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using this header extension are called jumbo grams. The use of jumbo grams is important for
supercomputer applications that must transfer gigabytes of data efficiently across the Internet.
The destination options header is intended for fields that need only be interpreted at the
destination host. In the initial version of IPv6, the only options defined are null options for
padding this header out to a multiple of 8 bytes, so initially it will not be used. It was included to
make sure that new routing and host software can handle it, in case someone thinks of a
destination option some day. The routing header lists one or more routers that must be visited on
the way to the destination. It is very similar to the IPv4 loose source routing in that all addresses
listed must be visited in order, but other routers not listed may be visited in between. The format
of the routing header is shown in Fig.

The extension header for routing

The first 4 bytes of the routing extension header contain four 1-byte integers. The Next header
and Header extension length fields were described above. The Routing type field gives the
format of the rest of the header. Type 0 says that a reserved 32-bit word follows the first word,
followed by some number of IPv6 addresses. Other types may be invented in the future, as
needed. Finally, the Segments left field keeps track of how many of the addresses in the list have
not yet been visited. It is decremented every time one is visited. When it hits 0, the packet is on
its own with no more guidance about what route to follow. Usually, at this point it is so close to
the destination that the best route is obvious. The fragment header deals with fragmentation
similarly to the way IPv4 does. The header holds the datagram identifier, fragment number, and a
bit telling whether more fragments will follow.

THE TRANSPORT LAYER

THE TRANSPORT SERVICE

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. To achieve this, the
transport layer makes use of the services provided by the network layer. The software and/or
hardware within the transport layer that does the work is called the transport entity. The
transport entity can be located in the operating system kernel, in a library package bound into
network applications, in a separate user process, or even on the network interface card. The first
two options are most common on the Internet. The (logical) relationship of the network,
transport, and application layers is illustrated in Fig.

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there are also two types of transport service. The connection-oriented transport service is similar
to the connection-oriented network service in many ways. In both cases, connections have three
phases: establishment, data transfer, and release. Addressing and flow control are also similar in
both layers. Furthermore, the connectionless transport service is also very similar to the
connectionless network service. However, note that it can be difficult to provide a connectionless
transport service on top of a connection-oriented network service, since it is inefficient to set up
a connection to send a single packet and then tear it down immediately afterwards. The obvious
question is this: if the transport layer service is so similar to the network layer service, why are
there two distinct layers? Why is one layer not Problems occur, that’s what? The users have no
real control over the network layer, so they cannot solve the problem of poor service by using
better routers or putting more error handling in the data link layer because they don’t own the
routers. The only possibility is to put on top of the network layer another layer that improves the
quality of the service. If, in a connectionless network, packets are lost or mangled, the transport
entity can detect the problem and compensate for it by using retransmissions. If, in a connection-
oriented network, a transport entity is informed halfway through a long transmission that its
network connection has been abruptly terminated, with no indication of what has happened to the
data currently in transit, it can set up a new network connection to the remote transport entity.
Using this new network connection, it can send a query to its peer asking which data arrived and
which did not, and knowing where it was, pick up from where it left off.

Transport Service Primitives

To allow users to access the transport service, the transport layer must provide some operations
to application programs, that is, a transport service interface. Each transport service has its own
interface. In this section, we will first examine a simple (hypothetical) transport service and its
interface to see the bare essentials. In the following section, we will look at a real example. The
transport service is similar to the network service, but there are also some important differences.
The main difference is that the network service is intended to model the service offered by real
networks, warts and all. Real networks can lose packets, so the network service is generally
unreliable.

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A quick note on terminology is now in order. For lack of a better term, we will use the term
segment for messages sent from transport entity to transport entity.TCP, UDP and other Internet
protocols use this term. Some older protocols used the ungainly name TPDU (Transport
Protocol Data Unit). That term is not used much anymore now but you may see it in older
papers and books the network entity similarly processes the packet header and then passes the
contents of the packet payload up to the transport entity. This nesting is illustrated in Fig. 6-3

Connection Establishment
Establishing a connection sounds easy, but it is actually surprisingly tricky. At first glance, it
would seem sufficient for one transport entity to just send a CONNECTION REQUEST segment
to the destination and wait for a CONNECTION ACCEPTED reply. The problem occurs when
the network can lose, delay, corrupt, and duplicate packets. This behaviour causes serious
complications. Imagine a network that is so congested that acknowledgements hardly ever get
back in time and each packet times out and is retransmitted two or three times. Suppose that the
network uses datagrams inside and that every packet follows a different route. Some of the
packets might get stuck in a traffic jam inside the network and take a long time to arrive. That is,
they may be delayed in the network and pop out much later, when the sender thought that they
had been lost. The worst possible nightmare is as follows. A user establishes a connection with a
bank, sends messages telling the bank to transfer a large amount of money to the account of a
not-entirely-trustworthy person. Unfortunately, the packets decide to take the scenic route to the
destination and go off exploring a remote corner of the network. The sender then times out and
sends them all again. This time the packets take the shortest route and are delivered quickly so
the sender releases the connection.

