UNit-4 Transport Layer and its protocols.pdf

Unit -4 Process-to-
Process Delivery
UDP, TCP, and SCTP

INTRODUCTION
oThe transport layer is responsible for the delivery of a message from
one process to another
oThe transport layer header must include a service – point –address
in the OSI model or port number in the TCP/IP (internet model)
oThe Internet model has three protocols at the transport layer:
UDP, TCP, and SCTP.
 UDP: Is the simplest of the three.
 TCP: A complex transport layer protocol.
 SCTP: The new transport layer protocol that is designed for specific
applications such as multimedia. a new reliable, message-oriented
transport layer protocol that combines the best features of UDP and
TCP

PROCESS-TO-PROCESS
DELIVERY
oThe Data link layer is responsible for delivery of frames between nodes over a
link node to node delivery using a MAC address to choose one node among
several.
o The Network layer is responsible for delivery of datagrams between two hosts
host to host delivery using an IP address to choose one host among millions.
o Real communication takes place between two processes (application
programs). We need process-to-process delivery.
oWe need a mechanism to deliver data from one of process running on the
source host to the corresponding process running on the destination host.
oThe Transport layer is responsible for process-to-process . We need a port
number, to choose among multiple processes running on the destination host.

CLIENT/SERVER PARADIGM
o process-to-process communication can be achieved through
client/server
oA process on the local host, called a client, needs services from a
process usually on the remote host, called a server.
oBoth processes (client and server) have the same name.
oFor example, to get the day and time from a remote machine, we need a Daytime
client process running on the local host and a Daytime server process running on
a remote machine.
oA remote computer can run several server programs at the same
time, just as local computers can run one or more client programs at
the same time.

PORT NUMBER
oIn the Internet model, the port numbers are 16-bit integers between 0 and
65,535.
oThe client program defines itself with a port number, chosen
randomly by the transport layer software running on the client host
oThe server process must also define itself with a port number This port
number, however, cannot be chosen randomly
oThe Internet uses port numbers for servers called well- known port
numbers.
oEvery client process knows the well-known port number of the
corresponding server process
o For example, while the Daytime client process, can use an ephemeral
(temporary) port number 52,000 to identify itself, the Daytime server process must
use the well-known (permanent) port number 13.

IPADDRESSES VERSUS PORT
NUMBERS
IP addresses and port numbers play different roles in selecting the final
destination of data.
The destination IP address defines the host among the different hosts
After the host has been selected, the port number defines one of the processes on
this particular host

SOCKET ADDRESSES
o Process-to-process delivery needs two identifiers, IP address and the port
number, at each end to make a connection.
oThe combination of an IP address and a port number is called a socket address.
oA transport layer protocol needs a pair of socket addresses: the client socket
address and the server socket address.
oThese four pieces of information are part of the IP header and the transport
layer protocol header.
• The IP header contains the IP addresses; the UDP or TCP header contains
the port numbers.

MULTIPLEXING
Upward Multiplexing
One IP address shared by multiple Transport Layer
processes (ports)
Downward Multiplexing
Use multiple virtual circuits to get more bandwidth
(e.g. join multiple ISDN lines to get a higher
bandwidth)

MULTIPLEXING AND DEMULTIPLEXING
Sender: multiplexing of
UDP datagrams.
UDP datagrams are
received from multiple
application programs.
A single sequence of
UDP datagrams is
passed to IP layer.
Receiver: demultiplexing
of UDP datagrams.
Single sequence of
UDP datagrams
received from IP layer.
UDP datagram
received is passed to
appropriate
application.

CONNECTIONLESS VERSUS
CONNECTION-ORIENTED SERVICE
oA transport layer protocol can either be connectionless or connection-
oriented.
oConnectionless Service
➢ In a connectionless service, the packets are sent from one party to another
with no need for connection establishment or connection release.
➢The packets are not numbered; they may be delayed or lost or may arrive out
of sequence.
➢There is no acknowledgment .
oConnection Oriented Service
➢In a connection-oriented service, a connection is first established between
the sender and the receiver.
➢Data are transferred.
➢At the end, the connection is released. ( virtual connection , not a physical
connection)

RELIABLE VERSUS UNRELIABLE
oThe transport layer service can be reliable or unreliable.
oIf the application layer program needs reliability, we use a reliable
transport layer protocol by implementing flow and error control at
the transport layer. This means a slower and more complex service.
oOn the other hand, if the application program does not need
reliability then an unreliable protocol can be used.
Note
oUDP is connectionless and unreliable;
oTCP and SCTP are connection oriented and reliable.
These three protocols can respond to the demands of the
application layer programs.

