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PART-A (UNIVERSITY QUESTIONS)


1. What are the uses of internetwork routing? (M/J-2013)

The uses of internetwork routing is connect one WAN to another WAN.

2. What is the purpose of subnetting? (M/J-2013)

Subnetting is three levels of hierarchy

Masking Masks

without subnetting Masks


with subnetting

Finding the subnetwork address

Boundary level masking


Non -boundary level masking

3. Differentiate ARP and RARP. (A/M-11)

ARP

The Address Resolution Protocol (ARP) is designed to convert IP addresses to MAC


addresses or vice versa.

RARP

Reverse Address Resolution Protocol (RARP) too is a protocol to find a certain


address.

4. Give the CIDR notation for class A, B and C? (A/M-2011)

5. What is Internet working? (N/D-2011)


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The physical and data link layers of a network operate locally. These two
layers are jointly responsible for data delivery on the network from one node to the
next.

6. What is IPV6? (N/D-2011)


In IPv6, the Internet protocol was extensively modified to accommodate the
unforeseen growth of the Internet. The format and the length of the IP address were
changed along with the packet format.

7. Why are protocols needed? (M/J-2012)


In networks, communication occurs between the entities in different systems.
Two entities cannot just send bit streams to each oth er and expect to be understood.
For communication, the entities must agree on a protocol. A protocol is a set of rules
that govern data communication.

8. What is the use of IP address? (M/J-2012)


The use of IP address is to identify the particular system or destination.

IP address is the 3-bit number for representing a host or system in the network.
One portion of the IP address indicates a networking and the other represents the
host in a network.

PART-A (IMPORTANT QUESTIONS)

1.What are the responsibilities of data link layer?


Specific responsibilities of data link layer include the following.
a) Framing
b) Physical addressing
c) Flow control d) Error control e) Access control

2. Mention the types of errors.


There are 2 types of errors
a) Single-bit error.
b) Burst-bit error.

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3. Define the following terms.


a) Single bit error: The term single bit error means that only one bit of a given data
unit (such as byte character/data un it or packet) is changed from 1 to 0 or from 0 to
1.
b) Burst error: Means that 2 or more bits in the data unit have changed from 1 to 0
from 0 to 1.

4. What is redundancy?
It is the error detecting mechanism, which means a shorter group of bits or
extra bits may be app ended at the destination of each unit.

5. List out the available detection methods.


There are 4 types of redundancy checks are used in data communication.
a) Vertical redundancy checks (VRC).
b) Longitudinal redundancy checks ( LRC).
c) Cyclic redundancy checks (CRC).
d) Checksum.

6. Write short notes on VRC.


The most common and least expensive mechanism for error detection is the
vertical redundancy check (VRC) often called a parity check. In this technique a
redundant bit called a parity bit, is appended to every data unit so, that the total
number of 0’s in the unit (including the parity bit) becomes even.

7. Write short notes on LRC.


In longitudinal redundancy check (LRC), a block of bits is divided into rows
and a redundant row of bits is added to the whole block.

8. Write short notes on CRC. The third and most powerful of the redundancy
checking techniques is the cyclic redundancy checks (CRC) CRC is based on binar y
division. Here a sequence of redundant bits, called the CRC remainder is appended
to the end of data unit.

9. Write short notes on CRC generator.


A CRC generator uses a modulo-2 division.
a) In the first step, the 4 bit divisor is subtracted from the first 4 bit of the dividend.

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b) Each bit of the divisor is subtracted from the corresponding bit of the dividend
without disturbing the next higher bit.

10. Write short notes on CRC checker.


A CRC checker functions exactly like a generator. After receiving the data
appended with the CRC it does the same modulo-2 division. If the remainder is all
0’s the CRC is dropped and the data accepted. Otherwise, the received stream o f
bits is discarded and the dates are resent.

11. Give the essential properties for polynomial.


A polynomial should be selected to have at least the following properties.
a) It should not be
b) It should be divisible by(x+1).

12. Define checksum.


The error detection method used by the higher layer protocol is called
checksum. Checksum is based on the concept of redundancy.

13. What are the steps followed in checksum generator?


The sender follows these steps
a) The units are divided into k sections each of n bits.
b) All sections are added together using 2’s complement to get the sum.
c) The sum is complemented and become the checksum.
d) The checksum is sent with the data.

14. List out the steps followed is checksum checker side.


The receiver must follow these steps
a) The unit is divided into k section each of n bits.
b) All sections are added together using 1’s complement to get
the sum.
c) The sum is complemented.
d) If the result is zero.

15. Write short notes on error correction.


It is the mechanism to correct the errors and it can be handled in 2 ways.
a) When an error is discov ered, the receiver can hav e the sender retransmit the
entire data unit.

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b) A receiver can use an error correcting coder, which automatically corrects certain
errors.

