21CS1403 – Computer Networks Unit 4

UNIT – IV : TRANSPORT LAYER

Introduction — Transport Layer Protocols — User Datagram Protocol —

Transmission Control Protocol — SCTP.

1. INTRODUCTION
 The transport layer is the fourth layer of the OSI model and is the core of the
Internet model.
 It responds to service requests from the session layer and issues service
requests to the network Layer.
 The transport layer provides transparent transfer of data between hosts.
 It provides end-to-end control and information transfer with the quality of
service needed by the application program.
 It is the first true end-to-end layer, implemented in all End Systems (ES).

TRANSPORT LAYER FUNCTIONS / SERVICES

 The transport layer is located between the network layer and the application
layer.
 The transport layer is responsible for providing services to the application
layer; it receives services from the network layer.
 The services that can be provided by the transport layer are
1. Process-to-Process Communication
2. Addressing : Port Numbers
3. Encapsulation and Decapsulation
4. Multiplexing and Demultiplexing
5. Flow Control
6. Error Control
7. Congestion Control

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Process-to-Process Communication
 The Transport Layer is responsible for delivering data to the appropriate
application process on the host computers.
 This involves multiplexing of data from different application processes,
i.e. forming data packets, and adding source and destination port
numbers in the header of each Transport Layer data packet.
 Together with the source and destination IP address, the port numbers
constitutes a network socket, i.e. an identification address of the
process-to-process communication.

Addressing: Port Numbers

 Ports are the essential ways to address multiple entities in the same
location.
 Using port addressing it is possible to use more than one network-based
application at the same time.
 Three types of Port numbers are used :
 Well-known ports - These are permanent port numbers. They range
between 0 to 1023.These port numbers are used by Server Process.
 Registered ports - The ports ranging from 1024 to 49,151 are not
assigned or controlled.
 Ephemeral ports (Dynamic Ports) – These are temporary port
numbers. They range between 49152–65535.These port numbers are
used by Client Process.

Encapsulation and Decapsulation

 To send a message from one process to another, the transport-layer
protocol encapsulates and decapsulates messages.
 Encapsulation happens at the sender site. The transport layer receives
the data and adds the transport-layer header.
 Decapsulation happens at the receiver site. When the message arrives at
the destination transport layer, the header is dropped and the transport
layer delivers the message to the process running at the application
layer.

Multiplexing and Demultiplexing

 Whenever an entity accepts items from more than one source, this is
referred to as multiplexing (many to one).
 Whenever an entity delivers items to more than one source, this is
referred to as demultiplexing (one to many).
 The transport layer at the source performs multiplexing
 The transport layer at the destination performs demultiplexing

Flow Control
 Flow Control is the process of managing the rate of data
transmission between two nodes to prevent a fast sender from
overwhelming a slow receiver.
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 It provides a mechanism for the receiver to control the transmission

speed, so that the receiving node is not overwhelmed with data from
transmitting node.

Error Control
 Error control at the transport layer is responsible for
1. Detecting and discarding corrupted packets.
2. Keeping track of lost and discarded packets and resending them.
3. Recognizing duplicate packets and discarding them.
4. Buffering out-of-order packets until the missing packets arrive.
 Error Control involves Error Detection and Error Correction

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 that
control the congestion and keep the load below the capacity.
 Congestion Control refers to techniques and mechanisms that can either
prevent congestion, before it happens, or remove congestion, after it has
happened
 Congestion control mechanisms are divided into two categories,
 Open loop - prevent the congestion before it happens.
 Closed loop - remove the congestion after it happens.
PORT NUMBERS
 A transport-layer protocol usually has several responsibilities.
 One is to create a process-to-process communication.
 Processes are programs that run on hosts. It could be either server or client.
 A process on the local host, called a client, needs services from a
process usually on the remote host, called a server.
 Processes are assigned a unique 16-bit port number on that host.
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 Port numbers provide end-to-end addresses at the transport layer

 They also provide multiplexing and demultiplexing at this layer.

ICANN (Internet Corporation for Assigned Names and Numbers) has divided
the port numbers into three ranges:
Well-known ports
Registered
Ephemeral ports (Dynamic Ports)

WELL-KNOWN PORTS
 These are permanent port numbers used by the servers.
 They range between 0 to 1023.
 This port number cannot be chosen randomly.
 These port numbers are universal port numbers for servers.
 Every client process knows the well-known port number of the
corresponding server process.
 For example, while the client process can use an ephemeral (temporary)
port number, 52,000 to identify itself, the server process must use the
well-known (permanent) port number 13.

Well Known Porta used with UDP

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EPHEMERAL PORTS (DYNAMIC PORTS)

 The client program defines itself with a port number, called the ephemeral
port number.
 The word ephemeral means “short-lived” and is used because the life of a
client is normally short.
 An ephemeral port number is recommended to be greater than 1023.
 These port number ranges from 49,152 to 65,535 .
 They are neither controlled nor registered. They can be used as temporary
or private port numbers.

REGISTERED PORTS
 The ports ranging from 1024 to 49,151 are not assigned or controlled.

TRANSPORT LAYER PROTOCOLS

 Three protocols are associated with the Transport layer.

