TCP IP Model

Employee name: Gowtham H R

Employee I D: 031
 Contents
 Introduction to TCP IP
1. TCP/IP architectural model

 Application Layer
1. Client and Server
2. Overview of HTTP

 Transport Layer
1.Connection less Transport : UDP
2. Connection oriented Transport : TCP
3. TCP Connection Management

 Network Layer
1. Network Service model
2. Routing Principles
3. Internet Protocol
 Data link Layer
1.Types of data link layer
2. Ethernet Data Link Layer
 Physical Layer
1. Physical Layer Functions
2. Examples of physical layer

Note :
The physical layer is also “special” in that it is the only
layer that really does not apply specifically to TCP/IP.
Even in studying TCP/IP, however, it is still important
To understand its significance and role in relation
To the other layers where TCP/IP protocols reside.
TCP/IP architectural
model
 The TCP/IP protocol suite is so named for two of its
most important protocols:
Transmission Control Protocol (TCP) and Internet Protocol
(IP).

 The TCP/IP Suite defines a set of rules to enable

computers to communicate over a network. TCP/IP
specifies how data should be formatted, addressed,
shipped, routed and delivered to the right destination.
The specification defines protocols for different types
of communication between computers and provides a
framework for more detailed standards.
TCP/IP is generally described as having four abstraction
layers : Application layer
Transport Layer
Network/Internet Layer
Link Layer
Physical Layer – This layer consists of
hardware parts
This diagram shows two Internet
hosts connected via two routers
and the corresponding layers used
at each hop.
Application layer stays at top.

Transport layer comes in 2nd

Position

Network layer is at 3rd place.

Data link is at 4th place

And at last it is physical layer

consists of Ethernet etc..
Application Layer
An application-layer protocol is only one part of a
network application.
This is the layer within which applications create user
data and communicate this data to other processes or
applications on another or the same host.

The communications partners are often called peers.

This is where the "higher level" protocols such as SMTP,

FTP,SSH, HTTP, etc. operate.
In particular, an application layer protocol defines:

• The types of messages exchanged, e.g., request

messages and response messages;

• The syntax of the various message types, i.e., the fields

in the message and how the fields are delineated;

• The semantics of the fields, i.e., the meaning of the

information in the fields;

• The rules for determining when and how a process

sends messages and responds to messages.
 Clients and Server
A network application protocol typically has two parts or
"sides", a client side and a server side. an application
involves two processes in two different
hosts communicating with each other over a network.
This figure illustrates socket communication between
two processes that communicate over the Internet.
What Services Does an Application
Layer Need?
The application layer needs few services from transport
protocol they are
1.Data loss
2.Bandwidth
3.Timing.

• Data Loss : Some applications, such as electronic

mail, file transfer, remote host access, Web document
transfers, and financial applications require fully
reliable data transfer
In particular, a loss of file data, or data in a financial
transaction, can have devastating consequences
Band width : Some applications must be able to
transmit data at a certain bandwidth in order to be
"effective".
If this amount of bandwidth is not available, the
application needs to either encode at a different rate (and
receive enough bandwidth to sustain this different coding
rate) or should give up receiving half of the needed
bandwidth is of no use to such a bandwidth-sensitive
Application.

Timing : The final service requirement is that of timing.

Interactive real-time applications, such as Internet
telephony, virtual environments, teleconferencing, and
multiplayer games require tight timing constraints on data
delivery in order to be "effective.“ if a a long delay between
taking an action and seeing the response from the
environment (e.g., from another player on the end of an
end-to-end connection) makes the application feel less
"realistic."
Services Provided by the Internet
Transport Protocols
The Internet (and more generally TCP/IP networks)
makes available two transport protocols to applications,
namely,

UDP (User Datagram Protocol) and

TCP (Transmission Control Protocol).

Each of these protocols offers a different service model to
the invoking applications.

• TCP Services : The TCP service model includes a

connection-oriented service and a reliable data transfer
service. When an application invokes TCP for its transport
protocol, the application receives both of these services
from TCP
Connection-oriented service: TCP has the client and
server exchange transport-layer control information with
each other before the application-level messages begin to
flow. This so-called handshaking procedure alerts the client
and server, allowing them to prepare for an onslaught of
packets.
After the handshaking phase, a TCP connection is said to
exist between the sockets of the two processes. The
connection is a full-duplex connection in that the two
processes can send messages to each other over the
connection at the same time. When the application is
finished sending messages, it must tear down the
connection.
The service is referred to as a "connection-oriented" service
rather than a "connection" service because the two
processes are connected in a very loose manner
• Reliable transport service: The communicating
processes can rely on TCP to deliver all messages sent
without error and in the proper order. When one side of
the application passes a stream of bytes into a socket, it
can count on TCP to deliver the same stream of data to
The receiving socket, with no missing or duplicate bytes.

