Transport Layer

Introduction to transport layer services:

A transport-layer protocol provides for logical communication between

application processes running on different hosts. Logical communication means
that from an application’s perspective, it is as if the hosts running the processes
were directly connected. Application processes use the logical communication
provided by the transport layer to send messages to each other.

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. This is done by
(possibly) breaking the application messages into smaller chunks and adding a
transport-layer header to each chunk to create the transport-layer segment.

The transport layer then passes the segment to the network layer at the sending
end system, where the segment is encapsulated within a network-layer packet (a
datagram) and sent to the destination. On the receiving side, the network layer
extracts the transport-layer segment from the datagram and passes the segment
up to the transport layer. The transport layer then processes the received
segment, making the data in the segment available to the receiving application.

Transport Layer Functions:

Functions that you can encounter in the Transport Layer are:

 Error Handling
 Flow Control
 Multiplexing
 Connection Set-up and Release
 Congestion Handling
 Segmentation and Reassembly
 Addressing

By: Er. Saurav Raj Pant, IOE – TU

Services Provided to the Upper Layers

 The ultimate goal of the transport layer is to provide efficient, reliable, and
cost-effective data transmission service to its users, normally processes in
the application layer.
 The software and/or hardware within the transport layer that does the
work is called the transport entity.
 Provide logical communication between application processes running on
different hosts.

By: Er. Saurav Raj Pant, IOE – TU

  Transport protocols run in end systems:
o send side: breaks application messages into segments, passes to
network layer
o receive side: reassembles segments into messages, passes to
application layer
 More than one transport protocol available to apps
o Internet: TCP and UDP

Addressing:

 TCP communication between two remote hosts is done by means of 16-bit

port numbers known as Transport Service Access Points.
 Port numbers ranges from 0 to 65535.
0 – 1023 = System port (Well-known ports)
1024 – 49151 = Registered ports
49152 – 65535 = dynamically assigned port numbers (Private port no.)

Some well known ports:

HTTP = 80 HTTPs = 443

SMTP = 25 DNS = 53

FTP data = 20 FTP control = 21 etc.

Transport Layer Protocols:

Connectionless Transport: User Datagram Protocol (UDP)

UDP is defined in RFC 768. UDP is the simplest transport layer

protocol as it does not provide any error correction mechanism as a result it is
also called as unreliable protocol. In 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. As it is connectionless protocol there is
no guarantee that the data packet sent by the sender will reach to the specified
host. Also in UDP there is no mechanism for acknowledgement so packets which
By: Er. Saurav Raj Pant, IOE – TU
are lost will never be retransmitted. As a result UDP is faster than TCP and it is
used where the loss of packet is tolerable.

UDP segment structure:

UDP Header

8 Bytes

Fig: UDP segment structure

As shown in figure, UDP segment structure contains five fields. A 2-byte source
port number, 2-byte destination port number, length field to specify total length
of UDP segment structure, checksum for error detection and application data field
for message transmission.

In the UDP segment structure, the UDP header contains 4 fields each of 2-bytes
as shown in previous figure.

1. Source port number: it contains the port number of the source process
that is sending the message.

2. Destination port number: it contains the port number of the destination

process so that message can be sent to the correct process running in the
destination host.

3. Length: The length field specifies the number of bytes in the UDP segment
(header plus data).

By: Er. Saurav Raj Pant, IOE – TU

 4. Checksum: The checksum is used by the receiving host to check whether
errors have been introduced into the segment. 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.

5. Application Data Field: The application data occupies the data field of the
UDP segment. For a streaming audio application, audio samples fill the data
field.

Advantages of UDP:

• It uses small packet size with small header (8 bytes). This fewer bytes in the
overhead makes UDP protocol need less time in processing the packet and
need less memory.

• It does not require connection to be established and maintained.

• Also absence of acknowledgement field in UDP makes it faster as it need

not have to wait on ACK or need not have to hold data in memory until they
are ACKed.

• It uses checksum with all the packets for error detection.

Disadvantages of UDP:

• It is connectionless and unreliable transport protocol. There is no

windowing and no function to ensure data is received in the same order as
it was transmitted.

• It does not use any error control. Hence if UDP detects any error in the
received packet, it silently drops it.

• There is no congestion control. Hence large number of users transmitting

lots of data via UDP can cause congestion and no one can do anything
about it.

• There is no flow control and no acknowledgement for received data.

By: Er. Saurav Raj Pant, IOE – TU

 • No retransmission of data packet if the packet is lost.

Where to use UDP?