Packet lifetime can be restricted to a known maximum using one (or more) of the following
techniques:
1. Restricted network design.

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2. Putting a hop counter in each packet.

3. Time stamping each packet.

The first technique includes any method that prevents packets from looping, combined with
some way of bounding delay including congestion over the (now known) longest possible path.
It is difficult, given that internets may range from a single city to international in scope. The
second method consists of having the hop count initialized to some appropriate value and
decremented each time the packet is forwarded. The network protocol simply discards any
packet whose hop counter becomes zero. The third method requires each packet to bear the time
it was created, with the routers agreeing to discard any packet older than some agreed-upon time.
This latter method requires the router clocks to be synchronized, which itself is a nontrivial task,
and in practice a hop counter is a close enough approximation to age.

TCP uses this three-way handshake to establish connections. Within a connection, a timestamp is
used to extend the 32-bit sequence number so that it will not wrap within the maximum packet
lifetime, even for gigabit-per-second connections. This mechanism is a fix to TCP that was
needed as it was used on faster and faster links. It is described in RFC 1323 and called PAWS
(Protection Against Wrapped Sequence numbers). Across connections, for the initial
sequence numbers and before PAWS can come into play, TCP originally use the clock-based
scheme just described. However, this turned out to have security vulnerability. The clock made it
easy for an attacker to predict the next initial sequence number and send packets that tricked the
three-way handshake and established a forged connection. To close this hole, pseudorandom
initial sequence numbers are used for connections in practice.

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the initial sequence numbers not repeat for an interval even though they appear random to an
observer. Otherwise, delayed duplicates can wreak havoc.

Connection Release
Releasing a connection is easier than establishing one. Nevertheless, there are more pitfalls than
one might expect here. As we mentioned earlier, there are two styles of terminating a connection:
asymmetric release and symmetric release Asymmetric release is the way the telephone system
works: when one party hangs up, the connection is broken. Symmetric release treats the
connection as two separate unidirectional connections and requires each one to be released
separately. Asymmetric release is abrupt and may result in data loss. Consider the scenario of
Fig. After the connection is established, host 1 sends a segment that arrives properly at host 2.
Then host 1 sends another segment. Unfortunately, host 2 issues a DISCONNECT before the
second segment arrives. The result is that the connection is released and data are lost.

Crash Recovery
If hosts and routers are subject to crashes or connections are long-lived (e.g., large software or
media downloads) recovery from these crashes becomes an issue. If the transport entity is
entirely within the hosts, recovery from network

and router crashes is straightforward. The transport entities expect lost segments all the time and
know how to cope with them by using retransmissions. A more troublesome problem is how to
recover from host crashes. In particular, it may be desirable for clients to be able to continue
working when servers crash and quickly reboot. To illustrate the difficulty, let us assume that
one host, the client, is sending a long file to another host, the file server, using a simple Stop-
and-wait protocol. The transport layer on the server just passes the incoming segments to the
transport user, one by one. Partway through the transmission, the server crashes. When it comes

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back up, its tables are reinitialized, so it no longer knows precisely where it was. In an attempt to
recover its previous status, the server might send a broadcast segment to all other hosts,
announcing that it has just crashed and requesting that its clients inform it of the status of all
open connections. Each client can be in one of two states: one segment outstanding, S1, or no
segments outstanding, S0. Based on only this state information, the client must decide whether to
retransmit the most recent segment.

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UNIT-V
UDP Protocol
UDP provides connectionless, unreliable, datagram service. Connectionless service means that
there is no logical connection between the two ends exchanging messages. Each message is an
independent entity encapsulated in a datagram.
UDP does not see any relation (connection) between consequent datagram coming from the same
source and going to the same destination.
UDP has an advantage: it is message-oriented. It gives boundaries to the messages exchanged.
An application program may be designed to use UDP if it is sending small messages and the
simplicity and speed is more important for the application than reliability.

User Datagram
UDP packets, called user datagram, have a fixed-size header of 8 bytes made of four fields, each
of 2 bytes (16 bits).
. The 16 bits can define a total length of 0 to 65,535 bytes. However, the total length needs to be
less because a UDP user datagram is stored in an IP datagram with the total length of 65,535
bytes. The last field can carry the optional checksum

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

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As mentioned previously, UDP provides a connection less service. This means that each user
datagram sent by UDP is an independent datagram. There is no relationship between the
different user data grams even if they are coming from the same source process and going to the
same destination program.

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.

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.

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

UDP Applications

UDP Features

Connectionless Service
As we mentioned previously,
 UDP is a connectionless protocol. Each UDP packet is independent from other packets
sent by the same application program. This feature can be considered as an advantage or
disadvanta8e depending on the application requirements.
 UDP does not provide error control; it provides an unreliable service. Most applications
expect reliable service from a transport-layer protocol. Although a reliable service is
desirable.

