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EC3401 NETWORKS AND SECURITY

MRK INSTITUTE OF TECHNOLOGY


(Approved by AICTE & Affiliated to Anna University, Chennai)
Kattumannarkoil – 608 301
An ISO 9001:2008 Certified Institution

UNIT – I: NETWORK MODELS AND DATALINK LAYER


(16 Marks Questions with Answers)

1. With a neat sketch, explain the architecture of an OSI seven layer model.(NOV / DEC 2017)
OR Draw the OSI network architecture and explain the function each layer in detail (NOV /
DEC 2016), (NOV / DEC 2015)
The ISO-OSI model is a seven layer architecture. It defines seven layers or levels in a complete
communication system.

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Physical layer:
 Data rate – Transmission rate represents send no of bits per second.
 Synchronizing – The sender and receiver must be synchronized at the bit level. Send and
Receive bits are in synchronize.
 Physical Topology – Defines how devices are connected to make a network.

Data Link Layer:


It is responsible for the node-to-node delivery. It has the response to make the physical layer as
error free for the upper layers and specific responsibilities of the data link layer. It deals the
followings:
 Framing – The data link layer divides the data packets received from the network layer into
manageable data units called frames.
 Physical addressing – The data link layer adds the physical address of the sender/receiver to
the frames which are transmitted from the data link layer to the network layer.
 Flow control – If the data produced by the sender is more than the receiver’s reception, then
congestion occurs in the network. To avoid this condition, by default the data link layer have
some flow control mechanism for the overwhelming of receiver side.
 Error control – The data link layer also have error control mechanism at bit level.
Example: Cyclic redundancy check
Network Layer:
This layer presents between the data link layer and the transport layer. It is responsible for the
source-to-destination delivery of a packet possibly across multiple It deals the followings:
 Logical Addressing – The data link layer handles the addressing problem locally by using the
physical address, outside the network we need one more address named logical address.

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 The network layer adds a header to the packet coming from the upper layer includes the
logical addresses of sender/receiver.
 Routing – In routing, some routing devices
Transport Layer:
The transport layer is responsible for end-to-end delivery of the entire message. For added
security, the transport layer may create a connection between the two end ports involves three steps:
 Connection establishment,
 Data transfer,
 Connection release.
The transport layer has more control over sequencing, flow and error detection and correction.
Application Layer:
Application layer provides user interfaces and support for services such as electronic mail,
remote file access and transfer, shared database management, and other types of distributed
information devices. It deals the following:
 File transfer, access, and management
 Mail Services
 Directory Services
Application layer protocols: Telnet, SMTP, SNMP, HTTP, FTP.
2. Discuss the approaches used for error detection in networking. (NOV / DEC 2017).
Whenever a message is transmitted, it may get scrambled by noise or data may get corrupted. To
avoid this, we use error-detecting codes which are additional data added to a given digital message
to help us detect if any error has occurred during transmission of the message.
Basic approach used for error detection is the use of redundancy bits, where additional bits are
added to facilitate detection of errors.
Some popular techniques for error detection are:
1. Simple Parity check
2. Two-dimensional Parity check
3. Checksum
4. Cyclic redundancy check

3. Explain the challenges faced in buildings a network. (APRIL / MAY 2017)

Typical causes that can be identified and treated, however, generally fall within one of the
following three categories:

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 Performance Degradation
 Host Identification
 Security Issues
Performance Degradation: Performance degradation refers to issues involving loss of speed and
data integrity due to poor transmissions.
Host Identification: Proper configuration is also essential to maintaining proper host identification.
Just as the post office cannot deliver messages without some form of addressing, neither can
computer networking hardware.
Security Issues: Network security issues involve maintaining network integrity, preventing
unauthorized users from infiltrating the system (viewing/stealing sensitive data, passwords, etc.), and
protecting the network denial of service attacks.

4. Obtain the 4-bit CRC code for the data bit sequence 10011011100 using the polynomial
x4+x2+1. (APRIL / MAY 2017)

5. With a
protocol graph, explain the architecture of internet. (APRIL / MAY 2017)
Communications between computers on a network is done through protocol suits. The most
widely used and most widely available protocol suite is TCP/IP protocol suite. A protocol suit
consists of a layered architecture where each layer depicts some functionality which can be carried
out by a protocol. Each layer usually has more than one protocol options to carry out the

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responsibility that the layer adheres to. TCP/IP is normally considered to be a 4 layer system. The 4
layers are as follows :
1. Application layer
2. Transport layer
3. Network layer
4. Data link layer

6. Consider a bus LAN with a number of equally spaced with a data rate of 9Mbps and a bus
length of 1km. what is the mean time to send a frame of 500 bits to another station, measured
from the beginning of transmission to the end of reception? Assume a propagation speed of
150m/s. if two stations begin to monitor and transmit at the same time, how long does it need
to wait before an interference is noticed? (APRIL / MAY 2017)

Assume a mean distance between stations of 0.375 km. This is an approximation based on the
following observation. For a station on one end, the average distance to any other station is 0.5 km.
For a station in the center, the average distance is 0.25 km. With this assumption, the time to send
equals transmission time plus propagation time.
T = (10^3 bits / 10^7 bps) + (375 m / 200 ×106 m / sec)= 102µ sec
At 1 Mbps 1 bit / 1Mbps = 1µsec Length = 200 m / µs * 1µsec =200m
At 40 Mbps 1 bit / 40 Mbps = 0.025 µsec Length = 200 m / µs * 0.025 µsec = 5m
7. Discuss in detail about the network performance measure (NOV / DEC 2016)
The following measures are often considered important:
 Bandwidth commonly measured in bits/second is the maximum rate that information can be
transferred
 Throughput is the actual rate that information is transferred
 Latency the delay between the sender and the receiver decoding it, this is mainly a function
of the signals travel time, and processing time at any nodes the information traverses
 Jitter variation in packet delay at the receiver of the information
 Error rate the number of corrupted bits expressed as a percentage or fraction of the total sent
8. Explain selective repeat ARQ flow control method. (NOV / DEC 2016)

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Selective Repeat ARQ
In Go-back-N ARQ, it is assumed that the receiver does not have any buffer space for its window
size and has to process each frame as it comes. This enforces the sender to retransmit all the frames
which are not acknowledged.

In Selective-Repeat ARQ, the receiver while keeping track of sequence numbers, buffers the
frames in memory and sends NACK for only frame which is missing or damaged. The sender in this
case, sends only packet for which NACK is received.
9. Explain any two error detection mechanism in detail. (MAY / JUNE 2016)
VRC: Vertical redundancy check (VRC) is an error-checking method used on an eight-bit ASCII
character. In VRC, a parity bit is attached to each byte of data, which is then tested to determine
whether the transmission is correct. VRC is considered an unreliable error-detection method because
it only works if an even number of bits is distorted.A vertical redundancy check is also called a
transverse redundancy check when used in combination with other error-controlling codes such as a
longitudinal redundancy check.
LRC: A longitudinal redundancy check (LRC) is an error-detection method for determining the
correctness of transmitted and stored data. LRC verifies the accuracy of stored and transmitted data
using parity bits. It is a redundancy check applied to a parallel group of bit streams. The data to be
transmitted is divided into transmission blocks into which additional check data is inserted. This
term is also known as a horizontal redundancy check.
10. Explain in detail about HDLC (MAY / JUNE 2016)
The HDLC protocol is a general purpose protocol which operates at the data link layer of
the OSI reference model. The protocol uses the services of a physical layer, and provides either

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a best effort or reliablecommunications path between the transmitter and receiver (i.e.
with acknowledged data transfer). The type of service provided depends upon the HDLC
mode which is used.
Each piece of data is encapsulated in an HDLC frame by adding a trailer and a header. The
header contains an HDLC address and an HDLC control field. The trailer is found at the end of the
frame, and contains a Cyclic Redundancy Check (CRC) which detects any errors which may occur
during transmission. The frames are separated by HDLC flag sequences which are transmitted
between each frame and whenever there is no data to be transmitted.

11. Explain in detail about PPP (MAY / JUNE 2016)


In computer networking, Point-to-Point Protocol (PPP) is a data link (layer 2) protocol used to
establish a direct connection between two nodes. It can provide connection authentication,
transmission encryption (using ECP, RFC 1968), and compression.
PPP is used over many types of physical networks including serial cable, phone line, trunk
line, cellular telephone, specialized radio links, and fiber optic links such as SONET. PPP is also
used over Internet access connections. Internet service providers (ISPs) have used PPP for
customer dial-up access to the Internet, since IP packets cannot be transmitted over a modem line on
their own, without some data link protocol.
Two derivatives of PPP, Point-to-Point Protocol over Ethernet (PPPoE) and Point-to-Point
Protocol over ATM (PPPoA), are used most commonly by Internet Service Providers (ISPs) to
establish a Digital Subscriber Line (DSL) Internet service connection with customers.
PPP is commonly used as a data link layer protocol for connection
over synchronous and asynchronous circuits, where it has largely superseded the older Serial Line
Internet Protocol (SLIP) and telephone company mandated standards (such as Link Access Protocol,
Balanced (LAPB) in the X.25 protocol suite). The only requirement for PPP is that the circuit
provided be duplex. PPP was designed to work with numerous network
layer protocols,including InternetProtocol (IP), TRILL,Novell's InternetworkPacketExchange (IPX),

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NBF, DECnet and AppleTalk. Like SLIP, this is a full Internet connection over telephone lines via
modem. It is more reliable than SlIP because it double checks to make sure that Internet
12. Explain the various flow control mechanisms. (NOV / DEC 2015)
Network services can be categorized as best-effort, connectionless services or reliable
connection- oriented services. In the Internet protocol suite, IP is a best-effort service and TCP is a
reliable service. IP provides basic packet forwarding while TCP implements flow controls, acknowl-
edgements, and retransmissions of lost or corrupted packets. This split in services "decentralizes" the
network and moves the responsibility for reliable delivery to end systems. TCP is an end-to-end
transport protocol, meaning that it runs in end systems, not the network. IP is a network protocol.
This topic discusses flow control in terms of TCP, but flow controls are used by many other
communication protocols, as mentioned later. The services offered by TCP include the following:
Flow-control mechanisms control packet flow so that a sender does not transmit more packets
than a receiver can process.
Reliable delivery mechanisms provide a way for a receiving system to acknowledge that it has
received a packet, and a way for the sender to know that it must retransmit a lost or corrupted packet.
Refer to "Reliable Data Delivery Services."
Congestion control mechanisms allow network systems to detect network congestion (a
condition in which there is more traffic on the network than can be handled by the network or
network devices) and throttle back their transmission to alleviate the congestion. Refer to
"Congestion Control Mechanisms."
13. Discuss in detail about Internet Architecture. (APRIL / MAY 2015)
The Internet's architecture is described in its name, a short from of the compound word "inter-
networking". This architecture is based in the very specification of the standard TCP/IP protocol,
designed to connect any two networks which may be very different in internal hardware, software,
and technical design. Once two networks are interconnected, communication with TCP/IP is
enabled end-to-end, so that any node on the Internet has the near magical ability to communicate
with any other no matter where they are. This openness of design has enabled the Internet
architecture to grow to a global scale.
In practice, the Internet technical architecture looks a bit like a multi-dimensional river system,
with small tributaries feeding medium-sized streams feeding large rivers. For example, an
individual's access to the Internet is often from home over a modem to a local Internet service
provider who connects to a regional network connected to a national network. At the office, a
desktop computer might be connected to a local area network with a company connection to a

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corporate Intranet connected to several national Internet service providers. In general, small local
Internet service providers connect to medium-sized regional networks which connect to large
national networks, which then connect to very large bandwidth networks on the Internet backbone.
Most Internet service providers have several redundant network cross-connections to other providers
in order to ensure continuous availability.
14. What is the need for error detection? Explain with typical examples. Explain methods used for
error detection and error correction. (APRIL / MAY 2015)
The main function of error detection is to make the probability of a receiving Data Link passing
up an incorrect frame to the client layer very, very small. This is known as the undetected error
probability; a possible goal for a system is one undetected error every 20 years. Recall that this is
necessary because most transport protocols and user application programs ignore the end-to-end
principle when it comes to the integrity of Data. End-to-end checksums are often not performed for
performance reasons; hence it is good engineering practice to require that the probability of
undetected errors on Data Links be low. Another reason, somewhat less important, is that
intermediate routers may make decisions based on bogus data and misdirect packets. Perhaps in the
future, if all workstations have hardware to do end-to-end checksums, the first reason will go away;
however, the second reason will remain.
Error-Detecting codes
Whenever a message is transmitted, it may get scrambled by noise or data may get corrupted. To
avoid this, we use error-detecting codes which are additional data added to a given digital message
to help us detect if an error occurred during transmission of the message. A simple example of error-
detecting code is parity check.
Error-Correcting codes
Along with error-detecting code, we can also pass some data to figure out the original message
from the corrupt message that we received. This type of code is called an error-correcting code.
Error-correcting codes also deploy the same strategy as error-detecting codes but additionally, such
codes also detect the exact location of the corrupt bit.
In error-correcting codes, parity check has a simple way to detect errors along with a
sophisticated mechanism to determine the corrupt bit location. Once the corrupt bit is located, its
value is reverted (from 0 to 1 or 1 to 0) to get the original message.

