Department of Computer Science & Engineering

COMPUTER NETWORK LAB

KCS-653

B.Tech. CSE 3rd Year (VI Sem.)

Academic Session 2023-24

 INDEX
S.NO List of Program Date Signature
To learn handling and configuration of networking hardware
1 like RJ-45 connector, CAT-6 cable, crimping tool, etc

2 Configuration of router, hub, switch etc. (using real devices or

simulators)
3 Implementation of Stop and Wait Protocol and Sliding
Window Protocol
4 Study of Socket Programming and Client – Server model

Write a code simulating ARP /RARP protocols.

5

6 Write a code simulating PING and TRACEROUTE commands

Create a socket for HTTP for web page upload and download
7

Write a program to implement RPC (Remote Procedure Call)

8
Implementation of Subnetting.
9

Applications using TCP Sockets like

a. Chat
10 b. File Transfer

Applications using TCP and UDP Sockets like

c. DNS
11 d. SNMP

Study of Network simulator (NS).and Simulation of

12
Congestion Control Algorithms using NS
13 Perform a case study about the different routing
algorithms to select the network path with its
optimum andeconomical during data transfer.
i. Link State routing
ii. Flooding
iii. Distance vector
14. Running and using services/commands like ping,
traceroute, arp, telnet, etc.
15. Network packet analysis using tools like Wireshark, tcpdump, etc.
16. Network simulation using tools like Cisco Packet Tracer, NetSim,
OMNeT++, NS2, NS3, etc
 EXPERIMENT-1

Objective: To learn handling and configuration of networking hardware like

RJ-45 connector, CAT-6 cable, crimping tool, etc.

Theory: Ethernet cable Color-coded wiring sequences exist as a cabling industry standard. It allows
cabling technicians to reliably predict how Ethernet cable is terminated on both ends so they can follow
other technicians' work without having to guess or spend time deciphering the function and connections
of each wire pair. Ethernet cable jack wiring follows the T568A and T568B standards.
There are two kinds of Ethernet cables you can make, Straight Through and Crossover.
Standard Cabling:
1. 10BaseT and 100BaseT are most common mode of LAN. You can use UTP category-5 cable for
both modes.

2. A straight cable is used to connect a computer to a hub

CROSSOVER CABLES - The purpose of a Crossover Ethernet cable is to directly connect one
computer to another computer (or device) without going through a router, switch or hub.
 Bulk RJ45 Crimpable Connectors for CAT-6

RJ-45 Crimping tool

A crimping tool is a device used to conjoin two pieces of metal by deforming one or both of them in a
way that causes them to hold each other. The result of the tool's work is called a crimp. A good
example of crimping is the process of affixing a connector to the end of a cable. For instance, network
cables and phone cables are created using a crimping tool (shown below) to join the RJ-45 and RJ-
11 connectors to the both ends of either phone
Cable Crimping steps:
1. Remove the outmost vinyl shield for 12mm at one end of the cable (we call this side A-side).
2. Arrange the metal wires in parallel
3. Insert the metal wires into RJ45 connector on keeping the metal wire arrangement.
4. Set the RJ45 connector (with the cable) on the pliers, and squeeze it tightly.
5. Make the other side of the cable (we call this side B-side) in the same way.
6. After you made it, you don't need to take care of the direction of the cable.

IO connector crimping: Run the full length of Ethernet cable in place, from
endpoint to endpoint, making sure to leave excess.

At one end, cut the wire to length leaving enough length to work, but not too much excess.
Strip off about 2 inches of the Ethernet cable sheath.
Align each of the colored wires according to the layout of the jack.
Use the punch down tool to insert each wire into the jack.
Repeat the above steps for the second RJ45 jack.
Testing the crimped cable using a cable tester:
Step 1 : Skin off the cable jacket 3.0 cm long cable stripper up to cable
Step 2: Untwist each pair and straighten each wire 190 0 1.5 cm long.
Step 3 : Cut all the wires
Step 4 : Insert the wires into the RJ45 connector right white orange left brown the pins facing up
Step 5 : Place the connector into a crimping tool, and squeeze hard so that the handle reaches its full
swing.
Step 6: Use a cable tester to test for proper continuity

Result:

Cable Crimping, Standard Cabling and Cross Cabling, IO connector crimping and testing the crimped
cable using a cable tester are done successfully
 EXPERIMENT-2
Objective: Configuration of router, hub, switch etc. (using real devices or
simulators)

Procedure: Following should be done to understand this practical.

1. Repeater: Functioning at Physical Layer. A Repeater is an electronic device that receives a

signal and retransmits it at a higher level and/or higher power, or onto the other side of an obstruction,
so that the signal can cover longer distances. Repeater have two ports ,so cannot be use to connect for
more than two devices

2. Hub: An Ethernet hub, active hub, network hub, repeater hub, hub or concentrator
is a device for connecting multiple twisted pair or fiber optic Ethernet devices together and making
them act as a single network segment. Hubs work at the physical layer (layer 1) of the OSI model. The
device is a form of multiport repeater. Repeater hubs also participate in collision detection, forwarding a
jam signal to all ports if it detects a collision.

3. Switch: A network switch or switching hub is a computer networking device that connects
network segments. The term commonly refers to a network bridge that processes and routes data at the
data link layer (layer 2) of the OSI model. Switches that additionally process data at the network layer
(layer 3 and above) are often referred to as Layer 3 switches or multilayer switches.

4. Bridge: A network bridge connects multiple network segments at the data link layer (Layer
2) of the OSI model. In Ethernet networks, the term bridge formally means a device that behaves
according to the IEEE 802.1D standard. A bridge and switch are very much alike; a switch being a
bridge with numerous ports. Switch or Layer 2 switch is often used interchangeably with bridge
.Bridges can analyze incoming data packets to determine if the bridge is able to send the given packet
to another segment of the network.

5. Router: A router is an electronic device that interconnects two or more computer networks,
and selectively interchanges packets of data between them. Each data packet contains address
information that a router can use to determine if the source and destination are on the same network, or
if the data packet must be transferred from one network to another. Where multiple routers are used in a
large collection of interconnected networks, the routers exchange information about target system
addresses, so that each router can build up a table showing the preferred paths between any two systems
on the interconnected networks.

6. Gate Way: In a communications network, a network node equipped for interfacing with
another network that uses different protocols.
• A gateway may contain devices such as protocol translators, impedance matching devices, rate
converters, fault isolators, or signal translators as necessary to provide system interoperability. It also
requires the establishment of mutually acceptable administrative procedures between both networks.
• A protocol translation/mapping gateway interconnects networks with different network protocol
technologies by performing the required protocol conversions.
 EXPERIMENT-3
Objective: Implementation of Stop and Wait Protocol and Sliding Window
Protocol.

Theory: It is the simplest flow control method in which the sender will send the packet and then wait
for the acknowledgement by the receiver that it has received the packet then it will send the next
packet.Stop and wait protocol is very easy to implement.

Total time taken to send is,

Ttotal = Tt(data) + Tp + Tq + Tprocess + Tt(ack) + Tp
( since, Tq and Tprocess = 0)
Ttotal = Tt(data) + 2Tp + Tt(ack)
Ttotal = Tt(data) + 2Tp
(when Tt(ack) is negligible)
Efficiency
= useful time / total cycle time
= Tt / (Tt+2Tp)
= 1 / (1+2a) [a = Tp/Tt]

Design
• Sender Site: The data link layer in the sender site waits for the network layer for a data
packet. It then checks whether it can send the frame. If it receives a positive notification from
the physical layer, it makes frames out of the data and sends it. It then waits for an
acknowledgement before sending the next frame.
 • Receiver Site: The data link layer in the receiver site waits for a frame to arrive. When it
arrives, the receiver processes it and delivers it to the network layer. It then sends an
acknowledgement back to the sender.

Algorithm: Sender Site Algorithm of Simplex Stop – and – Wait Protocol for Noiseless Channel
begin
canSend = True; //Allow the first frame to be sent
while (true) //check repeatedly
do
Wait_For_Event(); //wait for availability of packet
if ( Event(Request_For_Transfer) AND canSend) then
Get_Data_From_Network_Layer();
Make_Frame();
Send_Frame_To_Physical_Layer();
canSend = False;
else if ( Event(Acknowledgement_Arrival)) then
Receive_ACK();
canSend = True;
end if
end while
end

Algorithm: Receiver Site Algorithm of Simplex Stop – and – Wait Protocol for Noiseless Channel
begin
while (true) //check repeatedly
do
Wait_For_Event(); //wait for arrival of frame
if ( Event(Frame_Arrival) then
Receive_Frame_From_Physical_Layer();
Extract_Data();
Deliver_Data_To_Network_Layer();
Send_ACK();
end if
end while
end
Algorithm : Sliding Window Protocol

In computer networks sliding window protocol is a method to transmit data on a network. Sliding
window protocol is applied on the Data Link Layer of OSI model. At data link layer data is in the
form of frames. In Networking, Window simply means a buffer which has data frames that needs to
be transmitted.

Both sender and receiver agrees on some window size. If window size=w then after sending w frames
sender waits for the acknowledgement (ack) of the first frame.

As soon as sender receives the acknowledgement of a frame it is replaced by the next frames to be
transmitted by the sender. If receiver sends a collective or cumulative acknowledgement to sender
then it understands that more than one frames are properly received, for eg:- if ack of frame 3 is
received it understands that frame 1 and frame 2 are received properly.

In sliding window protocol the receiver has to have some memory to compensate any loss in
transmission or if the frames are received unordered .

Sliding window works in full duplex mode

It is of two types:-
1. Selective Repeat: Sender transmits only that frame which is erroneous or is lost.
2. Go back n: Sender transmits all frames present in the window that occurs after the error bit
including error bit also.
 Sliding Window Protocol Program in C
#include<stdio.h>

int main()
{
int w,i,f,frames[50];

printf("Enter window size: ");

scanf("%d",&w);

printf("\nEnter number of frames to transmit: ");

scanf("%d",&f);

printf("\nEnter %d frames: ",f);

for(i=1;i<=f;i++)
scanf("%d",&frames[i]);

printf("\nWith sliding window protocol the frames will be sent in the following manner (assuming no
corruption of frames)\n\n");
printf("After sending %d frames at each stage sender waits for acknowledgement sent by the receiver\n\n",w);

for(i=1;i<=f;i++)
{
if(i%w==0)
{
printf("%d\n",frames[i]);
printf("Acknowledgement of above frames sent is received by sender\n\n");
}
else
printf("%d ",frames[i]);
}
if(f%w!=0)
printf("\nAcknowledgement of above frames sent is received by sender\n");

return 0;
}

Output

Enter window size: 3

Enter number of frames to transmit: 5
Enter 5 frames: 12 5 89 4 6
With sliding window protocol the frames will be sent in the following manner (assuming no corruption
of frames)
After sending 3 frames at each stage sender waits for acknowledgement sent by the receiver
12 5 89
Acknowledgement of above frames sent is received by sender
46
Acknowledgement of above frames sent is received by sender
 Program 4
Objective: Study of Socket Programming and Client – Server model
DESCRIPTION:

❖ Socket

A socket is formally defined as an endpoint for communication between an application program,

and the underlying network protocols.