16. Mention the types of error correcting methods.


There are 2 error-correcting methods.
a) Single bit error correction
b) Burst error correction.

17. What is the purpose of hamming code?


A hamming code can be designed to correct burst errors of certain lengths. So
the simple strategy used by the hamming code to correct single bit errors must
be redesigned to be applicable for multiple bit correction.

18. Define flow control.


Flow control refers to a set of procedures used to restrict the amount of
data. The sender can send before waiting for acknowledgment.

19. What is a buffer?


Each receiving device has a block of memory called a buffer, reserved for
storing incoming data until they are processed.

20. Mention the categories of flow control.


There are 2 methods have been developed to control flow of data across
communication links.
a) Stop and wait- send one from at a time.
b) Sliding window- send several frames at a time.

21. What is the function of stop and wait flow control?


In this method, the sender sends one frame and waits for an acknowledgement
before sending.

22. Mention the advantage and disadvantage of stop and wait f low control.
Advantage: simplicity Disadvantage: inefficiency
.
23. Define ARQ.
Error control in the data link layer is based on Automatic repeat request
(ARQ), which means retransmission of data in 3 cases.

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a) Damaged frame
b) Lost frame
c) Lost acknowledgment.

24. Mention the function of go-back N-ARQ.


It is the popular mechanism for continuous transmission error control. In the
method, if our frame is lost or damaged, all frames sent since the last frame
acknowledged are retransmitted.

25. What is selective reject ARQ?


In selective reject ARQ only the specific damaged or lost frame is
retransmitted. If a frame is corrupted in transit, a NAK is returned and the frame is
resent out of sequence.

26. Define HDLC.


It is a bit-oriented data link protocol design ed to support both half-duplex
and full duplex communication over point to point and midpoint links.

27. List the types of stations is HDLC.


HDLC differentiates between 3 types of stations.
a) Primary b) Secondary c) Combined

28. Define configuration .


The bond configuration refers to the relationship of hardware devices on a link.

29. List the various ways of station configuration.


The stations are configured in 3 ways a) Unbalanced configuration
b) Symmetrical con figuration
c) Balanced configuration

30. What are the different communication modes in HDLC?


HDLC supports 3 modes of communication between stations.
a) Normal response mode (NRM)
b) Asynchronous response mode (ARM)
c) Asynchronous balanced mode (ABM)

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31. Mention the types of frames in HDLC.


There are 3 types of HDLC fr ames.
a) Information frames (I-frames)
b) Supervisory frames (S -frames)
c) Unnumbered frames (U-frames)

32. Give the usage of I, S, U frames.


I frames – used to transport user data and control information relating to user
data. S frames – used only to transport control information, primarily data link layer
and erro r controls. U frames – reserved for systems management.

33. Write the types of frame fields contained in HDLC.


Each frame in HDLC may contain up to 6 fields.
a) Beginning flag field b) An address field
c) A control field d) An information field
e) A frame check sequence (FCS) field f) An ending flag field.

34. What is meant by bit stuffing?


Bit stuffing is the process of adding one extra 0 whenever th ere are 5
consecutive in the data so that the receiver doesn’t mistake the data for a flag.

35. Define LAN. A Local Area Network (LAN) is a data communication system that
allows a number of independ ent devices to communicate directly with each other in
a limited geographic area.

36. Mention the various architecture in a LAN.


LAN is dominated by 4 architectures.
a) Ethernet
b) Token bus
c) Token ring
d) Fiber distributed data interface ( FDDI)

37. Define a standard 802.3


IEEE 802.3 supports a LAN standard originally developed by Xerox and later
extended by a joint venture between digital equipment corporations. Intel
Corporation and Xerox. This was called ‘Ethernet’.

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38. List the most command kinds of Base band 802.3 LAN.
a) 10 Base 5
b) 10 Base 2
c) 10 Base T d) 1 Base 5 e) 100 Base T

39. Mention the different kinds of Ethernet networks.


a) Switched Ethernet b) Fast Ethernet c) Gigabit Ethernet

40. Write short notes on FDDI.


Fiber distributed data interface is a local areas. Network protocol
standardized by ANSI and ITU-7. It supports data rates of 100 Mbps and provides a
high-speed alternative to Ethernet and token ring access method used here is token
passing.

41. Describe the three HDLC station types?


The three HDLC station types are:
Primary station : The primary station has the complete control of the link. The
Primary station sends commands to the secondary station.
Secondary station : The secondary station sends responses.
Combined station : The combined station is one which acts either as a primary or a
Secondary, depending upon the nature and direction of the transmission. Combined
station sends both commands and responses.

42. What is piggy backing?


Piggy backing means combining data to sent and acknowledgement of the

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PART-B
14. Illustrate with a neat sketch, the IPV4 datagram format. Compare the fields
in the main headers of IPV4 and IPV6. (A/M-2011)

IP version 4 (IPv4):

The packet format of IP version 4 (IPv4) is shown in Figure 2.3.