ERROR CONTROL
If the data link layer is reliable and has flow and error control, do we need this
at the transport layer, too? Yes

UDP
UDP is located between
the application layer and
the IP layer, and serves
as the intermediary
between the application
programs and the
network operations.

TCP/IP PROTOCOL SUITE 16
Figure 14.1 Position of UDP in the TCP/IP protocol suite

1. USER DATAGRAM PROTOCOL (UDP )
oUDP is a connectionless, unreliable transport protocol.
oIt does not add anything to the services of IP (very simple) except to
provide process-to process communication instead of host-to-host
communication.
o(when to use it?) If a process wants to send a small message and does
not care much about reliability, it can use UDP.
UDP packets, called user datagrams, have a fixed size header of 8 bytes.
https://www.youtube.com/watch?v=blV7WUZpkCE
UDP packets,
called user
datagrams, have
a fixed-size
header of 8
bytes.

UDP CHARACTERISTICS
End-to-End: an application sends/receives data to/from another
application.
Connectionless: Application does not need to preestablish communication
before sending data; application does not need to terminate
communication when finished.
Message-oriented: application sends/receives individual messages (UDP
datagram), not packets.
Best-effort: same best-effort delivery semantics as IP. I.e. message can be
lost, duplicated, and corrupted.
Arbitrary interaction: application communicates with many or one other
applications.
Operating system independent: identifying application does not depend
on O/S.

UDP HEADER
The UDP header consists of four fields each of 2 bytes in length:
Source Port (UDP packets from a client use this as a service access
point (SAP) to indicate the session on the local client that originated
the packet. UDP packets from a server carry the server SAP in this
field)
Destination Port (UDP packets from a client use this as a service
access point (SAP) to indicate the service required from the remote
server. UDP packets from a server carry the client SAP in this field)
UDP length (The number of bytes comprising the combined UDP
header information and payload data)
UDP Checksum (A checksum to verify that the end to end data has
not been corrupted by routers or bridges in the network or by the
processing in an end system.

UDP DATAGRAM FORMAT
Source Port - 16 bit port number
Destination Port - 16 bit port number
Length (of UDP header + data) - 16 bit count of octets
UDP checksum - 16 bit field. if 0, then there is no checksum, else it is a checksum
over a pseudo header + UDP data area

CHECKSUM AND PSEUDO HEADER
UDP uses a pseudo-header to verify that the UDP message
has arrived at both the correct machine and the correct port.
• Proto : IP protocol type code.
• Length : Length of the UDP datagram.

ENCAPSULATION AND LAYERING
UDP message is encapsulated into an IP datagram.
IP datagram in turn is encapsulated into a physical frame for
actually delivery.

The following is a dump of a UDP header in hexadecimal
format.
Example 14.1
a. What is the source port number?
b. What is the destination port number?
c. What is the total length of the user datagram?
d. What is the length of the data?
e. Is the packet directed from a client to a server or vice versa?
f. What is the client process?

Applications of UDP:
•Used for simple request response communication when size of data is less and
hence there is lesser concern about flow and error control.
•It is suitable protocol for multicasting as UDP supports packet switching.
•UDP is used for some routing update protocols like RIP(Routing Information
Protocol).
•Normally used for real time applications which can not tolerate uneven delays
between sections of a received message.
•Following implementations uses UDP as a transport layer protocol:
• NTP (Network Time Protocol)
• DNS (Domain Name Service)
• BOOTP, DHCP.
• NNP (Network News Protocol)
• Quote of the day protocol
• TFTP, RTSP, RIP, OSPF.
•Application layer can do some of the tasks through UDP-
• Trace Route
• Record Route
• Time stamp
•UDP takes datagram from Network Layer, attach its header and send it to the
user. So, it works fast.
•Actually UDP is null protocol if you remove checksum field.

2. TRANSMISSION CONTROL PROTOCOL(TCP)
oTCP, like UDP, is a process-to-process (program-to-
program) protocol uses port numbers.
oUnlike UDP, TCP is a connection oriented protocol; it
creates a virtual connection between two TCPs to send data.
oTCP uses flow and error control mechanisms at the
transport level.