15. Explain about the transmission modes available for data flow.
Three transmission modes are available for data flow.

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 Simplex : One way communication, unidirectional. Any one of the station can transmitand
other can receive.
 Half Duplex : Each station can both transmit and receive, but not at the same time. Whenone
is sending, another is receiving. It is two-directional traffic. Examples – Walkie-talkies.
 Full Duplex : Both stations can transmit and receive simultaneously
16. Name four network topologies and explain them giving all features.
 Mesh
 Star
 Ring
 Bus

17. Explain in detail about the components used in data communication


The process of transferring data from one location to another is called Data Communication. In
this process, data is transmitted from one location to another by using transmission media.
Components of Data Communication:
The basics components or elements of data communication system are as follows:
 Message
 Sender
 Receiver
 Medium or Communication Channel
 Encoder and Decoder
18. Explain the categories of networks.
Three categories of networks are generally we can refer while we speak about the networks.They
are,
 Local Area Network (LAN)
 Metropolitan Area network (MAN)
 Wide Area Network (WAN)
19. List and discuss the various requirements in designing the network
A traditional network planning methodology in the context of business decisions involves five
layers of planning, namely:
 Need assessment and resource assessment

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 Short-term network planning
 IT resource sourcing
 Long-term and medium-term network planning
 Operations and maintenance.
Each of these layers incorporates plans for different time horizons, i.e. the business planning
layer determines the planning that the operator must perform to ensure that the network will perform
as required for its intended life-span. The Operations and Maintenance layer, however, examines
how the network will run on a day-to-day basis.
The network planning process begins with the acquisition of external information. This includes:
 Forecasts of how the new network/service will operate;
 The economic information concerning costs; and
 The technical details of the network’s capabilities.
Planning a new network/service involves implementing the new system across the first four
layers of the OSI Reference Model.[1] Choices must be made for the protocols and transmission
technologies.
20. Discuss in detail about cost effective resource sharing
 No single library can satisfy all the information needs of its users.
 Due to literature explosion, it is not possible for a library to acquire all the bibliographic
materials at one place.
 Individual libraries cannot afford the cost of acquiring the materials, hardware and software
and the manpower required to maintain the modern information technologies,
 Gradual decrease in budgetary provisions for the library and information centre to provide
various information services,
 Governments cuts on funding the information activities in view of liberalization,
globalization and privatization of such activities,
 Availability of access facilities for various types of databases, electronic information and
increases in users and their demands for information.
 Inadequacy of infrastructure facilities, storage space, etc for individual libraries to cope with
the modern trends
21. Discuss in detail about the variable –size framing approach
The data link layer, on the other hand, needs to pack bits into frames, so that each frame is
distinguishable from another. Framing in the data link layer separates a message from one source to

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a destination, or from other messages to other destinations, by adding a sender address and a
destination address. The destination address defines where the packet is to go; the sender address
helps the recipient acknowledge the receipt.
Although the whole message could be packed in one frame, that is not normally done. One
reason is that a frame can be very large, making flow and error control very inefficient. When a
message is carried in one very large frame, even a single-bit error would require the retransmission
of the whole message. When a message is divided into smaller frames, a single-bit error affects only
that small frame.
Variable-Size Framing:
In variable-size framing, we need to define the end of the frame and the beginning of the next. There
are two approaches which are used for this purpose: A character-oriented approach and A bit-
oriented approach.
22. Compare and contrast byte stuffing and bit stuffing.
Byte stuffing is the process of adding 1 extra byte whenever there is a flag or escape character in
the text while bit stuffing is the process of adding one extra 0 whenever five consecutive 1s follow a
0 in the data, so that the receiver does not mistake the pattern 0111110 for a flag. Byte stuffing is
used at byte-oriented protocols and bit stuffing is used at bit-oriented protocols
Bit stuffing is used for various purposes, such as for bringing bit streams that do not necessarily
have the same or rationally related bit rates up to a common rate, or to fill buffers or frames. The
location of the stuffing bits is communicated to the receiving end of the data link, where these extra
bits are removed to return the bit streams to their original bit rates or form. Bit stuffing may be used
to synchronize several channels before multiplexing or to rate-match two single channels to each
other.
23. Which technique is used in byte oriented protocols
Character-Oriented Protocols
In character-oriented protocol, we add special characters (called flag) to distinguish beginning
and end of a frame. Usually flag has 8-bit length. The character-oriented protocols are popular only
with text data. While using character–oriented protocol another problem is arises, pattern used for
the flag may also part of the data to send. If this happens, the destination node, when it encounters
this pattern in the middle of the data, assumes it has reached the end of the frame. To deal with this
problem, a byte stuffing (also known as character stuffing) approach was included to character-
oriented protocol. In byte stuffing a special byte is add to the data part, this is known as escape
character (ESC). The escape characters have a predefined pattern. The receiver removes the escape

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character and keeps the data part. It cause to another problem, if the text contains escape characters
as part of data. To deal with this, an escape character is prefixed with another escape character. The
following figure explains everything we discussed about character stuffing.
24. Which technique is used in bit oriented protocols
Bit-Oriented Protocols
In a bit-oriented protocol, the data to send is a series of bits. In order to distinguish frames, most
protocols use a bit pattern of 8-bit length (01111110) as flag at the beginning and end of each frame.
Here also cause the problem of appearance of flag in the data part to deal with this an extra bit
added. This method is called bit stuffing. In bit stuffing, if a 0 and five successive 1 bits are
encountered, an extra 0 is added. The receiver node removes the extra-added zero.

25. Explain CRC error detection mechanism with example


CRC is an error-detecting code. Its computation resembles a long division operation in which the
quotient is discarded and the remainder becomes the result, with the important distinction that the
arithmetic used is the carry-less arithmetic of a finite field. The length of the remainder is always
less than or equal to the length of the divisor, which therefore determines how long the result can be.
The definition of a particular CRC specifies the divisor to be used, among other things.
Although CRCs can be constructed using any finite field, all commonly used CRCs employ the
finite field, the field of two elements, usually called 0 and 1, comfortably matching computer
architecture. An important reason for the popularity of CRCs for detecting the accidental alteration
of data is their efficiency guarantee. Typically, an n-bit CRC, applied to a data block or arbitrary
length, will detect any single error burst not longer than n bits and will detect a fraction 1-2-n of all
longer error bursts. Errors in both data transmission channels and magnetic storage media tend to be
distributed non-randomly, making CRC properties more useful than alternative schemes such as
multiple parity checks.

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UNIT – II: NETWORK LAYER PROTOCOLS


1. Explain the functions of wi-fi and Bluetooth in detail. (NOV / DEC 2017).
 Bluetooth and Wi-Fi are both methods that provide wireless communication, but the difference
between the two mainly stems from what they are designed to do and how they are used.
 The main difference is that Bluetooth is primarily used to connect devices without using cables,
while Wi-Fi provides high-speed access to the internet.
 Bluetooth is a wireless technology standard that is used to exchange data over short distances
(less than 30 feet), usually between personal mobile devices.
 Bluetooth has a wide variety of applications, and has boosted the convenience and functionality
of portable devices by providing a simple way for them to interact with other Bluetooth-enabled
devices.
 Wi-Fi has some similar applications to Bluetooth, such as setting up a network or printing and
transferring files.
 It is also a wireless standard, but rather than being designed to communicate between devices, it
serves to wirelessly connect devices to the internet or Ethernet networks such as a corporate local
area network (LAN).
 This hotspot can be a small area such as a single room, or may cover several miles if hotspots are
allowed to overlap. Wi-Fi is a trademark name used to refer to devices that employ IEEE
802.11 standards.
2. Explain the datagram forwarding in IP. (NOV / DEC 2017).

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Header contains all information needed to deliver datagram to destination computer Destination
address
 Source address
 Identifier
 Other delivery information
Router examines header of each datagram and forwards datagram along path to destination
Datagrams can have different sizes
 Header area usually fixed (20 octets) but can have options
 Data area can contain between 1 octet and 65,535 octets (216 - 1)
 Usually, data area much larger than header

3. Show and explain the ARP packet format for mapping IP addresses into Ethernet addresses.
Explain the physical properties of Ethernet 802.3 with neat diagram of Ethernet transceiver
and adapter. (NOV / DEC 2017).
 The Address Resolution Protocol (ARP) is a communications protocol used for discovering
the link layer address associated with a given IPv4 address, a critical function in Internet
Protocol (IP) computer networks.
 ARP is used for mapping a network address (e.g. an IPv4 address) to a physical address like
a MAC address.
 The Address Resolution Protocol is a request and response protocol whose messages are
encapsulated by a link layer protocol. It is communicated within the boundaries of a single
network, never routed across internetwork nodes.

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 ARP may also be used as a simple announcement protocol. This is useful for updating other
hosts' mappings of a hardware address when the sender's IP address or MAC address has
changed.
 Such an announcement, also called a gratuitous ARP message, is usually broadcast as an
ARP request containing the sender's protocol address (SPA) in the target field (TPA=SPA),
with the target hardware address (THA) set to zero.

4. Discuss the working of CSMA/CD protocol. (APRIL / MAY 2017)


 CSMA/CD works by detecting a collision in the medium and backing off (after transmitting a
jam signal) as necessary.
 Assume, we have a LAN with a number of hosts. When one host wants to transmit a frame, it
listens on the medium to check if the medium is busy.
 Note that absence of any current (or the presence of a residual background current) means no
one else is transmitting.
 Once the host starts transmitting the frame, it also monitors the current level to detect a
collision.
 If it detects a collision, it transmits a special jam signal so that all other receivers can know
there was a collision.
 At this point, the other sender will receive this jam signal and detect a CRC check failure,
and abort.
 After this, both senders enter an exponential backoff phase and retry transmission.
5. Explain the functions of MAC layer present in IEEE 802.11 with necessary diagrams.
(APRIL / MAY 2017)
 IEEE 802.11 is a set of media access control (MAC) and physical layer (PHY) specifications
for implementing wireless local area network (WLAN).
 The standard and amendments provide the basis for wireless network products using the Wi-
Fi brand.
 The 802.11 family consists of a series of half-duplex over-the-air modulation techniques that
use the same basic protocol.
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 the first wireless networking standard in the family, but 802.11b was the first widely
accepted one, followed by 802.11a, 802.11g, 802.11n, and 802.11ac.
 Other standards in the family (c–f, h, j) are service amendments that are used to extend the
current scope of the existing standard, which may also include corrections to a previous
specification.
6. Consider sending a 3500-byte datagram that has arrived at a router R 1 that needs to be sent
over a link that has an MTU size of 1000 bytes to R 2. Then it has to traverse a link with an
MTU of 600 bytes. Let the identification number of the original datagram be 465. How many
fragments are delivered at the destination? Show the parameters associated with each of these
fragments. (APRIL / MAY 2017)
Assume that the DF flag was not set : )
Assume that no optional fields of the IP header are in use (i.e. IP header is 20 bytes)
The original datagram was 3500 bytes, subtracting 20 bytes for header, that leaves 3480 bytes of
data.
Assume the ID of the original packet is 'x'
With an MTU of 465 bytes, 465 - 20 = 445 bytes of data may be transmitted in each packet
Therefore, ceiling(3480 / 445) = 7.8 packets are needed to carry the data.
The packets will have the following characteristics (NOTE: offset is measured in 8 byte blocks, you
don't need to specify Total_len)
Packet 1: ID=x, Total_len=500, MF=1, Frag_offset=0
Packet 2: ID=x, Total_len=500, MF=1, Frag_offset=60
Packet 3: ID=x, Total_len=500, MF=1, Frag_offset=120
Packet 4: ID=x, Total_len=500, MF=1, Frag_offset=180
Packet 5: ID=x, Total_len=500, MF=1, Frag_offset=240
Packet 6: ID=x, Total_len=500, MF=1, Frag_offset=300
Packet 7: ID=x, Total_len=120, MF=0, Frag_offset=360