Each packet comprises the header and data. The size of the header
is variable, with 20 bytes of fixed-length header and an options
field whose size is variable up to 40 bytes. A brief description of the
fields follows.

 Version specifies the IP version.


 Header length (HL) specifies the length of the header.
 Type of service specifies the quality-of-service (QoS) requirements of the
packet, such as priority level, delay, reliability, throughput, and cost.
 Total length specifies the total length of the packet in bytes, including the
header and data. A total of 16 bits are assigned to this field.
 Identification, flags, and fragment offset are used for packet fragmentation
and reassembly.
 Time to live specifies the maximum number of hops after which a packet must
be discarded.
 Protocol specifies the protocol used at the destination.
 Header checksum is a method of error detection
 Source address and destination address are 32-bit fields specifying the source
address and the destination address, respectively.
 Options is a rarely used variable-length field to specify security level,
timestamp, and type of route.
 Padding is used to ensure that the header is a multiple of 32 bits.

Figure 2.3. IP packet format

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Recall that the 16 bits in the total length field express the
total length of a packet. Hence, the total length of the packet is
limited to 216 bytes. However, the maximum packet size of 216
bytes is rarely used, since the packet size is limited by the physical
network capacity. The real physical network capacity per packet is
normally less than 10K and even gets smaller, to 1.5K when the
packet reaches a LAN. To accomplish packet partitioning, the
identification, flags, and fragment offset fields perform and keep
track of the packet-fragmentation process when needed.

IP Version 6 (IPv6):

The use of IPv4 has resulted in the exhaustion of the 32-bit address space to
the extent that IPv4 has run out of addressing spaces. Therefore, 128-bit address
spacing was introduced with Internet Protocol version 6 (IPv6). It enjoys tremendous
popularity because of its simplicity and flexibility in adapting to diverse network
technologies. Compatible with IPv4, IPv6 also supports real-time applications,
including those that require guaranteed QoS. Figure 2.9 shows the IPv6 header. A
brief description of the fields in the header follows.

 Version is the same as in IPv4, indicating the version number of the protocol.
 Traffic class specifies the priority level assigned to a packet.
 Flow label indicates the delay period within which application packets, such
as real-time video, must be delivered.
 Payload length is the 16-bit specification of the length of the data, excluding
the header.
 Next header specifies the type of extension header used. The functionality of
the option field in IPv4 is specified in the extension header. In addition, the
extension header is more flexible than the options field.
 Hop limit is the same as the time-to-live field in IPv4.
 Source address and destination address are each identified by a 128-bit field
address.

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Figure 2.9. An IPv6 packet format

The IPv4 and IPv6 header formats have some notable


differences. First, IPv6 uses a 128-bit address field rather than the
32-bit field in IPv4. The 128-bit field can support a maximum of 3.4
x 1038 IP addresses. IPv6 has a simpler header format, eliminating
the fragmentation, the checksum, and header length fields. The
removal of the checksum field in IPv6 allows for faster processing
at the routers without sacrificing functionality. In IPv6, error
detection and correction are handled at the data link and the TCP
layers. Note also that IPv6 can accommodate the QoS
requirements for some applications. Besides all these significant
advantages, IPv6 can provide built-in security features such as
confidentiality and authentication.

IPv6 Addressing Format


With its large address spacing, IPv6 network addressing is
very flexible. To efficiently represent the 128-bit address of IPv6 in
a compact form, hexadecimal digits are used. A colon separates
each of the four hexadecimal digits. For example, [2FB4 : 10AB :
4123 : CEBF : 54CD : 3912 : AE7B : 0932] can be a source address.
In practice, IPv6 addresses contain a lot of bits that are zero. The
address is commonly denoted in a more compact form. For
example, an address denoted by [2FB4 : 0000 : 0000 : 0000 : 54CD
: 3912 : 000B : 0932] can be compressed to [2FB4:::: 54CD:3912:B:
932].

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The network address space is classified into various types,


each of which is assigned a binary prefix. Currently, only a small
portion of the address space has been assigned, with the
remaining reserved for future use. One of the address types with a
leading byte of 1s is assigned for multicast; the rest of the
currently assigned types are used for unicast applications. Apart
from the unicast and multicast addresses, IPv6 introduces anycast
addresses. An anycast address is similar to a multicast address and
identifies a group of network devices for making connections.
However, unlike with multicast addressing, a packet needs to be
forwarded to any one device in the group. Anycast addresses share
the address space with unicast address types. IPv6 reserves some
addresses for special purposes.

