Bcs 052 solved assignment

Course Code :BCS-052
Course Title :Network Programming and Administration
Assignment Number :BCA(V)-052/Assign/14-15
Maximum Marks :100
Weightage :25%
Last Dates for Submission :15th October, 2014 (For July 2014 Session)
15th April, 2015 (For January 2015 Session)
This assignment has 4 questions of 80 marks. Rest 20 marks are for viva voce.
Question 1:
(a) What is IPv6? Explain its need and important features those are not available in IPv4.
Solution:
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).
The official name of IPng is IPv6, where IP stands for Internet Protocol and v6 stands for
version 6. Internet Protocol Version 6 (IPv6) is the successor to Internet Protocol Version
4 (IPv4). IPv6 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.
Important features those are not available in IPv4.
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.
While increasing the pool of addresses is one of the most often-talked about benefit of
IPv6, there are other important technological changes in IPv6 that will improve the IP
protocol:
- No more NAT (Network Address Translation)
- Auto-configuration
- No more private address collisions
- Better multicast routing
- Simpler header format
- Simplified, more efficient routing
- True quality of service (QoS), also called “flow labeling”
- Built-in authentication and privacy support
- Flexible options and extensions
- Easier administration (say good-bye to DHCP)

(b) Compare the sliding window protocol of data link layer and transport layer. Why flow
control is used at both layers? Justify your answer.
Solution:
Sliding Window protocol
Frames have sequence number 0 to maximum 2n - 1 (n bit field).
At any moment, the sender maintains a list of sequence numbers it is permitted to send
– these fall within the sending window. These are frames sent-but-no-ack and frames
not-yet-sent.
When new packet from Network layer comes in to send, it is given highest no, and upper
edge of window advanced by
1. When ack comes in, lower edge of window advanced by 1.
Receiver has receiving window – the frames it is permitted to accept.
Sliding window size 1. Sequence nos. 0 to 7.
(a) At start. Receiver waits for 0.
(b) Sender sends 0.
(c) Receiver receives 0. Waits for 1.
(d) Sender got ack for 0. Hasn’t got 1 from its Network layer yet.
More complex Data Link layer, as more freedom about the order in which it sends and
receives frames.
Sender may have n unacknowledged frames at any time (window size n).
Needs n buffers to hold them all for possible re-transmit.
If window grows to its maximum size, DA must shut off NA.
This is all hidden from NB – still receives packets in exact same order.
Sender window might grow as it receives more frames to send and still has ones un-
ack’ed.
Starts with nothing to send, then NA gives it frames to send.
Later, window may shrink as frames are ack-ed and NA has no more.
Receiver window constant size.
Receiver window size 1 means will only accept them in order.
Size n means will receive out of order (e.g. receive later ones after earlier frame is lost)
and then must buffer them before sending to NB (must send to NB in order).
e.g. DB has buffers to receive frames 0..7
Receives 1..7 in varying orders. Still waiting on 0. Can’t send frames to NB yet.
0 got lost and was re-sent. Eventually gets 0.
Can now send all of 0..7 to NB

and re-use these slots.
e.g. consider frames numbered 0..7 but DB only has 2 buffers
 Currently the sliding window is over 4,5
 If get 4 can send it to NB and move window to 5,6
 If get 5 have to wait for 4, then send both, and advance window to 6,7
Data Link Layer
The data link layer is the second layer in the OSI (open systems interconnection) seven-
layer reference model. It responds to service requests from the network layer above it
and issues service requests to the physical layer below it.
The data link layer is responsible for encoding bits into packets prior to transmission and
then decoding the packets back into bits at the destination. Bits are the most basic unit
of information in computing and communications. Packets are the fundamental unit of
information transport in all modern computer networks, and increasingly in other
communications networks as well.
The data link layer is also responsible for logical link control, media access control,
hardware addressing, error detection and handling and defining physical layer
standards. It provides reliable data transfer by transmitting packets with the necessary
synchronization, error control and flow control.
The data link layer is divided into two sublayers: the media access control (MAC) layer
and the logical link control (LLC) layer. The former controls how computers on the
network gain access to the data and obtain permission to transmit it; the latter controls
packet synchronization, flow control and error checking.
The data link layer is where most LAN (local area network) and wireless LAN
technologies are defined. Among the most popular technologies and protocols generally
associated with this layer are Ethernet, Token Ring, FDDI (fiber distributed data
interface), ATM (asynchronous transfer mode), SLIP (serial line Internet protocol), PPP
(point-to-point protocol), HDLC (high level data link control) and ADCCP (advanced data
communication control procedures).
The data link layer is often implemented in software as a driver for a network interface
card (NIC). Because the data link and physical layers are so closely related, many types
of hardware are also associated with the data link layer. For example, NICs typically
implement a specific data link layer technology, so they are often called Ethernet cards,
Token Ring cards, etc. There are also several types of network interconnection devices
that are said to operate at the data link layer in whole or in part, because they make
decisions about what to do with data they receive by looking at data link layer packets.
These devices include most bridges and switches, although switches also encompass
functions performed by the network layer.