7. Explain the working of DHCP protocol with its header format. (APRIL / MAY 2017)

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 All Dynamic Host Configuration Protocol (DHCP) messages include a FIXED format
section and a VARIABLE format section
 The fixed format section consists of several fields that are the same in every Dynamic Host
Configuration Protocol (DHCP) message.
 The variable format section in the Dynamic Host Configuration Protocol (DHCP) contains
"OPTIONS", which carry additional configuration parameters.
8. Explain the physical properties of Ethernet 802.3 with neat diagram of Ethernet transceiver
and adapter. (NOV / DEC 2016).
Ethernet and IEEE 802.3 specify similar technologies. Both are CSMA/CD LANs. Stations on a
CSMA/CD LAN can access the network at any time. Before sending data, CSMA/CD stations
“listen” to the network to see if it is already in use. If it is, the station wishing to transmit waits. If
the network is not in use, the station transmits. A collision occurs when two stations listen for
network traffic, “hear” none, and transmit simultaneously. In this case, both transmissions are
damaged, and the stations must retransmit at some later time. Backoff algorithms determine when
the colliding stations retransmit. CSMA/CD stations can detect collisions, so they know when they
must retransmit.
Both Ethernet and IEEE 802.3 LANs are broadcast networks. In other words, all stations see all
frames, regardless of whether they represent an intended destination. Each station must examine
received frames to determine if the station is a destination. If so, the frame is passed to a higher
protocol layer for appropriate processing.
Differences between Ethernet and IEEE 802.3 LANs are subtle. Ethernet provides services
corresponding to Layers 1 and 2 of the OSI reference model, while IEEE 802.3 specifies the
physical layer (Layer 1) and the channel-access portion of the link layer (Layer 2), but does not
define a logical link control protocol. Both Ethernet and IEEE 802.3 are implemented in hardware.
Typically, the physical manifestation of these protocols is either an interface card in a host computer
or circuitry on a primary circuit board within a host computer.
9. With a neat sketch explain about IP service model, packets fragmentation and reassembly
(NOV / DEC 2016)

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Data transmitted over an internet using IP is carried in messages called IP datagrams. Like all
network protocol messages, IP uses a specific format for its datagrams. We are of course looking
here at IP version 4 and so we will examine the IPv4 datagram format, which was defined in RFC
791 along with the rest of IPv4.
The IPv4 datagram is conceptually divided into two pieces: the header and the payload. The
header contains addressing and control fields, while the payload carries the actual data to be sent
over the internetwork. Unlike some message formats, IP datagrams do not have a footer following
the payload.
Even though IP is a relatively simple, connectionless, “unreliable” protocol, the IPv4 header
carries a fair bit of information, which makes it rather large. At a minimum, it is 20 bytes long, and
with options can be significantly longer.

10. Give the comparison between different wireless technologies? Enumerate 802.11 protocol stack
in detail. (MAY / JUNE 2016)
Bluetooth, WiFi and WiMAX:
Bluetooth, WiFi and WiMAX are wireless technologies which allow devices to inter-connect and
communicate with each other. Radio waves are electomagnetic waves and have different
frequencies. These technologies are radio frequencies. Similar to the analogue radio, or FM radio.
Bluetooth works on 2.45GHz frequency. WiFi works in two frequency bands 2.4GHz and 5GHz.

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WiMAX works in two frequency bands, 2 - 11GHz and 10 - 66GHz. See chart below for a
comparison of these technologies.
The protocol stack is an implementation of a computer networking protocol suite. The terms are
often used interchangeably. Strictly speaking, the suite is the definition of the protocols, and the
stack is the software implementation of them.
Individual protocols within a suite are often designed with a single purpose in mind.
This modularization makes design and evaluation easier. Because each protocol module usually
communicates with two others, they are commonly imagined as layers in a stack of protocols. The
lowest protocol always deals with "low-level", physical interaction of the hardware. Every higher
layer adds more features. User applications usually deal only with the topmost layers.

11. Write short note on DHCP. (MAY / JUNE 2016)


The Dynamic Host Configuration Protocol (DHCP) is a standardized network protocol used
on Internet Protocol (IP) networks. The DHCP is controlled by a DHCP server that dynamically
distributes network configuration parameters, such as IP addresses, for interfaces and services.
A router or a residential gateway can be enabled to act as a DHCP server. A DHCP server enables
computers to request IP addresses and networking parameters automatically, reducing the need for
a network administrator or a user to configure these settings manually. In the absence of a DHCP
server, each computer or other device (e.g., a printer) on the network needs to be statically (i.e.,
manually) assigned to an IP address.
12. Write a short note on ICMP. (MAY / JUNE 2016)
The Internet Control Message Protocol is part of the Internet Protocol Suite, as defined in RFC
792. ICMP messages are typically used for diagnostic or control purposes or generated in response
to errors in IP operations. ICMP errors are directed to the source IP address of the originating packet.
For example, every device (such as an intermediate router) forwarding an IP datagram first
decrements the time to live (TTL) field in the IP header by one. If the resulting TTL is 0, the packet
is discarded and an ICMP Time To Live exceeded in transit message is sent to the datagram's source
address.
Although ICMP messages are contained within standard IP packets, ICMP messages are usually
processed as a special case, distinguished from normal IP processing, rather than processed as a
normal sub-protocol of IP. In many cases, it is necessary to inspect the contents of the ICMP
message and deliver the appropriate error message to the application responsible for transmission of
the IP packet that prompted the sending of the ICMP message.

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Many commonly used network utilities are based on ICMP messages. The traceroute command
can be implemented by transmitting IP datagrams with specially set IP TTL header fields, and
looking for ICMP Time to live exceeded in transit (above) and "Destination unreachable" messages
generated in response. The related ping utility is implemented using the ICMP "Echo request" and
"Echo reply" messages.
13. Write short notes on Ethernet. (NOV / DEC 2015)
Ethernet is a family of computer networking technologies commonly used in local area
networks (LANs) and metropolitan area networks (MANs). It was commercially introduced in 1980
and first standardized in 1983 as IEEE 802.3,[2] and has since been refined to support higher bit
rates and longer link distances. Over time, Ethernet has largely replaced competing wired LAN
technologies such as token ring, FDDI and ARCNET.
The original 10BASE5 Ethernet uses coaxial cable as a shared medium, while the newer
Ethernet variants use twisted pair and fiber optic links in conjunction with hubs or switches. Over
the course of its history, Ethernet data transfer rates have been increased from the original
2.94 megabits per second (Mbit/s)[3] to the latest 100 gigabits per second (Gbit/s). The Ethernet
standards comprise several wiring and signaling variants of the OSI physical layer in use with
Ethernet.
Systems communicating over Ethernet divide a stream of data into shorter pieces called frames.
Each frame contains source and destination addresses, and error-checking data so that damaged
frames can be detected and discarded; most often, higher-layer protocols trigger retransmission of
lost frames. As per the OSI model, Ethernet provides services up to and including the data link layer
14. Write short note on wireless Lan. (NOV / DEC 2015)
A wireless local area network (WLAN) is a wireless computer network that links two or more
devices using a wireless distribution method (often spread-spectrum or OFDM radio) within
a limited area such as a home, school, computer laboratory, or office building. This gives users the
ability to move around within a local coverage area and yet still be connected to the network. A
WLAN can also provide a connection to the wider Internet.
Most modern WLANs are based on IEEE 802.11 standards and are marketed under the Wi-
Fi brand name.
Wireless LANs have become popular for use in the home, due to their ease of installation and
use. They are also popular in commercial complexes that offer wireless access to their customers
(often without charge). New York City, for instance, has begun a pilot program to provide city
workers in all five boroughs of the city with wireless Internet access.

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15. Explain in detail ARP. (NOV / DEC 2015)
The Address Resolution Protocol uses a simple message format containing one address
resolution request or response. The size of the ARP message depends on the upper layer and lower
layer address sizes, which are given by the type of networking protocol (usually IPv4) in use and the
type of hardware or virtual link layer that the upper layer protocol is running on. The message header
specifies these types, as well as the size of addresses of each. The message header is completed with
the operation code for request (1) and reply (2). The payload of the packet consists of four addresses,
the hardware and protocol address of the sender and receiver hosts.
The principal packet structure of ARP packets is shown in the following table which illustrates
the case of IPv4 networks running on Ethernet. In this scenario, the packet has 48-bit fields for the
sender hardware address (SHA) and target hardware address (THA), and 32-bit fields for the
corresponding sender and target protocol addresses (SPA and TPA). Thus, the ARP packet size in
this case is 28 bytes. The EtherType for ARP is 0x0806. (This appears in the Ethernet frame header
when the payload is an ARP packet. Not to be confused with PTYPE below, which appears within
this encapsulated ARP packet.)
16. Explain in detail about the access method and frame format used in Ethernet and token ring
(APRIL / MAY 2015)
Token Ring access is more deterministic, compared to Ethernet's contention-based CSMA/CD
Ethernet supports a direct cable connection between two network interface cards by the use of
a crossover cable or through auto-sensing if supported. Token ring does not inherently support this
feature and requires additional software and hardware to operate on a direct cable connection setup.
Token ring eliminates collision by the use of a single-use token and early token release to
alleviate the down time. Ethernet alleviates collision by carrier sense multiple access and by the use
of an intelligent switch; primitive Ethernet devices like hubs can precipitate collisions due to
repeating traffic blindly.
Token ring network interface cards contain all of the intelligence required for speed
autodetection, routing and can drive themselves on many Multistation Access Units (MAUs) that
operate without power (most MAUs operate in this fashion, only requiring a power supply
for LEDs). Ethernet network interface cards can theoretically operate on a passive hub to a degree,
but not as a large LAN and the issue of collisions is still present.
Token ring employs 'access priority' in which certain nodes can have priority over the token.
Unswitched Ethernet does not have provisioning for an access priority system as all nodes have
equal contest for traffic.

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17. Discuss the MAC layer functions of IEEE802.11 (APRIL / MAY 2015)
According to IEEE Std 802-2001 section 6.2.3 "MAC sublayer", the primary functions
performed by the MAC layer are:[1]
 Frame delimiting and recognition
 Addressing of destination stations (both as individual stations and as groups of stations)
 Conveyance of source-station addressing information
 Transparent data transfer of LLC PDUs, or of equivalent information in the Ethernet sublayer
 Protection against errors, generally by means of generating and checking frame check
sequences
 Control of access to the physical transmission medium
18. Briefly define key requirements of wireless LAN. (APRIL / MAY 2015)
 Throughput. The medium access-control (MAC) protocol should make as efficient use as
possible of the wireless medium to maximize capacity.
 Number of nodes. Wireless LANs may need to support hundreds of nodes across multiple
cells.
 Connection to backbone LAN. In most cases, interconnection with stations on a wired
backbone LAN is required. For infrastructure wireless LANs, this is easily accomplished
through the use of control modules that connect to both types of LANs. There may also need
to be accommodation for mobile users and ad hoc wireless networks.
 Service area. A typical coverage area for a wireless LAN has a diameter of 100 to 300 m.
 Battery power consumption. Mobile workers use battery-powered workstations that need to
have a long battery life when used with wireless adapters. This suggests that a MAC protocol
that requires mobile nodes to monitor access points constantly or engage in frequent
handshakes with a base station is inappropriate. Typical wireless LAN implementations have
features to reduce power consumption while not using the network, such as a sleep mode.
 Transmission robustness and security. Unless properly designed, a wireless LAN may be
interference-prone and easily eavesdropped. The design of a wireless LAN must permit
reliable transmission even in a noisy environment and should provide some level of security
from eavesdropping.
19. Discuss about the physical properties of Ethernet in detail
 Transmission Rate: 10 Mbps
 50 Ohm Coaxial cable: Maximum length 500 meters

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 Host Connections: Taps at least 2.5 meters apart
 Terminators: Placed at end of Ethernet cable to match line impedance absorb signals and
minimize reflections
 Repeaters: Maximum of four repeaters between nodes so maximum reach of 2500 meters
 Signal: Broadcast over entire network
 Bit Representation: Manchester Encoding
 Number of Hosts: Up to 1024 Hosts
20. Discuss in detail about the transmitter algorithm with example.
If adapter has a frame to send and line is idle it transmits frame immediately (connectionless - no
negotiation with other adapters).
Fairness of access mechanism: Adapter must wait at least 51.2 before transmitting another frame
to allow others access to network.
If two or more adapters begin transmitting at the same time Ethernet detects a frame collision.
The adapters must transmit a minimum 512 bit (64 Byte) frame before aborting the transmission.
The guaranteed minimum 51.2 jam time ensures that the collision can be detected over the maximum
network length of 1500 m (note maximum round-trip time-of-flight of signal in cable is only 15)
After collision, the adapter doubles the maximum wait time before retrying (exponential
backoff). The number of times the maximum wait time is doubled is limited to 10.
Adapter will report an error to the host if after 16 tries it still cannot transmit the frame.
21. Discuss in detail about WLAN configurations with neat diagram
Configuration:
Configure the Access Point
It can configure the AP with the use of any of these:
 GUI
 Command-line interface (CLI), after you establish a Telnet session
 The console port
In order to connect to the AP
through the console port,
connect a nine-pin, straight-
through DB-9 serial cable to the
RS-232 serial port on the AP and to
the COM port on a
computer. Set up a terminal
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emulator in order to communicate with the AP. Use these settings for the terminal emulator
connection:
 9600 baud
 8 data bits
 No parity
 1 stop bit
 No flow control