TCP also includes a congestion control mechanism, a

service for the general welfare of the Internet rather
than for the direct benefit of the communicating
processes.
The TCP congestion control mechanism throttles a process
(client or server) when the network is congested.
Overview of HTTP
• The Hypertext Transfer Protocol (HTTP), the Web's
application-layer protocol

• HTTP is implemented in two programs: a client

program and server program. The client program and
server programs, executing on different end systems, talk
to each other by exchanging HTTP messages.

• HTTP defines the structure of these messages and how

the client and server exchange the messages.
HTTP defines how Web clients (browsers) request Web
pages from servers (i.e., Web servers) and how servers
transfer Web pages to clients.

When a user requests a Web page (e.g., clicks on a

hyperlink), the browser sends HTTP request messages for
the objects in the page to the server. The server receives
the requests and responds with HTTP response messages
that contain the objects
HTTP Message Format
The HTTP specifications define the HTTP message
formats. There are two types of HTTP messages,
• Request messages and
• Response messages

Request message : This message is sent by client to

server requesting any web page or image

The format of the request message is shown in next slide

A typical HTTP request message:
GET /somedir/page.html HTTP/1.1
Connection: close
User-agent: Mozilla/4.0
Accept: text/html, image/gif, image/jpeg
Accept-language:fr
(extra carriage return, line feed)

General format of a request message :

The first line of a HTTP request message is called the
request line;
line
the subsequent lines are called the header lines.

The request line has three fields:

1.The method field,
2. The URL field, and
3.The HTTP version field.

The method field can take on several different values,

including GET, POST, and HEAD.

The GET method is used when the browser requests an object,

with the requested object identified in the URL field. In this
example, the browser is requesting the object
/somedir/page.html. (The browser doesn't have to specify
the host name in the URL field since the TCP connection is
already connected to the host (server) that serves the
requested file.)
Header line description:
Connection:close This is the header line, the browser
is telling the server that it doesn't want to use
Persistent connections; it wants the server to close
the connection after sending the requested object.
Thus the browser that generated this request message
implements HTTP/1.1 but it doesn't want to bother
with persistent connections.

The Useragent:
Useragent This header line specifies the user
agent, i.e., the browser type that is making the
request to the server . Here the user agent is The
HyperText Transfer Protocol Mozilla/4.0, a Netscape
browser. This header line is useful because the server
Can actually send different versions of the same
Object to different types of user agents.
The Accept: This header line tells the server the type of
objects the browser is prepared to accept. In this case, the
client is prepared to accept HTML text, a GIF image or a
JPEG image.
If the file /somedir/page.html contains a Java applet (and
who says it can't!), then the server shouldn't send the file,
since the browser can not handle that object type.

Accept-language: This header indicates that the user

prefers to receive a French version of the object, if such
an
Object exists on the server; otherwise, the server should
send its default version.
The general format for a response mess
age
HTTP/1.1 200 OK
Connection: close
Date: Thu, 06 Aug 1998 12:00:15 GMT
Server: Apache/1.3.0 (Unix)
Last-Modified: Mon, 22 Jun 1998 09:23:24 GMT
Content-Length: 6821
Content-Type: text/html
data data data data data ...

This message format has three section

1. status line
2. header lines, and 3.Entity body.
• Status line: The status line has three fields:
1.The protocol version field
2.A status code,
3.A corresponding status message.
In this example, the status line indicates that the server is
using HTTP/1.1 and that that everything is OK (i.e., the
server has found, and is sending, the requested object).

• The header lines: The server uses the

Connection: close header line to tell the client that it is
going to close the TCP connection after sending the
message.
The Server: This header line indicates that the message
was generated by an Apache Web server
The Content-Length: This header line indicates the number
of bytes in the object being sent.

The Content-Type:
Content-Type This header line indicates that the
object in the entity body is HTML text. (The object type is
officially indicated by the Content-Type: header and not by
the file extension.)

General format of a response message

Transport Layer
Transport Layer Resides between the application and
network layers, the transport layer is in the core of the
layered network architecture.
It has the critical role of providing communication services
directly to the Application Processes running on different
hosts.

Transport layers are contained in both the TCP/IP model

which is the foundation of the Internet, and the Open
Systems Interconnection (OSI) model of general
Networking.
The Internet, and more generally a TCP/IP network,
makes
available two distinct transport-layer protocols to the
application layer.