 They are
(1) UDP –User Datagram Protocol
(2) TCP – Transmission Control Protocol
(3) SCTP - Stream Control Transmission Protocol

 Each protocol provides a different type of service and should be used

appropriately.

UDP - UDP is an unreliable connectionless transport-layer protocol used for its

simplicity and efficiency in applications where error control can be provided by
the application-layer process.

TCP - TCP is a reliable connection-oriented protocol that can be used in any

application where reliability is important.

SCTP - SCTP is a new transport-layer protocol designed to combine some features

of UDP and TCP in an effort to create a better protocol for multimedia
communication.

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USER DATAGRAM PROTOCOL (UDP)

 User Datagram Protocol (UDP) is a connectionless, unreliable transport
protocol.

 UDP adds process-to-process communication to best-effort service provided

by IP.
 UDP is a very simple protocol using a minimum of overhead.
 UDP is a simple demultiplexer, which allows multiple processes on each
host to communicate.
 UDP does not provide flow control , reliable or ordered delivery.
 UDP can be used to send small message where reliability is not expected.
 Sending a small message using UDP takes much less interaction between
the sender and receiver.
 UDP allow processes to indirectly identify each other using an abstract
locator called port or mailbox

UDP PORTS
 Processes (server/client) are identified by an abstract locator known as port.
 Server accepts message at well known port.
 Some well-known UDP ports are 7–Echo, 53–DNS, 111–RPC, 161–SNMP, etc.
 < port, host > pair is used as key for demultiplexing.
 Ports are implemented as a message queue.
 When a message arrives, UDP appends it to end of the queue.
 When queue is full, the message is discarded.
 When a message is read, it is removed from the queue.
 When an application process wants to receive a message, one is removed
from the front of the queue.
 If the queue is empty, the process blocks until a message becomes available.

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UDP DATAGRAM (PACKET) FORMAT

 UDP packets are known as user datagrams .
 These user datagrams, have a fixed-size header of 8 bytes made of four
fields, each of 2 bytes (16 bits).

Source Port Number

 Port number used by process on source host with 16 bits long.
 If the source host is client (sending request) then the port number is an
temporary one requested by the process and chosen by UDP.
 If the source is server (sending response) then it is well known port
number.
Destination Port Number
 Port number used by process on Destination host with 16 bits long.
 If the destination host is the server (a client sending request)
then the port number is a well known port number.
 If the destination host is client (a server sending response) then
port number is an temporary one copied by server from the
request packet.
Length
 This field denotes the total length of the UDP Packet (Header plus data)
 The total length of any UDP datagram can be from 0 to 65,535 bytes.

Checksum
 UDP computes its checksum over the UDP header, the contents of the
message body, and something called the pseudoheader.
 The pseudoheader consists of three fields from the IP header—protocol
number, source IP address, destination IP address plus the UDP length
field.
Data
 Data field defines tha actual payload to be transmitted.
 Its size is variable.

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UDP SERVICES
Process-to-Process Communication
 UDP provides process-to-process communication using socket addresses,
a combination of IP addresses and port numbers.

Connectionless Services
 UDP provides a connectionless service.
 There is no connection establishment and no connection termination .
 Each user datagram sent by UDP is an independent datagram.
 There is no relationship between the different user datagrams even if they
are coming from the same source process and going to the same destination
program.
 The user datagrams are not numbered.
 Each user datagram can travel on a different path.

Flow Control
 UDP is a very simple protocol.
 There is no flow control, and hence no window mechanism.
 The receiver may overflow with incoming messages.
 The lack of flow control means that the process using UDP should
provide for this service, if needed.

Error Control
 There is no error control mechanism in UDP except for the checksum.
 This means that the sender does not know if a message has been lost or
duplicated.
 When the receiver detects an error through the checksum, the user
datagram is silently discarded.
 The lack of error control means that the process using UDP should provide
for this service, if needed.

Checksum
 UDP checksum calculation includes three sections: a pseudoheader, the
UDP header, and the data coming from the application layer.
 The pseudoheader is the part of the header in which the user datagram is
to be encapsulated with some fields filled with 0s.

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Pseudoheader for checksum Calculation

Figure shows the checksum calculation for a very small user datagram with only 7
bytes of data. Because the number of bytes of data is odd, padding is added for
checksum calculation. The pseudoheader as well as the padding will be dropped
when the user datagram is delivered to IP.

APPLICATIONS OF UDP
 UDP is used for management processes such as SNMP.
 UDP is used for route updating protocols such as RIP.
 UDP is a suitable transport protocol for multicasting. Multicasting
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capability is embedded in the UDP software

 UDP is suitable for a process with internal flow and error control
mechanisms such as Trivial File Transfer Protocol (TFTP).
 UDP is suitable for a process that requires simple request-response
communication with little concern for flow and error control.
 UDP is normally used for interactive real-time applications that cannot
tolerate uneven delay between sections of a received message.

PROBLEM 1:
SOLVED PROBLEMS

The following is the content of a UDP header in hexadecimal format.

CB84000D001C001C
(1) What is the source port number?
(2) What is the destination port number?
(3) What is the total length of the user datagram?
(4) What is the length of the data?
(5) Is the packet directed from a client to a server or vice versa?
(6) What is the client process?