22. Discuss in detail about the Bluetooth architectures, with neat diagrams.
Bluetooth communication occurs between a master radio and a slave radio. Bluetooth radios are
symmetric in that the same device may operate as a master and also the slave. Each radio has a 48-
bit unique device address (BD_ADDR) that is fixed.
Two or more radio devices together form ad-hoc networks called piconets. All units within a
piconet share the same channel. Each piconet has one master device and one or more slaves. There
may be up to seven active slaves at a time within a piconet. Thus, each active device within a piconet
is identifiable by a 3-bit active device address. Inactive slaves in unconnected modes may continue
to reside within the piconet.
A master is the only one that may initiate a Bluetooth communication link. However, once a link
is established, the slave may request a master/slave switch to become the master. Slaves are not
allowed to talk to each other directly. All communication occurs within the slave and the master.
Slaves within a piconet must also synchronize their internal clocks and frequency hops with that of
the master. Each piconet uses a different frequency hopping sequence. Radio devices used Time
Division Multiplexing (TDM). A master device in a piconet transmits on even numbered slots and
the slaves may transmit on odd numbered slots.

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23. Discuss circuit switching in detail


Circuit switching is a method of implementing a telecommunications network in which
two network nodes establish a dedicated communications channel (circuit) through the network
before the nodes may communicate. The circuit guarantees the full bandwidth of the channel and
remains connected for the duration of the communication session. The circuit functions as if the
nodes were physically connected as with an electrical circuit.
The defining example of a circuit-switched network is the early analog telephone network. When
a call is made from one telephone to another, switches within the telephone exchanges create a
continuous wire circuit between the two telephones, for as long as the call lasts.
Circuit switching contrasts with packet switching which divides the data to be transmitted
into packets transmitted through the network independently. In packet switching, instead of being
dedicated to one communication session at a time, network links are shared by packets from multiple
competing communication sessions, resulting in the loss of the quality of service guarantees that are
provided by circuit switching.
24. Discuss packet switching in detail.
Packet switching features delivery of variable bit rate data streams, realized as sequences of
packets, over a computer network which allocates transmission resources as needed using statistical
multiplexing or dynamic bandwidth allocation techniques. As they traverse network nodes, such as
switches and routers, packets are received, buffered, queued, and transmitted (stored and forwarded),
resulting in variable latency and throughput depending on the link capacity and the traffic load on
the network.
Packet switching contrasts with another principal networking paradigm, circuit switching, a
method which pre-allocates dedicated network bandwidth specifically for each communication
session, each having a constant bit rate and latency between nodes. In cases of billable services, such
as cellular communication services, circuit switching is characterized by a fee per unit of connection
time, even when no data is transferred, while packet switching may be characterized by a fee per unit
of information transmitted, such as characters, packets, or messages.
25. Discuss in detail about the virtual LAN with neat diagrams
A Virtual network is a computer network that consists of virtual network links, i.e. between the
computing devices there isn't a physical connection.Well known forms of network virtualization are

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virtual networks based on the virtual devices (for example the network based on the virtual devices
inside a hypervisor), protocol-based virtual networks (VLAN, VPN, VPLS, Virtual Wireless
network, etc.) and their combinations.
VLAN (Virtual LAN) is a logical LAN (local area network) based on the physical LAN that is
divided on a multiple logical LANs using a VLAN ID.VLAN can be on a VPN (virtual private
network). VPN consists of several remote end-points (such as routers, VPN gateways of software
clients) that are connected by the tunnel over another network. Two connected end points form the
PTP VPN (Point to Point Virtual Private Network), more than two end points form a Multipoint
VPN.
VPLS (Virtual Private LAN Service) is a specific type of Multipoint VPN. They can be
Transparent LAN Services (TLS) that provides geographic separation and Ethernet Virtual
Connection Services (EVCS) that provides geographic separation and VLAN subnetting.Virtual
Wireless network is a set of Wireless Access Points that behaves as one. Computer and Networks
Area for ConceptDraw PRO provides professional looking examples, the libraries with ready-to-use
predesigned vector stencils to help you create the Virtual Networks quick, easy and effective.

26. With a neat sketch, explain about IPv4 packet format


Internet Protocol being a layer-3 protocol (OSI) takes data Segments from layer-4 (Transport)
and divides it into packets. IP packet encapsulates data unit received from above layer and add to its
own header information.

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The encapsulated data is referred to as IP Payload. IP header contains all the necessary
information to deliver the packet at the other end.

27. What
is CIDR? Explain it with example.
CIDR notation is a compact representation of an IP address and its associated routing prefix. The
notation is constructed from an IP address, a slash ('/') character, and a decimal number. The number
is the count of leading 1 bits in the routing mask, traditionally called the network mask. The IP
address is expressed according to the standards of IPv4 or IPv6.
The address may denote a single, distinct interface address or the beginning address of an entire
network. The maximum size of the network is given by the number of addresses that are possible
with the remaining, least-significant bits below the prefix. The aggregation of these bits is often
called the host identifier.
Before the implementation of CIDR, IPv4 networks were represented by the starting address and
the subnet mask, both written in dot-decimal notation. Thus, 192.168.100.0/24 was often written as
192.168.100.0/255.255.255.0.
The number of addresses of a subnet may be calculated as 2address size − prefix size, in which
the address size is 128 for IPv6 and 32 for IPv4. For example, in IPv4, the prefix size /29 gives: 232
− 29 = 23 = 8 addresses.

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UNIT – III: TRANSPORT AND APPLICATION LAYERS


1. Discuss in detail about open source shortest path routing with neat diagrams. (NOV/DEC
2016)
Open Shortest Path First (OSPF) is a standard routing protocol that’s been used the world over
for many years. Supported by practically every routing vendor, as well as the open source
community, OSPF is one of the few protocols in the IT industry you can count on being available
just about anywhere you might need it.
Enterprise networks that outgrow a single site will often use OSPF to interconnect their
campuses and wide area networks (WANs).
If you’re considering a dynamic routing protocol because your network has outgrown static
routes, OSPF might seem a little daunting. It’s not quite as easy to set up as EIGRP so the temptation
might be to simply use EIGRP and avoid the intimidating terminology that comes along with a
complete understanding of OSPF.
My recommendation is not to let OSPF scare you. It’s true that OSPF in large implementations
can be complex. However, an OSPF configuration supporting smaller networks can be
comparatively simple.

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2. Discuss in detail about any two multicast routing with neat sketches. (NOV/DEC 2016)
(NOV/DEC 2015)
Multicast IP Routing protocols are used to distribute data (for example, audio/video streaming
broadcasts) to multiple recipients. Using multicast, a source can send a single copy of data to a
single multicast address, which is then distributed to an entire group of recipients.
A multicast group identifies a set of recipients that are interested in a particular data stream, and
is represented by an IP address from a well-defined range. Data sent to this IP address is forwarded
to all members of the multicast group.
Routers between the source and recipients duplicate data packets and forward multiple copies
wherever the path to recipients diverges. Group membership information is used to calculate the best
routers at which to duplicate the packets in the data stream to optimize the use of the network.
A source host sends data to a multicast group by simply setting the destination IP address of the
datagram to be the multicast group address. Any host can become a source and send data to a
multicast group. Sources do not need to register in any way before they can begin sending data to a
group, and do not need to be members of the group themselves.

3. Describe Distance vector routing. (MAY/JUNE 2016) (NOV/DEC 2015) (APRIL/MAY 2015)
In this routing scheme, each router periodically shares its knowledge about the entire network
with its neighbours.Each router has a table with information about network. These tables are updated
by exchanging information with the immediate neighbours.
It is also known as Belman-Ford or Ford-Fulkerson Algorithm. It is used in the original
ARPANET, and in the Internet as RIP. Neighboring nodes in the subnet exchange their tables
periodically to update each other on the state of the subnet (which makes this a dynamic algorithm).
If a neighbor claims to have a path to a node which is shorter than your path, you start using that
neighbor as the route to that node.
Distance vector protocols (a vector contains both distance and direction), such as RIP, determine
the path to remote networks using hop count as the metric. A hop count is defined as the number of
times a packet needs to pass through a router to reach a remote destination.
4. Explain about IPV6? Compare IPV4 and IPV6. (MAY/JUNE 2016)
IPv4 (Internet Protocol Version 4) is the fourth revision of the Internet Protocol (IP) used to to
identify devices on a network through an addressing system. The Internet Protocol is designed for
use in interconnected systems of packet-switched computer communication networks.

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IPv4 is the most widely deployed Internet protocol used to connect devices to the Internet. IPv4
uses a 32-bit address scheme allowing for a total of 2^32 addresses (just over 4 billion addresses).
With the growth of the Internet it is expected that the number of unused IPv4 addresses will
eventually run out because every device -- including computers, smartphones and game consoles --
that connects to the Internet requires an address.

A new Internet addressing system Internet Protocol version 6 (IPv6) is being deployed to fulfill
the need for more Internet addresses.

IPv6 (Internet Protocol Version 6) is also called IPng (Internet Protocol next generation) and it is
the newest version of the Internet Protocol (IP) reviewed in the IETF standards committees to
replace the current version of IPv4 (Internet Protocol Version 4).

IPv6 is the successor to Internet Protocol Version 4 (IPv4). It was designed as an evolutionary
upgrade to the Internet Protocol and will, in fact, coexist with the older IPv4 for some time. IPv6 is
designed to allow the Internet to grow steadily, both in terms of the number of hosts connected and
the total amount of data traffic transmitted.

IPv6 is often referred to as the "next generation" Internet standard and has been under
development now since the mid-1990s. IPv6 was born out of concern that the demand for IP
addresses would exceed the available supply.

5. Explain the shortest path algorithm with suitable illustrations. (APRIL/MAY 2015)
 It maintains a list of unvisited vertices.
 It chooses a vertex (the source) and assigns a maximum possible cost (i.e. infinity) to every
other vertex.
 The cost of the source remains zero as it actually takes nothing to reach from the source
vertex to itself.
 In every subsequent step of the algorithm it tries to improve(minimize) the cost for each
vertex. Here the cost can be distance, money or time taken to reach that vertex from the
source vertex. The minimization of cost is a multi-step process.
 For each unvisited neighbor (vertex 2, vertex 3, vertex 4) of the current vertex (vertex 1)
calculate the new cost from the vertex (vertex 1).
 For e.g. the new cost of vertex 2 is calculated as the minimum of the two ( (existing cost of
vertex 2) or (sum of cost of vertex 1 + the cost of edge from vertex 1 to vertex 2) )
 When all the neighbors of the current node are considered, it marks the current node as
visited and is removed from the unvisited list.
 Select a vertex from the list of unvisited nodes (which has the smallest cost) and repeat step
4.
 At the end there will be no possibilities to improve it further and then the algorithm ends
6. Explain about Switching and Forwarding.
Computer networks became more pervasive, more and more data and also less voice was
transmitted over telephone lines. Circuit Switching The telephone network used circuit switching.
Circuit switching operates by first reserving a complete route from the sender to the receiver. This

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bandwidth does not change during a call, and remains reserved even if the connection is not
transmitting any data but is still active.
Packet switching is similar to message switching using short messages. Any message exceeding
a network-defined maximum length is broken up into shorter units, known as packets, for
transmission; the packets, each with an associated header, are then transmitted individually through
the network. If you transmit from sender to receiver, all the network will do its best to get the packet
to the other end as fast as possible, but there are no guarantees on how fast that pac ket will arrive. A
packet contains three major fields: The header, the message, and redundancy check bits
Message Switching Packet switching has its origin in message switching. A message switch
typically operates in the store-and forward fashion

7. Explain about Routing areas.


Routing, in essence, is the act of finding a path from one place to another on which a packet can
travel. To find this path, we need algorithms. They will generally be distributed among many
routers, allowing them to jointly share information. Routing is said to contain three elements:
 Routing protocols, the things that allow information to be gathered and distributed
 Routing algorithms, to determine paths
 Routing databases to store information that the algorithm has discovered. The routing
database sometimes corresponds directly to routing table entries, sometimes not.