Even though TCP is the dominant protocol but due to its slow
performance, UDP is used where faster data transmission is required and little
data packet is tolerable. UDP is generally used in following situations:

1. For simple request and response communication where size of data is less
and hence there is lesser concern about flow and error control.

2. UDP is more preferable when using multicast or broadcast to transfer.

3. For transmission of real time packets normally in multimedia application i.e.

for audio and video streaming.

4. UDP is used for DNS querying. From host to DNS server the query is sent
through UDP protocol. If the query reaches to the DNS server a reply is
sent. If no reply is achieved by the host in a certain time interval, host
assumes that the query packet is lost. Hence, either it tries sending the
query to another name server, or it informs the invoking application that it
can’t get a reply. DNS will be very slow if it uses TCP instead of UDP.

5. UDP is used when the network layer is using RIP because it needs periodic
updation of routing table. Even if the updating packet is lost it does not
matter as RIP performs periodic updation so after some interval a new
updating packet will be sent and the previous loss of packet doesn’t make
any difference.

Connection Oriented Transport: Transmission Control Protocol(TCP)

Transmission Control Protocol (TCP) is a connection-oriented protocol that

computers use to communicate over the internet. It is one of the main protocols
in TCP/IP networks. TCP provides error-checking and guarantees delivery of data
and that packets will be delivered in the order they were sent. As a result it is
called as reliable protocol. TCP uses flow control and congestion control
By: Er. Saurav Raj Pant, IOE – TU
mechanism that ensures a sender is not overwhelming a receiver by sending too
many packets at once. Even if the packet is lost due to congestion the packet is
retransmitted by the sender.

TCP segment structure:

Fig: TCP segment structure

1. Source port:

A 16-bit field that contains the port number of the source process
that is sending the message.

2. Destination Port:

A 16-bit field that contains the port number of the destination

process that is receiving the message.

3. Sequence number:

A 32-bit number identifying the current position of the first data byte
in the segment within the entire byte stream for the TCP connection.

By: Er. Saurav Raj Pant, IOE – TU

4. Acknowledge number:

A 32-bit number identifying the next data byte the sender expects
from the receiver. Therefore, the number will be one greater than the most
recently received data byte.

5. Header length or offset:

A 4-bit field that specifies the total TCP header length in 32-bit
words. Without options the TCP header is always 20 bytes in length. The largest
TCP header may be is 60 bytes.

6. Reserved/Unused:

A 6-bit field is currently unused and is reserved for future use.

7. Control bits or flags:

The flag field contains 6 bits. The ACK bit is used to indicate that the
value carried in the acknowledgment field is valid; that is, the segment contains
an acknowledgment for a segment that has been successfully received. The RST,
SYN, and FIN bits are used for connection setup and teardown. PSH bit indicates
that the receiver should pass the data to the upper layer immediately. Finally, the
URG bit is used to indicate that there is data in this segment that the sending-side
upper-layer entity has marked as “urgent.”

8. Window:

A 16-bit integer is used by TCP for flow control in the form of a data
transmission window size. This number tells the sender how much data the
receiver is willing to accept.

9. Checksum:

A 16-bit checksum field for the error detection.

10. Urgent Pointer:

By: Er. Saurav Raj Pant, IOE – TU

 The 16-bit field tells the receiver when the last byte of urgent data in
the segment ends.

11. Options:

In order to provide additional functionality, several parameters may

be used between a TCP sender and receiver. Depending on the option(s) used, the
length of this field will vary in size. Options are generally used for various flow
control and congestion control techniques.

TCP Connection Establishment and Termination:

TCP Connection Establishment (3-Way Handshake):

The following scenario occurs when a TCP connection is established:

1. Client initiates the connection and send a "synchronize" (SYN) segment,

which tells server the client's initial sequence number for the data that the
client will send on the connection. Here, SYN flag is set.
2. The server must acknowledge (ACK) the client's SYN and the server must
also send its own SYN containing the initial sequence number for the data
that the server will send on the connection. The server sends its SYN and
the ACK of the client's SYN in a single segment. Here both ACK and SYN
flags are set
3. The client must acknowledge the server's SYN. Here, ACK flag is set.

The minimum number of packets required for this exchange is three; hence,
this is called TCP's three-way handshake.

Data Transmission:

 Once connection is established, continuous stream of data are sent by the

source in the form of TCP packets.
 When a packet of data is sent over TCP, the recipient must always
acknowledge what they received.