One of these protocols is UDP (UserDatagram Protocol),

which provides an unreliable, connectionless service to the
invoking application.
The second of the these protocols is TCP
(TransmissionControl Protocol), which provides a reliable,
connection-oriented service to the invoking application.
Connectionless Transport: UDP
The User Datagram Protocol (UDP) is one of the core
members of the Internet Protocol Suite, the set of
network protocols used for the Internet.

With UDP, computer applications can send messages, in

this case referred to as datagrams, to other hosts on an
Internet Protocol (IP) network without requiring prior
communications to set up special transmission channels
or data paths UDP uses a simple transmission model
without implicit hand-shaking dialogues for guaranteeing
reliability ,ordering, or data integrity.

Thus, UDP provides an unreliable service and datagrams

may arrive out of order, UDP assumes that error
checking
and correction is either not necessary or performed in the
application, avoiding the overhead of such processing at
the network interface level
Packet structure of UDP
• UDP provides a very simple interface between the Internet
Layer below and the Application Layer above.

• UDP provides no guarantees to the upper layer protocol for

message delivery and a UDP sender retains no state on UDP
messages once sent.

UDP adds only application multiplexing and checksumming of

the header and payload. If any kind of reliability for the
information transmitted is needed, it must be implemented in
upper layers.
This is the packet structure of the UDP protocol.
The UDP header consists of only 4 fields and Source port and
Checksum are optional.
• Source port 
– This field identifies the sending port when meaningful
and should be assumed to be the port to reply to if
needed. If not used, then it should be zero.
• Destination port 
– This field identifies the destination port and is
required.
• Length 
– A 16-bit field that specifies the length in bytes of the
entire datagram: header and data. The minimum
length is 8 bytes since that's the length of the
header. The field size sets a theoretical limit of
65,535 bytes (8 byte header + 65527 bytes of data)
for a UDP datagram. The length includes the UDP
header, so the minimum size for a UDP datagram is
8 (8 byte header with no data). The practical limit for
the data length which is imposed by the underlying
IPv4 protocol is 65,507 bytes.
• Checksum 
– The 16-bit checksum field is used for error-checking
of the header and data.
Checksum is the 16-bit one's complement of the one's
complement sum of a pseudo header of information from
the IP header, the UDP header, and the data, padded with
zero octets at the end (if necessary) to make a multiple of
two octets.
Connection-Oriented Transport:
TCP
The Internet Transport Layer has Connection oriented protocol
Called TCP- Transmission Control Protocol.

TCP relies on many of the underlying principles including

error detection, retransmissions, cumulative
acknowledgements, timers and header fields for sequence and
acknowledgement numbers.
A TCP connection provides for full duplex data transfer.

That is, application-level data can be transferred in both

directions between two hosts - if there is a TCP connection
between process A on one host and process B on another
host, then application-level data can flow from A to B at the
Same time as application-level data flows from B to A.

TCP connection is also always point-to-point, i.e.,

between
A single sender and a single receiver. So called
"multicasting" the transfer of data from
one sender to many receivers in a single send
operation -- is not possible with TCP.
Connection Establishment In TCP
• In the previous diagram Let us consider the sending of
data from the client process to the server process.

• The client process passes a stream of data through the

socket Once the data passes through the door, the data is
now in the hands of TCP running in the client.,

• TCP directs this data to the connection's send buffer,

which is one of the buffers that is set aside during the
initial three-way handshake. From time to time, TCP will
"grab" chunks of data from the send buffer.

• The maximum amount of data that can be grabbed and

placed in a segment is limited by the Maximum
Segment Size (MSS).
• TCP encapsulates each chunk of client data with TCP
header, thereby forming TCP segments.

 The segments are passed down to the network

layer ,where they are separately encapsulated within
network-layer IP datagrams.

 The IP datagrams are then sent into the network. When

TCP receives a segment at the other end, the segment's data
Is placed in the TCP connection's receive buffer.
The application reads the stream of data from this
buffer. Each side of the connection has its own send buffer
and its own receive buffer.
TCP segment Structure
Previous diagram shows the structure of the TCP segment.
As with UDP, the header includes source and destination
port numbers, that are used for
multiplexing/demultiplexing data from/to upper layer
Applications

• The32-bit sequence number field, and the 32-bit

acknowledgment number field are used by the TCP
sender and receiver in implementing a reliable data
transfer service.