8. Explain about RIP.


RIP (Routing Information Protocol) is a widely-used protocol for managing router information
within a self-contained network such as a corporate local area network (LAN) or an interconnected
group of such LANs.
Routing Information Protocol (RIP) is a protocol that routers can use to exchange network
topology information. It is characterized as an interior gateway protocol, and is typically used in
small to medium-sized networks. A router running RIP sends the contents of its routing table to each
of its adjacent routers every 30 seconds. When a route is removed from the routing table, it is
flagged as unusable by the receiving routers after 180 seconds, and removed from their tables after
an additional 120 seconds.
There are two versions of RIP (the managed switch supports both):
RIPv1 defined in RFC 1058.
RIPv2 defined in RFC 1723.
You can configure a given port to do the following:
Receive packets in either or both formats.Send packets formatted for RIPv1 or RIPv2, or send
RIPv2 packets to the RIPv1 broadcast address.Prevent any RIP packets from being received.
Prevent any RIP packets from being sent.

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9. Explain the two approaches of packet switching techniques.


Datagram approach
Virtual circuit approach
Switched virtual circuit(SVC)
Permanent virtual circuit(PVC)
Circuit – switched connection versus virtual – circuit connection
Path versus route
Dedicated versus shared

10. Explain IP addressing method.


Internetwork protocol (IP)
Datagram
Addressing
Classes
Dotted decimal notation
A sample internet

11. Define routing & explain distance vector routing and link state routing.
Distance vector routing
Sharing information
Routing table
Creating the table
Updating the table
Updating algorithm
Link state routing
Information sharing
Packet cost
Link state packet
Getting information about neighbors
Initialization
Link state database

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12. Define bridge and explain the type of bridges.
Bridges
Types of bridges
Simple bridge
Multiport bridge
Transparent bridge

13. Explain subnetting


Subnetting
Three levels of hierarchy
Masking
Masks without subnetting
Masks with subnetting
Finding the subnetwork address
Boundary level masking
Non-boundary level masking

14. Write short notes about repeaters, routers and gateways.


Repeaters
Routers
Routing concepts
Least-cost routing
Non adaptive routing
Adaptive routing
Packet lifetime
Gateways

15. Explain Link state routing protocol.


Two phases
Reliable flooding -Tell all routers what you know about your local topology
Path calculation (Dijkstra’s algorithm) -Each router computes best path over complete network
Motivation
Global information allows optimal route computation
Straightforward to implement and verify
Challenges
Packet loss
Out-of-order arrival
Solutions
Acknowledgments and retransmissions
Sequence numbers
Time-to-live for each packet
16. Explain LSP with example.

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Each node constructs a one-dimensional array containing the "distances"(costs) to all other nodes
and distributes that vector to its immediate neighbors.
The starting assumption for distance-vector routing is that each node knows the cost of the link to
each of its directly connected neighbors.
A link that is down is assigned an infinite cost.
Example.

17. Explain about OSPF.


OSPF allows the grouping of networks into a set, called an area. The topology of an area is
hidden from the rest of the Autonomous System. This technique minimizes the routing traffic
required for the protocol. When multiple areas are used, each area has its own copy of the
topological database.
Adjacency and Designated Routers
Link State Advertisements
OSPF Packet Types
Authentication of routing messages:
Additional hierarchy:
Load balancing

18. Discuss metrics in detail.


The original ARPANET routing metric measured the number of packets that were queued
waiting to be transmitted on each link, meaning that a link with 10 packets queued waiting to be
transmitted was assigned a larger cost weight than a link with 5packets queued for
transmission.First, each incoming packet was timestamped with its time of arrival at the router
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(ArrivalTime); its departure time from the router (DepartTime) was also recorded. Second, when the
link-level ACK was received from the other side, the node computed the delay for that packet as
Delay = (DepartTime− ArrivalTime) +TransmissionTime +Latency where TransmissionTime and
Latency were statically defined for the link an captured the link’s bandwidth and latency,
respectively. Notice that in this case, DepartTime − ArrivalTime represents the amount of time the
packet was delayed (queued) in the node due to load. If the ACK did not arrive, but instead the
packet timed out, then DepartTime was reset to the time the packet was retransmitted. In this case,
DepartTime − ArrivalTime captures the reliability of the link—the more frequent the retransmission
of packets, the less reliable the link, and the more we want to avoid it.
Finally, the weight assigned to each link was derived from the average delay experienced
 A highly loaded link never shows a cost of more than three times its cost when
idle;
 The most expensive link is only seven times the cost of the least expensive;
 A high-speed satellite link is more attractive than a low-speed terrestrial link;
 Cost is a function of link utilization only at moderate to high loads.
All these factors mean that a link is much less likely to be universally abandoned, since a
threefold increase in cost is likely to make the link unattractive for some paths

19. Explain interdomain routing in detail.


The interdomain routing problem is then one of having different ASs share reachability
information— descriptions of the set of IP addresses that can be reached via a given AS—with each
other.
A simple example to send traffic via AS X than via AS Y, but I’ll use AS Y if it is the only path,
and I never want to carry traffic from AS X to AS Y or vice versa. Such a policy would be typical
when I have paidmoney to both AS X and AS Y to connect my AS to the rest of the Internet, and AS
X is my preferred provider of connectivity with AS Y being the fallback. Because I view both AS X
and AS Y as providers (and presumably I paid them to play this role), I don’t expect to help them out
by carrying traffic between them across my network (this is called transit traffic).
Internet consists of an interconnection of multiple backbone networks they are usually called
service provider networks, and sites are connected to each other in arbitrary ways. Some large
corporations connect directly to one or more of the backbones, while others connect to smaller,
nonbackbone service providers.
Many service providers exist mainly to provide service to “consumers” (i.e., individuals with
PCs in their homes), and these providers must also connect to the backbone providers.
ASs into three types
Stub AS
Multihomed AS
Transit AS:

20. Explain about IPv6.


Motivation for a new version of IP is the same as the motivationfor the techniques described so
far in this section: to deal with scaling problems caused by the Internet’s massive growth.
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Support for real-time services;
 Security support
 Autoconfiguration (i.e., the ability of hosts to automatically configure themselves with such
information as their own IP address and domain name);
 Enhanced routing functionality, including support for mobile hosts.

Addresses and Routing


Address Space Allocation
Address Notation
Global Unicast Addresses
Packet Format
Autoconfiguration
21.Explain about IP has address space reserved for multicast address.
IP has a subrange of its address space reserved for multicast addresses. In IPv4, these addresses
are assigned in the class D address space, and IPv6 also has a portion of its address space reserved
for multicast group addresses.
There are 28 bits of possible multicast addresses in IPv4 when we ignore the prefix shared by all
multicast addresses.
This presents a problem when attempting to take advantage of hardware multicasting on a LAN.
Let’s take the case of Ethernet. Ethernet multicast addresses have only 23 bits when we ignore their
shared prefix. In other words, to take advantage of Ethernet multicasting, IP has to map 28-bit IP
multicast addresses into 23-bit Ethernet multicast addresses. This is implemented by taking the low-
order 23 bits of any IP multicast address to use as its Ethernet multicast address, and ignoring the
high-order 5 bits. Thus, 32 IP addresses map into each one of the Ethernet addresses.
When a host on an Ethernet joins an IP multicast group, it configures its Ethernet interface to
receive any packets with the corresponding Ethernet multicast address.
The receiving host to receive not only the multicast traffic it desired, but also traffic sent to any
of the other 31 IP multicast groups that map to the same Ethernet address, if they are routed to that
Ethernet. Therefore, IP at the receiving host must examine the IP header of any multicast packet to
determine whether the packet
22.Explain BGP in detail.
Interdomain routing protocol for the Internet
 Prefix-based path-vector protocol
 Policy-based routing based on AS Paths
 Evolved during the past 18 yearsA node learns multiple paths to destination Stores all of
the routes in a routing table
 Applies policy to select a single active route and may advertise the route to its neighbors
Incremental updates

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Announcement-Upon selecting a new active route, add node id to path and (optionally)
advertise to each neighbor
 Withdrawal-If the active route is no longer available send a withdrawal message to the
neighbors
BGP Attributes
 Local pref: Statically configured ranking of routes within AS
 AS path: ASs the announcement traversed
 Origin: Route came from IGP or EGP
 Multi Exit Discriminator: preference for where to exit network
 Community: opaque data used for inter-ISP policy
 Next-hop: where the route was heard from
23.Explain about PIM and illustrate its four modes.
Protocol-independent multicast (PIM) is a set of four specifications that define modes of Internet
multicasting to allow one-to-many and many-to-many transmission of information.
The four modes are:
 sparse mode (SM)
 dense mode (DM)
 source-specific multicast (SSM)
 bidirectional.
The term "protocol independent" means that PIM can function by making use of routing information
supplied by a variety of communications protocols.

24.What is DVMRP? Explain in detail?

The distance vector multicast routing protocol is multicast routing protocol that takes the routing
decision based upon the source address of the packet.
 This algorithm constructs the routing tree for a network.
 Whenever a router receives a packet, it forwards it to some of its ports based on the source
address of packet.
 The rest of the routing tree is made by downstream routers.
 In this way, routing tree is created from destination to source.
The protocol must achieve the following tasks:
 It must prevent the formation of loops in the network.
 It must prevent the formation of duplicate packets.
 It must ensure that the path traveled by a packet is the shortest from its source to the
router.
 It should provide dynamic membership.
25.Discuss briefly about NHRP.

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The NHRP (Next Hop Resolution Protocol) comes from the Internet world and is described in
RFC 1932. It allows to search the ATM corresponding to an IP address in an NBMA network of
several DSL. Specifically, NHRP allows the resolution of an IP address of a workstation on a remote
DSL NBMA address of a network (ATM address, frame relay, etc.).
Each LIS has a route server, called NHS (Next Hop Server), often located in a router. When a
client requests a connection, it is for the NHS to which it belongs LIS to obtain
routing information on the packet. If the local NHS can not solve the problem of location, it sends a
request to related NHS, and so on until reaching the LIS to which the recipient belongs.
This solution allows you to find a more direct route that passes through the different NHS, as
shown in Figure. Phase 1 is the address conversion request to the first DSL Router NHR, which
turns with the Phase 2 NHR DSL Router which depends the remote user. Phases 3 and 4
correspond to the return of the address conversion. With ATM the customer address opens a
virtual circuit to the remote: it is the phase 5. One can thus obtain a direct connection of two
ATM stations belonging to remote DSL without having to go back to IP the router

UNIT – IV: NETWORK SECURITY

1. Explain the various fields of the TCP header and the working of the TCP protocol. (NOV/DEC
2016)
Each TCP header has ten required fields totaling 20 bytes (160 bits) in size. They can also
optionally include an additional data section up to 40 bytes in size.
 source TCP port number (2 bytes)
 destination TCP port number (2 bytes)
 sequence number (4 bytes)
 acknowledgement number (4 bytes)
 TCP data offset (4 bits)
 reserved data (3 bits)
 control flags (up to 9 bits)
 window size (2 bytes)
 TCP checksum (2 bytes)
 urgent pointer (2 bytes)
 TCP optional data (0-40 bytes)
2. How is congestion controlled? Explain in detail about congestion control techniques in
transport layer. (NOV/DEC 2016) (MAY/JUNE 2016)
One of the main principles for congestion control is avoidance. TCP tries to detect signs of
congestion before it happens and to reduce or increase the load into the network accordingly. The

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alternative of waiting for congestion and then reacting is much worse because once a network
saturates, it does so at an exponential growth rate and reduces overall throughput enormously. It
takes a long time for the queues to drain, and then all senders again repeat this cycle. By taking a
proactive congestion avoidance approach, the pipe is kept as full as possible without the danger of
network saturation. The key is for the sender to understand the state of the network and client and to
control the amount of traffic injected into the system.

The amount of data that is to be sent to the remote peer on a specific connection is controlled by
two concurrent mechanisms:

 The congestion in the network - The degree of network congestion is inferred by the
calculation of changes in Round Trip Time (RTT): that is the amount of delay attributed to
the network. This is measured by computing how long it takes a packet to go from sender to
receiver and back to the client. This figure is actually calculated using a running smoothing
algorithm due to the large variances in time. The RTT value is an important value to
determine the congestion window, which is used to control the amount of data sent out to the
remote client. This provides information to the sender on how much traffic should be sent to
this particular connection based on network congestion.