Data link layer processing is faster than network layer processing because less analysis
of the packet is required.
(c) What are the corresponding protocols of the following TCP/IP protocols in the OSI
Model?Compare them.
 DNS
 FTP
 TFTP
Solution:
DNS
The DNS translates Internet domain and host names to IP addresses. DNS
automatically converts the names we type in our Web browser address bar to the IP
addresses of Web servers hosting those sites.
DNS implements a distributed database to store this name and address information for
all public hosts on the Internet. DNS assumes IP addresses do not change (are statically
assigned rather than dynamically assigned).
FTP
File Transfer Protocol (FTP) is a standard Internet protocol for transmitting files between
computers on the Internet. Like the Hypertext Transfer Protocol (HTTP), which transfers
displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP),
which transfers e-mail, FTP is an application protocol that uses the Internet’s TCP/IP
protocols. FTP is commonly used to transfer Web page files from their creator to the
computer that acts as their server for everyone on the Internet. It’s also commonly used
to download programs and other files to your computer from other servers.
TFTP
Trivial File Transfer Protocol (TFTP) is an Internet software utility for transferring files
that is simpler to use than the File Transfer Protocol (FTP) but less capable.
(d) Why do LANs tend to use broadcast networks? Why not use networks consisting of
multiplexers and switches.
Solution:
The computer systems within a Local Area Networks (LAN) are usually separated by a
small distance (usually < 100m) therefore high speed and efficient communication is
quite possible utilizing a shared broadcast carrier. The price of the carrier is minimal and
also the total cost is dominated by the expense of the network interface cards in every
computer system. Additionally, the Local Area Networks (LAN) users generally are
members of the same team where all users are usually trustworthy, so broadcast doesn’t
cause significant security and reliability risk.
Not use networks consisting of multiplexers and switches.
The main reason behind keeping away from a multiplexer and switch approach to LANs

is the fact that a central, costly “box” is needed. The production of Application Specific
Integrated Circuits (ASICs) has decreased the expense of switching boxes making
switch- dependent LANs possible, as well as in a few conditions the prominent method.
Question 2:
(a) Why would an application use UDP instead of TCP? Also, explain how can TCP handle
urgent data?
Solution:
UDP is good for sending messages from one system to another when the order isn’t
important and you don’t need all of the messages to get to the other machine. This is
why I’ve only used UDP once to write the example code for the faq. Usually TCP is a
better solution. It saves you having to write code to ensure that messages make it to
the desired destination, or to ensure the message ordering. Keep in mind that every
additional line of code you add to your project in another line that could contain a
potentially expensive bug.
If you find that TCP is too slow for your needs you may be able to get better
performance with UDP so long as you are willing to sacrifice message order and/or
reliability.
UDP must be used to multicast messages to more than one other machine at the same
time. With TCP an application would have to open separate connections to each of the
destination machines and send the message once to each target machine. This limits
your application to only communicate with machines that it already knows about.
It should be obvious with regard to the second point that if a TCP connection is used for
multiple request-reply exchanges, then the cost of the connection’s establishment and
teardown is amortized across all the requests and replies, and this is normally a better
design than using a new connection for each request-reply. Nevertheless, there are
applications that use a new TCP connection for each request-reply (e.g., the older
versions of HTTP), and there are applications in which the client and server exchange
one request-reply (e.g., the DNS) and then might not talk to each other for hours or
days.
We now list the features of TCP that are not provided by UDP, which means that an
application must provide these features itself, if they are necessary to the application.
We use the qualifier “necessary” because not all features are needed by all applications.
For example, dropped segments might not need to be retransmitted for a real-time audio
application, if the receiver can interpolate the missing data. Also, for simple request-reply
transactions, windowed flow control might not be needed if the two ends agree ahead of
time on the size of the largest request and reply.
 Positive acknowledgments, retransmission of lost packets, duplicate detection, and
sequencing of packets reordered by the network—TCP acknowledges all data,
allowing lost packets to be detected. The implementation of these two features
requires that every TCP data segment contain a sequence number that can then be
acknowledged. It also requires that TCP estimate a retransmission timeout value for
the connection and that this value be updated continually as network traffic between
the two end-systems changes.