2. TRANSMISSION CONTROL
PROTOCOL(TCP)
oThe sending and the receiving processes may not write or read data at the
same speed, TCP needs buffers for storage.
oTwo buffers, the sending buffer and the receiving buffer, one for each
direction.
oThese buffers are also necessary for flow and error control mechanisms used
by TCP.

A. TCP FEATURES
Numbering System:
There are two fields called the sequence number and the acknowledgment
number.
These two fields refer to the byte number and not the segment number.
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 0.
TCP generates a random number between 0 and 232 - 1 for the number of the first
byte.

Byte numbering is used for flow and error control. For
example: if the random number happens to be 1057 and the
total data to be sent are 6000 bytes, the bytes are numbered
from 1057 to 7056.
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
Each party uses an acknowledgment number to confirm the
bytes it has received.
The acknowledgment number defines the number of the next byte
that the party expects to receive.

The segment consists of a 20-60-byte header.
Source port address:
This is a 16-bit field , it 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, it defines 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 number of the next byte a party
expects to receive.
Header length: A 4-bit field that indicates the number of 4-byte words in the TCP
header. The length of the header can be between 20 and 60 bytes. Therefore, the
value of this field can be between 5 (5 x 4 =20) and 15 (15 x 4 =60).
TCP Segment Format

TCP SEGMENT FORMAT
Reserved. This is a 6-bit field reserved for future use.
Control: This field defines 6 different control bits or flags. One or
more of these bits can be set at a time. These bits enable flow control,
connection establishment and termination, connection abortion, and the
mode of data transfer in TCP

UNit-4 Transport Layer and its protocols.pdf

TCP SEGMENT FORMAT
Window size: Defines the size of the window, in bytes, that the other party must
maintain. the length of this field is 16 bits, which means that the maximum size of
the window is 65,535 bytes. This value is normally referred to as the receiving
window (rwnd) and is determined by the receiver. The sender must obey the
dictation of the receiver in this case.
Checksum: This 16-bit field contains the checksum. The inclusion of the checksum for
TCP is mandatory.
Options: There can be up to 40 bytes of optional information in the TCP header.

TCP SEGMENT FORMAT
Urgent data :
oThe data are presented from the application program to TCP as a
stream of bytes.
o Each byte of data has a position in the stream.
oIf the sending application program wants a piece of data to be
read out of order by the receiving application program., the
sending TCP creates a segment and inserts the urgent data at the
beginning of the segment.
oThe rest of the segment can contain normal data from the buffer
o

TCP SEGMENT FORMAT
Urgent data :
oThe urgent pointer field in the header (if its set) defines the end of
the urgent data and the start of normal data.
oWhen the receiving TCP receives a segment with the URG bit set, it
extracts the urgent data from the segment using the value of the urgent
pointer, and delivers them, out of order, to the receiving application
program.

C. TCP CONNECTION
A Connection-oriented transport protocol establishes a virtual path between
the source and destination.
In TCP, connection-oriented transmission requires three phases:
1. connection establishment
2. data transfer
3. connection termination

1) CONNECTION ESTABLISHMENT
TCP transmits data in full-duplex mode.
 When two TCPs in two machines are connected, they are able to send segments
to each other simultaneously.
Each party must initialize communication and get approval from the other party
before any data are transferred.
The connection establishment in TCP is called three way handshaking.

1) CONNECTION ESTABLISHMENT
Example:
Client-server communication using TCP as the transport layer protocol.
1.The server issues a request for a passive open: The server program
tells its TCP that it is ready to accept a connection.
2.The client program issues a request for an active open: A client that
wishes to connect to an open server tells its TCP that it needs to be
connected to that particular server. TCP can now start the three-way
handshaking process

THREE-WAY HANDSHAKING PROCESS
1.The client sends the first segment, a SYN segment, in which only the
SYN flag is set.
 This segment is for synchronization of sequence numbers. It consumes one sequence
number.
 When the data transfer starts, the sequence number is incremented by 1.
 The SYN segment carries no real data
2. The server sends the second segment, a SYN +ACK segment, with 2
flag bits set: SYN and ACK.
 This segment has a dual purpose. It is a SYN segment for communication in the other
direction and serves as the acknowledgment for the SYN segment.
 It consumes one sequence number.

