from which a

flow of data is coming.

Node III in the figure has more input data than it

can handle. It drops some packets in its input buffer

and informs node II to slow down. Node II, in turn,

may be congested because it is slowing down the

output flow of data. If node II is congested, it

informs node I to slow down, which in turn maycreate congestion. If so, node I informs the source

of data to slow down. This, in time, alleviates the

congestion. Note that the pressure on node III is

moved backward to the source to remove the

congestion.

Choke Packet:

A choke packet is a packet sent by a node to the

source to inform it of congestion.

Difference between the backpressure and choke

packet methods:

In backpressure, the warning is from one node to its

upstream node, although the warning may

eventually reach the source station. In the choke

packet method, the warning is from the router,

which has encountered congestion, to the source

station directly. The intermediate nodes through

which the packet has traveled are not warned.

Signaling

In implicit signaling, there is no communication

between the congested node or nodes and the

source. The source guesses that there is a

congestion somewhere in the network from other

symptoms. For example, when a source sends

several packets and there is no acknowledgment for

a while, one assumption is that the network is

congested. The delay in receiving an

acknowledgment is interpreted as congestion in the

network; the source should slow down.

iv) Explicit Signaling:

The node that experiences congestion can explicitly

send a signal to the source or destination. The

explicit signaling method, however, is different from

the choke packet method. In the choke packet

method, a separate packet is used for this purpose;

in the explicit signaling method, the signal is

included in the packets that carry data.

Backward Signaling: A bit can be set in a packet

moving in the direction opposite to the congestion.


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This bit can warn the source that there is congestion

and that it needs to slow down to avoid the

discarding of packets.

Forward Signaling: A bit can be set in a packet

moving in the direction of the congestion. This bitcan warn the destination that there is congestion.

The receiver in this case can use policies, such as

slowing down the acknowledgments, to alleviate

the congestion.

Congestion Control in TCP

Congestion Window:

The sender has two pieces of information: the

receiver-advertised window size and the congestion

window size. The actual size of the window is theminimum of these two.

Actual window size = minimum (rwnd, cwnd)

TCP's general policy for handling congestion is

based on three phases:

slow start

congestion avoidance

congestion detection.

Slow Start: Exponential Increase :

One of the algorithms used in TCP congestion

control is called slow start. This algorithm is based

on the idea that the size of the congestion window

(cwnd) starts with one maximum segment size

(MSS). The MSS is determined during connection

establishment by using an option of the same

name. The size of the window increases one MSS

each time an acknowledgment is received. In the

slow-start phase, the sender starts with a very slow

rate of transmission, but increases the rate rapidly

to reach a threshold. When the threshold is reached,

the data rate is reduced to avoid congestion. Finally

if congestion is detected, the sender goes back tothe slow-start or congestion avoidance phase based

on how the congestion is detected.

The rate is exponential as shown below:

The sender starts with cwnd =1 MSS. This means

that the sender can send only one segment. After

receipt of the acknowledgment for segment 1, the

size of the congestion window is increased by 1,

which means that cwnd is now 2. Now two more

segments can be sent. When each acknowledgment

is received, the size of the window is increased by 1

MSS. When all seven segments are acknowledged,

cwnd = 8.

Congestion Avoidance:

Additive Increase If we start with the slow-start

algorithm, the size of the congestion window

increases exponentially. To avoid congestion before

it happens, one must slow down this exponential

growth. TCP defines another algorithm called

congestion avoidance, which undergoes an additive

increase instead of an exponential one. When the

size of the congestion window reaches the slow-

start threshold, the slow-start phase stops and the

additive phase begins. In this algorithm, each time


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the whole window of segments is acknowledged

(one round), the size of the congestion window is

increased by 1.The rate is additive as shown below:

Congestion Detection:

Multiplicative Decrease If congestion occurs, the

congestion window size must be decreased. The

only way the sender can guess that congestion has

occurred is by the need to retransmit a segment.

