Module 2

Chapter 3

Transport Layer

Computer Networking: A Top Down

Approach
6th edition
Jim Kurose, Keith Ross
Addison-Wesley
Goals:
understand principles learn about Internet
behind transport layer transport layer
services: protocols:
 multiplexing,  UDP: connectionless
demultiplexing transport
 reliable data transfer  TCP: connection-
 flow control oriented reliable
transport
 congestion control
 TCP congestion control

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 Chapter 3 outline

3.1 Transport-layer 3.5 Connection-oriented

services transport: TCP
3.2 Multiplexing and  segment structure
demultiplexing  reliable data transfer
3.3 Connectionless  flow control
transport: UDP  connection management

3.4 Principles of reliable 3.6 Principles of congestion

data transfer control
3.7 TCP congestion control

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3.1 Introduction and Transport-Layer Services
•A transport-layer protocol provides for logical
communication between application processes running on
different hosts.

•As shown in Figure 3.1, transport-layer protocols are

implemented in the end systems but not in network routers.

•On the sending side, the transport layer converts the

application-layer messages it receives from a sending
application process into transport-layer packets, known as
transport-layer segments.
•Note that network routers act only on the network-layer
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• A Transport-layer protocol provides logical communication between
processes running on different hosts, a Network-layer protocol
provides logical communication between hosts.

Household
Analogy:
12 kids in Ann’s house sending letters to 12 kids in Bill’s house:
• Hosts =
Houses
• Processes =
Kids/cousins
• Application messages =
Letters in envelopes
• Transport-layer protocol =
Ann and Bill who mux/demux to in-house siblings
• Network-layer protocol =
Postal service

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 Internet transport-layer protocols
application
• reliable, in-order delivery transport
network
data link
(TCP) physical
network
– congestion control network data link

lo
data link physical

gi
physical

ca
– flow control

l
en
– connection setup

d-
en
d
tra
ns
po
• unreliable, unordered

rt
network
delivery: UDP data link
physical
application
transport
network
data link network
physical data link
physical

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 Chapter 3 outline

3.1 Transport-layer 3.5 connection-oriented

services transport: TCP
3.2 Multiplexing and  segment structure
demultiplexing  reliable data transfer
3.3 Connectionless  flow control
transport: UDP  connection management

3.4 Principles of reliable 3.6 principles of congestion

data transfer control
3.7 TCP congestion control

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3.2 Multiplexing and Demultiplexing

•At the destination host, the transport layer receives segments from
the network layer just below. The transport layer has the
responsibility of delivering the data in these segments to the
appropriate application process running in the host.

•The job of gathering data chunks at the source host from different
sockets, encapsulating each data chunk with header information to
create segments, and passing the segments to the network layer is
called multiplexing.

•At the receiving end, the transport layer examines these fields to
identify the receiving socket and then delivers the segment to
correct socket. This process is called demultiplexing.
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 Multiplexing/demultiplexing
multiplexing at sender:
handle data from multiple demultiplexing at receiver:
sockets, add transport header use header info to deliver
(later used for demultiplexing) received segments to correct
socket

application

application P1 P2 application socket

P3 transport P4
process
transport network transport
network link network
link physical link
physical physical

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• A process (as part of a network application) can have one or
more sockets, doors through which data passes from the network
to the process and through which data passes from the process to
the network.

• As shown in above Figure, the transport layer in the receiving

host does not actually deliver data directly to a process, but
instead to an intermediary socket.

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 How demultiplexing works

Figure 3.3 Source and destination port-number fields in a transport-layer segment

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• Each port number is a 16-bit number, ranging from 0 to 65535.
• The port numbers ranging from 0 to 1023 are called well-known port
numbers and are restricted, which means that they are reserved for
use by well-known application protocols.
• such as HTTP (which uses port number 80) and FTP (which uses port
number 21).
• The list of well-known port numbers is given in RFC 1700 and is
updated at [Link] [RFC 3232].

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• Connectionless Multiplexing and Demultiplexing

• Java program running in a host can create a UDP socket with the line
– DatagramSocket mySocket = new DatagramSocket();
• Python program running in a host can create a UDP socket with the
line
– clientSocket = socket(socket.AF_INET, socket.SOCK_DGRAM)
• When a UDP socket is created in this manner, the transport layer
automatically assigns a port number to the socket.
• In particular, the transport layer assigns a port number in the range
1024 to 65535 that is currently not being used by any other UDP port
in the host.
– DatagramSocket mySocket = new DatagramSocket(19157); // Java
– [Link]((‘’, 19157)) //Python
In this case, the application assigns a specific port number-namely
19157 to the UDP socket
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What is the purpose of the source port number?