The two modes of services available are

• Connection-oriented service
• Connection less service

Connectionless (UDP) vs Connection-Oriented (TCP) Servers

▪Programmer can choose a connection-oriented server or a connectionless server based on their

applications.
▪ In Internet Protocol terminology, the basic unit of data transfer is a datagram. This is basically
a header followed by some data. The datagram socket is connectionless.
▪ User Datagram Protocol (UDP):
1. Is a connectionless.
2. A single socket can send and receive packets from many different computers.
3. Best effort delivery.
4. Some packets may be lost some packets may arrive out of order.
▪ Transmission Control Protocol (TCP):
1. Is a connection-oriented.
2. A client must connect a socket to a server.
3. TCP socket provides bidirectional channel between client and server.
4. Lost data is re-transmitted.
5. Data is delivered in-order.
6. Data is delivered as a stream of bytes.
7. TCP uses flow control.
▪ It is simple for a single UDP server to accept data from multiple clients and reply.
▪ It is easier to cope with network problems using TCP.
CONNECTION-ORIENTED

Connection oriented model defines a reliable delivery service. The figure shows a sequence of
system calls for connection oriented communication. The server begins by carrying out a passive open
as follows. The socket call creates a TCP socket. The bind call then binds the well-known port number
of the server to the socket. The listen call turns the socket into a listening socket that can accept
incoming connections from clients. Finally, the accept call puts the server process to sleep until the
arrival of a client connection. The client does an active open. The socket call creates a socket on the
client side, and the connect call establishes the TCP connection to the server with the specified
destination socket address. When the TCP connection is completed, the accept function at the server
wakes up and returns the descriptor for the given connection, namely, the source IP address, the source
port number, destination IP address and destination port number. The client and server are now ready to
exchange information.

Socket system calls for connection-oriented service

▪ Following figure illustrates the example of client/server relationship of the socket APIs for
connection-oriented protocol (TCP).

CONNECTION-LESS

In a connection –less mode an application program sends its data immediately without waiting
for connection establishment. As a result the application program may waste its time by sending data
when the other end is not ready to accept it. Moreover,data may not arrive at the other end if the
network decides to discards it. If data arrives at the destination, it may not arrive in the same order as it
was transmitted.

The connectionless mode is often said to provide best effort service, since the network would
try its best to deliver the information but cannot guarantee the delivery.
The figure shows the sequence of system calls for a connectionless communication. No connection is
established prior to data transfer. The recvfrom call returns when a Complete UDP data gram has been
received.

Socket system calls for connection less service

▪ The following figure illustrates the example of client/server relationship of the socket APIs for a
connectionless protocol (UDP).

Simple Echo Client Server Using UDP:

stdin fgets Send to recvfrom

UDP UDP
Client Server

stdout fput recvfromsendto

❖ Types of sockets

In general, 3 types of sockets exist on most Unix systems:

➢ Stream sockets
➢ Datagram sockets and
➢ Raw sockets

Stream sockets are used for stream connections, i.e. connections that exist for a long duration. TCP
connections use stream sockets.
Datagram sockets are used for short-term connections that transfer a single packet across the network
before terminating. The UDP protocol uses such sockets, due to its connection-less nature.

Raw sockets are used to access low-level protocols directly, bypassing the higher protocols. They are
the means for a programmer to use the IP protocol, or the physical layer of the network, directly. Raw
sockets can therefor be used to implement new protocols on top of the low-level protocols. Naturally,
they are out of our scope.
Socket API functions

Create a new socket and return its descriptor.

socket()
Associate a socket with a port and address.
bind()
Establish queue for connection requests.
listen()
Accept a connection request.
accept()
Initiate a connection to a remote host.
connect()
Receive data from a socket descriptor.
recv()
Send data to a socket descriptor.
send()
Reads from files, devices, sockets etc.
read()
Writes to files, devices, sockets etc.
write()
“One-way” close of a socket descriptor.
close()
Allows you to cut off communication in a certain direction, or both ways just like
shutdown() close() does.
Socket API Functions

socket()

socket() creates an endpoint for communication and returns a file descriptor for the socket. socket()
takes three arguments:

• domain, which specifies the protocol family of the created socket. For example:
o PF_INET for network protocol IPv4 or
o PF_INET6 for IPv6.
o PF_UNIX for local socket (using a file).
• type, one of:
o SOCK_STREAM (reliable stream-oriented service or Stream Sockets)
 o SOCK_DGRAM (datagram service or Datagram Sockets)
o SOCK_SEQPACKET (reliable sequenced packet service), or
o SOCK_RAW (raw protocols atop the network layer).
• protocol specifying the actual transport protocol to use. The most common are IPPROTO_TCP,
IPPROTO_SCTP, IPPROTO_UDP, IPPROTO_DCCP. These protocols are specified in
<netinet/in.h>. The value “0” may be used to select a default protocol from the selected domain
and type.

The function returns -1 if an error occurred. Otherwise, it returns an integer representing the newly-
assigned descriptor.
Prototype
int socket(int domain,int type,int protocol);

bind()

bind() assigns a socket an address. When a socket is created using socket(), it is only given a protocol
family, but not assigned an address. This association with an address must be performed with the bind()
system call before the socket can accept connections to other hosts. bind() takes three arguments:

• sockfd, a descriptor representing the socket to perform the bind on

• my_addr, a pointer to a sockaddr structure representing the address to bind to.
• addrlen, a socklen_t field specifying the size of the sockaddr structure.

Bind() returns 0 on success and -1 if an error occurs.

Prototype

int bind(intsockfd,conststructsockaddr*my_addr,socklen_taddrlen);

listen()

After a socket has been associated with an address, listen() prepares it for incoming connections.
However, this is only necessary for the stream-oriented (connection-oriented) data modes, i.e., for
socket types (SOCK_STREAM, SOCK_SEQPACKET). listen() requires two arguments:

• sockfd, a valid socket descriptor.

• backlog, an integer representing the number of pending connections that can be queued up at
any one time. The operating system usually places a cap on this value. Once a connection is
accepted, it is dequeued. On success, 0 is returned. If an error occurs, -1 is returned.
Prototype

int listen(intsockfd,int backlog);

accept()

When an application is listening for stream-oriented connections from other hosts, it is notified of such
events (cf. select() function) and must initialize the connection using the accept() function. Accept()
creates a new socket for each connection and removes the connection from the listen queue. It takes the
following arguments:

• sockfd, the descriptor of the listening socket that has the connection queued.
• cliaddr, a pointer to a sockaddr structure to receive the client's address information.
• addrlen, a pointer to a socklen_t location that specifies the size of the client address structure
passed to accept(). When accept() returns, this location indicates how many bytes of the
structure were actually used.

The accept() function returns the new socket descriptor for the accepted connection, or -1 if an error
occurs. All further communication with the remote host now occurs via this new socket.

Datagram sockets do not require processing by accept() since the receiver may immediately respond to
the request using the listening socket.

Prototype

int accept(intsockfd,structsockaddr*cliaddr,socklen_t*addrlen);

connect()

The connect() system call connects a socket, identified by its file descriptor, to a remote host specified
by that host's address in the argument list.

Certain types of sockets are connectionless, most commonly user datagram protocol sockets. For these
sockets, connect takes on a special meaning: the default target for sending and receiving data gets set to
the given address, allowing the use of functions such as send() and recv() on connectionless sockets.

connect() returns an integer representing the error code: 0 represents success, while -1 represents an
error.

Prototype

int connect(intsockfd,conststructsockaddr*serv_addr,socklen_taddrlen);

❖ Sending and receiving data over a socket

After a connection is established, there are several ways to send information over the socket.

read()

The most common way of reading data from a socket is using the read () system call, which is defined
like this:

int read(int socket, char *buffer, intbuflen);

• socket - The socket from which we want to read.

• buffer - The buffer into which the system will write the data bytes.
• buflen - Size of the buffer, in bytes (actually, how much data we want to read).

The read system call returns one of the following values:

• 0 - The connection was closed by the remote host.

• -1 - The read system call was interrupted, or failed for some reason.
• n - The read system call put 'n' bytes into the buffer we supplied it with.

Note that read() might read less than the number of bytes we requested, due to unavailability of buffer
space in the system.

write()

The most common way of writing data to a socket is using the write() system call, which is defined like
this:

int write(int socket, char *buffer, intbuflen);

• socket - The socket into which we want to write.

• buffer - The buffer from which the system will read the data bytes.
• buflen - Size of the buffer, in bytes (actually, how much data we want to write).

The write system call returns one of the following values:

• 0 - The connection was closed by the remote host.

• -1 - The write system call was interrupted, or failed for some reason.
• n - The write system call wrote 'n' bytes into the socket.

Note that the system keeps internal buffers, and the write system call write data to those buffers, not
necessarily directly to the network. Thus, a successful write() doesn't mean the data arrived at the other
end, or was even sent onto the network. Also, it could be that only some of the bytes were written, and
not the actual number we requested. It is up to us to try to send the data again later on, when it's
possible, and we'll show several methods for doing just that.

sendto() and recvfrom() for DATAGRAM (UDP)

▪ Since datagram sockets aren’t connected to a remote host, we need to give the destination
address before we send a packet.
▪ The prototype is:
intsendto(intsockfd, const void *msg, intlen, unsigned int flags, const

structsockaddr *to, inttolen);

▪ This call is basically the same as the call to send() with the addition of two other pieces of
information.
▪ to is a pointer to a structsockaddr (which you’ll probably have as a structsockaddr_in and cast it
at the last minute) which contains the destination IP address and port.
▪ tolen can simply be set to sizeof(structsockaddr).
▪ Just like with send(), sendto() returns the number of bytes actually sent (which, again, might be
less than the number of bytes you told it to send!), or -1 on error.

▪ Equally similar are recv() and recvfrom().

The prototype of recvfrom() is:

intrecvfrom(intsockfd, void *buf, intlen, unsigned int flags, struct

sockaddr *from, int *fromlen);

▪ Again, this is just like recv() with the addition of a couple fields.
▪ from is a pointer to a local structsockaddr that will be filled with the IP address and port of the
originating machine.
▪ fromlen is a pointer to a local int that should be initialized to sizeof(structsockaddr). When the
function returns, fromlen will contain the length of the address actually stored in from.
recvfrom() returns the number of bytes received, or -1 on error (with errno set accordingly).
▪ Remember, if you connect() a datagram socket, you can then simply use send() and recv() for
all your transactions.
▪ The socket itself is still a datagram socket and the packets still use UDP, but the socket interface
will automatically add the destination and source information for you.
Socket Address

structsockaddr_in
structsockaddr_in {
u_charsin_len;
u_shortsin_family; // Address family
u_shortsin_port; // Port number
struct in_addrsin_addr; // Internet or IP address
char sin_zero[8]; // Same size as structsockaddr
};
• The sin_family field is the address family (always AF_INET for TCP and UDP).
• The sin_port field is the port number, and the sin_addr field is the Internet address. The
sin_zero field is reserved, and you must set it to hexadecimal zeroes.
• Data type structin_addr - this data type is used in certain contexts to contain an Internet host
address. It has just one field, named s_addr, which records the host address number as an
unsigned long int.
• sockaddr_in is a "specialized" sockaddr.
• sin_addr could be u_long.
• sin_addr is 4 bytes and 8 bytes are unused.
• sockaddr_in is used to specify an endpoint.
• The sin_port and sin_addr must be in Network Byte Order.
 EXPERIMENT-5
Objective: Write a code simulating ARP /RARP protocols
Theory:

Address Resolution Protocol (ARP) – Address Resolution Protocol is a communication protocol

used for discovering physical address associated with given network address. Typically, ARP is a
network layer to data link layer mapping process, which is used to discover MAC address for given
Internet Protocol Address.
In order to send the data to destination, having IP address is necessary but not sufficient; we also
need the physical address of the destination machine. ARP is used to get the physical address (MAC
address) of destination machine.