By: Er. Saurav Raj Pant, IOE – TU

  The source sends a packet with data and a sequence number. The second
computer acknowledges it by setting the ACK bit and increasing the
acknowledgement number by the length of the received data.
 Sequence number and acknowledgement numbers help the computers to
keep track of which data was successfully received, which data was lost,
and which data was accidentally sent twice

TCP Connection Termination

While it takes three segments to establish a connection, it takes four to terminate

a connection.

1. One application calls close first, and we say that this end performs the
active close. This end's TCP sends a FIN segment, which means it is
finished sending data.
2. The other end that receives the FIN performs the passive close. The
received FIN is acknowledged by TCP. The receipt of the FIN is also passed
to the application as an end-of-file (after any data that may have already
been queued for the application to receive), since the receipt of the FIN
means the application will not receive any additional data on the
connection.
3. Sometime later, the application that received the end-of-file will close its
socket. This causes its TCP to send a FIN.
4. The TCP on the system that receives this final FIN (the end that did the
active close) acknowledges the FIN.

By: Er. Saurav Raj Pant, IOE – TU

 Fig: Packets exchanged when a TCP connection is closed

TCP Operation:

TCP provides the following major services to the upper protocol layers:

 Connection-oriented data management to ensure the end to end

transfer of data across the network(s).
 Reliable data transfer to assure that all data is accurately received in
sequence and with no duplicates.
 Stream-oriented data transfer takes place between the sender
application and TCP and the receiving application and TCP.
 Prior to data transmission, hosts establish a virtual connection via a
synchronization process. The synch process is a 3-way “handshake”, which
ensures both sides are ready to transfer data and determines the initial
sequence numbers.
 Sequence numbers are reference numbers between the two devices.
 Sequence numbers give hosts a way to acknowledge what they have
received.

Connection Establishment and Connection Release:

• The three-way handshake was introduced to establish a connection.

By: Er. Saurav Raj Pant, IOE – TU
 • The first machine sends a CONNECTION REQUEST to host 2 containing a
sequence number.

• Host 2 replies with an ACK acknowledging the sequence number and

sending it’s own initial sequence number.

• Host 1 acknowledges Host 2’s choice of sequence number in the first data
packet.

Connection Release:

• We can release the connection symmetrically or asymmetrically.

• Asymmetric is just like a telephone – when one person ends the

connection, it is broken.

• Asymmetric release can cause data loss if one side sends data that is not
received before the disconnect.

• Symmetric – treats the connection like two unidirectional connections,

requiring each end of the connection to be released.

• Unfortunately, determining when the two sides are done is difficult.

• This problem is known as the two-army problem.

• Host 1 sends a Disconnect Request (DR) to Host 2.

• Host 2 replies with a DR and starts a timer just in case the reply is lost.

• Host 1 will ACK the DR from Host 2 and release the connection.

• When the ACK arrives, Host 2 will drop the connection.

• If the final ACK is lost, the time will take care of the disconnection.

By: Er. Saurav Raj Pant, IOE – TU

 Fig: Connection Release Protocols

By: Er. Saurav Raj Pant, IOE – TU

Port and Socket:

 Each process that wants to communicate with another process identifies itself to the TCP/IP
protocol suite by one or more ports.
 Usually a service is associated with a port (e.g. http on port 80).

Fig: Using Ports to identify services

Sockets:

 A socket is one endpoint of a two way communication link between two

programs running on the network.
 A pair of processes communicating over a network employs a pair of
sockets- one for each process.
 To an application, a socket is a file descriptor that lets the application
read/write from/to the network.
 Sockets are uniquely identified by an IP address, end-to-end protocol, and a
port number.
 The server waits for incoming client requests by listening to a specified
port. Once a request is received, the server accepts a connection from the
client socket to complete the connection.

By: Er. Saurav Raj Pant, IOE – TU

There are two types of sockets:

1. Datagram Socket: (Connection-less Socket)

 It provides unreliable, best- effort networking service
 Packets may be lost; may arrive out of order (uses UDP)
 applications: streaming audio/ video (real-player), …
2. Stream Socket: (Connection-Oriented Socket)
 It provides reliable, connected networking service
 Error free; no out- of- order packets (uses TCP)
 Applications: telnet/ ssh, http, …

Flow Control:

 Flow control at this layer is performed end-to-end rather than across a

single link.
 A sliding window is used to make data transmission more efficient as well
as to control the flow of data so that the receiver does not become
overwhelmed.
 Some points about sliding windows at the transport layer:
o The sender does not have to send a full window’s worth of data.
o An acknowledgment can expand the size of the window based on the
sequence number of the acknowledged data segment.
o The size of the window can be increased or decreased by the
receiver.
o The receiver can send an acknowledgment at anytime.