Extension Header
Extension headers are positioned between the header and the
payload. If multiple extension headers are used, they are
concatenated, as shown in Figure 2.10, making it mandatory for
them to be processed in the sequence in which they are listed.
Figure 2.10 specifies the sequence in which the extension headers
are to be listed.
Figure 2.10. Concatenated IPv6 extension header

Packet Fragmentation
In IPv6, fragmentation is permitted only at the source. The
result of this restriction is faster processing of packets at routers.
Before transmitting a packet, a host performs a maximum
transmission unit (MTU) discovery in the route of the packet. The
minimum MTU obtained determines the packet size and thus
requires the route from the host to the destination to remain
steady. If this minimum value of the physical network is less than
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the packet size to be transmitted, the intermediate router discards


the packet and sends an error message back to the source. In rare
cases, the packet needs to be fragmented, and the extension
header contains the fragmentation information.

2. Describe briefly the Internet control message protocol


(ICMP). (A/M-11, N/D-11, 12, M/J-2012)
In connectionless routing, routers operate autonomously. They
forward and deliver packets without requiring any coordination with
the source. In large communication networks, IP may not be able to
deliver a packet to its destination, owing to possible failures in the
connectivity of a destination. Besides the hardware failure, other
factors may be present to create this problem. For example, as
noted in Section 2.3.1, the time-to-live field in an IP packet
specifies the maximum number of hops after which a packet must
be discarded. If the counter of this field expires, packet delivery too
can become impossible.
Another issuerelated and equally importantis that a sender
cannot know whether a delivery failure is a result of a local or a
remote technical difficulty. With TCP/IP, routers in a network can
report errors through the Internet Control Message Protocol (ICMP).
An ICMP message is encapsulated in the data portion of an IP
datagram (packet). When an error occurs, ICMP reports it to the
originating source of the connection. This is compatible with the
fact that an IP datagram header itself specifies only the original
source and not any routers. The source must interpret the error.
One of the important ICMP messages is the redirect message.
In Figure 2.8, a source tries to send a message to a destination. But
R1 incorrectly sends the message to a wrong path (R1-R3-R4-R5-
R6) instead of to the short one (R1-R2-R6). In this case, if in the
middle of routing, R5 or R6 finds out about this error, it cannot
issue an ICMP message to R1 to correct the routing, as they do not
know the address of R1. Instead, they issue a redirect ICMP
message to the source.

Figure 2.8. With ICMP, a redirect message cannot be sent to R1, since R6 does
not know the address of R1.

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3. Describe any one routing algorithm. (N/D-2011)


Classification of routing algorithms:
 least-cost-path algorithms
 non-least-cost-path routing
 intradomain routing protocols
 interdomain routing protocols

Least-cost-path algorithms
The majority of Internet routing methods are based on least-
cost algorithms. In such algorithms, a link cost is proportional to
the links's current traffic load. However, the link cost may not
always be proportional to the current load. The link cost is defined
on both directions between each pair of nodes. Several least-cost-
path algorithms have been developed for packet-switched
networks. In particular, Dijkstra's algorithm and the Bellman-Ford
algorithm are the most effective and widely used algorithms.

Dijkstra's Algorithm

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Dijkstra's algorithm is a centralized routing algorithm that maintains information in a


central location. The objective is to find the least-cost path from a given source node
to all other nodes. This algorithm determines least-cost paths from a source node to a
destination node by optimizing the cost in multiple iterations.
Dijkstra's algorithm is as follows:

Begin Dijkstra's Algorithm


1. Define:
s=Source node
k=Set of visited nodes by the algorithm
.ij=Cost of the link from node i to node j
. ij=Cost of the least-cost path from node i to node j

2.Initialize:
k ={s}
sj = .sj for j s

3. Next node:
Find x k that .sx = min .sj for j k.
Add x to k.
4. Least-cost paths:
.sj = min(.sj, .sx + .xj) for j k

If any two nodes i and j are not connected directly, the cost for that link is
infinity, indicated by .ij =x. Steps 2 and 3 are repeated until paths are assigned to all
nodes. At step 1, k represents s, and .sj computes the cost of the least-cost path from
s to node j. At step 2, we want to find x among the neighbouring nodes but not in k
such that the cost is minimized. At step 3, we simply update the least-cost path. The
algorithm ends when all nodes have been visited and included in the algorithm.

Bellman-Ford Algorithm
The Bellman-Ford algorithm finds the least-cost path from a source to a destination
by passing through no more than links. The essence of the algorithm consists of the
following steps

Begin Bellman-Ford Algorithm


1. Define:

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s = Source node
.ij = Cost of the link from node i to node j
.ij( ) = Cost of the least-cost path from i to j with no more than links

2. Initialize:
.sj(0) = x, for all j s
ss( ) = 0, for all
3. Least-cost path:
for any node j s with predecessor node i:
.sj( + 1) = mini[.si( ) + .ij]

If any two nodes i and j are not connected directly, .ij( ) =x. At step 2, every value of
is initialized. At step 3, we increase the number of links in a sequence of iterations.
During each iteration, we find the least-cost path, given the value. The algorithm
ends when all nodes have been visited and included in the algorithm.