However, retransmission can occur in one of two

cases:

when a timer times out or when three ACKs are

received. In both cases, the size of the threshold is

dropped to one-half, a multiplicative decrease.

Most TCP implementations have two reactions:

I. If a time-out occurs, there is a stronger possibility

of congestion; a segment has probably been

dropped in the network, and there is no news about

the sent segments.In this case TCP reacts strongly:

a. It sets the value of the threshold to one-half of

the current window size.

b. It sets cwnd to the size of one segment.

c. It starts the slow-start phase again.

2. If three ACKs are received, there is a weaker

possibility of congestion; a segment may have been

dropped, but some segments after that may have

arrived safely since three ACKs are received. This is

called fast transmission and fast recovery. In this

case, TCP has a weaker reaction:

a. It sets the value of the threshold to one-half of

the current window size.

b. It sets cwnd to the value of the threshold (some

implementations add three segment sizes to the

threshold).

c. It starts the congestion avoidance phase

16) QUALITY OF SERVICE

Quality of service (QoS) is an internetworking issue

and can be defined as something a flow seeks to

attain.

Flow Characteristics:

Four types of characteristics are attributed to a flow:

reliability, delay, jitter, and bandwidth.

i) Reliability

Reliability is a characteristic that a flow needs. Lack

of reliability means losing a packet or

acknowledgment, which entails retransmission.

ii) Delay

Source-to-destination delay is another flowcharacteristic. Again applications can tolerate delay

in different degrees. In this case, telephony, audio

conferencing, video conferencing, and remote log-

in need minimum delay, while delay in file transfer

or e-mail is less important.

iii) Jitter

Jitter is the variation in delay for packets belonging

to the same flow. For example, if four packets

depart at times 0, 1, 2, 3 and arrive at 20, 21, 22, 23,all have the same delay, 20 units of time. On the

other hand, if the above four packets arrive at 21,

23, 21, and 28, they will have different delays: 21,22,

19, and 24. For applications such as audio and

video, the first case is completely acceptable; the

second case is not.

Jitter is defined as the variation in the packet delay.

High jitter means the difference between delays is

large; low jitter means the variation is small.

iv) Bandwidth


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Different applications need different bandwidths. In

video conferencing we need to send millions of bits

per second to refresh a color screen while the total

number of bits in an e-mail may not reach even a

million.

TECHNIQUES TO IMPROVE QoS

Techniques to improve the quality of service are:

Scheduling traffic shaping admission control resource reservation.

1) Scheduling:

Packets from different flows arrive at a switch or

router for processing. A good scheduling technique

treats the different flows in a fair and appropriate

manner. Several scheduling techniques are

designed to improve the quality of service. Three of

them are:

a) FIFO queuing b) priority queuing c) weighted

fair queuing.

a) FIFO Queuing:

In first-in, first-out (FIFO) queuing, packets wait in a

buffer (queue) until the node (router or switch) is

ready to process them. If the average arrival rate is

higher than the average processing rate, the queue

will fill up and new packets will be discarded.

b) Priority Queuing

In priority queuing, packets are first assigned to apriority class. Each priority class has its own queue.

The packets in the highest-priority queue are

processed first. Packets in the lowest-priority queue

are processed last.

A priority queue can provide better QoS than theFIFO queue because higher priority traffic, such as

multimedia, can reach the destination with less

delay.

c) Weighted Fair Queuing:

A better scheduling method is weighted fair

queuing. In this technique, the packets are still

assigned to different classes and admitted to

different queues. The queues, however, are

weighted based on the priority of the queues;

higher priority means a higher weight. The system

processes packets in each queue in a round-robin

fashion with the number of packets selected from

each queue based on the corresponding weight.

2) Traffic Shaping:

Traffic shaping is a mechanism to control the

amount and the rate of the traffic sent to the

network. Two techniques can shape traffic: leaky

bucket and token bucket.

a) Leaky Bucket


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If a bucket has a small hole at the bottom, the water

leaks from the bucket at a constant rate as long as

there is water in the bucket. The rate at which the

water leaks does not depend on the rate at which

the water is input to the bucket unless the bucket is

empty. The input rate can vary, but the output rateremains constant. Similarly, in networking, a

technique called leaky bucket can smooth out

bursty traffic. Bursty chunks are stored in the bucket

and sent out at an average rate.