•As shown in Figure 3.4, in the A-to-B segment the source port number
serves as part of a “return address”—when B wants to send a segment
back to A, the destination port in the B-to-A segment will take its value
from the source port value of the A-to-B segment.
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Connection-Oriented Multiplexing and Demultiplexing

•One difference between a TCP socket and a UDP socket is that a TCP
socket is identified by a four-tuple:
•(source IP address, source port number, destination IP address,
destination port number).
•Thus, when a TCP segment arrives from the network to a host, the host
uses all four values to direct (demultiplex) the segment to the appropriate
socket.

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• The TCP server application has a “welcoming socket,” that waits for
connection establishment requests from TCP clients (see Figure 2.29)
on port number 12000.
• The TCP client creates a socket and sends a connection establishment
request segment with the lines:
clientSocket = socket(AF_INET, SOCK_STREAM)
[Link]((serverName,12000))

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• The transport layer at the server notes the following four values in the
connection-request segment:
(1) the source port number in the segment,
(2) the IP address of the source host,
(3) the destination port number in the segment, and
(4) its own IP address.

This four values are illustrated in this diagram

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• Server B will still be able to correctly demultiplex the two
connections having the same source port number, since the two
connections have different source IP addresses.

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 Chapter 3 outline

3.1 transport-layer 3.5 connection-oriented

services transport: TCP
3.2 multiplexing and  segment structure
demultiplexing  reliable data transfer
3.3 connectionless  flow control
transport: UDP  connection management

3.4 principles of reliable 3.6 principles of congestion

data transfer control
3.7 TCP congestion control

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3.3 Connectionless Transport: UDP
•UDP takes messages from the application process, attaches source and
destination port number fields for the multiplexing/de multiplexing
service, adds two other small fields, and passes the resulting segment to
the network layer.
•The network layer encapsulates the transport-layer segment into an IP
datagram and then makes a best-effort attempt to deliver the segment to
the receiving host.
•If the segment arrives at the receiving host, UDP uses the destination
port number to deliver the segment’s data to the correct application
process.

•Note that with UDP there is no handshaking between sending and

receiving transport-layer entities before sending a segment. For this
reason, UDP is said to be connectionless.

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• Why an application developer would ever choose to build an
application over UDP rather than over TCP. Isn’t TCP always
preferable, since TCP provides a reliable data transfer service, while
UDP does not?
• Many applications are better suited for UDP for the following
reasons:
– UDP immediately pass segment, where as TCP not
– No connection establishment
– No connection state. (like- buffers, congestion-control parameters, and sequence
and acknowledgment number parameters)
– Small packet header overhead: The TCP segment has 20 bytes of header
overhead in every segment, whereas UDP has only 8 bytes of overhead.

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3.3.1 UDP Segment Structure

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• The UDP header has only four fields, each consisting of two bytes

1) Source port number field

2) Destination port number field
3) Checksum is used by the receiving host to check whether errors have
been introduced into the segment.
4) The length field specifies the length of the UDP segment, including
the header, in bytes

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3.3.2 UDP Checksum
•The UDP checksum provides for error detection. That is, the checksum
is used to determine whether bits within the UDP segment have been
altered (for example, by noise in the links or while stored in a router) as
it moved from source to destination.

•UDP at the sender side performs the 1s complement of the sum of all
the 16-bit words in the segment, with any overflow encountered during
the sum being wrapped.
Example, suppose that we have the following three 16-bit words:
–0 1 1 0 0 1 1 0 0 1 1 0 0 0 0 0
–0 1 0 1 0 1 0 1 0 0 1 1 0 1 0 1
–1 0 0 0 1 1 1 1 0 0 0 0 1 1 0 0
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• The sum of first two of these 16-bit words is
0110011001100000
0101010101010101
-----------------------------------
1011101110110101

• Adding the third word to the above sum gives

1011101110110101
1000111100001100
-----------------------------------
0100101011000010
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• the 1s complement of the sum 0100101011000010 is
1011010100111101, which becomes the checksum.