 Windowed flow control—A receiving TCP tells the sender how much buffer space it
has allocated for receiving data, and the sender cannot exceed this. That is, the
amount of unacknowledged data at the sender can never exceed the receiver’s
advertised window.
 Slow start and congestion avoidance—This is a form of flow control imposed by the
sender to determine the current network capacity and to handle periods of
congestion. All current TCPs must support these two features and we know from
experience (before these algorithms were implemented in the late 1980s) that
protocols that do not “back off” in the face of congestion just make the congestion
worse (e.g., [Jacobson 1988]). TCP
handle urgent data
When an interactive hits the DEL or CTRL-C key to break-off a remote computation that
has already begun, the sending application puts some control information in the data
stream and gives it to TCP along with the URGENT flag. This even causes TCP to stop
accumulating data and transmit everything it has for that connection immediately.
The receiving application is interrupted so it can stop whatever it was doing so that it can
read the data stream to find the urgent data.
(b) Write a connection-oriented client and server program in C language on UNIX
platform,
where client program interact with the Server as given below:
i- The client begins by sending a request; the server sends back a confirmation and its clock time
to the client.
ii -The client sends a number and server replies as the square of that number to the
client.
(C) What are the special IP addresses? Give the significance of these addresses.
Solution:
There are several IP addresses that are special in one way or another. These addresses
are for special purposes or are to be put to special use.
 Addresses significant to every IP subnet
 Network Address
 Broadcast Address
Addresses significant to individual hosts
 Loopback Address
Special Addresses of Global Significance
 Private Addresses
 Reserved Addresses
LOOPBACK ADDRESS (127.0.0.1)The 127.0.0.0 class ‘A’ subnet is used for only a
single address: the loopback address 127.0.0.1. This address is used to test the local
network interface device’s functionality. All network interface devices should respond to

this address. If you ping 127.0.0.1, you can be assured that the network hardware is
functioning and that the network software is also functioning.
Question 3:
(a) What is HTTP? Describe the various HTTP request methods using an example of each.
Solution:
Short for HyperText Transfer Protocol, HTTP is a set of standards that allow users of the
World Wide Web to exchange information found on web pages. When wanting to access
any web page enter http:// in front of the web address, which tells the browser to
communicate over HTTP. For example, the full URL for Computer Hope is
http://www.computerhope.com. Today’s modern browsers no longer require HTTP in
front of the URL since it is the default method of communication. However, it is still used
in browsers because of the need to access other protocols such as FTP through the
browser. Below are a few of the major facts on HTTP.
The term HTTP was coined by Ted Nelson.
HTTP commonly utilizes port 80, 8008, or 8080.
HTTP/0.9 was the first version of the HTTP and was introduced in 1991.
HTTP/1.0 is specified in RFC 1945 and introduced in 1996.
HTTP/1.1 is specified in RFC 2616 and officially released in January 1997.
HTTPS
Short for Hypertext Transfer Protocol over Secure, HTTPS is a secure method of
accessing or sending information across a web page. All data sent over HTTPS is
encrypted before it is sent, this prevents anyone from understanding that information if
intercepted. Because data is encrypted over HTTPS, it is slower than HTTP, which is
why HTTPS is only used when requiring login information or with pages that contain
sensitive information such as an online bank web page.
 HTTPS uses port 443 to transfer its information.
 HTTPS is first used in HTTP/1.1 and is defined in RFC 2616.
How to protect yourself and verify Internet data is secure while online.
Other related RFCs of interest
RFC 2068
HTTP status codes
Below is a listing of HTTP status codes currently defined by Computer Hope. These
codes enable a client accessing another computer or device over HTTP to know how to
proceed or not proceed.
1xx – 2xx 3xx – 4xx 5XX
100
101
102
200
201
301
302
304
400
401
500
501
503
505