 Client load - The rate at which the client can receive and process incoming traffic. The client
sends a receive window that provides information to the sender on how much traffic should
be sent to this connection based on client load.

3. Define UDP. Discuss the operations of UDP. Explain UDP checksum with one example.
(MAY/JUNE 2016)
The concept of UDP is simple. The UDP will not do the following. As a network administrator
you must be aware of these:
 The UDP will never show any acknowledgement to prove that the data was received.
 The UDP will never establish a connection before actually sending the data.
 The UDP does not give a guarantee that the messages will arrive.
 The UDP cannot detect the messages that were lost.
 The UDP will also not assure that the data will be received in the same format as they were
received.
 The UDP does not have any mechanism that manages flow of data between the devices and does
not handle congestion either.
The UDP messages will have the following parts:
 Source port - this is a 16 bit port number of the process that initially originated the UDP message
on the source device.
 Destination port - this is the second part of the message. It is a 16 bit-port number of the process
it is ultimate recipient of the message. This will generally be the port number of the client.
 Length -the length means the entire UDP datagram that includes data fields and the headers.
 Checksum - as we have already explained it before it is optional and generally 16 bit in size.
 Data - it is the variable part of the UDP. It encapsulates the higher layer messages that need to be
sent.

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4. With neat architecture, explain TCP in detail. (NOV/DEC 2015)
The TCP/IP protocol suite maps to a four-layer conceptual model known as the DARPA model,
which was named after the U.S. government agency that initially developed TCP/IP. The four layers
of the DARPA model are: Application, Transport, Internet, and Network Interface. Each layer in the
DARPA model corresponds to one or more layers of the seven-layer OSI model.
The TCP/IP protocol suite has two sets of protocols at the Internet layer: IPv4, also known as IP,
is the Internet layer in common use today on private intranets and the Internet. IPv6 is the new
Internet layer that will eventually replace the existing IPv4 Internet layer.

5. Explain TCP congestion control methods. (NOV/DEC 2015)

TCP uses a congestion window in the sender side to do congestion avoidance. The congestion
window indicates the maximum amount of data that can be sent out on a connection without being
acknowledged. TCP detects congestion when it fails to receive an acknowledgement for a packet
within the estimated timeout. In such a situation, it decreases the congestion window to one
maximum segment size (MSS), and under other cases it increases the congestion window by one
MSS. There also exists a congestion window threshold, which is set to half the congestion window
size at the time when a re-transmit was required.
The inherent assumption in this mechanism is that lack of an acknowledgement is due to network
congestion. If a packet, however, is lost by the network for reasons other than network congestion,
then waiting for the timer to run out is wasteful. This is a situation that may happen quite frequently
in wireless networks, and so to improve TCP performance, it is needed to pre-empt re-transmissions
before waiting for the timer to run out.
To guard against this scenario, Reno TCP [7] uses Fast Re-transmit and Fast Recovery
algorithms. Both these algorithms depend on counting duplicate acknowledgements sent by the data
receiver in response to each additional segment received following some missing data. Fast Re-
transmit detects loss of a segment when three duplicate acknowledgements are received, and re-

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transnits it. Fast Recovery algorithm attempts to estimate how much data is outstanding in the
network by counting duplicate acknowledgements.
6. Explain the various fields of the TCP header and the working of the TCP protocol.
(APRIL/MAY 2015)

Source port: 16 Bit number which identifies the Source Port number (Sending Computer's TCP
Port).

Destination port: 16 Bit number which identifies the Destination Port number (Receiving Port).

Sequence number: 32 Bit number used for byte level numbering of TCP segments. If you are
using TCP, each byte of data is assigned a sequence number. If SYN flag is set (during the
initial three way handshake connection initiation), then this is the initial sequence number. The
sequence number of the actual first data byte will then be this sequence number plus 1. For
example, let the first byte of data by a device in a particular TCP header will have its sequence
number in this field 50000. If this packet has 500 bytes of data in it, then the next packet sent by
this device will have the sequence number of 50000 + 500 + 1 = 50501.

Acknowledgment Number: 32 Bit number field which indicates the next sequence number that
the sending device is expecting from the other device.

Header Length: 4 Bit field which shows the number of 32 Bit words in the header. Also known
as the Data Offset field. The minimum size header is 5 words (binary pattern is 0101).

7. Explain the three way handshake protocol to establish the transport level connection.
(APRIL/MAY 2015)

To establish a connection, each device must send a SYN and receive an ACK for it from the
other device. Thus, conceptually, we need to have four control messages pass between the devices.
However, it's inefficient to send a SYN and an ACK in separate messages when one could
communicate both simultaneously. Thus, in the normal sequence of events in connection
establishment, one of the SYNs and one of the ACKs is sent together by setting both of the relevant
bits (a message sometimes called a SYN+ACK). This makes a total of three messages, and for this
reason the connection procedure is called a three-way handshake.

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8. Explain the duties of transport layer.

End to end delivery


Addressing Reliable delivery
Error control
Sequence control
Loss control
Duplication control
Flow control
Multiplexing

9. Explain UDP in detail

User Datagram Protocol(UDP)


Source port address
Destination port address
Total length
Checksum

10. Explain TCP in detail

Transmission Control Protocol(TCP)


Source port address
Destination port address
Sequence number
Acknowledgement number
Header length
Reserved
Control _ Window size
Check sum
Urgent pointer
Options and padding

11. Explain about congestion control.

Congestion avoidance
BECN

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FECN
Four situations
Discarding

12. Explain leaky bucket and token bucket algorithm

Leaky bucket algorithm


Leaky bucket
Switch controlling the output rate
Flowchart

13. Discuss about overview of TCP.

 Guarantees message delivery.


 Delivers messages in the same order they are sent.
 Delivers at most one copy of each message.
 Supports arbitrarily large messages.
 Supports synchronization between the sender and the receiver.
 Allows the receiver to apply flow control to the sender.
 Supports multiple application processes on each host.
 From below, the underlying network upon which the transport protocol operates has certain
limitations in the level of service it can provide. Some of the more typical limitations of the
network are that it may
 Drop messages.
 Reorder messages.
 Deliver duplicate copies of a given message.
 Limit messages to some finite size.
 Deliver messages after an arbitrarily long delay.

14. Explain about Segment Format of TCP.

TCP is a byte-oriented protocol, which means that the sender writes bytes into a TCP connection
and the receiver reads bytes out of the TCP connection

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15. Explain about 3-way handshakes in connection management in TCP

The algorithm used by TCP to establish and terminate a connection is called a three-way
handshake. The three-way handshake involves the exchange of three messages between the
client and the server, as illustrated by the timeline.
The idea is that two parties want to agree on a set of parameters, which, in the case of
opening a TCP connection, are the starting sequence numbers the two sides plan to use for their
respective byte streams. First, the client (the active participant) sends a segment to the server (the
passive participant) stating the initial sequence number it plans to use (Flags = SYN,
SequenceNum = x). then responds with a single segment that both acknowledges the client’s
sequence number (Flags = ACK, Ack = x + 1) and states its own beginning sequence number
(Flags = SYN, SequenceNum = y). That is, both the SYN and ACK bits are set in the Flags field
of this second message. Finally, the client responds with a third segment that acknowledges the
server’s sequence number (Flags = ACK, Ack = y + 1). The reason that each side acknowledges
a sequence number that is one larger than the one sent is that the Acknowledgment field actually

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identifies the “next sequence number expected,” there by implicitly acknowledging all earlier
sequence numbers.

16. Explain state transition diagram in TCP.

This shows only the states involved in opening a connection (everything above ESTABLISHED)
and in closing a connection (everything below ESTABLISHED). Everything that goes on while a
connection is open—that is, the operation of the sliding window algorithm—is hidden in the
ESTABLISHED state.
When opening a connection, the server first invokes a passive open operation on TCP, which causes
TCP to move to the LISTEN state. At some later time, the client does an active open, which causes
its end of the connection to send a SYN segment to the server and to move to the SYN_SENT state.
When the SYN segment arrives at the server, it moves to the SYN_RCVD state and responds with a
SYN+ACK segment.
on any one side there are three combinations of transitions that get a connection from the
ESTABLISHED state to the CLOSED state:
This side closes first:
ESTABLISHED→FIN_WAIT_1→FIN_WAIT_2→TIME_WAIT→CLOSED.
The other side closes first: ESTABLISHED→CLOSE_WAIT→LAST_ACK →CLOSED.
Both sides close at the same time: ESTABLISHED → FIN_WAIT_1
→CLOSING→TIME_WAIT→CLOSED.

17. How to implement reliable and ordered delivery in TCP.

How the sending and receiving sides of TCP interact with each other to implement reliable and
ordered delivery

TCP on the sending side maintains a send buffer and the receiving side, TCP maintains a receive
buffer. This buffer holds data that arrives out of order, as well as data that is in the correct order

At the sending side, three pointers are maintained into the send buffer, each with an obvious
meaning: LastByteAcked, LastByteSent, and Last-ByteWritten. Set of pointers (sequence numbers)
are maintained on the receiving side: LastByteRead, NextByteExpected, and LastByteRcvd

18. Discuss flow control in TCP with an example.

TCP uses an end-to-end flow control protocol to avoid having the sender send data too fast for
the TCP receiver to receive and process it reliably. Having a mechanism for flow control is essential
in an environment where machines of diverse network speeds communicate. For example, if a PC
sends data to a smartphone that is slowly processing received data, the smartphone must regulate the
data flow so as not to be overwhelmed.

Sender buffer size : MaxSendBuffer


Receive buffer size : MaxRcvBuffer
Receiving side
LastByteRcvd - NextBytteRead <= MaxRcvBuffer
AdvertisedWindow = MaxRcvBuffer - (LastByteRcvd - NextByteRead)

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Sending side
LastByteSent - LastByteAcked <= AdvertisedWindow
EffectiveWindow = AdvertisedWindow - (LastByteSent - LastByteAcked)
LastByteWritten - LastByteAcked <= MaxSendBuffer
Block sender if (LastByteWritten - LastByteAcked) + y > MaxSendBuffer
Always send ACK in response to an arriving data segment
Persist when AdvertisedWindow = 0

19. Discuss about Adaptive retransmission in TCP.

TCP guarantees reliable delivery and so it retransmits each segment if an ACK is not received in
a certain period of time. TCP sets this timeout as a function of the RTT it expects between the two
ends of the connection. Unfortunately, given the range of possible RTT's between any pair of hosts
in the Internet, as well as the variation in RTT between the same two hosts over time, choosing an
appropriate timeout value is not that easy. To address this problem, TCP uses an adaptive
retransmission mechanism. We describe this mechanism and how it has evolved over time.
Original Algorithm
Measure SampleRTT for each segment/ACK pair
Compute weighted average of RTT
EstimatedRTT = a*EstimatedRTT + b*SampleRTT, where a+b = 1
 a between 0.8 and 0.9
 b between 0.1 and 0.2
Set timeout based on EstimatedRTT
TimeOut = 2 * EstimatedRTT
Karn/Partridge Algorithm
Do not sample RTT when retransmitting
Double timeout after each retransmission
Jacobson/Karels Algorithm

New calculation for average RTT


Difference = SampleRTT - EstimatedRTT
EstimatedRTT = EstimatedRTT + ( d * Difference)
Deviation = Deviation + d ( |Difference| - Deviation)), where d is a fraction between 0 and 1
Consider variance when setting timeout value
Timeout = u * EstimatedRTT + q * Deviation, where u = 1 and q = 4

20. Explain about the TCP congestion control.

 Basic idea :: each source determines how much capacity is available to a given flow in the
network.

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 ACKs are used to ‘pace’ the transmission of packets such that TCP is “self-clocking”.
Additive Increase/Multiplicative Decrease
Sou Destin
rce ation

Slow start

Sourc Destinati
e on

Fast retransmit and Fast Recovery

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Sender Receiver
Packet 1
Packet 2
Packet 3 ACK 1

Packet 4 ACK 2

Packet 5 ACK 2

Packet 6
ACK 2
ACK 2

Retransmit
packet 3

ACK 6

21.Explain sliding window in TCP.

The sliding window serves several purposes:


(1) it guarantees the reliable delivery of data
(2) it ensures that the data is delivered in order,
(3) it enforces flow control between the sender and the receiver.

22.Explain detail about QoS


 Policing
 Iintegrated service
 Traffic Shaping
 Admission Control
 RSVP
 Differentiated Serviices/Qos

23.Explain about the RED algorithm.