• The 16-bit window size field is used for the purposes of

flow control that it is used to indicate the number of bytes
that a receiver is willing to accept.

• The 4-bit length field specifies the length of the TCP

header in 32-bit words. The TCP header can be of variable
length due to the TCP options field, discussed below.
• The optional and variable length options field is used
when a sender and receiver negotiate the maximum
segment size (MSS) or as a window scaling factor for use
in high-speed networks.

• The flag field contains 6 bits. The ACK bit is used to

indicate that the value carried in the acknowledgment field
is valid.
The RST, SYN and FIN bits are used for connection setup
and teardown, as we will discuss at the end of this section.
When the PSH bit is set, this is an indication that the
receiver should pass the data to the upper layer
immediately. Finally, the URG bit is used to indicate there
is data in this segment that the sending-side upper
layer entity has marked as ``urgent.'' The location of the
last byte of this urgent data is indicated by the 16-bit
urgent data pointer. TCP must inform the receiving-side
upper layer entity when urgent data exists and pass it a
pointer to the end of the urgent data.
TCP connection Management
Consider a process running in one host wants to initiate a
connection with another process in another host.

The host that is initiating the connection is called the client

host whereas the other host is called the server host. The
Client application process first informs the client TCP that it
wants to establish a connection to a process in the server.

client program does this by issuing the command:

Socket clientSocket = new Socket("hostname", "portnumber");

The TCP in the client then proceeds to establish a TCP
connection with the TCP in the server in the following
manner:

• Step 1.
The client-side TCP first sends a special TCP segment to the
server-side TCP. This special segment contains no
application-layer data.
It does, however, have one of the flag bits in the segment's
header the so-called SYN bit, set to 1.
For this reason, this special segment is referred to as a SYN
segment. In addition, the client chooses an initial sequence
number (client_isn) and puts this number in the sequence
number field of the initial TCP SYN segment. This segment is
encapsulated within an IP datagram and sent into the
Internet
• Step 2.
Once the IP datagram containing the TCP SYN segment
arrives at the server host (assuming it does arrive!), the
server extracts the TCP SYN segment from the datagram,
allocates the TCP buffers and variables to the connection,
and sends a connection-granted segment to client TCP.

This connection-granted segment also contains no

application-layer data. However, it does contain three
important pieces of information in the segment header.
First, the SYN bit is set to 1.
Second, the acknowledgment field of the TCP segment
header is set to isn+1.
Finally, the server chooses its own initial sequence number
(server_isn) and puts this value in the sequence number
field of the TCP segment header.
This connection granted segment is saying, in effect, "I
received your SYN packet to start a connection with your
initial sequence number, client_isn.
I agree to establish this connection. My own initial sequence
number is server_isn." The conenction-granted segment is
sometimes referred to as a SYNACK segment.
• Step 3.
Upon receiving the connection-granted segment, the client
also allocates buffers and variables to the connection. The
client host then sends the server yet another segment; this
last segment acknowledges the server's connection-granted
segment (the client does so by putting the value
server_isn+1 in the acknowledgment field of the TCP
segment header). The SYN bit is set to 0, since the
connection is established.
Network Layer
The network layer is responsible for the source-to-
destination (computer-to-computer or host-to-host)
delivery of a packet, possibly across multiple networks
(links). The network layer ensures that each packet gets
from its point of origin to its final destination.

The transport layer relies on the services of the network

layer, which provides a communication service between
hosts
The role of the network layer is thus deceptively simple to
transport packets from a sending host to a receiving host. To
do so, two important network layer functions can be
identified:
1.Path Determination
2.Switching.

Path Determination :The network layer must determine the

route or path taken by packets as they flow from a sender to a
receiver. The algorithms that calculate these paths are referred
to as routing algorithms.

Switching: When a packet arrives at the input to a router, the

router must move it to the appropriate output link.
Network Service Model
The network service model defines the characteristics of end-
to-end transport of data between one "edge" of the network
and the other, i.e., between sending and receiving end
systems.