Solution
(1) The source port number is the first four hexadecimal digits (CB84), which
means that the source port number is 52100.
(2) The destination port number is the second four hexadecimal digits (000D),
which means that the destination port number is 13.
(3) The third four hexadecimal digits (001C) define the length of the whole UDP
packet as 28 bytes.
(4) The length of the data is the length of the whole packet minus the length of
the header, or 28 − 8 = 20 bytes.
(5) Since the destination port number is 13 (well-known port), the packet is
from the client to the server.
(6) The client process is the Daytime.

PROBLEM 2 : (CHECKSUM)
What value is sent for the checksum in each one of the following hypothetical
situations?
(1) The sender decides not to include the checksum.
(2) The sender decides to include the checksum, but the value of the sum is all
1s.
(3) The sender decides to include the checksum, but the value of the sum is all
0s.
Solution
(1) The value sent for the checksum field is all 0s to show that the checksum
is not calculated.

(2) When the sender complements the sum, the result is all 0s; the sender
complements the result again before sending. The value sent for the
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checksum is all 1s. The second complement operation is needed to avoid

confusion with the case in part a.

(3) This situation never happens because it implies that the value of every term
included in the calculation of the sum is all 0s, which is impossible; some
fields in the pseudoheader have nonzero values.

TRANSMISSION CONTROL PROTOCOL (TCP)

TCP is a reliable, connection-oriented, byte-stream protocol.
TCP guarantees the reliable, in-order delivery of a stream of bytes. It is a
full-duplex protocol, meaning that each TCP connection supports a pair of
byte streams, one flowing in each direction.
TCP includes a flow-control mechanism for each of these byte streams that
allow the receiver to limit how much data the sender can transmit at a
given time.
TCP supports a demultiplexing mechanism that allows multiple application
programs on any given host to simultaneously carry on a conversation with
their peers.
TCP also implements congestion-control mechanism. The idea of this
mechanism is to prevent sender from overloading the network.
Flow control is an end to end issue, whereas congestion control is
concerned with how host and network interact.

TCP SERVICES
Process-to-Process Communication
 TCP provides process-to-process communication using port numbers.
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 bytes across the
Internet.
 The sending process produces (writes to) the stream and the
receiving process consumes (reads from) it.

Full-Duplex Communication
 TCP offers full-duplex service, where data can flow in both directions at the
same time.

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 Each TCP endpoint then has its own sending and receiving buffer, and
segments move in both directions.

Multiplexing and Demultiplexing

TCP performs multiplexing at the sender and demultiplexing at the receiver.

Connection-Oriented Service
 TCP is a connection-oriented protocol.

 A connection needs to be established for each pair of processes.

 When a process at site A wants to send to and receive data
from another process at site B, the following three phases
occur:
1. The two TCP’s establish a logical connection between them.
2. Data are exchanged in both directions.
3. The connection is terminated.

Reliable Service
 TCP is a reliable transport protocol.
 It uses an acknowledgment mechanism to check the safe and sound arrival
of data.

TCP SEGMENT
 A packet in TCP is called a segment.
 Data unit exchanged between TCP peers are called segments.
 A TCP segment encapsulates the data received from the application layer.
 The TCP segment is encapsulated in an IP datagram, which in turn is
encapsulated in a frame at the data-link layer.

 TCP is a byte-oriented protocol, which means that the sender writes bytes
into a TCP connection and the receiver reads bytes out of the TCP
connection.

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 TCP does not, itself, transmit individual bytes over the Internet.
 TCP on the source host buffers enough bytes from the sending process to
fill a reasonably sized packet and then sends this packet to its peer on the
destination host.
 TCP on the destination host then empties the contents of the packet into a
receive buffer, and the receiving process reads from this buffer at its
leisure.
 TCP connection supports byte streams flowing in both directions.
 The packets exchanged between TCP peers are called segments, since each
one carries a segment of the byte stream.
TCP PACKET FORMAT
 Each TCP segment contains the header plus the data.
 The segment consists of a header of 20 to 60 bytes, followed by data
from the application program.
 The header is 20 bytes if there are no options and up to 60 bytes if
it contains options.

SrcPort and DstPort―port number of source and destination process.

SequenceNum―contains sequence number, i.e. first byte of data segment.
Acknowledgment― byte number of segment, the receiver expects next.
HdrLen―Length of TCP header as 4-byte words.
Flags― contains six control bits known as flags.
 URG — segment contains urgent data.
 ACK — value of acknowledgment field is valid.
 PUSH — sender has invoked the push operation.
 RESET — receiver wants to abort the connection.
 SYN — synchronize sequence numbers during connection establishment.
 FIN — terminates the TCP connection.
Advertised Window―defines receiver’s window size and acts as flow control.
Checksum―It is computed over TCP header, Data, and pseudo header containing
IP fields (Length, SourceAddr & DestinationAddr).
UrgPtr ― used when the segment contains urgent data. It defines a value
that must be added to the sequence number.
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Options - There can be up to 40 bytes of optional information in the TCP header.