4. What is dynamic host configuration protocol? Explain in detail. (N/D-2011,


2012)
A different method of assigning addresses to a host is called Dynamic Host
Configuration Protocol (DHCP), whereby a host is allocated an IP address
automatically. DHCP allows a host to learn its subnet mask, the address of its first-
hop router, or even the address of other major local servers. Because this addressing
automation lets a host learn several key pieces of information in a network, DHCP is
sometimes called a plug-and-play protocol, whereby hosts can join or leave a
network without requiring configuration by network managers.
The convenience of this method of address assignment gives DHCP multiple
uses of IP addresses. If any ISP manager does not have a sufficient number of IP
addresses, DHCP is used to assign each of its connecting hosts a temporary IP
address. When a host joins or leaves, the management server must update its list of
available IP addresses. If a host joins the network, the server assigns an available IP
address; each time a host leaves, its address is included in the pool of available
addresses. DHCP is especially useful in mobile IP, with mobile hosts joining and
leaving an ISP frequently.

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To contact a DHCP server, a newly booted or attached host sends a


DHCPDISCOVER message to a special IP address (255.255.255.255) that is an IP
broadcast address. This means it will be received by all hosts and routers on that
network. (Routers do not forward such packets onto other networks, preventing
broadcast to the entire Internet.) In the simplest case, one of these nodes is the
DHCP server for the network. The server would then reply to the host that generated
the discovery message (all the other nodes would ignore it). However, it is not really
desirable to require one DHCP server on every network, because this still creates a
potentially large number of servers that need to be correctly and consistently
configured. Thus, DHCP uses the concept of a relay agent. There is at least one relay
agent on each network, and it is configured with just one piece of information: the IP
address of the DHCP server. When a relay agent receives a DHCPDISCOVER
message, it unicasts it to the DHCP server and awaits the response, which it will then
send back to the requesting client. The process of relaying a message from a host to
a remote DHCP server is shown in Figure 4.9.
Figure 4.10 shows the format of a DHCP message. The message is actually
sent using a protocol called the User Datagram Protocol (UDP) that runs over IP.
UDP is discussed in detail in the next chapter, but the only interesting thing it does
in this context is to provide a demultiplexing key that says, “This is a DHCP
packet.”
DHCP is derived from an earlier protocol called BOOTP, and some of the
packet fields are thus not strictly relevant to host configuration. When trying to
obtain configuration information, the client puts its hardware address (e.g., its
Ethernet address) in the chaddr field. The DHCP server replies by filling in the

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yiaddr (“your” IP address) field and sending it to the client. Other information such
as the default router to be used by this client can be included in the options field.
In the case where DHCP dynamically assigns IP addresses to hosts, it is clear that
hosts cannot keep addresses indefinitely, as this would eventually cause the server to
exhaust its address pool. At the same time, a host cannot be depended upon to give
back its address, since it might have crashed, been unplugged from the network, or
been turned off. Thus, DHCP allows addresses to be “leased” for some period of
time. Once the lease expires, the server is free to return that address to its pool. A
host with a leased address clearly needs to renew the lease periodically if in fact it is
still connected to the network and functioning correctly.

5. Write a note on addressing. (N/D-2011)


Four levels of addresses are used in an internet employing the TCP/IP protocols:

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Physical Addresses:
The physical address, also known as the link address, is the address of a node
as defined by its LAN or WAN. It is included in the frame used by the data link
layer. It is the lowest-level address.
The physical addresses have authority over the network (LAN or WAN). The
size and format of these addresses vary depending on the network. For example,
Ethernet uses a 6-byte (48-bit) physical address that is imprinted on the network
interface card (NIC). LocalTalk (Apple), however, has a I-byte dynamic address that
changes each time the station comes up.

Logical Addresses:
Logical addresses are necessary for universal communications that are
independent of underlying physical networks. Physical addresses are not adequate in
an internetwork environment where different networks can have different address

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formats. A universal addressing system is needed in which each host can be


identified uniquely, regardless of the underlying physical network.
The logical addresses are designed for this purpose. A logical address in the
Internet is currently a 32-bit address that can uniquely define a host connected to the
Internet. No two publicly addressed and visible hosts on the Internet can have the
same IP address.

The computer with logical address A and physical address 10 needs to send a
packet to the computer with logical address P and physical address 95. We use letters
to show the logical addresses and numbers for physical addresses, but note that both
are actually numbers, as we will see later in the chapter.
The sender encapsulates its data in a packet at the network layer and adds two
logical addresses (A and P). Note that in most protocols, the logical source address
comes before the logical destination address (contrary to the order of physical
addresses). The network layer, however, needs to find the physical address of the
next hop before the packet can be delivered. The network layer consults its routing
table (see Chapter 22) and finds the logical address of the next hop (router I) to be F.
The ARP discussed previously finds the physical address of router 1 that
corresponds to the logical address of 20. Now the network layer passes this address

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to the data link layer, which in tum, encapsulates the packet with physical
destination address 20 and physical source address 10.
The frame is received by every device on LAN 1, but is discarded by all
except router 1, which finds that the destination physical address in the frame
matches with its own physical address. The router decapsulates the packet from the
frame to read the logical destination address P. Since the logical destination address
does not match the router's logical address, the router knows that the packet needs to
be forwarded. The router consults its routing table and ARP to find the physical
destination address of the next hop (router 2), creates a new frame, encapsulates the
packet, and sends it to router 2.