• At the receiver, all four 16-bit words are added, including the
checksum.
• If no errors are introduced into the packet, then clearly the sum
at the receiver will be 1111111111111111.
• If one of the bits is a 0, then we know that errors have been
introduced into the packet.

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 Chapter 3 outline

3.1 transport-layer 3.5 connection-oriented

services transport: TCP
3.2 multiplexing and  segment structure
demultiplexing  reliable data transfer
3.3 connectionless  flow control
transport: UDP  connection management

3.4 principles of reliable 3.6 principles of congestion

data transfer control
3.7 TCP congestion control

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3.4 Principles of Reliable Data Transfer

•Figure 3.8 illustrates the framework for our study of reliable

data transfer.
•The service abstraction provided by the upper-layer entities is
that of a reliable channel through which data can be
transferred.
•With a reliable channel, no transferred data bits are corrupted
(flipped from 0 to 1, or vice versa) or lost, and all are
delivered in the order in which they were sent.
•This is precisely the service model offered by TCP to the
Internet applications that invoke it.

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 Reliable data transfer: getting started

rdt_send(): called from above, deliver_data(): called by

(e.g., by app.). Passed data to rdt to deliver data to upper
deliver to receiver upper layer

send receive
side side

udt_send(): called by rdt, rdt_rcv(): called when packet

to transfer packet over arrives on rcv-side of channel
unreliable channel to receiver
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• Figure 3.8(b) illustrates the interfaces for our data transfer protocol.

• The sending side of the data transfer protocol will be invoked from
above by a call to rdt_send().
• It will pass the data to be delivered to the upper layer at the receiving
side.

• On the receiving side, rdt_rcv() will be called when a packet arrives

from the receiving side of the channel.
• When the rdt protocol wants to deliver data to the upper layer, it will
do so by calling deliver_data().

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3.4.1 Building a Reliable Data Transfer Protocol

•Reliable Data Transfer on a perfectly reliable channel: rdt1.0

•The finite-state machine (FSM) definitions for the rdt1.0 sender and
receiver are shown in Figure 3.9.
•The FSM in Figure 3.9(a) defines the operation of the sender, while the
FSM in Figure 3.9(b) defines the operation of the receiver.

Figure 3.9 rdt1.0 – A protocol for a completely reliable channel

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 Figure 3.9 rdt1.0 – A protocol for a completely reliable channel

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• The event causing the transition is shown above the horizontal line
labeling the transition, and the actions taken when the event occurs
are shown below the horizontal line.
• When no action is taken on an event, or no event occurs and an
action is taken, we’ll use Λ the symbol below or above the
horizontal.

• The sending side of rdt simply accepts data from the upper layer via
the rdt_send(data) event, creates a packet containing the data (via the
action make_pkt(data)) and sends the packet into the channel.
• On the receiving side of rdt, receives a packet from the underlying
channel via the rdt_rcv(packet) event, removes the data from the
packet (via the action extract (packet, data)) and passes the data up to
the upper layer (via the action deliver_data(data)).

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• Reliable Data Transfer over a Channel with Bit Errors: rdt2.0

In a computer network setting, reliable data transfer protocols based

retransmission are known as ARQ (Automatic Repeat reQuest)
protocols.
Three additional protocol capabilities are required in ARQ
protocols to handle the presence of bit errors:
• Error detection
• Receiver feedback
• Retransmission

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02/26/24
02/26/24
 Chapter 3 outline

3.1 transport-layer services 3.5 connection-oriented

3.2 multiplexing and transport: TCP
demultiplexing  segment structure
 reliable data transfer
3.3 connectionless transport:
 flow control
UDP
 connection management
3.4 principles of reliable data
transfer 3.6 principles of congestion
control
3.7 TCP congestion control

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 TCP: Overview RFCs: 793,1122,1323, 2018, 2581

• point-to-point:  full duplex data:

– one sender, one receiver  bi-directional data flow in
• reliable, in-order byte same connection
 MSS: maximum segment
steam:
size
– no “message boundaries”
 connection-oriented:
• pipelined:
 handshaking (exchange of
– TCP congestion and flow
control msgs) inits sender,
control set window size receiver state before data
exchange
 flow controlled:
 sender will not overwhelm
receiver
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 3.5.2 TCP segment structure
32 bits
URG: urgent data counting
(generally not used) source port # dest port #
by bytes
sequence number of data
ACK: ACK #
valid acknowledgement number (not segments!)
head not
PSH: push data now len used
UAP R S F receive window
(generally not used) # bytes
checksum Urg data pointer
rcvr willing
RST, SYN, FIN: to accept
options (variable length)
connection estab
(setup, teardown
commands)
application
Internet data
checksum (variable length)
(as in UDP)

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• Sequence Numbers and Acknowledgment Numbers

The acknowledgment number that Host A puts in its

segment is the sequence number of the next byte Host A
is expecting from Host B.