TCP Windows and Flow Control:

 Data often is too large to be sent in a single segment.

 TCP splits the data into multiple segments.
 TCP provides flow control through “windowing” to set the pace of how
much data is sent at a time
o how many bytes per window, and how many windows between
ACKs.

By: Er. Saurav Raj Pant, IOE – TU

  Sliding window refers to the fact that the window size is negotiated
dynamically during the TCP session.
 If the source receives no acknowledgment, it knows to retransmit at a
slower rate.

Reliable Delivery

Sequence and ACK Numbers:

 Each TCP segment is numbered before transmission so that the receiver

will be able to properly reassemble the bytes in their original order.
 They also identify missing data pieces so the sender can retransmit them.
 Only the missing segments need to be re-transmitted.

Positive Ack and Retransmission:

 Source sends packet, starts timer, and waits for ACK.

 If timer expires before source receives ACK, source retransmits the packet
and restarts the timer.

By: Er. Saurav Raj Pant, IOE – TU

Error Control:

 Mechanisms for error control are based on error detection and

retransmission.
 Error detections are performed using algorithms implemented in software,
such as checksum.
 We already have error handling at the data link layer, why do we need it at
the transport layer?

Multiplexing and De-Multiplexing:

By: Er. Saurav Raj Pant, IOE – TU

  Gathering data from multiple application processes of sender,
enveloping that data with header and sending them as a whole to the
intended receiver is called as multiplexing.
 Delivering received segments at receiver side to the correct app layer
processes is called as de-multiplexing.

How multiplexing and De-multiplexing is done?

For sending data from an application at sender side to an application at the

destination side, sender must know the IP address of destination and port
number of the application (at the destination side) to which he wants to
transfer the data. 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.

For demultiplexing, host receives IP datagram which contains source and

destination IP address along with source and destination port number.
Receiving host will uses these IP addresses and port numbers to direct
segment to appropriate socket.

Connection-less Demultiplexing:

 UDP socket identified by two tuple:

o (destination IP address, destination port number)
 When host receives UDP segment:
o checks destination port number in segment
o directs UDP segment to socket with that port number

By: Er. Saurav Raj Pant, IOE – TU

 Fig: Connection-less Demultiplexing

Connection Oriented De-multiplexing:

 TCP socket identified by 4- tuple:

o source IP address
o source port number
o destination IP address
o destination port number
 receiving host uses all four values to direct segment to appropriate socket
 Server host may support many simultaneous TCP sockets:
o each socket identified by its own 4-tuple
 Web servers have different sockets for each connecting Client
o non-persistent HTTP will have different socket for each request

Fig: Connection-Oriented De-multiplexing

By: Er. Saurav Raj Pant, IOE – TU

 Congestion Control:

 Too many packets present in the network cause packet delay and loss
that degrades performance. This situation is called congestion.
 Congestion control refers to the mechanisms and techniques to control
the congestion and keep the load below the capacity.

Effects of Congestion

 As delay increases, performance decreases.

 If delay increases, retransmission occurs, making situation worse.

Congestion Control Techniques can be classified into two categories:

Open loop congestion control policies are applied to prevent congestion before it
happens. The congestion control is handled either by the source or the
destination. Policies adopted by open loop congestion control are:

1. Retransmission Policy:
It is the policy in which retransmission of the packets are taken care.
If the sender feels that a sent packet is lost or corrupted, the packet
needs to be retransmitted. This transmission may increase the
congestion in the network. To prevent congestion, retransmission

By: Er. Saurav Raj Pant, IOE – TU

 timers must be designed to prevent congestion and also able to
optimize efficiency.
2. Window Policy:
The type of window at the sender side may also affect the congestion.
Several packets in the Go-back-n window are resent, although some
packets may be received successfully at the receiver side. This duplication
may increase the congestion in the network and making it worse.
Therefore, selective repeat window should be adopted as it sends the
specific packet that may have been lost.
3. Discarding Policy:
A good discarding policy adopted by the routers is that the routers may
prevent congestion and at the same time partially discards the corrupted or
less sensitive package and also able to maintain the quality of a message.
In case of audio file transmission, routers can discard less sensitive packets
to prevent congestion and also maintain the quality of the audio file.
4. Acknowledgment Policy:
Since acknowledgement are also the part of the load in network, the
acknowledgment policy imposed by the receiver may also affect
congestion. Several approaches can be used to prevent congestion related
to acknowledgment.
The receiver should send acknowledgement for N packets rather than
sending acknowledgement for a single packet. The receiver should send a
acknowledgment only if it has to sent a packet or a timer expires.
5. Admission Policy:
An admission policy, which is a quality-of-service mechanism, can also
prevent congestion in virtual-circuit networks. Switches in a flow first check
the resource requirement of a flow before admitting it to the network. A
router can deny establishing a virtual- circuit connection if there is
congestion in the network or if there is a possibility of future congestion.