Port Addresses
The IP address and the physical address are necessary for a quantity of data to
travel from a source to the destination host. However, arrival at the destination host
is not the final objective of data communications on the Internet. A system that sends
nothing but data from one computer to another is not complete. Today, computers
are devices that can run multiple processes at the same time. The end objective of
Internet communication is a process communicating with another process. For
example, computer A can communicate with computer C by using TELNET. At the
same time, computer A communicates with computer B by using the File Transfer
Protocol (FTP). For these processes to receive data simultaneously, we need a
method to label the different processes. In other words, they need addresses. In the
TCPIIP architecture, the label assigned to a process is called a port address. A port
address in TCPIIP is 16 bits in length.

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6. Write short notes on:


(i) TCP/IP protocol.
(ii) Network model.
(iii) Datagram network.
(iv) Virtual circuit network. (M/J-2012)

(i) TCP/IP protocol.

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TCP/IP is a hierarchical protocol made up of interactive modules, each of


which provides a specific functionality; however, the modules are not necessarily
interdependent. Whereas the OSI model specifies which functions belong to each of
its layers, the layers of the TCP/IP protocol suite contain relatively independent
protocols that can be mixed and matched depending on the needs of the system. The
term hierarchical means that each upper-level protocol is supported by one or more
lower-level protocols.
At the transport layer, TCP/IP defines three protocols: Transmission Control
Protocol (TCP), User Datagram Protocol (UDP), and Stream Control Transmission
Protocol (SCTP). At the network layer, the main protocol defined by TCP/IP is the
Internetworking Protocol (IP); there are also some other protocols that support data
movement in this layer.

Physical and Data Link Layers


At the physical and data link layers, TCPIIP does not define any specific
protocol. It supports all the standard and proprietary protocols. A network in a
TCPIIP internetwork can be a local-area network or a wide-area network.

Network Layer
At the network layer (or, more accurately, the internetwork layer), TCP/IP
supports the Internetworking Protocol. IP, in turn, uses four supporting protocols:
ARP, RARP, ICMP, and IGMP.

Internetworking Protocol (IP)


The Internetworking Protocol (IP) is the transmission mechanism used by the
TCP/IP protocols. It is an unreliable and connectionless protocol-a best-effort
delivery service. The term best effort means that IP provides no error checking or
tracking. IP assumes the unreliability of the underlying layers and does its best to get
a transmission through to its destination, but with no guarantees.
IP transports data in packets called datagrams, each of which is transported
separately. Datagrams can travel along different routes and can arrive out of
sequence or be duplicated. IP does not keep track of the routes and has no facility for
reordering datagrams once they arrive at their destination.
The limited functionality of IP should not be considered a weakness, however.
IP provides bare-bones transmission functions that free the user to add only those
facilities necessary for a given application and thereby allows for maximum
efficiency.

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Address Resolution Protocol


The Address Resolution Protocol (ARP) is used to associate a logical address
with a physical address. On a typical physical network, such as a LAN, each device
on a link is identified by a physical or station address, usually imprinted on the
network interface card (NIC). ARP is used to find the physical address of the node

Reverse Address Resolution Protocol


The Reverse Address Resolution Protocol (RARP) allows a host to discover
its Internet address when it knows only its physical address. It is used when a
computer is connected to a network for the first time or when a diskless computer is
booted.

Internet Control Message Protocol


The Internet Control Message Protocol (ICMP) is a mechanism used by hosts
and gateways to send notification of datagram problems back to the sender. ICMP
sends query and error reporting messages.

Internet Group Message Protocol


The Internet Group Message Protocol (IGMP) is used to facilitate the
simultaneous transmission of a message to a group of recipients.

Transport Layer
Traditionally the transport layer was represented in TCP/IP by two protocols:
TCP and UDP. IP is a host-to-host protocol, meaning that it can deliver a packet
from one physical device to another. UDP and TCP are transport level protocols
responsible for delivery of a message from a process (running program) to another
process. A new transport layer protocol, SCTP, has been devised to meet the needs of
some newer applications.

User Datagram Protocol


The User Datagram Protocol (UDP) is the simpler of the two standard TCPIIP
transport protocols. It is a process-to-process protocol that adds only port addresses,
checksum error control, and length information to the data from the upper layer.