There are two types of services provided by network layer

they are
Datagram service model
Virtual-circuit service model
Datagram Subnet
In Datagram model each time an host system wants to send a
packet, it stamps the packet with the address of the
destination end system, and then pops the packet into the
Network, the Packet switches route a packet towards its
destination by examining the packet's destination address,
indexing a routing table with the destination address, and
forwarding the packet in the direction of the destination using
routers and these routing tables can be modified at any time,
a series of packets sent from one end system to another may
follow different paths through the network and may arrive out
of order

This datagram service is connectionless Service

Virtual-circuit service model
A virtual-circuit packet network behaves much like a
telephone network, which uses "real circuits" as opposed to
"virtual circuits". There are three identifiable phases in a
virtual circuit:
• VC setup: During the setup phase, the sender contacts
the
network layer, specifies the receiver address, and waits
for the network to setup the VC. The network layer
determines the path between sender and receiver, i.e., the
Series of links and switches through which all packets of the
VC will travel, this typically involves updating tables in each
of the packet switches in the path. During VC setup, the
network layer may also reserve resources (e.g., bandwidth)
along the path of the VC.
• Data transfer. Once the VC has been established, data
can begin to flow along the VC.
• Virtual circuit teardown. This is initiated when the
sender (or receiver) informs the network layer of its desire
To terminate the VC. The network layer will then typically
inform the end system on the other side of the network of
the call termination, and update the tables in each of the
packet switches on the path to indicate that the VC no
longer exists.

Connection setup at the transport layer only involves the

two end systems. The two end systems agree to
communicate and together determine the parameters of
their transport level connection before data actually
begins to flow on the transport level connection. Although
the two end systems are aware of the transport-layer
connection, the switches within the network are completely
oblivious to it. On the otherhand, with a virtual-circuit
network layer, packet switches are involved in virtual-circuit
setup, and each packet switch is fully aware of all the VCs
passing through it.
Routing Principles
In order to transfer packets from a sending host to the
destination host, the network layer must determine the path
or route that the packets are to follow. Whether the network
layer provides a datagram service or a virtual circuit service
the network layer must nonetheless determine the path for a
packet. This is the job of the network layer routing
protocol.

At the heart of any routing protocol is the algorithm (the

"routing algorithm") that determines the path for a packet.
There are 2 main types of routing protocols
link state algorithms
distance vector algorithm.

Link State Algorithm : link-state routing protocol is one of

the two main classes of routing protocols used in packet-
switched networks for computer communications .
The link state algorithm we present below is known as
Dijkstra's algorithm,
• It computes the least cost path from one node (the source,
Which we will refer to as A) to all other nodes in the network.
Dijkstra's algorithm is iterative and has the property that after
the kth iteration of the algorithm, the least cost paths are
known to k destination nodes, and among the least cost paths
to all destination nodes, these k path will have the k smallest
costs. Let us define the following notation:

• c(i,j): link cost from node i to node j. If nodes i and j are

not directly connected, then c(i,j) = infty. We will assume for
simplicity that c(i,j) equals c(j,i).

• D(v): the cost of path from the source node to destination

v that has currently (as of this iteration of the algorithm) the
least cost.

• p(v): previous node (neighbor of v) along current least cost

path from source to v

• N: set of nodes whose shortest path from the source is

definitively known
let us consider the network in Figure and compute the shortest
path from A to all possible destinations.

A tabular summary of the algorithm's computation is shown in

Table where each line in the table gives the values of the
algorithms variables at the end of the iteration. Let us consider
the few first steps in detail:
In the initialization step, the currently known least path costs
from A to its directly attached neighbors, B, C and D are initialized
to 2, 5 and 1 respectively. Note in particular that the cost to C is set
to 5 (even though we will soon see that a lesser cost path does
indeed exists) since this is cost of the direct (one hop) link from A
to C. The costs to E and F are set to infinity since they are not
directly connected to A.
In the first iteration, we look among those nodes not yet
added to the set N and find that node with the least cost as
of
the end of the previous iteration. That node is D, with a cost
of 1, and thus D is added to the set N. Line 12 of the LS
algorithm is then performed to update D(v) for all nodes v,
yielding the results shown in the second line (step 1) in Table
4.2-1. The cost of the path to B is unchanged. The cost of the
path to C (which was 5 at the end of the initialization)
through node D is found to have a cost of 4. Hence
this lower cost path is selected and C's predecessor along the
shortest path from A is set to D. Similarly, the cost to E
(through D) is computed to be 2, and the table is updated
accordingly.
• In the second iteration, nodes B and E are found to have
the shortest path costs (2), and we break the tie arbitrarily
and add E to the set N so that N now contains A, D, and E. The
cost to the remaining nodes not yet in N, i.e., nodes B, C and
F,
are updated via line 12 of the LS algorithm , yielding the
results shown in the third row in the above table.