A simple leaky bucket implementation is shown in

Figure. A FIFO queue holds the packets. If the

traffic consists of fixed-size packets , the process

removes a fixed number of packets from the queue

at each tick of the clock. If the traffic consists of

variable-length packets, the fixed output rate mustbe based on the number of bytes or bits.

The following is an algorithm for variable-length

packets:

1. Initialize a counter to n at the tick of the clock.

2. Ifn is greater than the size of the packet, send the

packet and decrement the

counter by the packet size. Repeat this step until n

is smaller than the packet size.

3. Reset the counter and go to step 1.

b) Token Bucket:

The leaky bucket is very restrictive. It does not

credit an idle host. For example, if a host is not

sending for a while, its bucket becomes empty. Now

if the host has bursty data, the leaky bucket allows

only an average rate. The time when the host was

idle is not taken into account. On the other hand,

the token bucket algorithm allows idle hosts to

accumulate credit for the future in the form of

tokens. For each tick of the clock, the system sends

n tokens to the bucket. The system removes one

token for every cell (or byte) of data sent. For

example, ifn is 100 and the host is idle for 100 ticks,

the bucket collects 10,000 tokens. Now the host can

consume all these tokens in one tick with 10,000cells, or the host takes 1000 ticks with 10 cells per

tick. In other words, the host can send bursty data

as long as the bucket is not empty.

3) Resource Reservation:

A flow of data needs resources such as a buffer,

bandwidth, CPU time, and so on. The quality ofservice is improved if these resources are reserved

beforehand. one QoS model called Integrated

Services, which depends heavily on resource

reservation to improve the quality of service.

4) Admission Control

Admission control refers to the mechanism used by

a router, or a switch, to accept or reject a flow based

on predefined parameters called flow specifications.

Before a router accepts a flow for processing, it

checks the flow specifications to see if its capacity

(in terms of bandwidth, buffer size, CPU speed, etc.)

and its previous commitments to other flows can

handle the new flow.

14) Transmission Control Protocol (TCP)

TCP is a connection oriented protocol; it creates a

virtual connection between two TCPs to send data.

In addition, TCP uses flow and error control

mechanisms at the transport level

TCP Services


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i) Process-to-Process Communication: TCP

provides process-to-process communication using

port numbers.

Well-known ports used by TCP:

ii) Stream Delivery Service:

TCP is a stream-oriented protocol. TCP allows the

sending process to deliver data as a stream of bytes

and allows the receiving process to obtain data as a

stream of bytes. TCP creates an environment in

which the two processes seem to be connected by

an imaginary "tube" that carries their data across

the Internet. The sending process produces (writes

to) the stream of bytes, and the receiving process

consumes (reads from) them.

iii) Sending and Receiving Buffers:

Because the sending and the receiving processes

may not write or read data at the same speed, TCP

needs buffers for storage. There are two buffers, the

sending buffer and the receiving buffer, one for

each direction. One way to implement a buffer is to

use a circular array of I-byte locations.

At the sending site, the buffer has

three types of chambers. The white section contains

empty chambers that can be filled by the sending

process (producer). The gray area holds bytes that

have been sent but not yet acknowledged. TCP

keeps these bytes in the buffer until it receives an

acknowledgment. The colored area contains bytes

to be sent by the sending TCP.

The operation of the buffer at the

receiver site is simpler. The circular buffer is divided

into two areas (shown as white and colored). The

white area contains empty chambers to be filled by

bytes received from the network. The colored

sections contain received bytes that can be read by

the receiving process. When a byte is read by the

receiving process, the chamber is recycled and

added to the pool of empty chambers.

iv) Segments

TCP groups a number of bytes together into a

packet called a segment. TCP adds a header to each

segment (for control purposes) and delivers the

segment to the IP layer for transmission. Thesegments are encapsulated in IP datagrams and

transmitted. This entire operation is transparent to

the receiving process.