Transmission Control Protocol


The Transmission Control Protocol (TCP) provides full transport-layer services to

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applications. TCP is a reliable stream transport protocol. The term stream, in this
context, means connection-oriented: A connection must be established between both
ends of a transmission before either can transmit data.
At the sending end of each transmission, TCP divides a stream of data into
smaller units called segments. Each segment includes a sequence number for
reordering after receipt, together with an acknowledgment number for the segments
received. Segments are carried across the internet inside of IP datagrams. At the
receiving end, TCP collects each datagram as it comes in and reorders the
transmission based on sequence numbers.

Stream Control Transmission Protocol


The Stream Control Transmission Protocol (SCTP) provides support for
newer applications such as voice over the Internet. It is a transport layer protocol
that combines the best features of UDP and TCP.

Application Layer
The application layer in TCPIIP is equivalent to the combined session, presentation,
and application layers in the OSI modeL Many protocols are defined at this layer.

(ii) Virtual circuit network


A virtual-circuit network is a cross between a circuit-switched network and a
datagram network. It has some characteristics of both.

1. As in a circuit-switched network, there are setup and teardown phases in addition


to the data transfer phase.
2. Resources can be allocated during the setup phase, as in a circuit-switched
network, or on demand, as in a datagram network.
3. As in a datagram network, data are packetized and each packet carries an address
in the header. However, the address in the header has local jurisdiction (it defines
what should be the next switch and the channel on which the packet is being canied),
not end-to-end jurisdiction. The reader may ask how the intermediate switches know
where to send the packet if there is no final destination address carried by a packet.
The answer will be clear when we discuss virtual-circuit identifiers in the next
section.
4. As in a circuit-switched network, all packets follow the same path established
during the connection.
5. A virtual-circuit network is normally implemented in the data link layer, while a

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circuit-switched network is implemented in the physical layer and a datagram


network in the network layer. But this may change in the future.

Figure 8.10 is an example of a virtual-circuit network. The network has switches that
allow traffic from sources to destinations. A source or destination can be a computer,
packet switch, bridge, or any other device that connects other networks.

Figure 8.10 Virtual-circuit network

Addressing
In a virtual-circuit network, two types of addressing are involved: global and
local (virtual-circuit identifier).

Global Addressing
A source or a destination needs to have a global address-an address that can be
unique in the scope of the network or internationally if the network is part of an
international network. However, we will see that a global address in virtual-circuit
networks is used only to create a virtual-circuit identifier, as discussed next.
Virtual-Circuit Identifier
The identifier that is actually used for data transfer is called the virtual-circuit
identifier (Vel). A vel, unlike a global address, is a small number that has only switch
scope; it is used by a frame between two switches. When a frame arrives at a switch,
it has a VCI; when it leaves, it has a different VCl. Figure 8.11 shows how the VCI
in a data frame changes from one switch to another. Note that a VCI does not need to
be a large number since each switch can use its own unique set of VCls.

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7. Perform a comparative study between the ISO-OSI model and TCP/IP


reference model. (M/J-2012)
The OSI model is a layered framework for the design of network systems that
allows communication between all types of computer systems. It consists of seven
separate but related layers, each of which defines a part of the process of moving
information across a network (see Figure 2.2). An understanding of the fundamentals
of the OSI model provides a solid basis for exploring data communications

Layered Architecture
The OSI model is composed of seven ordered layers: physical (layer 1), data
link (layer 2), network (layer 3), transport (layer 4), session (layer 5), presentation
(layer 6), and application (layer 7). Figure 2.3 shows the layers involved when a
message is sent from device A to device B. As the message travels from A to B, it
may pass through many intermediate nodes. These intermediate nodes usually
involve only the first three layers of the OSI model.
In developing the model, the designers distilled the process of transmitting
data to its most fundamental elements. They identified which networking functions

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had related uses and collected those functions into discrete groups that became the
layers. Each layer defines a family of functions distinct from those of the other
layers. By defining and localizing functionality in this fashion, the designers created
an architecture that is both comprehensive and flexible. Most importantly, the OSI
model allows complete interoperability between otherwise incompatible systems.
Within a single machine, each layer calls upon the services of the layer just
below it. Layer 3, for example, uses the services provided by layer 2 and provides
services for layer 4. Between machines, layer x on one machine communicates with
layer x on another machine. This communication is governed by an agreed-upon
series of rules and conventions called protocols. The processes on each machine that
communicate at a given layer are called peer-to-peer processes. Communication
between machines is therefore a peer-to-peer process using the protocols appropriate
to a given layer.

Peer-to-Peer Processes
At the physical layer, communication is direct: In Figure 2.3, device A sends a
stream of bits to device B (through intermediate nodes). At the higher layers,
however, communication must move down through the layers on device A, over to
device B, and then

back up through the layers. Each layer in the sending device adds its own
information to the message it receives from the layer just above it and passes the
whole package to the layer just below it.