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Typical Applications
The following shows some typical applications that can benefit more from the services of UDP

 UDP is suitable for a process that requires simple request-response communication with
little concern for flow and error control

 UDP is suitable for a process with internal flow- and error-control mechanisms. For
example, the Trivial File Transfer Protocol (TFIP)

 UDP is a suitable transport protocol for multicasting. Multicasting capability is embedded

in the UDP software

 UDP is used for management processes such as SNMP

 UDP is used for some route updating protocols such as Routing Information Protocol
(RIP)

 UDP is normally used for interactive real-time applications that cannot tolerate uneven
delay between sections of a received message

TRANSMISSION CONTROL PROTOCOL

Transmission Control Protocol (TCP) is a connection-oriented, reliable protocol. TCP
explicitly defines connection establishment, data transfer, and connection teardown phases to
provide a connection-oriented service.

TCP Services

Process-to-Process Communication
As with UDP, TCP provides process-to-process communication using port numbers. We have
already given some of the port numbers used by TCP.

Stream Delivery Service

In UDP, a process sends messages with predefined boundaries to UDP for delivery. UDP adds its
own header to each of these messages and delivers it to IP for transmission.

TCP, on the other hand, 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.

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Sending and Receiving Buffers

Because the sending and the receiving processes may not necessarily write or read data at the
same rate, TCP needs buffers for storage.

There are two buffers, the sending buffer and the receiving buffer, one for each direction.

 At the sender, the buffer has three types of chambers. The white section contains empty
chambers that can be filled by the sending process (producer).

 The colored area holds bytes that have been sent but not yet acknowledged.

 The TCP sender keeps these bytes in the buffer until it receives an acknowledgment. The
shaded area contains bytes to be sent by the sending TCP.

 The operation of the buffer at the receiver is simpler. The circular buffer is divided into
two areas (shown as white and colored).

 The white area contains empty chambers to be filled by bytes received from the network.

 The colored sections contain received bytes that can be read by the receiving process.
When a byte is read by the receiving process, the chamber is recycled and added to the
pool of empty chambers.

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Segments
 Although buffering handles the disparity between the speed of the producing and
consuming Processes, we need one more step before we can send data.

 The network layer, as a service provider for TCP, needs to send data in packets, not as a
stream of bytes. At the transport layer, TCP groups a number of bytes together into a
packet called a segment.

 The segments are encapsulated in an IP datagram and transmitted. This entire operation is
transparent to the receiving process.

Format
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.

Source port address This is a 16-bit field that defines the port number of the application program
in the host that is sending the segment.

Destination port address This is a 16-bit field that defines the port number of the application
program in the host that is receiving the segment.

Sequence number This 32-bit field defines the number assigned to the first byte of data
contained in this segment.

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Acknowledgment number This 32-bit field defines the byte number that the receiver of the
segment is expecting to receive from the other party.

Header length This 4-bit field indicates the number of 4-byte words in the TCP header. The
length of the header can be between 20 and 60 bytes.

A TCP Connection
 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.

 TCP operates at a higher level. TCP uses the services of IP to deliver individual segments
to the receiver, but it controls the connection itself.

 In TCP, connection-oriented transmission requires three phases: connection

establishment, data transfer, and connection termination.

Connection Establishment

TCP transmits data in full-duplex mode. When two TCPs in two machines are connected, they
are able to send segments to each other simultaneously.

Three- Way Handshaking

The connection establishment in TCP is called three-way handshaking. an application program,
called the client, wants to make a connection with another application program, called the server,

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using TCP as the transport-layer protocol The process starts with the server. The server program
tells its TCP that it is ready to accept a connection. This request is called a passive open.
Although the server TCP is ready to accept a connection from any machine in the world, it
cannot make the connection itself.

The client program issues a request for an active open. A client that wishes to connect
to an open server tells its TCP to connect to a particular server.

 A SYN segment cannot carry data, but it consumes one sequence number.
 A SYN + ACK segment cannot carry data, but it does consume one sequence number.

 An ACK segment, if carrying no data, consumes no sequence number.

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HTTP
The Hyper Text Transfer Protocol (HTTP) is used to define how the client-server programs can
be written to retrieve web pages from the Web.

An HTTP client sends a request; an HTTP server returns a response. The server uses the port
number 80; the client uses a temporary port number.

Non persistent versus Persistent Connections

If the web pages, objects to be retrieved, are located on different servers, we do not have any
other choice than to create a new TCP connection for retrieving each object. If some of the
objects are located on the same server, we have two choices: to retrieve each object using a new
TCP connection

Non persistent Connections

In a non persistent connection, one TCP connection is made for each request/response.
The following lists the steps in this strategy:

1. The client opens a TCP connection and sends a request.

2. The server sends the response and closes the connection.
3. The client reads the data until it encounters an end-of-file marker; it then closes the
connection.

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