THREE-WAY HANDSHAKING PROCESS
3. The client sends the third segment. This is just an ACK
segment.
 It acknowledges the receipt of the second segment with the ACK flag
and acknowledgment number field.
 The sequence number in this segment is the same as the one in the
SYN segment.
 The ACK segment does not consume any sequence numbers.

2) DATA TRANSFER
After connection is established, bidirectional data transfer can take place.
The client and server can both send data and acknowledgments
The acknowledgment is piggybacked with the data.
Example:
After connection is established,(shown before) the client sends 2000 bytes
of data in two segments.
The server then sends 2000 bytes in one segment.
The client sends one more acknowledgment segment.
The first three segments carry both data and acknowledgment, but the last
segment carries only an acknowledgment because there are no more data to
be sent.

3) CONNECTION TERMINATION
1. The client TCP, after receiving a close command from the client
process, sends the first segment, a FIN segment in which the FIN flag is
set.
 A FIN segment can include the last chunk of data sent by the client, or it can be just a
control segment
 If it is only a control segment, it consumes only one sequence number.
2. The server TCP, after receiving the FIN segment, sends the second
segment, a FIN +ACK segment, to confirm the receipt of the FIN
segment from the client and at the same time to announce the closing
of the connection in the other direction.
 This segment can also contain the last chunk of data from the server.
 If it does not carry data, it consumes only one sequence number.

3) CONNECTION TERMINATION
3. The client TCP sends the last segment, an ACK segment, to confirm
the receipt of the FIN segment from the TCP server.
 This segment contains the acknowledgment number, which is 1 plus the sequence
number received in the FIN segment from the server.
 This segment cannot carry data and consumes no sequence numbers.

Any of the two parties involved in exchanging data (client or server) can close
the connection using three-Way Handshaking

D. FLOW CONTROL
TCP uses a sliding window to handle flow control.
TCP sliding window is of variable size.
The window is opened, closed, or shrunk.
 Opening a window means moving the right wall to the right.
 Closing the window means moving the left wall to the right.
 Shrinking the window means moving the right wall to the left.
The size of the window at one end is determined by the lesser of two
values: receiver window (rwnd) or congestion window (cwnd).