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• Host A rcvd all bytes 0 to 535 (0-999) from B
• A is waiting for 536 byte to rcv
• Host A puts ack 536

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 3.5.3 Round-Trip Time Estimation and Timeout

What is the length of the timeout interval?

Soln:
The timeout should be larger than the connection’s round-trip time
(RTT), that is, the time from when a segment is sent until it is
acknowledged. Otherwise, unnecessary retransmissions would be
sent.

• Estimating the Round-Trip Time

0.125
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 Fig: RTT samples and RTT estimates

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3.5.4 Reliable Data Transfer

• TCP creates rdt service on top of IP’s unreliable

service
– pipelined segments
– cumulative acks
– single retransmission timer
• retransmissions triggered by:
– timeout events
– duplicate acks

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 TCP sender events:
data rcvd from app: timeout:
 create segment with seq #  retransmit segment that
 seq # is byte-stream caused timeout
number of first data byte  restart timer
in segment ack rcvd:
 start timer if not already  if ack acknowledges
running previously unacked
 think of timer as for segments
oldest unacked  update what is known
segment to be ACKed
 expiration interval:  start timer if there are
TimeOutInterval still unacked segments
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 TCP: retransmission scenarios
Host A Host B Host A Host B

SendBase=92
Seq=92, 8 bytes of data Seq=92, 8 bytes of data

Seq=100, 20 bytes of data

timeo

timeo
ACK=100
X
ut

ut
ACK=100
ACK=120

Seq=92, 8 bytes of data Seq=92, 8

SendBase=100 bytes of data
SendBase=120
ACK=100
ACK=120

SendBase=120

lost ACK scenario premature timeout

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 TCP: retransmission scenarios
Host A Host B

Seq=92, 8 bytes of data

Seq=100, 20 bytes of data

ACK=100
timeo

X
ut

ACK=120

Seq=120, 15 bytes of data

cumulative ACK
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 Fig: Retransmission due to a lost acknowledgment
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 Premature timeout

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 A cumulative acknowledgment avoids retransmission of the
first segment
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Doubling the Timeout Interval
Each time TCP retransmits, it sets the next timeout interval to twice the
previous value, rather than deriving it from the last EstimatedRTT

Fast Retransmit
Retransmitting the missing segment before that segment’s timer expires.

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3.5.5 Flow Control

• A speed-matching service.

• TCP provides a flow-control service to its applications to

eliminate the possibility of the sender overflowing the
receiver’s buffer.

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LastByteRead

LastByteRcvd

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• Because TCP is not permitted to overflow the allocated buffer, we
must have
LastByteRcvd – LastByteRead <= RcvBuffer

• The receive window, denoted rwnd is set to the amount of spare

room in the buffer:
rwnd = RcvBuffer – [LastByteRcvd – LastByteRead]

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3.5.6 TCP Connection Management

Let’s first take a look at how a TCP connection is established.

•Suppose a process running in one host (client) wants to initiate

a connection with another process in another host (server). The
client application process first informs the client TCP that it
wants to establish a connection to a process in the server.
•Step-1
•Step-2
•Step-3

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Life cycle of a TCP connection
• During the life of a TCP connection, the TCP protocol running in
each host makes transitions through various TCP states.
• Figure 3.41 illustrates a typical sequence of TCP states that are
visited by the client TCP.
• The client TCP begins in the CLOSED state.

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Principles of TCP Congestion Control

The Causes and the Costs of Congestion

Scenario 1: Two Senders, a Router with Infinite Buffers

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Scenario 2: Two Senders and a Router with Finite Buffers

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Network-Assisted Congestion-Control Example:

ATM ABR Congestion Control

Asynchronous transfer mode (ATM) available bit-rate (ABR)

ATM ABR—a protocol that takes a network-assisted approach toward

congestion control.

• explicit forward congestion indication (EFCI) bit.

02/26/24 • congestion indication (CI) bit
• no increase (NI) bit
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 Slow Start

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