 RED is based on DECbit, and was designed to work well with TCP.
 RED implicitly notifies sender by dropping packets.
 Drop probability is increased as the average queue length increases.
 (Geometric) moving average of the queue length is used so as to detect long term congestion,
yet allow short term bursts to arrive.

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24.Explain about the DECbit algorithm.


 Each packet has a “Congestion Notification” bit called the DECbit in its header.
 If any router on the path is congested, it sets the DECbit.
 Set if average queue length >= 1 packet, averaged since the start of the previous busy cycle.
 To notify the source, the destination copies DECbit into ACK packets.
 Source adjusts rate to avoid congestion.
 Counts fraction of DECbits set in each window.
 If <50% set, increase rate additively.
 If >=50% set, decrease rate multiplicatively.
25.Discuss Approaches to QoS.
Integrated Services
Architecture for providing QOS guarantees in IP networks for individual application sessions
resource reservation: routers maintain state info (a la VC) of allocated resources, QoS r
 Network wide control
 Admission Control
 Absolute guarantees
 Traffic Shaping
 Reservations
 RSVP
 Differentiated Services
 Router based control
 Per hop behavior
 Resolves contentions
 Hot spots
 Relative guarantees
 Traffic policing
 At entry to network

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UNIT – V: HARDWARE SECUTITY

1. Explain the function of Internet Message Access Protocol (IMAP) with a state diagram.
[Nov/Dec 2017]
IMAP (Internet Message Access Protocol) is a standard email protocol that stores email
messages on a mail server, but allows the end user to view and manipulate the messages as though they
were stored locally on the end user's computing device(s).
The Internet Message Access Protocol is an Application Layer Internet protocol that allows an e-
mail client to access e-mail on a remote mail server. The current version, IMAP version 4 revision 1
(IMAP4rev1), is defined by RFC 3501. An IMAP server typically listens on well-known port 143.
IMAP over SSL (IMAPS) is assigned well-known port number 993.
IMAP supports both on-line and off-line modes of operation. E-mail clients using IMAP
generally leave messages on the server until the user explicitly deletes them. This and other
characteristics of IMAP operation allow multiple clients to manage the same mailbox. Most e-
mail clients support IMAP in addition to Post Office Protocol (POP) to retrieve messages; however,

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fewer e-mail services support IMAP. IMAP offers access to the mail storage. Clients may store local
copies of the messages, but these are considered to be a temporary cache.

2. List and explain the various HTTP request operations. [Nov/Dec 2017]
HTTP defines a set of request methods to indicate the desired action to be performed for
a given resource. Although they can also be nouns, these request methods are sometimes referred
to as HTTP verbs. Each of them implements a different semantic, but some common features are
shared by a group of them: e.g. a request method can be safe, idempotent, or cacheable.

GET : The GET method requests a representation of the specified resource. Requests
using GET should only retrieve data.

HEAD : The HEAD method asks for a response identical to that of a GET request, but without the
response body.

POST : The POST method is used to submit an entity to the specified resource, often causing a
change in state or side effects on the server

PUT : The PUT method replaces all current representations of the target resource with the request
payload.

DELETE : The DELETE method deletes the specified resource.

CONNECT : The CONNECT method establishes a tunnel to the server identified by the target
resource.

OPTIONS : The OPTIONS method is used to describe the communication options for the target
resource.

TRACE : The TRACE method performs a message loop-back test along the path to the target
resource.

3. What is domain name system (DNS)? Explain. [Nov/Dec 2017]


DNS stands for Domain Name System. The main function of DNS is to translate domain names
into IP Addresses, which computers can understand. It also provides a list of mail servers which accept
Emails for each domain name.

4. Brief about the important of simple network management protocol (SNMP). [Nov/Dec 2017]
Simple Network Management Protocol (SNMP) is an Internet Standard protocol for collecting
and organizing information about managed devices on IP networks and for modifying that information
to change device behavior. Devices that typically support SNMP include cable modems, routers,
switches, servers, workstations, printers.

SNMP is widely used in network management for network monitoring. SNMP exposes
management data in the form of variables on the managed systems organized in a management
information base (MIB) which describe the system status and configuration. These variables can then be
remotely queried (and, in some circumstances, manipulated) by managing applications.

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Three significant versions of SNMP have been developed and deployed. SNMPv1 is the original
version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in
performance, flexibility and security.
An SNMP-managed network consists of three key components:

 Managed devices
 Agent – software which runs on managed devices
 Network management station (NMS) – software which runs on the manager

SNMPv1 specifies five core protocol data units (PDUs). Two other
PDUs, GetBulkRequest and InformRequest were added in SNMPv2 and the Report PDU was added in
SNMPv3.
All SNMP PDUs are constructed as follows:

IP UDP versio PDU- request- error- error- variable


community
header header n type id status index bindings

5. Describe how SMTP transfers message from one host to another with suitable illustration.
[April/May 2017]
SMTP (Simple Mail Transfer Protocol) is a TCP/IP protocol used in sending and receiving
e-mail. In other words, users typically use a program that uses SMTP for sending e-mail and either
POP3 or IMAP for receiving e-mail.

SMTP provides a set of codes that simplify the communication of email messages between
email servers (the network computer that handles email coming to you and going out). It's a kind of
shorthand that allows a server to break up different parts of a message into categories the other server
can understand. When you send a message out, it's turned into strings of text that are separated by the
code words (or numbers) that identify the purpose of each section.

SMTP provides those codes, and email server software is designed to understand what they
mean. As each message travels towards its destination, it sometimes passes through a number of
computers as well as their individual MTAs. As it does, it's briefly stored before it moves on to the next
computer in the path. Think of it as a letter going through different hands as it winds its way to the right
mailbox.

6. Explain IMAP with its state transition diagram. [April/May 2017]

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 IMAP (Internet Message Access Protocol) is a standard email protocol that stores email
messages on a mail server, but allows the end user to view and manipulate the messages as
though they were stored locally on the end user's computing device(s).
 Most implementations of IMAP support multiple logins; this allows the end user to
simultaneously connect to the email server with different devices. For example, the end user
could connect to the mail server with his Outlook iPhone app and his Outlook desktop client
at the same time.
 The details for how to handle multiple connections are not specified by the protocol but are
instead left to the developers of the mail client.
 Even though IMAP has an authentication mechanism, the authentication process can easily
be circumvented by anyone who knows how to steal a password by using a protocol
analyzerbecause the client’s username and password are transmitted as clear text.
 In an Exchange Server environment, administrators can work around this security flaw by
using Secure Sockets Layer (SSL) encryption for IMAP.

7. List the elements of network management

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Simple Network Management Protocol (SNMP) is an application–layer protocol
defined by the Internet Architecture Board (IAB) in RFC1157 for exchanging ... A managed
device or the network element is a part of the network that requires some form of monitoring and
management e.g. routers, switches, servers. Manage LAN, WAN, Bandwidth, VoIP Agent less
tool, deploys in minutes.
An SNMP-managed network consists of three key components: managed devices, agents,
and network-management systems (NMSs).
8. Explain the operation of SNMP protocol in detail. [April/May 2017]

A managed device is a network node that contains an SNMP agent and that resides on a
managed network. Managed devices collect and store management information and make this
information available to NMSs using SNMP. Managed devices, sometimes called network
elements, can be routers and access servers, switches and bridges, hubs, computer hosts, or
printers.

An agent is a network-management software module that resides in a managed device.


An agent has local knowledge of management information and translates that information into a
form compatible with SNMP.

9. Discuss the functions performed by of DNS. Give example. [April/May 2017]


DNS stands for Domain Name System. The main function of DNS is to translate domain
names into IP Addresses, which computers can understand. It also provides a list of mail servers which
accept Emails for each domain name.
Each domain name in DNS will nominate a set of name servers to be authoritative for its
DNS records. This is where all other name servers will be pointed when looking for information about
the domain name.
Name servers are a program or computer server that implements a name-service protocol.
This is where the zone file is stored and your DNS records are stored within.
A zone file is a small set of instructions that points domain names to IP addresses. A zone
file is made up of 'records' such as A Records and MX Records.

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This record is used to point your domain name at an IP address. If you need to set up
an A record, your provider would provide you with an IP address that will look something like
194.154.164.196.
If you need to set up a CNAME record, your provider would provide you with a URL
address that will look something like web.me.com.
This is used to specify which mail servers are responsible for a particular domain name.
One special MX Record feature is priority numbers, which provide information to the querying mail
server about which mail server should be used first. The next entry is tried only when the mail server
with the highest priority is not available.

10. Give detailed notes on DNS operation. (NOV / DEC 2016) (NOV / DEC 2015)
The Domain Name System (DNS) translates Internet domain and host names to IP addresses and
vice versa.On the Internet, DNS automatically converts between the names we type in our Web
browser address bar to the IP addresses of Web servers hosting those sites. Larger corporations also
use DNS to manage their own company intranet. Home networks use DNS when accessing the
Internet but do not use it for managing the names of home computers.
DNS Works:
DNS is a client/server network communication systems: DNS clients send requests to and
receive responses from DNS servers. Requests containing a name, that result in an IP address being
returned from the server, are called forward DNS lookups. Requests containing an IP address and
resulting in a name, called reverse DNS lookups, are also supported. DNS implements a distributed
database to store this name and last-known address information for all public hosts on the Internet.
The DNS database resides on a hierarchy of special database servers. When clients like Web
browsers issue requests involving Internet host names, a piece of software (usually built into the
network operating system) called the DNS resolver first contacts a DNS server to determine the
server's IP address. If the DNS server does not contain the needed mapping, it will in turn forward
the request to a different DNS server at the next higher level in the hierarchy.
After potentially several forwarding and delegation messages are sent within the DNS hierarchy,
the IP address for the given host eventually arrives at the resolver, that in turn completes the request
over Internet Protocol.
DNS additionally includes support for caching requests and for redundancy. Most
network operating systems support configuration of primary, secondary, and tertiary DNS servers,
each of which can service initial requests from clients.

11. Explain in detail about SNMP messages. (NOV / DEC 2016) (MAY / JUNE 2016)
Simple Network Management Protocol (SNMP) is an Internet-standard protocol for collecting
and organizing information about managed devices on IP networks and for modifying that
information to change device behavior. Devices that typically support SNMP include cable
modems, routers, switches, servers, workstations, printers, and more.[1]
SNMP is widely used in network management for network monitoring. SNMP exposes
management data in the form of variables on the managed systems organized in a management

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information base (MIB) which describe the system status and configuration. These variables can
then be remotely queried (and, in some circumstances, manipulated) by managing applications.
Three significant versions of SNMP have been developed and deployed. SNMPv1 is the original
version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in
performance, flexibility and security.
SNMP is a component of the Internet Protocol Suite as defined by the Internet Engineering Task
Force (IETF). It consists of a set of standards for network management, including an application
layer protocol, a database schema, and a set of data objects.

12. Illustrate the role of POP3 in electronic mail Applications (NOV / DEC 2016)
POP3 is designed to delete mail on the server as soon as the user has downloaded it. However,
some implementations allow users or an administrator to specify that mail be saved for some period
of time. POP can be thought of as a "store-and-forward" service.
An alternative protocol is Internet Message Access Protocol (IMAP). IMAP provides the user
more capabilities for retaining e-mail on the server and for organizing it in folders on the server.
IMAP can be thought of as a remote file server.
POP and IMAP deal with the receiving of e-mail and are not to be confused with the Simple
Mail Transfer Protocol (SMTP), a protocol for transferring e-mail across the Internet. You send e-
mail with SMTP and a mail handler receives it on your recipient's behalf. Then the mail is read
using POP or IMAP.
13. Explain HTTP with an example. (MAY / JUNE 2016)
HTTP is a Stateless Protocol
HTTP is called a stateless protocol because each command is executed independently, without
any knowledge of the commands that came before it. This is the main reason that it is difficult to
implement Web sites that react intelligently to user input. This shortcoming of HTTP is being
addressed in a number of new technologies, including ActiveX, Java, JavaScript and cookies.
HTTP Status Codes are Error Messages
Errors on the Internet can be quite frustrating — especially if you do not know the difference
between a 404 error and a 502 error. These error messages, also called HTTP status codes are
response codes given by Web servers and help identify the cause of the problem.