Before sending the IP packet, the MAC address of destination must be known. If not so, then sender
broadcasts the ARP-discovery packet requesting the MAC address of intended destination. Since
ARP-discovery is broadcast, every host inside that network will get this message but the packet will
be discarded by everyone except that intended receiver host whose IP is associated. Now, this
receiver will send a unicast packet with its MAC address (ARP-reply) to the sender of ARP-
discovery packet. After the original sender receives the ARP-reply, it updates ARP-cache and start
sending unicast message to the destination.
Reverse Address Resolution Protocol (RARP) –

Reverse ARP is a networking protocol used by a client machine in a local area network to request its
Internet Protocol address (IPv4) from the gateway-router’s ARP table. The network administrator
creates a table in gateway-router, which is used to map the MAC address to corresponding IP
address.
When a new machine is setup or any machine which don’t have memory to store IP address, needs an
IP address for its own use. So the machine sends a RARP broadcast packet which contains its own
MAC address in both sender and receiver hardware address field.

A special host configured inside the local area network, called as RARP-server is responsible to reply
for these kind of broadcast packets. Now the RARP server attempt to find out the entry in IP to MAC
address mapping table. If any entry matches in table, RARP server send the response packet to the
requesting device along with IP address.
• LAN technologies like Ethernet, Ethernet II, Token Ring and Fiber Distributed Data Interface
(FDDI) support the Address Resolution Protocol.
• RARP is not being used in today’s networks. Because we have much great featured protocols
like BOOTP (Bootstrap Protocol) and DHCP( Dynamic Host Configuration Protocol).

C Program To Simulate ARP/RARP:

//ARP SERVER

#include<stdio.h>
#include<sys/types.h>
#include<sys/shm.h>
#include<string.h>

main()
{
int shmid, a, i;
char *ptr, *shmptr;
shmid=shmget(3000,10,IPC_CREAT | 0666);
shmptr=shmat(shmid,NULL,0);
ptr=shmptr;
for(i=0;i<3;i++)
{
puts("enter the mac");
scanf("%s",ptr);
a=strlen(ptr);
printf("string length:%d",a);
ptr[a]= ' ' ;
puts("enter ip");
ptr=ptr+a+1;
scanf("%s",ptr);
ptr[a]='\n' ;
ptr= ptr+a+1;
}
ptr[strlen(ptr)]= '\0';
printf("\n ARP table at serverside is=\n%s", shmptr);
shmdt(shmptr);
}

ARP table at serverside is

a.b.c.d 1.2.3.4
e.f.g.h 5.6.7.8
i.j.k.l 9.1.2.3

//ARP CLIENT

#include<stdio.h>
#include<string.h>
#include<sys/types.h>
#include<sys/shm.h>
main()
{
int shmid,a;
char *ptr, *shmptr;
char ptr2[51], ip[12], mac[26];
shmid=shmget(3000,10,0666);
shmptr=shmat(shmid,NULL,0);
puts("the arp table is");
printf("%s",shmptr);
printf("\n1.ARP\n 2.RARP\n 3.EXIT\n");
scanf("%d",&a);
switch(a)
{
case 1:
puts("enter ip address");
scanf("%s",ip);
ptr=strstr(shmptr, ip);
ptr-=8;
sscanf(ptr,"%s%*s",ptr2);
printf("mac addr is %s",ptr2);
break;
case 2:
puts("enter mac addr");
scanf("%s",mac);
ptr=strstr(shmptr, mac);
sscanf(ptr,"%*s%s",ptr2);
printf("%s",ptr2);
break;
case 3:
exit(1);
}}

SAMPLE INPUT OUTPUT:

the arp table is

a.b.c.d 1.2.3.4
e.f.g.h 5.6.7.8
i.j.k.l 9.1.2.3

1.ARP
2.RARP
3.EXIT
enter your choice: 1
enter ip address: 1.2.3.4
mac addr is a.b.c.d

enter your choice:2

enter mac address: e.f.g.h
ip addr is 5.6.7.8
 EXPERIMENT-6
Objective: Write a code simulating PING and TRACEROUTE commands

Theory:
In computer networks, data is sent in small blocks known as packets. Each packet is transmitted
individually and may also follow a different route to reach the destination. Once all these packets of
the original message reach the destination, they are re-assembled to form the original message. But,
sometimes, it may happen that the webserver is down, network congestion, or some other technical
glitch is there, that may prevent the message from reaching the destination. To diagnose such
congestions and network failures, we use two common programs namely Ping and Traceroute.
Ping – It is a utility that helps one to check if a particular IP address is accessible or not. Ping works
by sending a packet to the specified address and waits for the reply. It also measures round trip time
and reports errors.
Ping is also used in checking if the computers on a local network are active. For this, the user has to
go in command prompt and type: ping 127.0.0.1, and if the address is active, the ping would return a
message like this :
Pinging 127.0.0.1 with 32 bytes of data
Reply from 127.0.0.1: bytes=32 time<10ms TTL=32
Reply from 127.0.0.1: bytes=32 time<10ms TTL=32
Reply from 127.0.0.1: bytes=32 time<10ms TTL=32
Reply from 127.0.0.1: bytes=32 time<10ms TTL=32
The IP address 127.0.0.1 is the address of the local host and would receive a ping reply even if the
sender is not connected to the internet.
Traceroute – It is a utility that traces a packet from your computer to the host, and will also show the
number of steps (hops) required to reach there, along with the time by each step. Traceroute works by
sending the packets of data with low survival time (Time to Live – TTL) which specifies how many
steps (hops) can the packet survive before it is returned. When a packet can’t reach the final
destination and expires at an intermediate step, that node returns the packet and identifies itself. So,
by increasing the TTL gradually, Traceroute is able to identify the intermediate hosts. If any of the
hops come back with “Request timed out”, it denotes network congestion and a reason for slow
loading Web pages and dropped connections.

The main difference between Ping and Traceroute is that Ping is a quick and easy utility to tell if the
specified server is reachable and how long will it take to send and receive data from the server
whereas Traceroute finds the exact route taken to reach the server and time taken by each step (hop).
Algorithm:

Server

Step1: Start the program.

Step2: Import necessary packages.
Step3: Initialize the ping server with both sockets as null value.
Step4: Start the server socket.
Step5: At the client give the IP address of the server(by using ifconfig command in command
prompt).
Step6: The client program is then started by starting socket.
Step7: At the receiver end, the client is pinged and traced. Step8: Stop the program.

Client
Step1: Start the program.
Step2: Import necessary packages.
Step3: Initialize the ping client with both sockets as null value.
Step4: Start the socket.
Step5: Get the IP address of the server.
Step6: Ping the server.
Step7: At the receiver end, the server is pinged and traced.
Step8: Stop the program.

pingclient.java
/*…localhostport name and 5555-port number…*/
Socket s=new Socket("127.0.0.1",5555);

/*… Get an input file handle from the socket and read the input…*/
DataInputStream dis=new DataInputStream(s.getInputStream()); PrintStream out=new
PrintStream(s.getOutputStream()); while(c<4){
…returns the current time in milliseconds…*/
t1=System.currentTimeMillis();
str="Welcome to network programming
world"; out.println(str);
/*…readline() method read a line of text…*/
System.out.println(dis.readLine());
t2=System.currentTimeMillis();
System.out.println(";TTL="+(t2-t1)+"ms"); c++;

pingserver.java

/*…ServerSocket object is used to establish the communication with clients…*/

 ServerSocketss=new ServerSocket(5555);
/*…accept(): Used to accept the client request…*/
Socket
s=ss.accept(); int
c=0; while(c<4) {
/*… Get an input file handle from the socket and read the input…*/
DataInputStream dis=new DataInputStream(s.getInputStream());
PrintStream out=new PrintStream(s.getOutputStream());
/*…readline() method read a line of text…*/
String str=dis.readLine();
out.println("Reply from"+InetAddress.getLocalHost()+";Length"+str.length()); c++;

Sample Output:

Viva questions:
1. Dose ping can measure round trip time?

2. What is ping sweep to trace the path from source to destination?

4. Difference between traceroute and ping.

5.What utility is used to find the number of routers between a source and destination?

6. Which protocol does ping use?

7. What is the ‘ping’ command useful for?

8. How do you trace route an ip address?

 EXPERIMENT-7

Objective: Create a socket for HTTP for web page upload and download

Concept:
Concurrent Server: The server can be iterative, i.e. it iterates through each client and serves one
request at a time. Alternatively, a server can handle multiple clients at the same time in parallel,
and this type ofa server is called a concurrent server.

Algorithm:
Server

Step 1: Create a socket and bind to the address. Leave socket unconnected.
Step 2 : Leave socket in passive mode, making it ready for use by a server.
Step 3: Repeatedly call accept to receive the next request from a client to handle the response with
thethrough socket.
Client
Step 1: Begin with a connection passed from the server (i.e., a socket for the connection).
Step 2: Use input streams; get the message from user to be given to the server.
Step 3: Use input streams read message given by server and print it.
Step 4: Use output streams to write message to the server.
Step 5: Close the connection and exit, i.e., slave terminates after handling all requests from one
client.

Sample Program:
ConServer.java
/*… Register service on port 8020…*/
ServerSocketss=new ServerSocket(8500);
System.out.println("Waiting for client...");
while(true)

/*... ServerSocket in order to listen for and accept connections from clients...*/
{Socket s=ss.accept();
/*…getInputStream()-This method take the permission to write the data from client program to
server program and server program to client program…*/
BufferedReader br=new BufferedReader(new InputStreamReader(s.getInputStream()));
cli_name=br.readLine();
System.out.println("\nCLIENT NAME: "+cli_name);
no=Integer.parseInt(br.readLine());
sq=no*no;
PrintWriter pw=new PrintWriter(s.getOutputStream(),true);
pw.println(sq);
System.out.println("OUTPUT - The square of "+no+" is "+sq);}}}

ConClient1.java

{Socket s=new Socket("localhost",8500);

BufferedReader br=new BufferedReader(new InputStreamReader(System.in));

/*...Integer.parseInt() java method is used primarily in parsing a String method

argument intoan Integer object. The Integer object is a wrapper class for the int primitive
data type ...*/

int num=Integer.parseInt(br.readLine());

/* …getOutputStream()-This method is used to take the permission to read data from client
system by the server or from the server system by the client…*/
PrintWriterpw=new PrintWriter(s.getOutputStream(),true);
pw.println("Client 1");
pw.println(num);
BufferedReader br1=new BufferedReader(new InputStreamReader(s.getInputStream()));
intsqu=Integer.parseInt(br1.readLine());
System.out.println("Square of "+num+" is "+squ+"\n");

ConClient2.java
Socket s=new Socket("localhost",8500);
BufferedReader br=new BufferedReader(new InputStreamReader(System.in));
System.out.println("\nCLIENT 2:\nEnter the number to find square: ");
intnum=Integer.parseInt(br.readLine());
PrintWriter pw=new PrintWriter(s.getOutputStream(),true);
pw.println("Client 2");
 pw.println(num);
BufferedReader br1=new BufferedReader(new InputStreamReader(s.getInputStream()));
intsqu=Integer.parseInt(br1.readLine());
System.out.println("Square of "+num+" is "+squ+"\n"); s.close();

Sample Output:
 EXPERIMENT-8

Objective: Write a program to implement RPC (Remote Procedure Call)

Theory: Remote Procedure Call (RPC) is a powerful technique for constructing distributed,
client-server based applications. It is based on extending the conventional local procedure calling
so that the called procedure need not exist in the same address space as the calling procedure.
The two processes may be on the same system, or they may be on different systems with a network
connecting them.