TCP CONNECTION MANAGEMENT
 TCP is connection-oriented.
 A connection-oriented transport protocol establishes a logical path
between the source and destination.
 All of the segments belonging to a message are then sent over this logical
path.
 In TCP, connection-oriented transmission requires three
phases: Connection Establishment, Data Transfer and
Connection Termination.

Connection Establishment
 While opening a TCP connection the two nodes(client and server) want to
agree on a set of parameters.
 The parameters are the starting sequence numbers that is to be used
for their respective byte streams.
 Connection establishment in TCP is a three-way handshaking.

1. Client sends a SYN segment to the server containing its initial sequence
number (Flags = SYN, SequenceNum = x)
2. Server responds with a segment that acknowledges client’s segment and
specifies its initial sequence number (Flags = SYN + ACK, ACK = x + 1
SequenceNum = y).
3. Finally, client responds with a segment that acknowledges server’s sequence
number (Flags = ACK, ACK = y + 1).
 The reason that each side acknowledges a sequence number that is one
larger than the one sent is that the Acknowledgment field actually
identifies the “next sequence number expected,”
 A timer is scheduled for each of the first two segments, and if the
expected response is not received, the segment is retransmitted.

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Data Transfer
 After connection is established, bidirectional data transfer can take
place.
 The client and server can send data and acknowledgments in both
directions.

 The data traveling in the same direction as an acknowledgment are

carried on the same segment.
 The acknowledgment is piggybacked with the data.

Connection Termination
 Connection termination or teardown can be done in two ways :
Three-way Close and Half-Close

Three-way Close—Both client and server close simultaneously.

 Client sends a FIN segment.

 The FIN segment can include
last chunk of data.
 Server responds with FIN +
ACK segment to inform its
closing.
 Finally, client sends an
ACK segment

Half-Close—Client stops sending but receives data.

 Client half-closes the
connection by sending a
FIN segment.
 Server sends an ACK segment.
 Data transfer from client to
the server stops.
 After sending all data, server
sends FIN segment to client,
which is acknowledged by
the client.

STATE TRANSITION DIAGRAM

 To keep track of all the different events happening during connection
establishment, connection termination, and data transfer, TCP is specified
as the finite state machine (FSM).
 The transition from one state to another is shown using directed lines.

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 States involved in opening and closing a connection is shown above and

below ESTABLISHED state respectively.
 States Involved in TCP :

Opening a TCP Connection

1. Server invokes a passive open on TCP, which causes TCP to move to LISTEN
state
2. Client does an active open, which causes its TCP to send a SYN segment to
the server and move to SYN_SENT state.
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3. When SYN segment arrives at the server, it moves to SYN_RCVD state and
responds with a SYN + ACK segment.
4. Arrival of SYN + ACK segment causes the client to move to ESTABLISHED
state and sends an ACK to the server.
5. When ACK arrives, the server finally moves to ESTABLISHED state.

Closing a TCP Connection

 Client / Server can independently close its half of the connection or
simultaneously.
Transitions from ESTABLISHED to CLOSED state are:
One side closes:
ESTABLISHED → FIN_WAIT_1 → FIN_WAIT_2 → TIME_WAIT →
CLOSED
Other side closes:
ESTABLISHED → CLOSE_WAIT → LAST_ACK → CLOSED
Simultaneous close:
ESTABLISHED → FIN_WAIT_1 → CLOSING → TIME_WAIT → CLOSED

SOLVED EXAMPLE
Suppose a TCP connection is transferring a file of 5000 bytes. The first byte is
numbered 10001.What are the sequence numbers for each segment if data are
sent in five segments, each carrying 1000 bytes?

Solution
The following shows the sequence number for each segment:

TCP FLOW CONTROL

TCP flow control balances the rate a producer creates data with the rate a
consumer can use the data.
TCP uses a variant of sliding window known as adaptive flow control that:
 guarantees reliable delivery of data
 ensures ordered delivery of data
 enforces flow control at the sender
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Advertised Window
To achieve flow control, TCP uses the AdvertisedWindow field.
Receiver advertises its window size to the sender using AdvertisedWindow
field.
Sender thus cannot have unacknowledged data greater than
AdvertisedWindow.

Send Buffer
 Sending TCP maintains send buffer which contains 3 segments
(1) acknowledged data
(2) unacknowledged data
(3) data to be transmitted.
 Send buffer maintains three pointers
(1) LastByteAcked, (2) LastByteSent, and (3) LastByteWritten
such that:
LastByteAcked ≤ LastByteSent ≤ LastByteWritten
 A byte can be sent only after being written and only a sent byte can be
acknowledged.

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 Bytes to the left of LastByteAcked are not kept as it had been acknowledged.
Receive Buffer
 Receiving TCP maintains receive buffer to hold data even if it arrives out-of-
order.
 Receive buffer maintains three pointers namely
(1) LastByteRead, (2) NextByteExpected, and (3)
LastByteRcvd such that:
LastByteRead ≤ NextByteExpected ≤ LastByteRcvd + 1
 A byte cannot be read until that byte and all preceding bytes have been
received.
 If data is received in order, then NextByteExpected = LastByteRcvd + 1
 Bytes to the left of LastByteRead are not buffered, since it is read by the
application.