202
204
205
206
207
402
403
404
405
406
407
408
409
410
413
414
416
HTTP request methods
Minimally, the response should be a 200 OK and have an Allow header with a list of
HTTP methods that may be used on this resource. As an authorized user on an API, if
you were to request OPTIONS /users/me, you should receive something like…
200 OK
Allow: HEAD,GET,PUT,DELETE,OPTIONS
…
Server: Apache/2.4.1 (Unix) OpenSSL/1.0.0g
Allow: GET,HEAD,POST,OPTIONS,TRACE
Content-Type: httpd/unix-directory
…
It could be an HTML page with documentation, but that’s sort of unpractical because
users don’t click the “get options” button in their browsers before visiting a page.
Machines may though.
APIs should be taking advantage of this. There are many benefits to be gained from
producing machine readable docs at every endpoint. It would be a boon for automatic
client generation for web services. Communication between web services could be much
more resilient if they had a codified way to check their abilities against each other.
At the very least, services should be responding with a 200 and the Allow header. That’s
just correct web server behavior. But there’s really no excuse for JSON APIs not to be
returning a documentation object. To use GitHub as example again, on the issues
endpoint, a request like OPTIONS /repos/:user/:repo/issues should respond with a body
like…
{
“POST”: {
“description”: “Create an issue”,
“parameters”: {
“title”: {
“type”: “string”

“description”: “Issue title.”,
“required”: true
},
“body”: {
“type”: “string”,
“description”: “Issue body.”,
},
“assignee”: {
“type”: “string”,
“description” “Login for the user that this issue should be assigned to.”
},
“milestone”: {
“type”: “number”,
“description”: “Milestone to associate this issue with.”
},
“labels”: {
“type”: “array/string”
“description”: “Labels to associate with this issue.”
}
},
“example”: {
“title”: “Found a bug”,
“body”: “I’m having a problem with this.”,
“assignee”: “octocat”,
“milestone”: 1,
“labels”: [
"Label1",
"Label2"
]
}
}
}
(b) What is domain name? How is a domain name translated to an equivalent IP address?
Explain using an example.
Solution:
Domain name
Every time you visit a website, you are interacting with the largest distributed database in
the world. This massive database is collectively known as the DNS, or the Domain
Name System. Without it, the Internet as we know it would be unable to function. The
work that the DNS does happens so seamlessly and instantaneously that you are
usually completely unaware that it’s even happening. The only time that you’ll get an
inkling about what the DNS is doing is when you’re presented with an error after trying to
visit a website. Learn more about what the DNS is, how it works and why it’s so critical
by reading on below.