1. The calling environment is suspended, procedure parameters are transferred across the network to
the environment where the procedure is to execute, and the procedure is executed there.
2. When the procedure finishes and produces its results, its results are transferred back to the calling
environment, where execution resumes as if returning from a regular procedure call.

The Programmer's Interface to RPC

This section addresses the C interface to RPC and describes how to write network applications using
RPC. For a complete specification of the routines in the RPC library, see the rpc and related man pages.
Simplified Interface

The simplified interface is the easiest level to use because it does not require the use of any other RPC
routines. It also limits control of the underlying communications mechanisms. Program development at
this level can be rapid, and is directly supported by the rpcgen compiler. For most applications, rpcgen
and its facilities are sufficient. Some RPC services are not available as C functions, but they are
available as RPC programs. The simplified interface library routines provide direct access to the RPC
facilities for programs that do not require fine levels of control.

Routines such as rusers are in the RPC services library librpcsvc. rusers.c, below, is a program that
displays the number of users on a remote host. It calls the RPC library routine, rusers.

Program:
#include"rpc/rpc.h"
#include"square.h"
#include"stdio.h"
#include"stdlib.h"
#include"math.h"

square_out *squareproc_1_svc(square_in *inp,struct svc_req *rqstp)

{
static square_out out;
out.res1= inp->arg1 * inp->arg1;
return(&out);
}

// CLIENT FILENAME: client.c

#include"errno.h"
#include"rpc/rpc.h"
#include"square.h"
#include"stdio.h"
#include"stdlib.h"
#include"math.h"

int main(int argc,char **argv)

{
CLIENT *cl;
square_in in;
square_out *outp;
f(argc!=3)
{
printf("\n\n error:insufficient arguments!!!");
exit(-1);
}

cl=clnt_create(argv[1],SQUARE_PROG,SQUARE_VERS,"tcp");
in.arg1=atol(argv[2]);

if(cl==NULL)
{
printf("\nerror:%s",strerror(errno));
exit(-1);
}

if((outp=squareproc_1(&in,cl))==NULL)
{
printf("\nerror :%s",clnt_sperror(cl,argv[1]));
exit(-1);
}

printf("\n\n result is : %ld",outp->res1);

exit(0);
}

// .h FILENAME: square.h

struct square_in
{
/*input arg*/
long arg1;
};
 struct square_out
{
/*op result*/
long res1;
};

program SQUARE_PROG
{
version SQUARE_VERS
{
square_out SQUAREPROC(square_in)=1; /*proc no=1*/
}=1; /*version no*/
}=0x31230000;/*prog no*/

Output:
[root@localhost~]#rpcgen -C square.x
[root@localhost~]#cc -c client.c -o client.o
[root@localhost~]#cc -c square_clnt.c -o square_clnt.o
[root@localhost~]#cc -c square_xdr.c -o square.xdr.o
[root@localhost~]#cc -o client client.o square_clnt.o square_xdr.o
[root@localhost~]#cc -c client.c server.c square_xdr.c
[root@localhost~]#cc -c server.c -o server.o
[root@localhost~]#cc -c square_svc.c -o square_svc.o
[root@localhost~]#cc -o server server.o square_svc.o square_xdr.o
[root@localhost~]#./server &
[1] 2264
[root@localhost~]#./client localhost 4
result is: 16

EXPERIMENT-9
Objective: Implementation of Subnetting
Theory:
If an organization was granted a large block in class A or B, it could divide the addresses
into several contiguous groups and assign each group to smaller networks (called subnets) or, in
 rare cases, share part of the addresses with neighbours.

Algorithm:

Step1: Get the input from the user by using scanner method.
Step 2: Read the input by using nextLine() and store it.
Step 3: Split the string based on string by using
split(“\\”)Step4 : Convert it into binary.
Step 5: calculating the network mask by using math and logarithmic
Step 6: get the first address by ANDding the last n bits with 0.
Step7 : get the last address by ANDding the last n bits with 1.

Sample Coding:

//…Calculation of mask…//

int bits = (int)Math.ceil(Math.log(n)/Math.log(2));

/*eg if address = 120, log 120/log 2 gives log to the base 2 => 6.9068, ceil gives us upper integer */
System.out.println("Number of bits required for address = "+bits);
int mask = 32-bits;
System.out.println("The subnet mask is = "+mask);
//…Calculation of first address and last address…//
intfbip[] = new int[32];
for(int i=0; i<32;i++) fbip[i] = (int)bip.charAt(i)-48;
//convert cahracter 0,1 to integer 0,1
for(int i=31;i>31-bits;i--)//Get first address by ANDing last n bits with 0 fbip[i] &= 0;
String fip[] = {"","","",""};
for(int i=0;i<32;i++)
fip[i/8] = new String(fip[i/8]+fbip[i]);
System.out.print("First address is = ");
for(int i=0;i<4;i++)
{
System.out.print(Integer.parseInt(fip[i],2));
 if(i!=3) System.out.print(".");
}

Sample Output:

Viva questions:

1. What are the advantages of subnetting?

2. Your router has the following IP address on Ethernet0: 172.16.2.1/23. Which of the following
can bevalid host IDs on the LAN interface attached to the router?
a. 172.16.1.100 2) 172.16.1.198 3) 172.16.2.255 4) 172.16.3.0
b. 1 only
c. 2 and 3 only
d. 3 and 4 only
e. None of the above
3. A network administrator is connecting hosts A and B directly through their Ethernet
interfaces, as shown in the illustration. Ping attempts between the hosts are unsuccessful. What
can be done to provide connectivity between the hosts?
4. What is the maximum number of IP addresses that can be assigned to hosts on a local subnet
that uses the 255.255.255.224 subnet mask?
5. If an Ethernet port on a router were assigned an IP address of 172.16.112.1/25, what would
be the valid subnet address of this host?
6. What are the network address, broadcast address, and the subnet mask for a host with the IP
Address below?IP Address: 101. 39. 85. 201/ 23 Network Address: 101.39.84.0 Subnet Mask:
 255.255.254.0 Broadcast Address: 101.39.85.255
7. On a VLSM network, which mask should you use on point-to-point WAN links in order to
reduce the waste of IP addresses?
8. To test the IP stack on your local host, which IP address would you ping?

EXPERIMENT-10(a)
Objective: Applications using TCP Sockets like Chat
A server program to establish the socket connection with the client for performing chat.
A client program which on establishing a connection with the server for performing chat.
Concept:
It uses TCP socket communication .We have a server as well as a client.
Both can be run in the same machine or different machines. If both are running in the machine, the
address to be given at the client side is local host address.
If both are running in different machines, then in the client side we need to specify the ip address of
machine in which server application is running.

Algorithm:
Server
Step1: Start the program and create server and client sockets.
Step2: Use input streams to get the message from user.
Step3: Use output streams to send message to the client.
Step4: Wait for client to display this message and write a new one to be displayed by the server.
Step5: Display message given at client using input streams read from socket.
Step6: Stop the program.
Client
Step1: Start the program and create a client socket that connects to the required host and port.
Step2: Use input streams read message given by server and print it.
Step3: Use input streams; get the message from user to be given to the server.
Step4: Use output streams to write message to the server.
Step5: Stop the program.

Sample Program: GossipServer.java

ServerSocketsersock = new serverSocket(3000);

System.out.println("Server ready for chatting");
Socket sock = sersock.accept( );

/*…reading from keyboard (keyRead object)…*/

BufferedReader keyRead = new BufferedReader(new InputStreamReader(System.in));

/*…sending to client (pwrite object)…*/

OutputStreamostream = sock.getOutputStream();
PrintWriterpwrite = new PrintWriter(ostream,
true);
/*… receiving from server ( receiveRead object)…*/
InputStreamistream = sock.getInputStream();
BufferedReader receiveRead = new BufferedReader(new InputStreamReader(istream));
String receiveMessage, sendMessage;
while(true)
{if((receiveMessage = receiveRead.readLine()) != null)
{System.out.println(receiveMessage); }

GossipClient.java
Socket sock = new Socket("127.0.0.1", 3000);

/*…reading from keyboard (keyRead object)…*/

BufferedReader keyRead = new BufferedReader(new InputStreamReader(System.in));

/*…sending to client (pwrite object)…*/

OutputStreamostream = sock.getOutputStream();

PrintWriterpwrite = new PrintWriter(ostream, true);

/*… receiving from server ( receiveRead object)…*/

InputStreamistream = sock.getInputStream();

BufferedReader receiveRead = new BufferedReader(new InputStreamReader(istream));

System.out.println("Start the chitchat, type and press Enter key");

String receiveMessage, sendMessage;
while(true)
{sendMessage = keyRead.readLine(); /*... keyboard reading
...*/ pwrite.println(sendMessage); /*... sending to server...*/
pwrite.flush(); /*... flush the data...*/

if((receiveMessage = receiveRead.readLine()) != null) /*...receive from server...*/

{System.out.println(receiveMessage);} /*... displaying at DOS prompt...*/

Sample Output:

EXPERIMENT-10(b)
Objective: Applications using TCP Sockets like file Transfer

Algorithm

Server
 Step1: Import java packages and create class file server.
Step2: Create a new server socket and bind it to the port.
Step3: Accept the client connection
Step4: Get the file name and stored into the BufferedReader.
Step5: Create a new object class file and realine.
Step6: If file is exists then FileReader read the content until EOF is reached.
Step7: Stop the program.

Client

Step1: Import java packages and create class file server.

Step2: Create a new server socket and bind it to the port.
Step3: Now connection is established.
Step4: The object of a BufferReader class is used for storing data content which has been
retrieved from socket object.
Step5: The content of file is displayed in the client window and the connection is closed.
Step6: Stop the program.

Sample Code:Se.java

{/*… Register service on port 15123…*/

ServerSocketserverSocket = new ServerSocket(15123);

/*… Wait and accept a connection…*/

Socket socket = serverSocket.accept();
System.out.println("Accepted connection : " + socket);

/*… enter the source file name which is to be transferred to client…*/

File transferFile = new File ("sunithanandhinifiletransfer.doc");
byte [] bytearray = new byte [(int)transferFile.length()];
/*…FileInputStream is meant for reading streams of raw bytes such as image data…*/
FileInputStream fin = new FileInputStream(transferFile);
BufferedInputStream bin = new BufferedInputStream(fin);bin.read(bytearray,0,bytearray.length);

/*… Get a communication stream associated with the

socket…*/OutputStreamos = socket.getOutputStream();
System.out.println("Sending Files...");
os.write(bytearray,0,bytearray.length);
os.flush();socket.close();

Cl.java
/*… Open your connection to a server, at port 15123…*/
Socket socket = new Socket("127.0.0.1",15123); byte [] bytearray = new byte [filesize];
InputStream is = socket.getInputStream();

/*… enter the destinationfile name which is to be transferred …*/

FileOutputStream fos = new
FileOutputStream("kalpanasonikafiletransfer.doc"); BufferedOutputStream
bos = new BufferedOutputStream(fos);
bytesRead = is.read(bytearray,0,bytearray.length);
currentTot = bytesRead;

do { bytesRead = is.read(bytearray, currentTot, (bytearray.length-currentTot)); if(bytesRead>= 0)

currentTot += bytesRead;}
while(bytesRead> -1);
bos.write(bytearray, 0 , currentTot);
bos.flush(); bos.close(); socket.close(); } }

Sample Output:
 EXPERIMENT-11
Objective: Applications using TCP and UDP Sockets like DNS & SNMP
Theory:

1. The DNS client program sends a request to a DNS server to map the e-mail address
to thecorresponding IP address.
2. When the Internet was small, mapping was done by using a host file. The host file had
only twocolumns: name and address.
 3. The host that needs mapping can contact the closest computer holding the needed
information. This method is used by the Domain Name System (DNS).
Algorithm:
Server

Step1: Start the program.