Working of Flow Control in TCP

 Size of send and receive buffer is MaxSendBuffer and MaxRcvBuffer
respectively.
 Sending TCP prevents overflowing of send buffer by maintaining
LastByteWritten − LastByteAcked ≤ MaxSendBuffer
 Receiving TCP avoids overflowing its receive buffer by maintaining
LastByteRcvd − LastByteRead ≤ MaxRcvBuffer
 Receiver throttles the sender by having AdvertisedWindow based on free
space available for buffering.
AdvertisedWindow = MaxRcvBuffer − ((NextByteExpected − 1) –
LastByteRead)
 Sending TCP adheres to AdvertisedWindow by computing EffectiveWindow
that limits how much data it should send.
EffectiveWindow = AdvertisedWindow − (LastByteSent −
LastByteAcked)
 When data arrives, LastByteRcvd moves to its right and AdvertisedWindow
shrinks.
 Receiver acknowledges only, if preceding bytes have arrived.
 AdvertisedWindow expands when data is read by the
application. o If data is read as fast as it arrives then
AdvertisedWindow = MaxRcvBuffer
o If data is read slowly, it eventually leads to a AdvertisedWindow of size
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0.
 AdvertisedWindow field is designed to allow sender to keep the pipe
full.
TCP TRANSMISSION
 TCP has two mechanism to trigger the transmission of a segment.
 They are
o Silly Window Syndrome - Uses Maximum Segment Size (MSS)
o Nagle’s Algorithm - Uses Timeout
Silly Window Syndrome
 When either the sending application program creates data slowly or the
receiving application program consumes data slowly, or both, problems
arise.
 Any of these situations results in the sending of data in very small
segments, which reduces the efficiency of the operation.
 This problem is called the silly window syndrome.

 The sending TCP may create a silly window syndrome if it is serving an

application program that creates data slowly, for example, 1 byte at a time.
 The application program writes 1 byte at a time into the buffer of the sending
TCP.
 The result is a lot of 1-byte segments that are traveling through an internet.
 The solution is to prevent the sending TCP from sending the data byte by
byte.
 The sending TCP must be forced to wait and collect data to send in a larger
block.

Nagle’s Algorithm
 If there is data to send but is less than MSS, then we may want to wait some
amount of time before sending the available data

 If we wait too long, then it may delay the process.

 If we don’t wait long enough, it may end up sending small segments resulting
in Silly Window Syndrome.

 The solution is to introduce a timer and to transmit when the timer expires

 Nagle introduced an algorithm for solving this problem

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TCP CONGESTION CONTROL

 Congestion occurs if load (number of packets sent) is greater than capacity
of the network (number of packets a network can handle).
 When load is less than network capacity, throughput increases
proportionally.
 When load exceeds capacity, queues become full and the routers
discard some packets and throughput declines sharply.
 When too many packets are contending for the same link
o The queue overflows
o Packets get dropped
o Network is congested
 Network should provide a congestion control mechanism to deal with
such a situation.
 TCP maintains a variable called CongestionWindow for each connection.
 TCP Congestion Control mechanisms are:
 Additive Increase / Multiplicative Decrease (AIMD)
 Slow Start
 Fast Retransmit and Fast Recovery
Additive Increase / Multiplicative Decrease (AIMD)
 TCP source initializes CongestionWindow based on congestion level in the
network.

 Source increases CongestionWindow when level of congestion goes down and

o decreases the same when level of congestion goes up.

 TCP interprets timeouts as a sign of congestion and reduces the rate of

transmission.

 On timeout, source reduces its CongestionWindow by half, i.e.,

multiplicative decrease.
 For example, if CongestionWindow = 16 packets, after timeout it is 8.

 Value of CongestionWindow is never less than maximum segment size (MSS).

 When ACK arrives CongestionWindow is incremented marginally, i.e.,

additive increase.

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Increment = MSS ×(MSS/ CongestionWindow)

CongestionWindow += Increment

 For example, when ACK arrives for 1 packet, 2 packets are sent. When ACK
for both packets arrive, 3 packets are sent and so on.
 CongestionWindow increases and decreases throughout lifetime of the
connection.

 When CongestionWindow is plotted as a function of time, a saw-tooth

pattern results.

Slow Start
 Slow start is used to increase CongestionWindow exponentially from a cold
start.
 Source TCP initializes CongestionWindow to one packet.
 TCP doubles the number of packets sent every RTT on successful
transmission.
 When ACK arrives for first packet TCP adds 1 packet to CongestionWindow
and sends two packets.
 When two ACKs arrive, TCP increments CongestionWindow by 2 packets
and sends four packets and so on.
 Instead of sending entire permissible packets at once (bursty traffic),
packets are sent in a phased manner, i.e., slow start.
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 Initially TCP has no idea about congestion, henceforth it

increases CongestionWindow rapidly until there is a timeout.
 On timeout:
 CongestionThreshold= CongestionWindow/ 2
 CongestionWindow = 1
 Slow start is repeated until CongestionWindow reaches CongestionThreshold
and thereafter 1 packet per RTT.