A lot of what has been discussed may be a bit confusing, so lets do a real life example.
In the flowchart below labeled Figure 1, you will see a computer trying to connect to
www.google.com and the steps it takes.
Example
1. A User opens a web browser and tries to connect to www.google.com. The
operating system not knowing the IP Address for www.google.com, asks the
ISP’s DNS Server for this information.
2. The ISP’s DNS Server does not know this information, so it connects to a
Root Server to find out what name server, running somewhere in the world,
know the information about google.com.
3. The Root Server tells the ISP’s DNS Server to contact a particular name
server that knows the information about google.com.
4. The ISP’s DNS Server connects to Google’s DNS server and asks for the IP
Address for www.google.com.
5. Google’s DNS Server responds to the ISP’s DNS server with the appropriate
IP Address.
6. The ISP’s DNS Server tells the User’s operating system the IP Address for
google.com.
7. The operating system tells the Web Browser the IP Address for
www.google.com.
1. The web browser connects and starts communication with www.google.com.
(c) What is a mail server? Briefly explain specifying the protocols involved, how a sender
can send a mail to the server and the recipient retrieves it from the server?
Solution:
Mail server
A mail server is the computerized equivalent of your friendly neighborhood mailman.
Every email that is sent passes through a series of mail servers along its way to its
intended recipient. Although it may seem like a message is sent instantly – zipping from
one PC to another in the blink of an eye – the reality is that a complex series of transfers
takes place. Without this series of mail servers, you would only be able to send emails to
people whose email address domains matched your own – i.e., you could only send
messages from one example.com account to another example.com account.
Types of Mail Servers

Mail servers can be broken down into two main categories: outgoing mail servers and
incoming mail servers. Outgoing mail servers are known as SMTP, or Simple Mail
Transfer Protocol, servers. Incoming mail servers come in two main varieties. POP3, or
Post Office Protocol, version 3, servers are best known for storing sent and received
messages on PCs’ local hard drives. IMAP, or Internet Message Access Protocol,
servers always store copies of messages on servers. Most POP3 servers can store
messages on servers, too, which is a lot more convenient.
The Process of Sending an Email
Now that you know the basics about incoming and outgoing mail servers, it will be easier
to understand the role that they play in the emailing process. The basic steps of this
process are outlined below for your convenience.
Step #1: After composing a message and hitting send, your email client – whether it’s
Outlook Express or Gmail – connects to your domain’s SMTP server. This server can be
named many things; a standard example would be smtp.example.com.
Step #2: Your email client communicates with the SMTP server, giving it your email
address, the recipient’s email address, the message body and any attachments.
Step #3: The SMTP server processes the recipient’s email address – especially its
domain. If the domain name is the same as the sender’s, the message is routed directly
over to the domain’s POP3 or IMAP server – no routing between servers is needed. If
the domain is different, though, the SMTP server will have to communicate with the other
domain’s server.
Step #4: In order to find the recipient’s server, the sender’s SMTP server has to
communicate with the DNS, or Domain Name Server. The DNS takes the recipient’s
email domain name and translates it into an IP address. The sender’s SMTP server
cannot route an email properly with a domain name alone; an IP address is a unique
number that is assigned to every computer that is connected to the Internet. By knowing
this information, an outgoing mail server can perform its work more efficiently.
Step #5: Now that the SMTP server has the recipient’s IP address, it can connect to its
SMTP server. This isn’t usually done directly, though; instead, the message is routed
along a series of unrelated SMTP servers until it arrives at its destination.
Step #6: The recipient’s SMTP server scans the incoming message. If it recognizes the
domain and the user name, it forwards the message along to the domain’s POP3 or
IMAP server. From there, it is placed in a sendmail queue until the recipient’s email
client allows it to be downloaded. At that point, the message can be read by the
recipient.
How Email Clients are Handled
Many people use web-based email clients, like Yahoo Mail and Gmail. Those who
require a lot more space – especially businesses – often have to invest in their own
servers. That means that they also have to have a way of receiving and transmitting
emails, which means that they need to set up their own mail servers. To that end,
programs like Postfix and Microsoft Exchange are two of the most popular options. Such
programs facilitate the preceding process behind the scenes. Those who send and
receive messages across those mail servers, of course, generally only see the “send”