Step2: Create the socket for the server.
Step3: Bind the socket to the port.
Step4: Listen for the incoming client connection.
Step5: Receive the IP address from the client to be resolved.
Step6: Get the domain name for the client.
Step7: Check the existence of the domain in the server.
Step8: If domain matches then send the corresponding address to the client.
Step9: Stop the program execution

Client
Step1: Start the Program.
Step2: Create the socket for the client.
Step3: Connect the socket to the Server.
Step4: Send the host name to the server to be resolved.
Step5: If the server corresponds then print the address and terminate the process

Sample Program:
Clientdns12.java

/*... datagram socket is the sending or receiving point for a packet delivery service.
DatagramSocket client=new DatagramSocket();

/*...InetAddress class provides methods to get the IP of any host name...*/

InetAddress addr=InetAddress.getByName("127.0.0.1");
byte[] sendbyte=new byte[1024];
byte[] receivebyte=new byte[1024];
BufferedReader in=new BufferedReader(new InputStreamReader(System.in));
System.out.println("Enter the DOMAIN NAME or IP adress:");
String str=in.readLine(); sendbyte=str.getBytes();
 /*…send the data to the server(data,length,ip address and port number)…*/ DatagramPacket
sender=new DatagramPacket(sendbyte,sendbyte.length,addr,1309); client.send(sender);
DatagramPacket receiver=new DatagramPacket(receivebyte,receivebyte.length);
client.receive(receiver);
String s=new String(receiver.getData());
System.out.println("IP address or DOMAIN NAME: "+s.trim());

Serverdns12.java
DatagramSocket server=new
DatagramSocket(1309); while(true)
{byte[] sendbyte=new byte[1024];
byte[] receivebyte=new byte[1024];

/*..receiving the packet from client…*/

DatagramPacket receiver=new
DatagramPacket(receivebyte,receivebyte.length);
server.receive(receiver);
String str=new String(receiver.getData()); String
s=str.trim();
//System.out.println(s);
InetAddressaddr=receiver.getAddress(); int
port=receiver.getPort();

/*... specify the IP address to map with its domain name...*/

String ip[]={"165.165.80.80","165.165.79.1"};
/*domain name...*/
String name[]={"www.skct.edu","www.sonika.com"};
for(int i=0;i<ip.length;i++)
{if(s.equals(ip[i]))
{sendbyte=name[i].getBytes();
DatagramPacket sender=new DatagramPacket(sendbyte,sendbyte.length,addr,port);
server.send(sender);
break;}
else if(s.equals(name[i])){
sendbyte=ip[i].getBytes();
DatagramPacket sender=new DatagramPacket(sendbyte,sendbyte.length,addr,port);
 server.send(sender);

Sample Output:

APPLICATIONS (SNMP)
Concept:

1. Simple Network Management Protocol (SNMP) is a framework for managing devices

in an internet using TCP/IP.
2. It provides a set of fundamental operations for monitoring and maintaining an internet.
3. SNMP uses the concept of manager and agent
➢ A manager is a host that runs the SNMP client program.
➢ A managed station called an agent, is a router that runs the SNMP server program

Algorithm:
Step 1: Using start method the system is used to receive SNMP request.
Step 2: The assigned Port no 162 are used to send and reciver trap.
Step 3: Set the address using the format ("udp:127.0.0.1/161")
Step 4: Create a variable binding and add the object identifier in OID format.
Step 5: Create the Protocol data unit object
Step 6: Listen enable listening for SNMP packet by using listen ().
Step 8: Get method asynchronous GET request PDU is send to the given target.
Step 7: Send a new get request for single OID and return response event for the request only,
 if notimeout has occurred.
Step 8: Send a new get request for multiple OIDS and return response event for the request only,
if notimeout has occurred.
Step 9: Create target method which contains information about where the data should be
fetched and how to return.
Step 10: If response is not NULL then no time out has occurred and the response was
successfully delivered.

Sample Code:
publicSNMPManager(String add)
{address = add;
public static void main(String[] args) throws IOException {

/*...Port 161 is used for Read and Other operations, Port 162 is used for the
trap generation ...*/
SNMPManager client = new
SNMPManager("udp:127.0.0.1/161"); client.start();

/*...OID - .1.3.6.1.2.1.1.1.0 =>SysDec, OID - .1.3.6.1.2.1.1.5.0 =>SysName...*/

String sysDescr = client.getAsString(new OID(".1.3.6.1.2.1.1.1.0"));
System.out.println(sysDescr);}

/*... the listen() method listens for answers...*/

private void start() throws IOException {
TransportMapping transport = new
DefaultUdpTransportMapping(); snmp = new Snmp(transport);
transport.listen();}

/*...Method which takes a single OID and returns the response from the agent
as a String...*/
public String getAsString(OID oid) throws IOException {
ResponseEvent event = get(new OID[]{oid});
returnevent.getResponse().get(0).get Variable().toString();}
/*...This method is capable of handling multiple OIDs paramoids Eturn throws
IOException ...*/
publicResponseEvent get(OID oids[]) throws IOException { PDU pdu = new
PDU(); for (OID oid : oids) {
pdu.add(new VariableBinding(oid));}
pdu.setType(PDU.GET);
ResponseEvent event = snmp.send(pdu, getTarget(), null); if(event != null) { return event;}
throw new RuntimeException("GET timed out");}

/*... This method returns a Target, which contains information about where
the data should be
fetched and how to return ...*/

private Target getTarget() {

Address targetAddress = GenericAddress.parse(address); CommunityTarget
target = new CommunityTarget();
target.setCommunity(new OctetString("public"));
target.setAddress(targetAddress); target.setRetries(2);
target.setTimeout(1500);
target.setVersion(SnmpConstants.version2c); return
target;}}

Sample Output:

Hardware: x86 Family 6 Model 23 Stepping 10 AT/AT COMPATIBLE –

Software: Windows 2000 Version 5.1 (Build 2600 Multiprocessor Free)

EXPERIMENT-12
Objective: Study of Network simulator (NS).and Simulation of Congestion
Control Algorithms using NS

NET WORK SIMULATOR (NS2)

Ns overview
 Ns programming: A Quick start
Case study I: A simple Wireless network
Case study II: Create a new agent in Ns
Ns Status
Periodical release (ns-2.26, Feb 2003)
Platform support
FreeBSD, Linux, Solaris, Windows and Mac

Ns Functionalities
Routing, Transportation, Traffic sources,queuing disciplines, QoS

Wireless
Ad hoc routing, mobile IP, sensor-MAC Tracing, visualization and various utilitiesNS(Network
Simulators) Most of the commercial simulators are GUI driven, while some network simulators are
CLI driven. The network model / configuration describe the state of the network (nodes, routers,
switchesand links) and the events (data transmissions, packet error etc.). The important outputs of
simulations are the trace files. Trace files log every packet, every event that occurred in the
simulation and are used for analysis. Network simulators can also provide other tools to facilitate
visual analysis of trends and potential trouble spots.
Most network simulators use discrete event simulation, in which a list of pending "events" is
stored, and those events are processed in order, with some events triggering future events such as
the event of the arrival of a packet at one node triggering the event of the arrival of that packet at a
downstream node. Simulation of networks is a very complex task. For example, if congestion is
high, then estimation of the average occupancy is challenging because of high variance. To
estimate the likelihood of a buffer overflow in a network, the time required for an accurate answer
can be extremely large. Specialized techniques such as "control variants" and "importance
sampling" have been developed to speed simulation.

Examples of network simulators

There are many both free/open-source and proprietary network simulators. Examples of notable
network simulation software are, ordered after how often they are mentioned in research papers:
ns (open source)
OPNET (proprietary software)
NetSim (proprietary software)
Uses of network simulators

Network simulators serve a variety of needs. Compared to the cost and time involved in setting up
an entire test bed containing multiple networked computers, routers and data links, network
simulators are relatively fast and inexpensive. They allow engineers, researchers to test scenarios
that might be particularly difficult or expensive to emulate using real hardware - for
instance,simulating a scenario with several nodes or experimenting with a new protocol in the
network. Network simulators are particularly useful in allowing researchers to test new networking
protocols or changes to existing protocols in a controlled and reproducible environment. A typical
network simulator encompasses a wide range of networking technologies and can help the users to
build complex networks from basic building blocks such as a variety of nodes and links. With the
help of simulators, one can design hierarchical networks using various types of nodes like
computers, hubs, bridges, routers, switches, links, mobile units etc. Various types of Wide Area
Network (WAN) technologies like TCP, ATM, IP etc. and Local Area Network (LAN)
technologies like Ethernet, token rings etc., can all be simulated with a typical simulator and the
user can test, analyse various standard results apart from devising some novel protocol or strategy
for routing etc. Network simulators are also widely used to simulate battlefield networks in
Network-centric warfare There are a wide variety of network simulators, ranging from the very
simple to the very complex. Minimally, a network simulator must enable a user to represent a
network topology, specifying the nodes on the network, the links between those nodes and the
traffic between the nodes. More complicated systems may allow the user to specify everything
about the protocols used to handle traffic in a network. Graphical applications allow users to easily
visualize the workings of their simulated environment. Text-based applications may provide a less
intuitive interface, but may permit more advanced forms of customization.
Packet loss occurs when one or more packets of data travelling across a computer network fail to
reach their destination. Packet loss is distinguished as one of the three main error types
encountered in digital communications; the other two being bit error and spurious packets
caused due to noise. Packets can be lost in a network because they may be dropped when a
queue in the network node overflows. The amount of packet loss during the steady state is
another important property of a congestion control scheme. The larger the value of packet loss,
the more difficult it is for transport layer protocols to maintain high bandwidths, the sensitivity
to loss of individual packets, as well as to frequency and patterns of loss among longer packet
sequences is strongly dependent on the application itself.

Throughput

This is the main performance measure characteristic, and most widely used. In communication
networks, such as Ethernet or packet radio, throughput or network throughput is the average rate
of successful message delivery over a communication channel. The throughput is usually
measured in bit sper second (bit/s orbps), and sometimes in data packet sper second or data
packets per time slot This measure how soon the receiver is able to get a certain amount of data
send by the sender. It is determined as the ratio of the total data received to the end to end delay.
 Throughput is an important factor which directly impacts the network performance

Delay

Delay is the time elapsed while a packet travels from one point e.g., source premise or network
ingress to destination premise or network degrees. The larger the valueof delay, the more
difficult it is for transport layer protocols to maintain high bandwidths. We will calculate end to
end delay

Queue Length

A queuing system in networks can be described as packets arriving for service, waiting for service
if it is not immediate, and if having waited for service, leaving the system after being served. Thus
queue length is very important characteristic to determine that how well the active queue
management of the congestion control algorithm has been working.