 The congestion window trace will look like

Fast Retransmit And Fast Recovery

 TCP timeouts led to long periods of time during which the connection went
dead while waiting for a timer to expire.
 Fast retransmit is a heuristic approach that triggers retransmission of a
dropped packet sooner than the regular timeout mechanism. It does not
replace regular timeouts.
 When a packet arrives out of order, receiving TCP resends the same
acknowledgment (duplicate ACK) it sent last time.
 When three duplicate ACK arrives at the sender, it infers that corresponding
packet may be lost due to congestion and retransmits that packet. This is
called fast retransmit before regular timeout.
 When packet loss is detected using fast retransmit, the slow start phase is
replaced by additive increase, multiplicative decrease method. This is
known as fast recovery.
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 Instead of setting CongestionWindow to one packet, this method uses the

ACKs that are still in pipe to clock the sending of packets.
 Slow start is only used at the beginning of a connection and after regular
timeout. At other times, it follows a pure AIMD pattern.

 For example, packets 1 and 2 are received whereas packet 3 gets lost.
o Receiver sends a duplicate ACK for packet 2 when packet 4 arrives.
o Sender receives 3 duplicate ACKs after sending packet 6 retransmits
packet 3.
o When packet 3 is received, receiver sends cumulative ACK up to packet
6.
 The congestion window trace will look like

TCP VERSIONS
Three versions of TCP EXISTS :
Taho TCP, Reno TCP & New Reno TCP.

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Taho TCP:
The early TCP is known as Taho TCP.
Taho TCP uses only two different algorithms in their congestion policy:
Slow start and Congestion avoidance.

Reno TCP
 A newer version of TCP, called Reno TCP, added a new state to the
congestion- control FSM, called the fast-recovery state.
 This version uses the two signals of congestion : time-out and three
duplicate ACKs, differently.
 Normally TCP moves to the slow-start state
 On the other hand, if three duplicate ACKs arrive, TCP moves to the fast-
recovery state and remains there as long as more duplicate ACKs arrive.
 The fast-recovery state is a state somewhere between the slow-start
and the congestion-avoidance states.

NewReno TCP
 The later version of TCP, is called NewReno TCP.
 It makes an extra optimization on the Reno TCP.
 In this version, TCP checks to see if more than one segment is lost in the
current window when three duplicate ACKs arrive.
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 When TCP receives three duplicate ACKs, it retransmits the lost segment
until a new ACK arrives.
 NewReno TCP retransmits this segment to avoid receiving more and more
duplicate ACKs for it.

TCP CONGESTION AVOIDANCE

 Congestion avoidance mechanisms prevent congestion before it actually
occurs.
 These mechanisms predict when congestion is about to happen and then to
reduce the rate at which hosts send data just before packets start being
discarded.
 TCP creates loss of packets in order to determine bandwidth of the
connection.
 Routers help the end nodes by intimating when congestion is likely to occur.
 Congestion-avoidance mechanisms are:
o DEC bit - Destination Experiencing Congestion Bit
o RED - Random Early Detection

DEC Bit (Destination Experiencing Congestion Bit)

 The first mechanism developed for use on the Digital Network Architecture
(DNA).
 The idea is to evenly split the responsibility for congestion control between
the routers and the end nodes.
 Each router monitors the load it is experiencing and explicitly notifies the
end nodes when congestion is about to occur.
 This notification is implemented by setting a binary congestion bit in the
packets that flow through the router; hence the name DECbit.
 The destination host then copies this congestion bit into the ACK it sends
back to the source.
 The Source checks how many ACK has DEC bit set for previous window
packets.
 If less than 50% of ACK have DEC bit set, then source increases its
congestion window by 1 packet
 Otherwise, decreases the congestion window by 87.5%.
 Finally, the source adjusts its sending rate so as to avoid congestion.
 Increase by 1, decrease by 0.875 rule was based on AIMD for stabilization.
 A single congestion bit is added to the packet header.

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 Using a queue length of 1 as the trigger for setting the congestion bit.
 A router sets this bit in a packet if its average queue length is greater than
or equal to 1 at the time the packet arrives.
Computing average queue length at a router using DEC bit

 Average queue length is measured over a time interval that includes the
Average Queue Length = last busy + last idle cycle + current busy cycle.

 It calculates the average queue length by dividing the curve area with time
interval.

RED(Random Early Detection)

 The second mechanism of congestion avoidance is called as
Random Early Detection (RED).

 Each router is programmed to monitor its own queue length, and when it
detects that there is congestion, it notifies the source to adjust its congestion
window.

 RED differs from the DEC bit scheme by two ways:

 In DECbit, explicit notification about congestion is sent to source, whereas

RED implicitly notifies the source by dropping a few packets.

 DECbit may lead to tail drop policy, whereas RED drops packet based on drop
probability in a random manner.
 Drop each arriving packet with some drop probability whenever
the queue length exceeds some drop level. This idea is called early
random drop.

Computation of average queue length using RED

AvgLen = (1 − Weight) × AvgLen + (Weight × SampleLen)

where 0 < Weight < 1
SampleLen – is the length of the queue when a
sample measurement is made.
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 The queue length is measured every time a new packet arrives at the gateway.