and “receive” parts of the process.
At the end of the day, a mail server is a computer that helps move files along to their
intended destinations. In this case, of course, those files are email messages. As easy
as they are to take for granted, it’s smart to have a basic grasp of how mail servers work.
(d) Draw the IP datagram header format. ―IP datagram has a checksum field still its
called unreliable protocol.‖ Justify.
Solution:
IP is the workhorse protocol of the TCP/IP protocol suite. All TCP, UDP, ICMP, and
IGMP data gets transmitted as IP datagrams (Figure 1.4). A fact that amazes many
newcomers to TCP/IP, especially those from an X.25 or SNA background, is that IP
provides an unreliable, connectionless datagram delivery service.
By unreliable we mean there are no guarantees that an IP datagram successfully gets to
its destination. IP provides a best effort service. When something goes wrong, such as a
router temporarily running out of buffers, IP has a simple error handling algorithm: throw
away the datagram and try to send an ICMP message back to the source. Any required
reliability must be provided by the upper layers (e.g., TCP).
The term connectionless means that IP does not maintain any state information about
successive datagrams. Each datagram is handled independently from all other
datagrams. This also means that IP datagrams can get delivered out of order. If a source
sends two consecutive datagrams (first A, then B) to the same destination, each is
routed independently and can take different routes, with B arriving before A.
In this chapter we take a brief look at the fields in the IP header, describe IP routing, and
cover subnetting. We also look at two useful commands: ifconfig and netstat. We leave a
detailed discussion of some of the fields in the IP header for later when we can see
exactly how the fields are used. RFC 791 [Postel 1981a] is the official specification of IP.
IP Header
shows the format of an IP datagram. The normal size of the IP header is 20 bytes,
unless options are present.
IP datagram, showing the fields in the IP header.
We will show the pictures of protocol headers in TCP/IP as in Figure The most

significant bit is numbered 0 at the left, and the least significant bit of a 32-bit value is
numbered 31 on the right.
The 4 bytes in the 32-bit value are transmitted in the order: bits 0-7 first, then bits 8-15,
then 16-23, and bits 24-31 last. This is called big endian byte ordering, which is the byte
ordering required for all binary integers in the TCP/IP headers as they traverse a
network. This is called the network byte order. Machines that store binary integers in
other formats, such as the little endian format, must convert the header values into the
network byte order before transmitting the data.
The current protocol version is 4, so IP is sometimes called IPv4. Section 3.10
discusses some proposals for a new version of IP.
The header length is the number of 32-bit words in the header, including any
options. Since this is a 4-bit field, it limits the header to 60 bytes. In Chapter 8 we’ll
see that this limitation makes some of the options, such as the record route
option, useless today. The normal value of this field (when no options are present)
is 5.
The type-of-service field (TOS) is composed of a 3-bit precedence field (which is
ignored today), 4 TOS bits, and an unused bit that must be 0. The 4 TOS bits are:
minimize delay, maximize throughput, maximize reliability, and minimize monetary
cost.
Recommended values for type-of-service field.
Only 1 of these 4 bits can be turned on. If all 4 bits are 0 it implies normal service. RFC
1340 [Reynolds and Postel 1992] specifies how these bits should be set by all the
standard applications. RFC 1349 [Almquist 1992] contains some corrections to this RFC,
and a more detailed description of the TOS feature.
shows the recommended values of the TOS field for various applications. In the final
column we show the hexadecimal value, since that’s what we’ll see in the tcpdump
output later in the text.
Question 4:
(a) What are the NTFS, FAT, HPFS file systems? Compare and contrast between these file
systems.
Solution:
File Systems (FAT, HPFS, NTFS)
At the BIOS level, a disk partition contains sectors numbered 0, 1, etc. Without additional
support, each partition would be one large dataset. Operating systems add a directory