EXPERIMENT-13
Objective: Perform a case study about the different routing algorithms to select the
network path with its optimum and economical during data transfer. i. Link State
routing ii. Flooding iii. Distance vector
 a) LINK STATE ROUTING
Routing is the process of selecting best paths in a network. In the past, the term routing was
also used to mean forwarding network traffic among networks. However this latter function is
muchbetter described as simply forwarding. Routing is performed for many kinds of networks,
including the telephone network (circuit switching), electronic data networks (such as the Internet),
and transportation networks. This article is concerned primarily with routing in electronic data
networks using packet switching technology .In packet switching networks, routing directs packet
forwarding (the transit of logically addressed network packets from their source toward their
ultimate destination) through intermediate nodes. Intermediate nodes are typically network
hardware devices such as routers, bridges, gateways, firewalls, or switches. General-purpose
computers can also forward packets and perform routing, though they are not specialized hardware
and may suffer from limited performance. The routing process usually directs forwarding on the
basis of routing tables which maintain a record of the routes to various network destinations. Thus,
constructing routing tables, which are held in the router's memory, is very important for efficient
routing. Most routing algorithms use only one network path at a time. Multipath routing techniques
enable the use of multiple alternative paths. In case of overlapping/equal routes, the following
elements are considered in order to decide which routes get installed into the routing table (sorted
by priority):
1. Prefix-Length: where longer subnet masks are preferred (independent of whether it is within a
routing protocol or over different routing protocol)
2. Metric: where a lower metric/cost is preferred (only valid within one and the same routing
protocol)
3. Administrative distance: where a lower distance is preferred (only valid between different
routing protocols) Routing, in a more narrow sense of the term, is often contrasted with bridging
in its assumption that network addresses are structured and that similar addresses imply proximity
within the network. Structured addresses allow a single routing table entry to represent the route to
a group of devices. In large networks, structured addressing (routing, in the narrow sense)
outperforms unstructured addressing (bridging). Routing has become the dominant form of
addressing on the Internet. Bridging is still widely used within localized environments.

b) FLOODING
Flooding is a simple routing algorithm in which every incoming packet is sent through every
outgoing link except the one it arrived on Flooding is used in bridging and in systems such as
Usenet and peer-to-peer file sharing and as part of some routing protocols, including OSPF,
DVMRP, and those used in ad-hoc wireless networks. There are generally two types of flooding
available, Uncontrolled Flooding and Controlled Flooding. Uncontrolled Flooding is the fatal law
of flooding. All nodes have neighbors and route packets indefinitely. More than two neighbors
create a broadcast storm. Controlled Flooding has its own two algorithms to make it reliable,
 SNCF (Sequence Number Controlled Flooding) and RPF (Reverse Path Flooding). In SNCF, the
node attaches its own address and sequence number to the packet, since every node has a memory
of addresses and sequence numbers. If it receives a packet in memory, it drops it immediately
while in RPF, the node will only send the packet forward. If it is received from the next node, it
sends it back to the sender.

Algorithm
There are several variants of flooding algorithm. Most work roughly as follows:
1. Each node acts as both a transmitter and a receiver.

2. Each node tries to forward every message to every one of its neighbours except the source node.
This results in every message eventually being delivered to all reachable parts of the network.
Algorithms may need to be more complex than this, since, in some case, precautions have to be
taken to avoid wasted duplicate deliveries and infinite loops, and to allow messages to eventually
expire from the system. A variant of flooding called selective flooding partially addresses these
issues by only sending packets to routers in the same direction. In selective flooding the routers
don't send every incoming packet on every line but only on those lines which are going
approximately in the right direction.

Advantages
• packet can be delivered, it will (probably multiple times).
• Since flooding naturally utilizes every path through the network, it will also use the shortest path.
• Thisalgorithm is very simple to implement.

Disadvantages
Flooding can be costly in terms of wasted bandwidth. While a message may only have one
destination it has to be sent to every host. In the case of a ping flood or a denial of service attack, it
can be harmful to the reliability of a computer network. Messages can become duplicated in the
network further increasing the load on the networks bandwidth as well as requiring an increase in
processing complexity to disregard duplicate messages. Duplicate packets may circulate forever,
unless certain precautions are taken: Use a hop count or a time to live count and include it with
each packet. This value should take into account the number of nodes that a packet may have to
pass through on the way to its destination.
c) DISTANCE VECTOR ROUTING PROTOCOL USING NS2
In computer communication theory relating to packet-switched networks, a distance vector
routing protocol is one of the two major classes of routing protocols, the other major class being
the link-state protocol. Distance-vector routing protocols use the Bellman–Ford algorithm, Ford–
Fulkerson algorithm, or DUAL FSM (in the case of Cisco Systems protocols) to calculate paths.
A distance-vector routing protocol requires that a router informs its neighbours of topology changes
periodically. Compared to link-state protocols, which require a router to inform all the nodes in a
network of topology changes, distance-vector routing protocols have less computational complexity
 and message overhead. The term distance vector refers to the fact that the protocol manipulates
vectors (arrays) of distances to other nodes in the network. The vector distance algorithm was the
original ARPANET routing algorithm and was also used in the internet under the name of RIP
(Routing Information Protocol). Examples of distance-vector routing protocols include RIPv1 and
RIPv2 and IGRP.

Method:
Routers using distance-vector protocol do not have knowledge of the entire path to a
destination. Instead they use two methods:
1. Direction in which router or exit interface a packet should be forwarded.

2. Distance from its destination

Distance-vector protocols are based on calculating the direction and distance to any link in a
network. "Direction" usually means the next hop address and the exit interface. "Distance" is a
measure of the cost to reach a certain node. The least cost route between any two nodes is the route
with minimum distance. Each node maintains a vector (table) of minimum distance to every
node. The cost of reaching a destination is calculated using various route metrics. RIP uses the hop
count of the destination whereas IGRP takes into account other information such as node delay and
available bandwidth. Updates are performed periodically in a distance-vector protocol where all or
part of a router's routing table is sent to all its neighbors that are configured to use the same
distance-vector routing protocol. RIP supports cross-platform distance vector routing whereas
IGRP is a Cisco Systems proprietary distance vector routing protocol. Once a router has this
information it is able to amend its own routing table to reflect the changes and then inform its
neighbors of the changes. This process has been described as routing by rumor‘ because routers are
relying on the information they receive from other routers and cannot determine if the information
is actually valid and true. There are a number of features which can be used to help with
instability and inaccurate routing information.
EGP and BGP are not pure distance-vector routing protocols because a distance-vector protocol
calculates routes based only on link costs whereas in BGP, for example, the local route preference
value takes priority over the link cost.

Count-to-infinity problem
The Bellman–Ford algorithm does not prevent routing loops from happening and suffers from the
count to infinity problem. The core of the count-to-infinity problem is that if A tells B that it has a
path somewhere, there is no way for B to know if the path has B as a part of it. To see the problem
clearly, imagine a subnet connected like A–B–C–D–E–F, and let the metric between the routers be
"number of jumps". Now suppose that A is taken offline. In the vector-update-process B notices
that the route to A, which was distance 1, is down – B does not receive the vector update from A.
The problem is, B also gets an update from C, and C is still not aware of the fact that A is down –
so it tells B that A is only two jumps from C (C to B to A), which is false. This slowly propagates
through the network until it reaches infinity (in which case the algorithm corrects itself, due to the
 relaxation property of Bellman–Ford).

EXPERIMENT-14
Objective: Running and using services/commands like ping, traceroute, arp, telnet.

Tracert: Determines the path taken to a destination by sending Internet Control Message Protocol
(ICMP) Echo Request messages to the destination with incrementally increasing Time to Live (TTL)
field values. The path displayed is the list of near-side router interfaces of the routers in the path
between a source host and a destination. The near-side interface is the interface of the router that is
closest to the sending host in the path. Used without parameters, tracert displays help.
This diagnostic tool determines the path taken to a destination by sending ICMP Echo Request
messages with varying Time to Live (TTL) values to the destination. Each router along the path is
required to decrement the TTL in an IP packet by at least 1 before forwarding it.

Effectively, the TTL is a maximum link counter. When the TTL on a packet reaches 0, the router is
expected to return an ICMP Time Exceeded message to the source computer. Tracert determines the
path by sending the first Echo Request message with a TTL of 1 and incrementing the TTL by 1 on
each subsequent transmission until the target responds or the maximum number of hops is reached. The
maximum number of hops is 30 by default and can be specified using the -h parameter.

The path is determined by examining the ICMP Time Exceeded messages returned by intermediate
routers and the Echo Reply message returned by the destination. However, some routers do not return
Time Exceeded messages for packets with expired TTL values and are invisible to the tracert
command. In this case, a row of asterisks (*) is displayed for that hop.

Examples:

To trace the path to the host named www.google.co.in use following command

tracert www.google.co.in

To trace the path to the host named www.google.com and prevent the resolution of each
IP address to its name, type:
tracert -d www.google.com

To trace the path to the host named www.google.com and use the loose source route
10.12.0.1-10.29.3.1-10.1.44.1, type:
tracert -j 10.12.0.1 10.29.3.1 10.1.44.1 www.google.com
Syntax
tracert [-d] [-h MaximumHops] [-j HostList] [-w Timeout] [TargetName]

Parameters

-d Prevents tracert from attempting to resolve the IP addresses of intermediate routers to their names.
This can speed up the display of tracert results.
-h MaximumHops Specifies the maximum number of hops in the path to search for the target (destination).
The default is 30 hops.
-j HostList Specifies that Echo Request messages use the Loose Source Route option in the IP header with
the set of intermediate destinations specified in HostList. With loose source routing, successive
intermediate destinations can be separated by one or multiple routers. The maximum number of addresses
or names in the host list is 9. The HostList is a series of IP addresses (in dotted decimal notation) separated
by spaces.
-w Timeout Specifies the amount of time in milliseconds to wait for the ICMP Time Exceeded or Echo
Reply message corresponding to a given Echo Request message to be received. If not received within
the time-out, an asterisk (*) is displayed. The default time-out is 4000 (4 seconds).

Ping
Verifies IP-level connectivity to another TCP/IP computer by sending Internet Control Message
Protocol (ICMP) Echo Request messages. The receipt of corresponding Echo Reply messages are
displayed along with round-trip times. Ping is the primary TCP/IP command used to troubleshoot
connectivity, reachability, and name resolution.
You can use ping to test both the computer name and the IP address of the computer. If pinging the IP
address is successful, but pinging the computer name is not, you might have a name resolution
problem. In this case, ensure that the computer name you are specifying can be resolved through the
local Hosts file by using Domain Name System (DNS) queries or through NetBIOS name resolution
techniques.

To test a TCP/IP configuration by using the ping command:

➢ To quickly obtain the TCP/IP configuration of a computer, open Command Prompt, and then
type ipconfig . From the display of the ipconfig command, ensure that the network adapter for
the TCP/IP configuration you are testing is not in a Media disconnected state.
➢ At the command prompt, ping the loopback address by typing ping 127.0.0.1
➢ Ping the IP address of the computer.
➢ Ping the IP address of the default gateway. If the ping command fails, verify that the default
gateway IP address is correct and that the gateway (router) is operational.
➢ Ping the IP address of a remote host (a host that is on a different subnet). If the ping command
fails, verify that the remote host IP address is correct, that the remote host is operational, and
that all of the gateways (routers) between this computer and the remote host are operational.
➢ Ping the IP address of the DNS server. If the ping command fails, verify that the DNS server IP
address is correct, that the DNS server is operational, and that all of the gateways (routers)
between this computer and the DNS server are operational.