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At layer I the entire package is converted to a form that can be transmitted to


the receiving device. At the receiving machine, the message is unwrapped layer by
layer, with each process receiving and removing the data meant for it. For example,
layer 2 removes the data meant for it, then passes the rest to layer 3. Layer 3 then
removes the data meant for it and passes the rest to layer 4, and so on.

Interfaces Between Layers


The passing of the data and network information down through the layers of
the sending device and back up through the layers of the receiving device is made
possible by an interface between each pair of adjacent layers. Each interface defines
the information and services a layer must provide for the layer above it. Well-defined
interfaces and layer functions provide modularity to a network. As long as a layer
provides the expected services to the layer above it, the specific implementation of
its functions can be modified or replaced without requiring changes to the
surrounding layers.
Organization ofthe Layers
The seven layers can be thought of as belonging to three subgroups. Layers I,
2, and 3-physical, data link, and network-are the network support layers; they deal
with the physical aspects of moving data from one device to another (such as
electrical specifications, physical connections, physical addressing, and transport
timing and reliability). Layers 5, 6, and 7-session, presentation, and application-can
be thought of as the user support layers; they allow interoperability among unrelated
software systems. Layer 4, the transport layer, links the two subgroups and ensures
that what the lower layers have transmitted is in a form that the upper layers can use.
The upper OSI layers are almost always implemented in software; lower layers are a
combination of hardware and software, except for the physical layer, which is
mostly hardware.
In Figure 2.4, which gives an overall view of the OSI layers, D7 means the
data unit at layer 7, D6 means the data unit at layer 6, and so on. The process starts
at layer 7 (the application layer), then moves from layer to layer in descending,
sequential order. At each layer, a header, or possibly a trailer, can be added to the
data unit. Commonly, the trailer is added only at layer 2. When the formatted data
unit passes through the physical layer (layer 1), it is changed into an electromagnetic
signal and transported along a physical link.

8. Compare Address resolution protocol and RARP. (M/J-2013)


Address Resolution Protocol (ARP)

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The Address Resolution Protocol (ARP) is designed to convert IP addresses to


MAC addresses or vice versa. Suppose that a user wants to transmit a packet to its
destination. If it does not know the link-layer address of the destination, the sender
broadcasts an ARP packet requesting the link-layer address given the IP address.
The destination is denoted by its IP address in the ARP request. Only the destination
replies with its link-layer address. The sender then stores this address in the local
ARP table for its subsequent use. Each networking device, such as a host or a router,
has an interface, or adapter. Each adapter has a table that keeps the MAC address of
each device within the network it is attached to. MAC addresses are listed in the
adapter's ARP table.
Figure 5.4 shows that LAN1 has several users. A user with IP address
133.176.8.55 and adapter MAC address AB-49-9B-25-B1-66 wants to send a frame
within its LAN to user 133.176.8.56 with adapter MAC address 11-40-33-55-A3-57.
In this case, original packets are converted to frames in its adapter with the
destination address AB-49-9B-25-B1-66. Now suppose that the same sender wants
to send frames outside its LAN to a user in LAN3 with IP address 198.34.7.25
connected through a four-port router. But assume that the sender does not know the
MAC address of the destination. In this case, the sender must broadcast an ARP
packet to all users, but only the destination user replies to the ARP query, providing
its adapter MAC address of 33-BA-76-55-A3-BD. This new address is used for the
router to forward the message of the sending user to the destination.
Figure 5.4. ARP packets and MAC and port adapters

Reverse Address Resolution Protocol (RARP)


Reverse Address Resolution Protocol (RARP) finds the logical address for a
machine that knows only its physical address. Each host or router is assigned one or
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more logical (IP) addresses, which are unique and independent of the physical
(hardware) address of the machine. To create an IP datagram, a host or a router needs
to know its own IP address or addresses. The IP address of a machine is usually read
from its configuration file stored on a disk file.
However, a diskless machine is usually booted from ROM, which has
minimum booting information. The ROM is installed by the manufacturer. It cannot
include the IP address because the IP addresses on a network are assigned by the
network administrator.
The machine can get its physical address (by reading its NIC, for example),
which is unique locally. It can then use the physical address to get the logical address
by using the RARP protocol. A RARP request is created and broadcast on the local
network. Another machine on the local network that knows all the IP addresses will
respond with a RARP reply. The requesting machine must be running a RARP client
program; the responding machine must be running a RARP server program.
There is a serious problem with RARP: Broadcasting is done at the data link
layer. The physical broadcast address, allis in the case of Ethernet, does not pass the
boundaries of a network. This means that if an administrator has several networks or
several subnets, it needs to assign a RARP server for each network or subnet. This is
the reason that RARP is almost obsolete. 1\vo protocols, BOOTP and DHCp, are
replacing RARP.

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