1. CONGESTION
Congestion: the load on the network is greater than the capacity of the network
Congestion control: the mechanisms to control the congestion and keep the load
below the capacity
Congestion occurs because routers and switches have queues- buffers that hold
the packets before and after processing
The rate of packet arrival > packet processing time → input queue longer
The packet departure time < packet processing time → output queue longer
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CONGESTION CONTROL
Congestion control refers to techniques and mechanisms that can either
prevent congestion, before it happens, or remove congestion, after it has
happened.
Two broad categories: open-loop congestion control (prevention) and
closed-loop congestion control (removal).
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A. OPEN LOOP CONTROL: PREVENTION
In open-loop congestion control, policies are applied to prevent congestion before it
happens.
list of policies that can prevent congestion:
1. Retransmission policy and timers must to be designed to optimize efficiency and at the
same time prevent congestion
2. Window policy: The type of window at the sender may also affect congestion. Selective
Repeat is better than Go-back-N
3. Acknowledgement policy: The acknowledgment policy imposed by the receiver may also
affect congestion.
4. Discard policy: A good discarding policy by the routers may prevent congestion and at the
same time may not harm the integrity of the transmission.
5. Admission policy: Switch first check the resource requirement of a flow before admitting it
to the network
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B. CLOSED-LOOP CONTROL: REMOVAL
Closed-loop congestion control mechanisms try to alleviate congestion after it happens.
Several mechanisms have been used by different protocols.
1. Back pressure: inform the previous upstream router to reduce the rate of outgoing packets
if congested
2. Choke point: a packet sent by a router to the source to inform it of congestion, similar to
ICMP’s source quench packet
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3. 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.
4. Explicit signaling: The node that experiences congestion can explicitly
send a signal to the source or destination. Backward signaling /
Forward signaling
 Backward Signaling A bit can be set in a packet moving in the direction
opposite to the congestion. 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 bit can 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.
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CONGESTION CONTROL IN TCP
TCP assumes that the cause of a lost segment is due to congestion in the
network.
If the cause of the lost segment is congestion, retransmission of the segment
does not remove the cause—it aggravates it.
The sender has two pieces of information: the receiver-advertised window
size and the congestion window size
TCP Congestion window
 Actual window size = minimum (rwnd, cwnd)
(where rwnd = receiver window size, cwnd = congestion window size)
TCP Congestion Policy
Based on three phases: 1. slow start, 2. congestion avoidance, and 3.
congestion detection
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TCP CONGESTION POLICY
1. Slow Start: Exponential Increase
 In the slow-start algorithm, the size of the congestion
window increases exponentially until it reaches a
threshold
The sender keeps track of a variable named
ssthresh (slow-start threshold).
When the size of window in bytes reaches
this threshold, slow start stops and the next
phase starts (additive phase begins).
In most implementations the value of ssthresh
is 65,535 bytes.
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2. Congestion Avoidance:
Additive
 Increase the size of the congestion
window increases additively until
congestion is detected
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CONGESTION CONTROL
As Internet can be considered as a Queue of packets,
where transmitting nodes are constantly adding packets
and some of them (receiving nodes) are removing packets
from the queue.
So, consider a situation where too many packets are
present in this queue (or internet or a part of internet),
such that constantly transmitting nodes are pouring
packets at a higher rate than receiving nodes are
removing them.
This degrades the performance, and such a situation is
termed as Congestion.
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CONGESTION CONTROL
Main reason of congestion is more number of packets into
the network than it can handle.
So, the objective of congestion control can be summarized
as to maintain the number of packets in the network below
the level at which performance falls off dramatically.
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CONGESTION CONTROL
The nature of a Packet switching network can be
summarized in following points:
A network of queues
At each node, there is a queue of packets for each
outgoing channel
If packet arrival rate exceeds the packet transmission
rate, the queue size grows without bound
When the line for which packets are queuing
becomes more than 80% utilized, the queue length
grows alarmingly
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CONGESTION CONTROL
When the number of packets dumped into the
network is within the carrying capacity, they all are
delivered, expect a few that have too be rejected
due to transmission errors).
And then the number delivered is proportional to the
number of packets sent.
However, as traffic increases too far, the routers are
no longer able to cope, and they begin to lose
packets.
This tends to make matter worse. At very high traffic,
performance collapse completely, and almost no
packet is delivered.
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CONGESTION CONTROL
insufficient memory
Slow processors
bursty nature of traffic
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REGULATING FLOW CONTROL
The bursty traffic in the network results in congestion
Traffic shaping reduces congestion and thus helps the carrier live up to
its guarantees
Traffic shaping is about regulating the average rate (and burstiness)
of data transmission
70

Congestion control refers to the mechanisms and
techniques used to control congestion and keep the traffic
below the capacity of the network. the congestion control
techniques can be broadly classified two broad
categories:
Open loop: Protocols to prevent or avoid congestion,
ensuring that the system (or network under consideration)
never enters a Congested State. • Close loop: Protocols
that allow system to enter congested state, detect it, and
remove it.
71