structure to break the partition up into smaller files, assign names to each file, and
manage the free space available to create new files.
The directory structure and methods for organizing a partition is called a File System.
Different File Systems reflect different operating system requirements or different
performance assumptions. Unix, for example, has the convention that lowercase and
uppercase are different in file names, so “sample.txt” and “Sample.txt” are two different
files. DOS and the systems that descend from it (Windows 95, OS/2, and Windows NT)
ignore case differences when finding file names. Some File Systems work better on
small machines, others work better on large servers.
Each partition is assigned a type (in the MBR for primary partitions, in the Extended
Partition directory for logical volumes). When the partition is formatted with a particular
File System, the partition type will be updated to reflect this choice.
The same hard disk can have partitions with File Systems belonging to DOS, OS/2, NT,
and Linux (or other Unix clones). Generally, an operating system will ignore partitions
whose type ID represents an unknown file system type. It is fairly easy (given a big
enough disk) to install all of the different operating systems and all of the File System
types. There are a few rules to make things simple.
Each File System is described in detail in a separate section.
 FAT File System
The FAT File system is used by DOS and is supported by all the other operating
systems. It is simple, reliable, and uses little storage.
 VFAT
VFAT is an alternate use of the FAT file system available in Windows 95 and Windows
NT 3.5. It allows files to have longer names than the “8.3″ convention adopted by DOS.
VFAT stores extra information in the directory that older DOS and OS/2 systems can
ignore.
 HPFS
HPFS is used by OS/2 and is supported by Windows NT. It provides better performance
than FAT on larger disk volumes and supports long file names. However, it requires
more memory than FAT and may not be a reasonable choice on systems with only 8
megs of RAM.
 NTFS
NTFS provides everything. It supports long file names, large volumes, data security, and
universal file sharing. A departmental NT file server will probably have all its partitions
formatted for NTFS. Because the other operating systems cannot use it, NTFS is less
attractive on personal desktop workstations or portables.
File Systems and Disk Letters
DOS and Windows 95 can only boot from the C: disk. Technically, the C: letter will be
assigned to the first Primary Partition on the first hard disk that has a FAT file system. In
no case can DOS boot from a second hard disk or a logical volume in the extended

partition. However, if as the system comes up, the DOS boot sector and DOS files turn
out to be on the second Primary Partition on the first hard disk, then this will not be a
problem so long as the first partition has a non-FAT file system. DOS simply ignores
primary partitions that are formatted for other operating systems.
Some people exploit this feature. They put an HPFS or NTFS file system on the first
Primary Partition, and a FAT file system on the second. This can produce confusion.
When the other operating system boots up, it will now assign letter C: to its first partition,
and the disk that DOS calls “C:” will become “D:” on the other system. If the two systems
share application programs, it becomes very difficult to configure INI files as the drive
letter keeps changing back and forth. It is a simpler and safer strategy to accept the view
that the first Primary Partition on the first hard disk should be formatted with the FAT file
system and should be the C: drive in every operating system.
Choosing a File System
The performance problems with FAT have been greatly reduced by various strategies to
use Cache memory and to periodically DEFRAG the disk. FAT is the only system fully
supported by DOS and Windows 95. It is also a perfectly acceptable choice under
Windows NT and OS/2. FAT systems require the least memory and are the best choice
on small machines.
Although it is simpler to manage a few larger volumes, FAT performance degrades with
volume size. The distance between the directory and the data increases the disk
movement, and larger allocation units waste space. A good rule of thumb would limit
FAT volumes to a maximum of 255 megabytes.
FAT proven to be quite reliable and is fairly immune to damage. When the system
crashes, FAT can “misplace” disk space that was being allocated to a file. CHKDSK (or
Microsoft’s newer SCANDISK) will recover the missing space. Less frequently a really
serious error could leave the same sector of disk space assigned to two different files.
Such “crosslinked” files are damaged, and once this occurs the entire volume is suspect.
The preferred recovery would be to back everything up, reformat the volume, and
restore the data. Crosslinked files could be produced by a damaged operating system,
or by a hardware problem in the disk subsystem itself.
HPFS is supported by OS/2 and Windows NT. Although it is not officially supported by
DOS or Windows 95, there are shareware drivers (such as AMOS3) that can provide
these systems with at least Read-Only access to HPFS files. Since OS/2 does not
support VFAT, it cannot use long file names on a FAT volume. Many OS/2 software
packages require long file names. An OS/2 system with enough memory and disk space
should have at least one HPFS volume to support such packages.
Only Windows NT can use data on an NTFS volume. NTFS is required to provide full
security on an NT File Server, and to support Macintosh datasets. On desktop
workstations that run other operating systems as well as NT, NTFS is probably more
trouble than it is worth.
A good general principle is to put FAT volumes first on a disk, then HPFS, and finally
NTFS. All the systems will see the FAT volumes and will assign them disk letters. With
device drivers for DOS, all the system will see the HPFS volumes as well. The NTFS
volumes will only be available to Windows NT and will be ignored by the other systems.