When the LS algorithm terminates, we have for each node, its

predecessor along the least cost path from the source node.
For each predecessor, we also have its predecessor and so in
this manner we can construct the entire path from the source
to all destinations.
• Advantages:

Nodes send information about only their attached links

Fast convergence after change

• Disadvantages:
Each node "knows" the whole topology!
Dijkstra's running time grows with topology
Flooding consumes bandwidth
Internet Protocol
The Internet Protocol (IP) is a protocol used For
communicating data across a packet-switched internetwork
using the Internet Protocol Suite, also referred to as TCP/IP.
IP is the primary protocol in the Internet Layer of the Internet
Protocol Suite and has the task of delivering distinguished
protocol datagrams (packets) from the source host to the
destination host solely based on their addresses. For this
purpose the Internet Protocol defines addressing methods and
structures for datagram encapsulation.
The network protocol in the Internet is called the Internet
Protocol, or more commonly, the IP Protocol. There are
currently two versions of the IP protocol in use today
1.IPv4
2.IPv6
IP Addressing
Each IP address is 32 bits long (equivalently, four bytes)
long.
IP addresses are typically written in so-called "dot-decimal
notation", whereby each byte of the address is written in its
decimal form and is separated by a period. For example, a
Typical IP address would be 193.32.216.9. The 193 is the
decimal equivalent for the first 8 bits of the address; the 32
is the decimal equivalent for the second 8 bits of the
address,
etc. Thus, the address 193.32.216.9 in binary notation is:

11000001 00100000 11011000 00001001

This figure provides an example of IP addressing and interfaces. In
this figure there is one router which interconnects three LANs.
First, the router has threes interfaces, labeled 1, 2 and 3. Each of
the router interfaces has its own IP address.

Each host also has its own interface and IP address. Second, all of
the interfaces attached to LAN 1, including a router interface, have
an IP address of the form 223.1.1.xxx . Similarly, all the
interfaces attached to LAN 2 and LAN 3 have IP addresses of the
form 223.1.2.xxx and 233.1.3.xxx, respectively. In other
words, each address has two parts: the first part that specifies the
network; and the second part that addresses a specific host on the
network.
IP datagram Format
The key fields in the IPv4 datagram are the following:

 Version Number: These 4 bits specify the IP protocol

version of the datagram. By looking at the version number,
The router can then determine how to interpret the remainder
of the IP datagram. Different versions of IP use different
datagram formats. The datagram format for the "current“
version of IP, IPv4,

 Header Length: Because an IPv4 datagram can contain a

variable number of options (which are included in the IPv4
datagram header) these 4 bits are needed to determine
where in the IP datagram the data actually begins. Most IP
datagrams do not contain options so the typical IP datagram
has a 20 byte header.

 Datagram Length: This is the total length of the IP

datagram (header plus data) measured in bytes. Since this
field is 16 bits long, the theoretical maximum size of the IP
datagram to 65,535 bytes. However, datagrams are rarely
Greater than 1500 bytes, and are often limited in size to 576
bytes.
 TOS: The type of service (TOS) bits were included in the
IPv4 header to allow different "types" of IP datagrams to be
distinguished from each other, presumably so that they could
be handled differently in times of overload. When the
network is overloaded, for example, it would be useful to be
able to distinguish network control datagrams from datagrams
Carrying data (e.g., HTTP messages). It would also be useful
To distinguish real-time datagrams (e.g., used by an IP
Telephony application) from non-real-time traffic (e.g., FTP).

 Identifier, Flags, Fragmentation Offset: These three

fields have to do with so-called IP fragmentation.

 Time-to-live: The time-to-live (TTL) field is included to

insure that datagrams do not circulate forever in the network.
This field is decremented by one each time the datagram is
processed by a router. If the TTL field reaches 0, the datagram
must be dropped.
 Protocol: This field is only used when an IP datagram
reaches its final destination. The value of this field indicates
The transport-layer protocol at the destination to which the
Data portion of this IP datagram will be passed. For example,
A value of 6 indicates that the data portion is passed to TCP,
while a value of 17 indicates that the data is passed to UDP.

 Source and Destination IP Address: These fields carry

the 32 bit IP address of the source and final destination for
this IP datagram. The use and importance of the destination
address is clear. The source IP address (along with the
source and destination port numbers) is used at the
destination host to direct the application data in the proper
socket.
 Header Checksum: The header checksum aids a router in
detecting bit errors in a received IP datagram. The header
checksum is computed by treating each 2 bytes in the header
as a number and summing these numbers using 1's
complement arithmetic, the 1's complement of this sum, known
as the Internet checksum, is stored in the checksum field.
A router computes the Internet checksum for each received IP
datagram and detects an error condition if the checksum
carried in the datagram does not equal the computed checksum.
Routers typically discard datagrams for which an error has been
detected. Note that the checksum must be recomputed and
restored at each router, as the TTL field, and possibly options
fields as well, may change.