The first category of solutions or protocols attempt to solve the problem
by a good design, at first, to make sure that it doesn’t occur at all.
Once system is up and running midcourse corrections are not made.
These solutions are somewhat static in nature, as the policies to control
congestion don’t change much according to the current state of the
system. Such Protocols are also known as Open Loop solutions.
These rules or policies include deciding upon when to accept traffic,
when to discard it, making scheduling decisions and so on.
Main point here is that they make decision without taking into
consideration the current state of the network.
The open loop algorithms are further divided on the basis of whether
these acts on source versus that act upon destination.
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Traffic Shaping
Traffic shaping controls the rate at which packets are
sent (not just how many)
At connection set-up time, the sender and carrier
negotiate a traffic pattern (shape)
Two traffic shaping algorithms are:
 Leaky Bucket
 Token Bucket
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THE LEAKY BUCKET ALGORITHM
The Leaky Bucket Algorithm used to control rate in a network. It is
implemented as a single-server queue with constant service time. If the
bucket (buffer) overflows then packets are discarded.
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THE LEAKY BUCKET ALGORITHM
(a) A leaky bucket with water. (b) a leaky bucket with
packets.
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Leaky Bucket Algorithm (contd.)
The leaky bucket enforces a constant output rate regardless of
the burstiness of the input. Does nothing when input is idle.
The host injects one packet per clock tick onto the network. This
results in a uniform flow of packets, smoothing out bursts and
reducing congestion.
When packets are the same size (as in ATM cells), the one
packet per tick is okay. For variable length packets though, it is
better to allow a fixed number of bytes per tick.
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TOKEN BUCKET ALGORITHM
In contrast to the LB, the Token Bucket (TB) algorithm, allows the
output rate to vary, depending on the size of the burst.
In the TB algorithm, the bucket holds tokens. To transmit a
packet, the host must capture and destroy one token.
Tokens are generated by a clock at the rate of one token
every t sec.
Idle hosts can capture and save up tokens (up to the max. size
of the bucket) in order to send larger bursts later.
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TOKEN BUCKET ALGORITHM (CONTD.)
(a) Before (b) After
78
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TOKEN BUCKET OPERATION
TB accumulates fixed size tokens in a token bucket
Transmits a packet (from data buffer, if any are
there) or arriving packet if the sum of the token
sizes in the bucket add up to packet size
More tokens are periodically added to the bucket
(at rate t). If tokens are to be added when the
bucket is full, they are discarded
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TOKEN BUCKET PROPERTIES
Does not bound the peak rate of small bursts, because bucket may
contain enough token to cover a complete burst size
Performance depends only on the sum of the data buffer size and the
token bucket size
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TOKEN BUCKET - EXAMPLE
2 tokens of size 100 bytes added each second to
the token bucket of capacity 500 bytes
 Avg. rate = 200 bytes/sec, burst size = 500 bytes
 Packets bigger than 500 bytes will never be sent
 Peak rate is unbounded – i.e., 500 bytes of burst can be
transmitted arbitrarily fast
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Leaky Bucket vs Token Bucket
LB discards packets; TB does not. TB discards tokens.
With TB, a packet can only be transmitted if there are enough
tokens to cover its length in bytes.
LB sends packets at an average rate. TB allows for large bursts
to be sent faster by speeding up the output.
TB allows saving up tokens (permissions) to send large bursts.
LB does not allow saving.
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3. Congestion Detection: Multiplicative Decrease
An implementation reacts to congestion detection in one of two ways:
 If detection is by time-out, a new slow start phase starts
 If detection is by three ACKs, a new congestion avoidance phase starts
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2. QUALITY OF SERVICE (QOS)
Flow Characteristics:
Reliability:
 needed by flow, Lack of reliability means losing a packet or
acknowledgment, which entails retransmission.
Delay:
 applications can tolerate delay in different degrees.
Jitter:
 the variation in delay for packets belonging to the same flow
 High jitter means the difference between delays is large; low jitter
means the variation is small.
Bandwidth:
 Different applications need different bandwidths.
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QOS TECHNIQUES
Four common techniques that can be used to improve the quality of
service :
 Scheduling: A good scheduling technique treats the different flows in a
fair and appropriate manner.
 Traffic shaping: Leaky bucket, token bucket
 Resource reservation
 Admission control: accept or reject a flow based on predefined
parameters called flow specification
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SCTP
In computer networking, the Stream Control Transmission Protocol (SCTP) is a
transport-layer protocol, serving in a similar role to the popular protocols TCP
and UDP. It is standardized by IETF in RFC 4960.

SCTP
SCTP adds to the mix is multi-homing. Multi-homing allows the
two endpoints of a connection to declare multiple interfaces (IP
addresses). Providing and alternate route for the data in case the
current interface in use fails for some reason.
The second feature is multi-streaming. Rather than use a single
stream of data, SCTP can create multiple streams that can be used
independently.
This doesn’t really improve the speed of the medium but it allows
the data to arrive concurrently; minimizing the wait time for
pages to completely load. This feature also prevents control
packets from getting blocked by data packets, like what usually
happens in TCP; thereby improving data control.
TCP’s 3-way handshake initiation, SCTP uses a 4-way handshake
that allocates resources near the end of the entire handshake

SCTP
Data in TCP comes in packets. Packets have a specific size and a
long stream would be divided to fit while short ones are spliced
together. This means that message framing must be provided at
the application layer to fully identify separate messages. SCTP
implements message framing and each message would always
have the same size when it comes out as it came in.
Summary:
1. SCTP is better at multi-homing than TCP
2. SCTP has multi-streaming while TCP doesn’t
3. SCTP has initiation protection while TCP doesn’t
4. SCTP has message framing while TCP doesn’t
5. Ordered delivery is optional with SCTP but not with TCP

UNit-4 Transport Layer and its protocols.pdf

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UNit-4 Transport Layer and its protocols.pdf