 RED has two queue length thresholds that trigger certain activity:
MinThreshold and MaxThreshold

 When a packet arrives at a gateway it compares Avglen with these two

values according to the following rules.

STREAM CONTROL TRANSMISSION PROTOCOL (SCTP)

 Stream Control Transmission Protocol (SCTP) is a reliable, message-
oriented transport layer protocol.
 SCTP has mixed features of TCP and UDP.
 SCTP maintains the message boundaries and detects the lost data,
duplicate data as well as out-of-order data.
 SCTP provides the Congestion control as well as Flow control.
 SCTP is especially designed for internet applications as well as
multimedia communication.
SCTP SERVICES
Process-to-Process Communication
 SCTP provides process-to-process communication.
Multiple Streams
 SCTP allows multistream service in each connection, which is called
association in SCTP terminology.
 If one of the streams is blocked, the other streams can still deliver their
data.

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Multihoming
 An SCTP association supports multihoming service.
 The sending and receiving host can define multiple IP addresses in each end
for an association.
 In this fault-tolerant approach, when one path fails, another interface can be
used for data delivery without interruption.

Full-Duplex Communication
 SCTP offers full-duplex service, where data can flow in both directions at
the same time. Each SCTP then has a sending and receiving buffer and
packets are sent in both directions.

Connection-Oriented Service
 SCTP is a connection-oriented protocol.
 In SCTP, a connection is called an association.
 If a client wants to send and receive message from server , the steps are :
o Step1: The two SCTPs establish the connection with each other.
o Step2: Once the connection is established, the data gets
exchanged in both the directions.
o Step3: Finally, the association is terminated.

Reliable Service
 SCTP is a reliable transport protocol.
 It uses an acknowledgment mechanism to check the safe and sound arrival of
data.
SCTP Features
The following shows the general features of SCTP.

Transmission Sequence Number (TSN)

 The unit of data in SCTP is a data chunk, which may or may not have a one-to-
one relationship with the message coming from the process because of
fragmentation.
 Data transfer in SCTP is controlled by numbering the data chunks.
 SCTP uses a transmission sequence number (TSN) to number the data
chunks.
 TSN in its header.
Stream Identifier (SI)
 In SCTP, there may be several streams in each association. Each stream in
SCTP needs to be identified using a stream identifier (SI).

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 Each data chunk must carry the SI in its header so that when it arrives at the
destination, it can be properly placed in its stream. The SI is a 16-bit number
starting from 0.
Stream Sequence Number (SSN)
 When a data chunk arrives at the destination SCTP, it is delivered to the
appropriate stream and in the proper order. This means that, in addition to an
SI, SCTP defines each data chunk in each stream with a stream sequence
number (SSN).

SCTP PACKET FORMAT

An SCTP packet has a mandatory general header and a set of blocks called chunks.

General Header
 The general header (packet header) defines the end points of each association
to which the packet belongs
 It guarantees that the packet belongs to a particular association
 It also preserves the integrity of the contents of the packet including the header
itself.
 There are four fields in the general header:
o Source port
This field identifies the sending port.
Destination port
This field identifies the receiving port that hosts use to route the
packet to the appropriate endpoint/application.
Verification tag
A 32-bit random value created during initialization to distinguish stale
packets from a previous connection.
Checksum
The next field is a checksum. The size of the checksum is 32 bits.
SCTP uses CRC-32 Checksum.

Chunks
 Control information or user data are carried in chunks.
 Chunks have a common layout.
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 The first three fields are common to all chunks; the information field
depends on the type of chunk.

The type field can define up to 256 types of chunks. Only a few have been
defined so far; the rest are reserved for future use.
 The flag field defines special flags that a particular chunk may need.
 The length field defines the total size of the chunk, in bytes, including the
type, flag, and length fields.
Types of Chunks
 An SCTP association may send many packets, a packet may contain
several chunks, and chunks may belong to different streams.
 SCTP defines two types of chunks - Control chunks and Data chunks.
 A control chunk controls and maintains the association.
 A data chunk carries user data.

SCTP ASSOCIATION
 SCTP is a connection-oriented protocol.
 A connection in SCTP is called an association to emphasize multihoming.
 SCTP Associations consists of three phases:
o Association Establishment
o Data Transfer
o Association Termination

Association Establishment
 Association establishment in SCTP requires a four-way handshake.
 In this procedure, a client process wants to establish an association with a
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server process using SCTP as the transport-layer protocol.

 The SCTP server needs to be prepared to receive any association (passive
open).
 Association establishment, however, is initiated by the client (active open).

 The client sends the first packet, which contains an INIT chunk.

 The server sends the second packet, which contains an INIT ACK chunk.
The INIT ACK also sends a cookie that defines the state of the server at this
moment.
 The client sends the third packet, which includes a COOKIE ECHO chunk.
This is a very simple chunk that echoes, without change, the cookie sent by
the server. SCTP allows the inclusion of data chunks in this packet.
 The server sends the fourth packet, which includes the COOKIE ACK
chunk that acknowledges the receipt of the COOKIE ECHO chunk. SCTP
allows the inclusion of data chunks with this packet.