• Options: The optional options fields allows an IP header to

be extended. Header options were meant to be used rarely
hence the decision to save overhead by not including the
information in options fields in every datagram header.
• Data (payload): Finally, we come to the last, and most
important field - the raison d'être for the datagram in the first
place! In most circumstances, the data field of the IP
datagram contains the transport-layer segment (TCP or UDP)
to be delivered to the destination. However, the data field
can
carry other types of data, such ICMP messages
 IPv6
In the early 1990's the Internet Engineering Task force began an
effort to develop a successor to the IPv4 protocol. A prime
motivation for this effort was the realization that the 32-bit IP
address space was beginning to be used up, with new networks
and IP nodes being attached to the Internet (and being
allocated unique IP addresses) at a breathtaking rate. To
respond
to this need of a large IP address space, a new IP protocol, IPv6,
was developed. The designers of IPv6 also took this opportunity
To tweak and augment other aspects of IPv4, based on the
accumulated operational experience with IPv4.
IPv6 Packet Format
The format of the IPv6 packet is shown in Figure
The following packet fields are defined in IPv6:

 version. This four bit field identifies the IP version number.

Not surprisingly, IPv6 carries a value of "6" in this field. Note
that putting a "4" in this field does not create a valid IPv4
packet.

 priority. This four bit field is similar in spirit to the ToS field
we saw in IP version 4. states that values 0 through 7 are to be
used for priority among traffic that is congestion controlled (i.e.,
for which the source will back off on detection of congestion),
while values 8 through 15 are used for non-congestion
controlled traffic, such as constant bit rate real-time
traffic.

 flow label. As discussed above, this field is used

to identify a "flow" of packets.
 payload length. This 16-bit value is treated as an
Unsigned integer given the number of bytes in the IPv6 packet
Following the fixed length, 40 byte packet header.

 next header. This field identifies the protocol to which the

contents (data field) of this packet will be delivered (e.g., to
TCP or UDP). The field uses the same values as the Protocol
field in the IPv4 header.

 hop limit. The contents of this field are decremented by

one by each router that forward the packet. If the hop limit
count reaches zero, the packet is discarded.
 source and destination address. An IP v6 address has
the following structure:

 data. This is the payload portion of the IPv6 packet. When

the packet reaches its destination, the payload will be removed
from the IP packet and passed on to the protocol specified in
the next header field.
Data link Layer
• Provides a well-defined service
interface to the network layer.
Application
Layer
• Determines how the bits of the
physical layer are grouped into
frames (framing). Transport
Layer
Network
• Deals with transmission errors Layer
(CRC and ARQ).
(Data) Link
Layer
• Regulates the flow of frames.

• Performs general link layer Logical Link

management. Control (LLC) Sublayer in
Local Area
Media Access Networks
Control (MAC)
The main tasks of the data link layer are:

• Transfer data from the network layer of one

machine to the network layer of another
machine
• Convert the raw bit stream of the physical
layer into groups of bits (“frames”)

Network Network
Layer Layer
Data Link Data Link
Layer Layer
Physical Physical
Layer Layer
Two types of networks at the data
link layer
– Broadcast Networks: All stations share a single
communication channel
– Point-to-Point Networks: Pairs of hosts (or
routers) are directly connected

Broadcast Network Point-to-Point Network

Typically, local area networks (LANs) are broadcast and

wide area networks (WANs) are point-to-point
 Local Area Networks
• Local area networks (LANs) connect computers within a
building or a enterprise network Almost all LANs are broadcast
Networks Typical topologies of LANs are bus or ring or star
We will work with Ethernet LANs. Ethernet has a bus or star
topology.

Bus LAN Ring LAN

 MAC sublayer
In any broadcast network, the stations must ensure that
only one station transmits at a time on the shared
communication channel The protocol that determines who
can transmit on a broadcast channel are called Medium
Access Control (MAC) protocol The MAC protocol are
Implemented in the MAC sublayer which is the
lower sublayer of the data link layer The higher portion of
the data link layer is often called Logical Link
Control (LLC)

to Network Layer

Logical Link
Control
Medium Access

Layer
Data Link Control

to Physical Layer
Ethernet - Data-Link Layer
• It will be useful to discuss a real data-link layer.
• Ethernet (really IEEE 802.3) is widely used.
• Supported by a variety of physical layer implementations.