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14. Explain in detail about web service architecture. (MAY / JUNE 2016) (APRIL / MAY 2015)
Web Service Roles
There are three major roles within the web service architecture:
Service Provider
This is the provider of the web service. The service provider implements the service and makes
it available on the Internet.
Service Requestor
This is any consumer of the web service. The requestor utilizes an existing web service by
opening a network connection and sending an XML request.
Service Registry
This is a logically centralized directory of services. The registry provides a central place where
developers can publish new services or find existing ones. It therefore serves as a centralized
clearing house for companies and their services.
15. Write short notes on the following: (APRIL / MAY 2015)
Email: E-mail (electronic mail) is the exchange of computer-stored messages by
telecommunication. (Some publications spell it email; we prefer the currently more established
spelling of e-mail.) E-mail messages are usually encoded in ASCII text. However, you can also
send non-text files, such as graphic images and sound files, as attachments sent in binary streams.
E-mail was one of the first uses of the Internet and is still the most popular use. A large percentage
of the total traffic over the Internet is e-mail. E-mail can also be exchanged between online service
provider users and in networks other than the Internet, both public and private.
E-mail can be distributed to lists of people as well as to individuals. A shared distribution list can
be managed by using an e-mail reflector. Some mailing lists allow you to subscribe by sending a
request to the mailing list administrator. A mailing list that is administered automatically is called
a list server.
E-mail is one of the protocols included with the Transport Control Protocol/Internet Protocol
(TCP/IP) suite of protocols. A popular protocol for sending e-mail is Simple Mail Transfer
Protocol and a popular protocol for receiving it is POP3. Both Netscape and Microsoft include an e-
mail utility with their Web browsers.
HTTP: HTTP is called a stateless protocol because each command is executed independently,
without any knowledge of the commands that came before it. This is the main reason that it is
difficult to implement Web sites that react intelligently to user input. This shortcoming of HTTP is
being addressed in a number of new technologies, including ActiveX, Java, JavaScript and cookies.
HTTP Status Codes are Error Messages

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Errors on the Internet can be quite frustrating — especially if you do not know the difference
between a 404 error and a 502 error. These error messages, also called HTTP status codes are
response codes given by Web servers and help identify the cause of the problem.
16. Explain the message transfer using simple mail transfer protocol. (APRIL / MAY 2015)
SMTP stands for Simple Mail Transfer Protocol. It is a TCP/IP protocol that specifies
how computers exchange electronic mail. It works with post office protocol (POP).

SMTP is used to upload mail directly from the client to an intermediate host, but only computers
constantly connected such as Internet Service Providers (ISP) to the Internet can use SMTP to
receive mail. The ISP servers then offload the mail to the users to whom they provide the Internet
service.

SMTP uses TCP port number 25 for his service. Therefore e-mail is delivered from source to
destination by having the source machine established a TCP connection to port 25 of the destination
machine.

SMTP Commands
 SMTP commands are sent from the client to the server.
 Each command consists of a keyword or command name followed by zero or more
argument. It means some keywords do not contain any argument.
The format of command is:
 Keyword: argument(s)
 There are 14 different SMTP commands listed in the table below:
SMTP Responses
 SMTP responses are sent from server to client.
 Each response begins with a three digit code and may be followed by additional
textual information.
 The leading digits indicate the category of the response.

17. Write short notes on SNMP. (APRIL / MAY 2015)


Simple Network Management Protocol (SNMP) is an Internet-standard protocol for collecting
and organizing information about managed devices on IP networks and for modifying that
information to change device behavior. Devices that typically support SNMP include cable
modems, routers, switches, servers, workstations, printers, and more.
SNMP is widely used in network management for network monitoring. SNMP exposes
management data in the form of variables on the managed systems organized in a management
information base (MIB) which describe the system status and configuration. These variables can
then be remotely queried (and, in some circumstances, manipulated) by managing applications.
Three significant versions of SNMP have been developed and deployed. SNMPv1 is the original
version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in
performance, flexibility and security.
An SNMP-managed network consists of three key components:

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 Managed device
 Agent — software which runs on managed devices
 Network management station (NMS) — software which runs on the manager

18. Discuss about world wide web in detail.


WWW (World Wide Web):
The World Wide Web (WWW), or the web, is a repository of information spread all over the
world and linked together. The WWW has a unique combination of flexibility, portability, and user
friendly features that distinguish it from other services provided by the Internet.
The WWW is a distributed client-server service, in which a client using a browser can access a
service using a server. However, the service provided is distributed over many locations called web
sites.

19. Write short notes on FTP.


It is standard mechanism provided by TCP/IP for copying a file from one host to another.
Transferring files from one computer to another is one of the most common tasks expected from
a networking or internetworking environment.
FTP differs from other client-server applications in that it establishes two connections between
the hosts. One connection is used for data transfer, the other for control information (commands and
responses). Separation of commands and data transfer makes FTP more efficient. The control
connection uses very simple rules of communication.

20. Explain about the IMAP.


Developed after POP and attempts to fix POP deficiencies
 allows keeping all mail on the server
 allows mail categorization via folder metaphor
 mail is easily flagged (answered, draft, deleted, seen, recent); this isn’t the same on all
servers
 provides for multiple connections to the server
IMAP - process
 make connection
 send user credentials (userid and password)
 repeat until done – send a command – read response
 disconnect
IMAP - Commands
 login list status examine select create, delete, rename fetch store close expunge copy idle
lsub, subscribe, unsubscribe logout capability, getquotaroot, getacl

21. Explain about the MIME.


MIME: Multipurpose Internet Mail Extensions
MIME – Version: Version of MIME being used
Content Description: A human readable description of what’s in the message

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Content Type: Type of message
Example: Still images: image/gif, image/jpeg
Text: text/ rich text ,marked up texts
Application: – application/ postscripts ,application/ network
Also enables structuring of multiport type – - Message carrying more than one data type
structures
Mechanism for encoding:
Email contains only ASCII
Encoding – base 64
Map three bytes of original into 4 ASCII characters
Each 6-bit maps to a valid ASCII characteruc, lc, 10 digits + and /

22. Explain about the Traditional applications.


 SMTP: Simple Mail Transfer Protocol is used to exchange electronic mail.
 HTTP: HyperText Transport Protocol is used to communicate between web browsers and
web servers.
 DNS: Domain Name System protocol is used to query name servers and send the responses.
 SNMP: Simple Network Management Protocol is used to query (and sometimes modify) the
state of remote network nodes. The second point is that since all of the application protocols
described in this section follow the same request/reply communication pattern, we would
expect that they are all built on top of an RPC transport protocol
23. Explain about the SOAP.
 SOAP stands for Simple Object Access Protocol
 SOAP is a communication protocol
 SOAP is for communication between applications
 SOAP is a format for sending messages
 SOAP communicates via Internet
 SOAP is platform independent
 SOAP is language independent
 SOAP is based on XML
 SOAP is simple and extensible
 SOAP allows you to get around firewalls
 SOAP is a W3C recommendation
Some of the advantages of leveraging SOAP include:
 It is platform and language independent.
 SOAP provides simplified communications through proxies and firewalls, as mentioned
above.
 It has the ability to leverage different transport protocols, including HTTP and SMTP, as well
as others.
Some disadvantages of leveraging SOAP include:
 SOAP is typically much slower than other types of middleware standards, including CORBA.
This due to the fact that SOAP uses a verbose XML format. You need to fully understand the
performance limitations before building applications around SOAP.

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 SOAP is typically limited to pooling, and not event notifications, when leveraging HTTP for
transport. What's more, only one client can use the services of one server in typical situations.
 Again, when leveraging HTTP as the transport protocol, there tends to be firewall latency due
to the fact that the firewall is analyzing the HTTP transport. This is due to the fact that HTTP
is also leveraged for Web browsing, and many firewalls do not understand the difference
between the use of HTTP within a Web browser, and the use of HTTP within SOAP.
 SOAP has different levels of support, depending upon the programming language supported.
For example, SOAP support within Python and PHP is not as strong as it is
within Java and .NET.

24. Explain about the WSDL in web services.


 WSDL stands for Web Services Description Language
 WSDL is written in XML
 WSDL is an XML document
 WSDL is used to describe Web services
 WSDL is also used to locate Web services
 WSDL is a W3C recommendation
WSDL Describes Web Services
 WSDL stands for Web Services Description Language.
 WSDL is a document written in XML. The document describes a Web service. It specifies
the location of the service and the operations (or methods) the service exposes.
 WSDL is a W3C Recommendation
 WSDL became a W3C Recommendation 26. June 2007.

25. Explain about the DNS.


A distributed database used by TCP/IP applications – translates between symbolic names and IP
addresses – provides mail routing information
Top Level Domains (TLD)
Domain Description:.gov U.S. government, .org other organizations, .edu educational institutions
Country Code Domains:.us USA, .eg Egypt, .in India
Domain Names : Organizations apply for names within a top-level domain: – uni-ulm.de
unesco.org – hp.com

26. Explain about Mail reader and Mail transfer in SNMP in detail.
Components:
User agents: mail readers,composing, editing, reading the mail messages (Pine, Netscape
Messenger, Outlook) outgoing & incoming messages are stored on server
Message transfer agents: Mail Transfer Agent, used to move mail (ex. Send mail Mail Transfer
Agent l)
Mail Delivery Agent: used to deliver mail
Mail Delivery Agent Sendmail, POP (Post Office Protocol [RFC 1939]), IMAP (Internet Mail
Access Protocol [RFC 1730])

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UNEXPECTED QUESTIONS

27. Discuss briefly about UDDI.


Universal Description, Discovery, and Integration (UDDI) provides the definition of a set of
services supporting the description and discovery of (1) businesses, organizations, and other Web
Services providers, (2) the Web Services they make available, and (3) the technical interfaces which
may be used to access those services. The idea is to "discover" organizations and the services that
organizations offer, much like using a phone book or dialing information.

UDDI is based on a common set of industry standards, including HTTP, XML, XML Schema,
and SOAP. It provides an infrastructure for a Web Services-based software environment for both
publicly available services and services only exposed internally within an organization. The UDDI
Business Registry system consists of three directories:

 UDDI white pages: basic information such as a company name, address, and phone
numbers, as well as other standard business identifiers like Dun & Bradstreet and tax
numbers.
 UDDI yellow pages: detailed business data, organized by relevant business classifications.
The UDDI version of the yellow pages classifies businesses according to the newer NAICS
(North American Industry Classification System) codes, as opposed to the SIC (Standard
Industrial Classification) codes.
 UDDI green pages: information about a company's key business processes, such as
operating platform, supported programs, purchasing methods, shipping and billing
requirements, and other higher-level business protocols.

28. Write short notes on RESTful web services.


REST stands for REpresentational State Transfer. REST is a web standards based architecture
and uses HTTP Protocol for data communication. It revolves around resources where every
component is a resource and a resource is accessed by a common interface using HTTP standard
methods.
HTTP Methods
The following HTTP methods are most commonly used in a REST based architecture.
 GET − Provides a read only access to a resource.

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 PUT − Used to create a new resource.
 DELETE − Used to remove a resource.
 POST − Used to update an existing resource or create a new resource.
 OPTIONS − Used to get the supported operations on a resource.

29. What are the elements containing in SOAP message?Explain?

A SOAP message is an ordinary XML document containing the following elements

 Envelope − Defines the start and the end of the message. It is a mandatory element.
 Header − Contains any optional attributes of the message used in processing the message,
either at an intermediary point or at the ultimate end-point. It is an optional element.
 Body − Contains the XML data comprising the message being sent. It is a mandatory
element.
 Fault − An optional Fault element that provides information about errors that occur while
processing the message.

30. Elaborate Service Oriented Architecture.
A service-oriented architecture is essentially a collection of services. These services
communicate with each other. The communication can involve either simple data passing or it could
involve two or more services coordinating some activity. Some means of connecting services to each
other is needed.

SERVICES
If a service-oriented architecture is to be effective, we need a clear understanding of the term
service. A service is a function that is well-defined, self-contained, and does not depend on the
context or state of other services.

CONNECTIONS
The technology of Web Services is the most likely connection technology of service-oriented
architectures. The following figure illustrates a basic service-oriented architecture. It shows a service
consumer at the right sending a service request message to a service provider at the left. The service
provider returns a response message to the service consumer. The request and subsequent response
connections are defined in some way that is understandable to both the service consumer and service
provider. How those connections are defined is explained in Web Services Explained. A service
provider can also be a service consumer.

31. Describe in detail about SOAP element fault.


If an error occurs during processing, the response to a SOAP message is a SOAP fault element
in the body of the message, and the fault is returned to the sender of the SOAP message.

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The SOAP fault mechanism returns specific information about the error, including a predefined
code, a description, and the address of the SOAP processor that generated the fault.
Below mentioned are few important points about SOAP fault element to take note of
 A SOAP message can carry only one fault block.
 Fault is an optional part of a SOAP message.
 For HTTP binding, a successful response is linked to the 200 to 299 range of status
codes.
SOAP Fault is linked to the 500 to 599 range of status codes.

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