Closed-loop congestion control mechanisms try to alleviate congestion after it

happens. Several mechanisms have been used by different protocols.
By: Er. Saurav Raj Pant, IOE – TU
 1. Back-pressure:
The technique of backpressure refers to a congestion control mechanism in
which a congested node stops receiving data from the immediate upstream
node or nodes. This may cause the upstream node or nodes to become
congested, and they, in turn, reject data from their upstream nodes or
nodes. And so on. Backpressure is a node-to-node congestion control that
starts with a node and propagates, in the opposite direction of data flow, to
the source. The backpressure technique can be applied only to virtual
circuit networks, in which each node knows the upstream node from which
a flow of data is coming.

2. Choke Packet:
A choke packet is a packet sent by a node to the source to inform it of
congestion. Note the difference between the backpressure and choke
packet methods. In backpressure, the warning is from one node to its
upstream node, although the warning may eventually reach the source
station. In the choke packet method, the warning is from the router, which
has encountered congestion, to the source station directly. The
intermediate nodes through which the packet has traveled are not warned.
Figure shows the idea of a choke packet.

3. Implicit Signaling:
In implicit signaling, there is no communication between the congested
node or nodes and the source. The source guesses that there is congestion

By: Er. Saurav Raj Pant, IOE – TU

 somewhere in the network from other symptoms. For example, when a
source sends several packets and there is no acknowledgment for a while,
one assumption is that the network is congested. The delay in receiving an
acknowledgment is interpreted as congestion in the network; the source
should slow down.

4. Explicit Signaling:
The node that experiences congestion can explicitly send a signal to the
source or destination. The explicit signaling method, however, is different
from the choke packet method. In the choke packet method, a separate
packet is used for this purpose; in the explicit signaling method, the signal is
included in the packets that carry data. Explicit signaling can occur in either
the forward or the backward direction.
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.

Congestion Control Algorithms:

Also called traffic shaping algorithms, as they help regulate the data transmission
and reduce congestion.

Leaky Bucket:

This algorithm is used to control the rate at which traffic is sent to the
network and shape the burst traffic to a steady traffic stream. Consider a bucket
with a small hole at the bottom, whatever may be the rate of water pouring into
the bucket, the rate at which water comes out from that small hole is constant.

By: Er. Saurav Raj Pant, IOE – TU

Once the bucket is full, any additional water entering it spills over the sides and is
lost.

The same idea of leaky bucket can be applied to packets:

 When the host has to send a packet, the packet is thrown into the
bucket.
 The bucket leaks at a constant rate, meaning the network interface
transmits packets at a constant rate.
 Bursty traffic is converted to a uniform traffic by the leaky bucket.
 If a packet arrives when the bucket is full, the packet must be
discarded.
 A FIFO queue is used for holding the packets.

By: Er. Saurav Raj Pant, IOE – TU

Token Bucket:

The leaky bucket algorithm described above, enforces a rigid pattern at the
output stream, irrespective of the pattern of the input. For many application it is
better to allow the output to speed up somewhat when a larger burst arrives than
to loose the data. Token bucket algorithm provides such a solution. In this
algorithm, bucket holds the token generated at regular intervals. Main steps of
this algorithm can be described as follows:

 In regular intervals token are thrown into the bucket.

 The bucket has a maximum capacity.
 If there is a ready packet, a token is removed from the bucket, and the
packet is sent
 If there is no token in the bucket, the packet cannot be sent. The packet
waits in a queue until a token is thrown into the bucket.
 The token bucket algorithm is less restrictive than the leaky bucket
algorithm, in a sense that it allows bursty traffic. However the limit of burst
is restricted by the number of tokens available in the bucket at a particular
instant of time.
 The implementation of basic token bucket algorithm is simple; a variable is
used just to count the tokens. This counter is incremented every t seconds
and is decremented whenever a packet is sent. Whenever this counter
reaches to zero, no further packet is sent out.

By: Er. Saurav Raj Pant, IOE – TU

 Fig: token bucket (a) before (b) after

By: Er. Saurav Raj Pant, IOE – TU