ARP
Displays and modifies entries in the Address Resolution Protocol (ARP) cache, which contains one
or more tables that are used to store IP addresses and their resolved Ethernet or Token Ring physical
addresses. There is a separate table for each Ethernet or Token Ring network adapter installed on your
computer.

Syntax

arp [-a [InetAddr] [-N IfaceAddr]] [-g [InetAddr] [-N IfaceAddr]] [-d InetAddr [IfaceAddr]]
[-s InetAddr EtherAddr [IfaceAddr]]
Parameters

Used without parameters displays help

-a [InetAddr] Displays current ARP cache tables for all interfaces. To display the ARP
[-N IfaceAddr] cache entry for a specific IP address, use arp -a with the InetAddr
parameter, where InetAddr is an IP address. To display the ARP cache
table for a specific interface, use the -N IfaceAddr parameter where
IfaceAddr is the IP address assigned to the interface.
The -N parameter is case-sensitive.
-g [InetAddr] Identical to -a.
[-N IfaceAddr]
-d InetAddr [IfaceAddr] Deletes an entry with a specific IP address, where InetAddr is the IP
address. To delete an entry in a table for a specific interface, use the
IfaceAddr parameter where IfaceAddr is the IP address assigned to the
interface. To delete all entries, use the asterisk (*) wildcard character in
place of InetAddr.
-s InetAddr EtherAddr [IfaceAddr] Adds a static entry to the ARP cache that resolves the IP address
InetAddr to the physical address EtherAddr. To add a static ARP cache
entry to the table for a specific interface, use the IfaceAddr parameter
where IfaceAddr is an IP address assigned to the interface.

Examples:

To display the ARP cache tables for all interfaces use following command

arp -a
 To display the ARP cache table for the interface that is assigned the IP address 192.168.42.171

TELNET
The telnet command is used for connection and communication with a remote or local host via
the Telnet TCP/IP protocol.
You can enter a domain or IP address and try connecting to it via the chosen port. In case the port is
not specified, telnet utility tries to connect via the default port 23.

The command is really useful in cases when you need to check whether the needed port is open on
your computer and on the side of the remote host.
How to use Telnet
For Windows

Telnet is disabled on Windows by default. To enable it, perform these steps:

1. Press the Start button > Control Panel:
2. Go to the Programs section:

3. Select Programs and Features > Turn Windows Features on or off:

 4. Scroll down the list available in the Windows Features window > check Telnet Client option
> press OK > wait a few moments for the changes to be applied

5. Telnet is enabled now, so we can run it in the same way as other commands:

➢ Select the Start button > click on the Run option.

➢ In the command line, type in cmd and press Enter.
➢ After that, type in the following and press Enter:

telnet [domain name or IP] [port number]

for example:
 telnet namecheap.com 80

If you see the blank output after that, you have connected successfully. To quit you can press CTRL +
C or any key:

As a result of successful telnet, we can conclude, that the entered domain or IP exists, and the chosen
port is open on your computer and on the side of the target host.

If connection has not been established, the following error will appear:

Possible reasons for it are:

➢ domain or IP does not exist/not available/entered with a typo

➢ the chosen port is closed by a security software or via network configurations on your computer
or on the side of the host you connect to
➢ connection/networking issue has occurred

EXPERIMENT-15
Objective: Network packet analysis using tools like Wireshark, tcpdump, etc.
Theory:

tcpdump

The fundamental tool of almost all network traffic collection is tcpdump. It is an open-source
application that comes installed on almost all Unix-like operating systems. Tcpdump is an excellent
collection tool and comes complete with a very complex filtering language. It’s essential to know how
to filter the data at collection time to end up with a manageable chunk of data to analyze. Capturing
all data from a network device on even a moderately busy network can create too much data to analyze
efficiently.

In some rare cases, allowing tcpdump to output its capture directly to your screen may be enough to
find what you’re looking for. For example, in writing this article, captured some traffic and noticed that
machine was sending traffic to an IP address . It turns out that machine was sending data to a Google IP
address of 172.217.11.142.

It seems that even when Chrome is not running in the foreground it remains running as a service. it
would not have necessarily noticed this without a packet analysis to tip me off. it re-captured some
more tcpdump data but this time told tcpdump to write the data to a file that itopened in Wireshark
(more on that later). Here’s that entry:

Tcpdump is a favorite tool among sysadmins because it is a command-line tool. This means that it
doesn’t require a full-blown desktop to run. It is unusual for production servers to provide a desktop
because of the resources that would take, so command-line tools are preferred. As with many advanced
tools, tcpdump has a very rich and arcane language that takes some time to master.

Key Features:

• Command line tool

• Packet capture capabilities
 • Free to use

A few of the very basic commands involve selecting the network interface from which to collect data,
and writing that data to a file so it can be exported for analysis elsewhere. The -i and -w switches are
used for this.

# tcpdump -i eth0 -w tcpdump_packets

tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 262144 bytes
^C51 packets captured

This produces a capture file:

file tcpdump_packets
tcpdump_packets: tcpdump capture file (little-endian) - version 2.4 (Ethernet, capture length 262144)

The standard TCP capture file is a pcap file. It is not text so it can only be read by an analysis program
that knows how to read pcap files.

Wireshark

Wireshark is probably the next best-known tool in any sysadmin’s toolkit. It can not only capture data,
but also provides some advanced analysis tools. Adding to its appeal, Wireshark is open source, and
has been ported over to almost every server operating system that exists. Starting life named Ethereal,
Wireshark now runs everywhere, including as a standalone portable app.

If you’re analyzing traffic on a server with a desktop installed, Wireshark can do it all for you. The
collected packets can then be analyzed all in one spot. However, desktops are not common on servers,
so in many cases, you’ll want to capture the network data packets remotely and then pull the resulting
pcap file into Wireshark.

At first launch, Wireshark allows you to either load an existing pcap file, or start capturing. If you elect
to capture network traffic, you can optionally specify filters to pare down the amount of data Wireshark
collects. Since its analysis tools are so good, it’s less important to ensure you surgically identify the
data at collection time with Wireshark. If you don’t specify a filter, Wireshark will simply collect all
network data that your selected interface observes.
One of the most useful tools Wireshark provides is the ability to follow a stream. It’s probably most
useful to think of a stream as an entire conversation. In the screenshot below we can see a lot of data
has been captured, but what is most interested in is that Google IP address. itcan right-click it
and Follow the TCP Stream to see the entire conversation.

If you’ve captured traffic elsewhere, you can import the pcap file using Wireshark’s File -
> Open dialogue. The same filters and tools that can be used for natively captured network data are
available for imported files.
 EXPERIMENT-16
Objective: Network simulation using tools like Cisco Packet Tracer, NetSim,
OMNeT++, NS2, NS3, etc

Theory:
Simulation is a very important technology in modern time. Computer assisted simulation can model
hypothetical and real-life objects or activities on a computer to study the well-designed structure. A network
simulator is a system of implementing the network on the computer through which the performance of the
network is calculated. The computer assisted simulation technologies are applied in the simulation of
networking algorithms. The functional network field is narrower than general simulation and it is natural that
more specific requirements will be placed on network simulations.

Network simulator allows the researchers to test the scenarios that are difficult or expensive to
simulate in real world. Design of various network topologies using nodes, hosts, hubs, bridges, routers and
mobile units etc. is possible. The network simulators are of various types which can be compared on the basis
of: range (simple to the complex), specification of nodes, links and traffic between the nodes. Specifying
about the protocols used to handle traffic in a network, user friendly applications (allow users to easily
visualize the simulated environment.), text-based applications (permit more advanced forms of
customization) and programming-oriented tools (providing a programming framework that customizes to
create an application that simulates the networking environment to be tested).

Network simulators are used by people from different areas such as academic researchers,
industrialized sectors and Quality Assurance (QA) to design, simulate and analyze the performance of
different network protocols. They can also be used to evaluate the outcome of the different parameters of the
protocols being studied. Normally a network simulator comprises of wide range of networking technologies
and protocols that help users to build complex networks from basic building blocks like clusters of nodes and
links. With their help, different network topologies can be designed using various types of nodes such as end-
hosts, network bridges, routers, hubs, optical link-layer devices and mobile units.

Concepts in network simulation

Generally, network simulators try to represent the real world networks and it is a useful technique, given that
the activities of a network can be modeled by calculating the interaction between the different network
components (they can be end-host or network entities such as routers, packets or physical links) using
mathematical formulas. They can also be modeled by actually or virtually capturing and playing back
experimental observations from real networks. Upon receipt of the observation data from simulation
experiments, the behavior of the network and protocols supported are analyzed in a series of offline test
experiments. All types of attributes can also be modified in a controlled manner to assess how the network can
behave under different parameter combinations. Another feature of network simulation worth noticing is that
the simulation program can be used and analyzed together with various strategy, links, applications etc.
Typically, users can then adapt the simulator to fulfill their exact needs. Simulators support the most popular
protocols and networks such as WLAN, TCP and WSN.
 Simulators
Most of the commercial simulators are Graphical User Interface (GUI) driven, while some
network simulators are Command-Line Interface (CLI). The design of the network describes the state of the
network (nodes, routers, switches and links) and the events (data transfer, transmission delay, packet error
etc.). The major output of simulation is the trace files which log every packet and event that occurred during
simulation and is used for analysis. Also provides other tools to facilitate visual analysis of trends and
potential trouble spots. Most of the network simulators are discrete event, in which the list of pending "events"
are stored and processed in order. Some events triggers the future events (i.e.) the event of the arrival of a
packet at one node triggering the event of the arrival of that packet at a downstream node.

Simulation of networks is a very difficult task. For example, if blocking is high, then evaluation of the
average occupancy is challenging because of high variance. To evaluate the probability of buffer overflow in a
network, the time required for a precise answer can be enormously large. Techniques like "control variants"
and “sampling" have been developed to speed simulation.

List of Network Simulators

There are many both free/open-source and proprietary simulators. Examples of notable simulation
software are ordered based on how frequently they are

NS2 (Network Simulator 2)

1. NS3 (Network Simulator 3)
2. OPNET.
3. OMNeT++.
4. NetSim.

Uses of Network Simulators

Network simulators serve a variety of requirements. Simulators are relatively fast and economical
when compared to the cost and time involved in setting up an entire bed containing multiple network
computers, data links and routers. They authorize researchers to test scenarios that might be particularly
difficult or expensive to emulate using a real hardware - for instance simulating a scenario with several nodes
or experimenting with a new protocol in the network. Simulators are mainly useful in allowing researchers to
test new networking protocols or changes to existing protocols in controlled and reproducible surroundings.

A typical simulator encompasses a wide range of networking technologies and can help the users to
build complex networks from basic building blocks such as selection of nodes and links. Various types of
nodes in Hierarchical networks resembling computers, hubs, bridges, routers, links, switches mobile units etc
can be designed with the help of simulators.