Data Transfer
 The whole purpose of an association is to transfer data between two ends.
 After the association is established, bidirectional data transfer can take
place.
 The client and the server can both send data.
 SCTP supports piggybacking.
 Types of SCTP data Transfer :
1. Multihoming Data Transfer
 Data transfer, by default, uses the primary address of the
destination.
 If the primary is not available, one of the alternative addresses is
used.
 This is called Multihoming Data Transfer.

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2. Multistream Delivery
 SCTP can support multiple streams, which means that the sender
process can define different streams and a message can belong to
one of these streams.
 Each stream is assigned a stream identifier (SI) which uniquely
defines that stream.
 SCTP supports two types of data delivery in each stream: ordered
(default) and unordered.

Association Termination
 In SCTP,either of the two parties involved in exchanging data (client or
server) can close the connection.
 SCTP does not allow a “half closed” association. If one end closes the
association, the other end must stop sending new data.
 If any data are left over in the queue of the recipient of the termination
request, they are sent and the association is closed.
 Association termination uses three packets.

SCTP FLOW CONTROL

 Flow control in SCTP is similar to that in TCP.
 Current SCTP implementations use a byte-oriented window for flow control.

Sender Side
 The sender has one buffer (queue) and three variables: curTSN, rwnd, and
inTransit.
 We assume each chunk is 100 bytes long. The buffer holds the chunks
produced by the process that either have been sent or are ready to be sent.
 The first variable, curTSN, refers to the next chunk to be sent.
 All chunks in the queue with a TSN less than this value have been sent, but
not acknowledged; they are outstanding.
 The second variable, rwnd, holds the last value advertised by the receiver (in
bytes).
 The third variable, inTransit, holds the number of bytes in transit, bytes sent
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but not yet acknowledged.

 The following figure shows the queue and variables at the sender site.

 A chunk pointed to by curTSN can be sent if the size of the data is less
than or equal to the quantity rwnd - inTransit.
 After sending the chunk, the value of curTSN is incremented by 1 and now
points to the next chunk to be sent.
 The value of inTransit is incremented by the size of the data in the
transmitted chunk.
 When a SACK is received, the chunks with a TSN less than or equal to the
cumulative TSN in the SACK are removed from the queue and discarded.
The sender does not have to worry about them anymore.
 The value of inTransit is reduced by the total size of the discarded chunks.
 The value of rwnd is updated with the value of the advertised window in the
SACK.

Receiver Side
 The receiver has one buffer (queue) and three variables.
 The queue holds the received data chunks that have not yet been read by
the process.
 The first variable holds the last TSN received, cumTSN.
 The second variable holds the available buffer size; winsize.
 The third variable holds the last accumulative acknowledgment, lastACK.
 The following figure shows the queue and variables at the receiver site.

 When the site receives a data chunk, it stores it at the end of the buffer
(queue) and subtracts the size of the chunk from winSize.
 The TSN number of the chunk is stored in the cumTSN variable.
 When the process reads a chunk, it removes it from the queue and adds
the size of the removed chunk to winSize (recycling).
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
When the receiver decides to send a SACK, it checks the value of lastAck; if
it is less than cumTSN, it sends a SACK with a cumulative TSN number
equal to the cumTSN.
 It also includes the value of winSize as the advertised window size.
SCTP ERROR CONTROL
 SCTP is a reliable transport layer protocol.
 It uses a SACK chunk to report the state of the receiver buffer to the sender.
 Each implementation uses a different set of entities and timers for the
receiver and sender sites.
Sender Side
 At the sender site, it needs two buffers (queues): a sending queue
and a retransmission queue.
 Three variables were used - rwnd, inTransit, and curTSN as described in
the previous section.
 The following figure shows a typical design.

The sending queue holds chunks 23 to 40.

The chunks 23 to 36 have already been sent, but not acknowledged;
they are outstanding chunks.
 The curTSN points to the next chunk to be sent (37).
 We assume that each chunk is 100 bytes, which means that 1400 bytes of
data (chunks 23 to 36) is in transit.
 The sender at this moment has a retransmission queue.
 When a packet is sent, a retransmission timer starts for that packet (all
data chunks in that packet).
 Some implementations use one single timer for the entire association, but
other implementations use one timer for each packet.
Receiver Side
 The receiver stores all chunks that have arrived in its queue including the
out-of- order ones. However, it leaves spaces for any missing chunks.
 It discards duplicate messages, but keeps track of them for reports to the
sender.

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 The following figure shows a typical design for the receiver site and the
state of the receiving queue at a particular point in time.

 The available window size is 1000 bytes.

 The last acknowledgment sent was for data chunk 20.
 Chunks 21 to 23 have been received in order.
 The first out-of-order block contains chunks 26 to 28.
 The second out-of-order block contains chunks 31 to 34.
 A variable holds the value of cumTSN.
 An array of variables keeps track of the beginning and the end of each
block that is out of order.
 An array of variables holds the duplicate chunks received.
 There is no need for storing duplicate chunks in the queue and they will be
discarded.

SCTP CONGESTION CONTROL

 SCTP is a transport-layer protocol with packets subject to congestion in the

network.
 The SCTP designers have used the same strategies for congestion control
as those used in TCP.
NOTE : REFER TCP CONGESTION CONTROL

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