• Standard: 802.3, Ethernet II (DIX)

• Speed: 10Mbps -10 Gbps

• Most popular physical layers for Ethernet:
• 10Base5 Thick Ethernet: 10 Mbps coax cable
• 10Base2 Thin Ethernet: 10 Mbps coax cable
• 10Base-T 10 Mbps Twisted Pair
• 100Base-TX 100 Mbps over Category 5 twisted pair
• 100Base-FX 100 Mbps over Fiber Optics
• 1000Base-FX 1Gbps over Fiber Optics
• 10000Base-FX 1Gbps over Fiber Optics (for wide area links)
An Ethernet Frame
• The preamble is a sequence of alternating 1s and 0s used for
synchronization.
• CRC is Cyclic Redundancy Check

Ethernet Addressing :

Each interface looks at every frame and inspects the destination

address. If the address does not match the hardware address of
the interface (or the broadcast address), the frame is discarded.
Some interfaces can also be programmed to recognize multicast
addresses.

Destination Source
Preamble Len DATA CRC
Address Address
 IEEE 802 Standards
IEEE 802 is a family of standards for LANs, which defines an
LLC and several MAC sublayers
Difference for our purposes: Ethernet and 802.3 use differe
nt methods to encapsulate an IP datagram

IEEE 802 standard

IEEE Higher
Reference Layer
802.1 Model
Logical Link
802.2 Control Data Link
Medium Layer
Access
802.11
802.3

802.4

802.5

Control
Physical Physical
Layer Layer
Point-to-Point links
• Many data link connections are point-to-point serial links:
– Dial-in or DSL access connects hosts to access routers
– Routers are connected by
high-speed point-to-point links

• Here, IP hosts and routers are connected by a serial cable

• Data link layer protocols for point-to-point links are simple:

– Main role is encapsulation of IP datagrams
– No media access control needed

Router
Access
Router
Router
Modems

Router Router
Dial-Up Access
Point-to-Point Links
• The frame format of PPP is as shown in the figure

flag addr ctrl protocol data CRC flag

7E FF 03 7E
1 1 1 2 <= 1500 2 1

0021 IP datagram

C021 link control data

8021 network control data

• PPP works in duplex circuit

• Usual maximum frame size is 1500
Switched Networks

Some data link technologies can be used to build

complete networks, with their own addressing, routing,
and forwarding mechanisms. These networks are often
called switched networks.
At the IP layer, a switched network may like a point-to-
point link or like a broadcast link
Some switched networks have a complete protocol suite
R1 R2

Point-to-point link Point-to-point link

H2

Switched network
of data link layer
switches

H1 R3

Ethernet
Physical Layer
The physical layer is also “special” in that it is the only
layer that really does not apply specifically to TCP/I
P.
Even in studying TCP/IP, however, it is still important
To understand its significance and role in relation
To the other layers where TCP/IP protocols reside.
Physical Layer Functions
The following are the main responsibilities of the physical
layer :

• Definition of Hardware Specifications: The details of

operation of cables, connectors, interface cards and other
hardware devices are generally a function of the physical layer

• Encoding and Signaling: The physical layer is responsible

for various encoding and signaling functions that transform
the data from bits that reside within a computer or other
device into signals that can be sent over the network.

• Data Transmission and Reception: After encoding the

data appropriately, the physical layer actually transmits the
data, and of course, receives it. Note that this applies equally
to wired and wireless networks, even if there is no tangible
cable in a wireless network!
Examples of physical Layer
• Twisted pair
• Coaxial cable
• Optical fiber

Twisted Pair : Two insulated wires arranged in a spiral pattern

Copper or steel coated with copper The signal is transmitted
through one wire and a ground reference is transmitted in the
other wire. Typically twisted pair is installed in building
telephone wiring.
Co-axial cable : Discussion divided into two basic categories
for coax used in LANs:
– 50-ohm cable [baseband]
– 75-ohm cable [broadband or single channel baseband]
In general, coax has better noise immunity for higher
frequencies than twisted pair.
Coaxial cable provides much higher bandwidth than twisted
pair

Center
conductor
Dielectric Braided Outer
material outer cover
conducto
r
Optical fiber :
A thin flexible medium capable of conducting
optical rays. Optical fiber consists of a very fine cylinder of
glass (core) surrounded by concentric layers of glass (cladding).
a signal-encoded beam of light (a fluctuating beam) is
transmitted by total internal reflection.
Total internal reflection occurs in the core because it has a
higher optical density (index of refraction) than the cladding.
Attenuation in the fiber can be kept low by controlling the
impurities in the glass

light
cladding jacket

core
It ends here

Thank you. . .