Various types of Wide Area Network (WAN) technologies like TCP, ATM, IP etc. and Local Area
Network (LAN) technologies like Ethernet, token rings etc., can be imitated with a simulator and the user can
examine various standard results apart from devising some novel protocol or routing strategy. Network
simulators are widely used to simulate battlefield networks in Network-centric warfare.
 There are ample varieties of simulators, ranging from simple to complex. A simple simulator must
enable a user to represent network topology, to specify nodes on the network, the links and the traffic between
the nodes. More complex systems may permit the user to specify everything about the protocols used to
handle traffic in a network. User friendly applications permit users to envision easily the working mechanism
of their simulated situation. Text-based applications offer a less sensitive interface, but permits more advanced
forms of customization.
Overview of Network Simulators
Currently there are many network simulators that have different features in different aspects. Short
lists of the current network simulators include NS-2, NS-3, OPNET, OMNeT++, NETSIM, QualNet, and J-
Sim. These network simulators are selected for discussion regarding their features, advantages and
restrictions.

NS2
The Ns2 is a discrete event simulator targeted at packet level networking research and
provides substantial support to simulate group of protocols like TCP, UDP, FTP and HTTP. It
comprises of two simulation tools. Ns-2 is primarily UNIX based and fully simulates a layered wire
or wireless network from the physical radio transmission channel to high-level applications. The
simulator is written in C++ and a script language called OTcl.

C++: C++ is fast to run but slower to change, making it suitable for detailed protocol implementation.

Otcl: OTcl runs much slower but can be changed very quickly (and interactively), making it
ideal for simulation configuration. Ns provides glue to make objects and variables appear on both
languages.

NS2 uses an OTcl interpreter by which the user writes an OTcl script that defines the
network, (number of nodes and links) the transaction in the network (sources destinations, type of
traffic) and the type of protocols used. The outcome of the simulation is a trace file that can be used
for data processing (calculate delay, throughput etc). To visualize the simulation, a program called
Network Animator (NAM) is used. It visualizes the packets as they propagate throughout the
network. The ns- 2 simulator has numerous features that make it suitable for our simulations.

• A network environment for ad-hoc networks,

• Wireless channel modules (e.g.802.11),
• Routing along multiple paths,
• Mobile hosts for wireless cellular networks.
• Download of ns-2 source code is possible and can be compiled for multiple platforms.
 Advantages:
1. NS2 has large number of available models, realistic mobility models, powerful and flexible
scripting and simulation setup, large user community and ongoing development.
2. NS2 provides an easy traffic and movement pattern by including an efficient energy model.
3. It provides a set of randomized mobility model and there are several projects to bring
advanced mobility models to the simulators.
4. Complex scenarios can be easily tested.
5. Popular for its modularity.

Limitations:
1. NS2 needs to be recompilation every time if there is a change in the user code.
2. Real system is too complex to model i.e. complicated infrastructure.

NS3
The ns-3 simulator is a discrete-event network simulator for Internet systems, targeted
primarily for research and learning purpose. The ns-3 project, started in 2006, is open-source free
software, licensed under the GNU GPLv2 license. It will rely on the current contributions of the
community to develop new models, debug or maintain the existing ones, and share the results. Ns3 is
mainly used on LINUX systems and not limited to internet based systems alone.

C++: implementation of simulation and core model.Ns-3 is built as a library which may be
statically or dynamically linked to a C++ main program. These libraries describe the beginning of
simulation and their topology.
Python: C++ wrapped by Python. Python programs to import an “ns3” module. The features of NS3
simulator are given below.

1. Modular, documented core

2. C++ programs and Python scripting
3. Alignment with real systems
4. Software integrations
5. Virtualization and test bed integration
6. Attribute system
7. Updated models
Advantages:
1. High modularity than its ancestor NS2.
2. Support simulation for TCP, UDP, ICMP, IPv4, multicast routing, P2P and CSMA protocols.
3. Support for ported code should make model validation easier and more credible.
4. Much more flexible than any other simulators.
5. Wide range of use in both optimization and expansion of the existing networks.

Limitations:
1. NS3 still suffers from lack of credibility.
2. NS3 is intended to replicate the successful mode of NS2 in which various organizations
contributed to the models and components based on the framework of NS2.
3. NS3 needs a lot of specialized maintainers in order to avail the merits of NS3 as the
commercial OPNET network simulators.
4. Active maintainers are required to respond to the user questions, bug reports and help to Test &
validate the system.

OMNET++
It is a component-based, modular and open architecture discrete event simulator framework.
The most common use of OMNeT++ is for simulation of networks, but it is also used for queuing
network simulations and other areas as well. It is licensed under its own Academic Public License,
which permits GNU Public License like freedom but only in noncommercial settings. It provides
component architecture for models.

C++: The C++ class library comprises of simulation kernel and utility classes (for random
number generation, statistics collection, topology discovery etc) -- this one is used to create
simulation components (simple modules and channels); infrastructure to assemble simulations from
these components and configure (NED language, ini files); runtime user interfaces or environments
for simulations (Tkenv, Cmdenv); an Eclipse-based simulation IDE for designing, running and
evaluating simulations; extension interfaces for real-time simulation, emulation, MRIP, parallel
distributed simulation, database connectivity and so on.

The OMNeT++ components include:

1. Simulation kernel library

2. Compiler for the NED topology description language (nedc)
3. Graphical network editor for NED files (GNED)
4. GUI for simulation execution, links into simulation executable (Tkenv)
5. Command-line user interface for simulation execution (Cmdenv)
6. Graphical output vector plotting tool (Plove)
7. Graphical output scalars visualization tool (Scalars)
 8. Model documentation tool (opp_neddoc)
9. Utilities (random number seed generation tool, make file creation tool, etc.)
10. Documentation, sample simulations, etc.
Advantages:
➢ Provides a powerful GUI environment.
➢ Tracing and debugging are much easier than other simulators.
➢ Accurately models most hardware and include the modeling of physical phenomena.
Limitations:
➢ It does not offer a great variety of protocols and very few protocols have been implemented,
leaving users with significant background work.
➢ Poor analysis and management of typical performance. The mobility extension is relatively
incomplete

NETSIM

NetSim is a discrete event simulator developed by Tetcos in 1997, in association with Indian
Institute of Science. It has also been featured with Computer Networks and Internets V edition by
Dr. Douglas Comer, published by Prentice Hall. It has an object-oriented system simulating
environment to support simulation and analysis of voice and data communication scenarios for High
Frequency Global Communication Systems (HFGCS).

Java: It creating fast, platform independent software that could be used in simple, consumer
electronic products. Java designed for simple, efficient, platform-independent program for creating
WWW-based programs. Using Java one can create small programs called applets that are entrenched
into an HTML document and viewable on any Java-compatible browser. Java applets are compiled
into a set of byte-codes, or machine-independent processing instructions.
Features:
• NetSim modeling and simulation are supported for Aloha, Slotted Aloha, Token Ring/Bus,
Ethernet CSMA/CD, Fast Ethernet, WLAN - IEEE 802.11 a/b/g/n and e, X.25, Frame Relay,
TCP, UDP, IPv4 and IPv6, Routing - RIP, OSPF, BGP,MPLS, MANET, GSM, CDMA,
Wire-less Sensor Network, Zigbee, Cognitive radio)[5].
• It simulates a wide variety of Cisco routers, including 2500 series, 2600 series, 2800 series,
and 3600 series, as well as the Cisco Catalyst 1900 series, 2900 series, and 3500 series
switches. Protocol libraries are available as open C code for user modification. This can help
to avoid the time consuming process such as encoding, customization and configuring
commercial simulators to meet customer specific needs. Along with the Boson Virtual Packet
Technology engine NetSim utilizes Boson’s proprietary Router & Simulator EROUTER
software technologies, to produce individual packets. These packets are routed and switched
through the simulated network, allowing the simulator to build an appropriate virtual routing
 table and simulate proper networking. Other simulation products on the market do not support
this level of functionality.

• It can be used to create a simulation of the topology of corporate network and help practice
trouble-shooting without using devices on the production network.
Advantages:
1. NetSim has a GUI which features drag and drop functionality for devices, links etc. i.e.
Modeling in NetSim is simple and user friendly.
2. It has a built in analysis framework that provides intra and inter-protocol performance
comparison with graphical options.
3. Data packet and control packet flow can be visual-ized through NetSim built-in packet
animator.
4. It is easy to learn all about NetSim.

Limitations:

1. NetSim is a single process discrete event simulator. A single event queue is used for the
simulation which at any given time contains one entry for each station on the network.
2. Free version of NetSim is not available.

Cisco Packet Tracer

Cisco Packet Tracer is a powerful network simulation program that allows students to experiment
with network behavior and ask “what if” questions. As an integral part of the Networking Academy
comprehensive learning experience, Packet Tracer provides simulation, visualization, authoring,
assessment, and collaboration capabilities to facilitate the teaching and learning of complex
technology concepts.

Packet Tracer supplements physical equipment in the classroom by allowing students to create a
network with an almost unlimited number of devices, encouraging practice, discovery, and
troubleshooting. The simulation-based learning environment helps students develop 21st century
skills such as decision making, creative and critical thinking, and problem solving. Packet Tracer
complements the Networking Academy curricula, allowing instructors to easily teach and
demonstrate complex technical concepts and networking systems design. Instructors can customize
individual or multiuser activities, providing hands-on lessons for students that offer value and
relevance in their classrooms. Students can build, configure, and troubleshoot networks using virtual
equipment and simulated connections, alone or in collaboration with other students. Packet Tracer
offers an effective, interactive environment for learning networking concepts and protocols. Most
importantly, Packet Tracer helps students and instructors create their own virtual “network worlds”
for exploration, experimentation, and explanation of networking concepts and technologies.
 Packet Tracer’s drag-and-drop interface allows students to configure and validate system
architecture

Key Features Packet Tracer Workspaces:

Cisco Packet Tracer has two workspaces—logical and physical. The logical workspace allows users
to build logical network topologies by placing, connecting, and clustering virtual network devices.
The physical workspace provides a graphical physical dimension of the logical network, giving a
sense of scale and placement in how network devices such as routers, switches, and hosts would look
in a real environment. The physical view also provides geographic representations of networks,
including multiple cities, buildings, and wiring closets.
Packet Tracer Modes:
Cisco Packet Tracer provides two operating modes to visualize the behavior of a network—real-time
mode and simulation mode. In real-time mode the network behaves as real devices do, with
immediate real-time response for all network activities. The real-time mode gives students a viable
alternative to real equipment and allows them to gain configuration practice before working with real
equipment.
In simulation mode the user can see and control time intervals, the inner workings of data transfer, and
the propagation of data across a network. This helps students understand the fundamental concepts
behind network operations.

Layer Cisco Packet Tracer Supported Protocols

Application • FTP , SMTP, POP3, HTTP, TFTP, Telnet, SSH, DNS, DHCP, NTP, SNMP, AAA,
ISR VOIP, SCCP config and calls ISR command support, Call Manager Express
Transport • TCP and UDP, TCP Nagle Algorithm & IP Fragmentation,RTP

Network • BGP, IPv4, ICMP, ARP, IPv6, ICMPv6, IPSec, RIPv1/ v2/ng, Multi-Area OSPF,
EIGRP, Static Routing, Route Redistribution, Multilayer Switching, L3 QoS, NAT, CBAL, ,
Zone-based policy firewall and Intrusion Protection,System on the ISR, GRE VPN, IPSec
VPN
Network Access/ • Ethernet (802.3), 802.11, HDLC, Frame Relay, PPP, PPPoE, STP, RSTP, VTP, DTP,
Interface CDP, 802.1q, PAgP, L2 QoS, SLARP, Simple WEP, WPA, EAP