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v) Full-Duplex Communication

TCP offers full-duplex service, in which data can flow

in both directions at the same time. Each TCP then

has a sending and receiving buffer, and segments

move in both directions.

vi) Connection-Oriented Service:

TCP, unlike UDP, is a connection-oriented protocol.

When a process at site A wants to send and receive

data from another process at site B, the following

occurs:

1. The two TCPs establish a connection between

them.

2. Data are exchanged in both directions.

3. The connection is terminated.

Note that this is a virtual connection, not a physical

connection.

TCP is a reliable transport protocol. It uses an

acknowledgment mechanism to check the safe and

sound arrival of data.

TCP Features:

To provide the services mentioned, TCP has severalfeatures.

a) Numbering System:

Byte Number

The bytes of data being transferred in each

connection are numbered by TCP. TCP numbers all

data bytes that are transmitted in a connection.

Numbering is independent in each direction. When

TCP receives bytes of data from a process, it stores

them in the sending buffer and numbers them. The

numbering does not necessarily start from O.

Instead, TCP generates a random number between

0 and 232 - 1 for the number of the first byte.

Sequence Number

After the bytes have been numbered, TCP assigns a

sequence number to each segment that is being

sent. The sequence number for each segment is the

number of the first byte carried in that segment.

Acknowledgment Number

Communication in TCP is full duplex; when a

connection is established, both parties can send and

receive data at the same time. Each party numbers

the bytes, usually with a different starting byte

number. The sequence number in each direction

shows the number of the first byte carried by thesegment. Each party also uses an acknowledgment

number to confirm the bytes it has received.

However, the acknowledgment number defines the

number of the next byte that the party expects to

receive. In addition, the acknowledgment number is

cumulative, which means that the party takes the

number of the last byte that it has received, safe

and sound, adds I to it, and announces this sum as

the acknowledgment number.The term cumulative

here means that if a party uses 5643 as an

acknowledgment number, it has received all bytes

from the beginning up to 5642. Note that this does

not mean that the party has received 5642 bytes

because the first byte number does not have to

start from O.

b) Flow Control

TCP, unlike UDP, provides flow control. The receiver

of the data controls the amount of data that are to

be sent by the sender. This is done to prevent the

receiver from being overwhelmed with data. Thenumbering system allows TCP to use a byte-

oriented flow control.

c) Error Control

To provide reliable service, TCP implements an error

control mechanism. Although error control

considers a segment as the unit of data for error

detection (loss or corrupted segments), error

control is byte-oriented.

d) Congestion Control:


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TCP, unlike UDP, takes into account congestion in

the network. The amount of data sent by a sender is

not only controlled by the receiver (flow control),

but is also detennined by the level of congestion in

the network.

TCP segment format:

The segment consists of a 20- to 60-byte header,

followed by data from the application program.

Source port address: This is a 16-bit field that

defines the port number of the application program

in the host that is sending the segment.

Destination port address: This is a 16-bit field thatdefines the port number of the application program

in the host that is receiving the segment.

Sequence number: This 32-bit field defines the

number assigned to the first byte of data contained

in this segment.

Acknowledgment number: This 32-bit field

defines the byte number that the receiver of the

segment is expecting to receive from the other

party.

Header length : This 4-bit field indicates the

number of 4-byte words in the TCP Header.

Reserved: This is a 6-bit field reserved for future

use.

Control: This field defines 6 different control bits or

flags.

Window size : This field defines the size of the

window, in bytes, that the other party must

maintain.

Checksum: This 16-bit field contains the checksum.

Urgent pointer: This l6-bit field, which is valid onlyif the urgent flag is set, is used when the segment

contains urgent data.

Options : There can be up to 40 bytes of optional

information in the TCP header.

Documents

CN Notes by Mohan