(b) Describe the activities to be performed at every layer in the TCP/IP model when
information flows from layer to another layer.
Solution:
The theory and idea behind having standards accepted, ratified, and agreed upon by
nations around the world, is to ensure that the system from Country A will be easily
integrated with the system from Country B with little effort. It also helps to make
specification for industries to create goods and services that conform to the standard
and by providing competition to the same product, decrease prices for products that
must match the minimum standards. Comparisons are made easier in this way for
products made by competing groups that must meet or exceed the minimum accepted
specified standards.
A protocol is more like a language that can be shared by many people. A protocol may
become a standard, if all of the players in the game that would like to use that protocol
all politically agree that it shall be the protocol of choice for use in, and between nations.
When the protocol is ratified by the governing bodies as the shared and agreed upon
system, it becomes an official standard..
The ISO looked to create a simple model for networking. They took the approach of
defining layers that rest in a stack formation, one layer upon the other. Each layer would
have a specific function, and deal with a specific task. Much time was spent in creating
their model called “The ISO OSI Seven Layer Model for Networking”. In this model, they
have 7 layers, and each layer has a special and specific function.
ISO OSI Seven Layer Model
Described:
 7.) Application Layer:
The Application Layer can include things like File Transfers, and display formatting.
HTTP is an example of an Application Layer protocol. Commonly known protocols
considered by many to be part of the application layer actually may be considered to
occupy the Session,Presentation, and Application Layers. For example an examination
of an NFS file mount with files being copied defies simple categorization within the ISO
OSI 7 Layer Reference Model. Is NFS an Application Layer Protocol? Well, files are
copied, so we see that the Application Layer may be included. However, synchronization
in file transfers takes place to some extent and session are created and torn down on
demand as files are transferred. This suggests that it could also be part of the session
layer or maybe presentation layer.
 6.) Presentation Layer
Other than data sent/accepted to/from the application layer and Session Layer, this layer
is reserved for certain kinds of data manipulation or consistent data types being

encapsulated for transmission. Translations could possibly be made between ASCII and
Unicode or even EBCIDIC if hexadecimal values for letter were being transmitted.
It is the presentation layer that is also able to exchange messages and often dynamically
create a syntax that is shared by it and its peer layer service on the remote stack.
It is possible for something like a database translation system that could provide a
consistent presentation service for an application program performing database queries
to operate here. There are some parts of OBDC that may fit into the presentation layer in
this respect.
Other examples of translation that might be made to “fit” in this layer include vt100,
vt220, HTML and codes for translation of data to be presented. FOr example, in html,
“&” can be used to represent “&” and this is effectively a modification of data being
displayed. (The end user often does not see the escape codes used to display an
inverse letter, or odd symbol.)
 5.) Session Layer:
This section is one of the most often misunderstood sections since it does not
have an obvious separate protocol when people try to apply it to a common
layered system that may use TCP/IP or IPX/SPX. Often with these protocols
and protocols on top of these protocols, layer boundaries are not so obvious.
In examining what services are supposed to take place here according to the
ISO in this OSI 7 Layer Reference model for Networking we can see a short
list.
Deal with creating a session, transmission of data, and then tear down of the created
session. Sessions are created and terminated at the request of thePresentation Layer as
it has data needing to be passed on to a different location.
Part of the Session creation process includes dealing with cases of Half Duplex sessions
where only a transmission or reception may take place and working out a turn sharing
system to ensure both sides get opportunities to transmit as they need to relay data. In
the case of Full Duplex support, a discovery process may be needed to allow this layer
to know that bi-directional conversations make take place at the same time.
Another service that is offered as a part of the Session Layer might include data
synchronization. Checksums may also be included at the Session Layer as a part of
data synchronization. A checksum is performed after each packet is transmitted to see if
applying the data from the packet to the file or stream being moved or transmitted would
cause it to have the same checksum as the file on the remote location up to that point. If
it is, then the new data may be added to the local machine being transferred from the
remote site. This is a form of error correction for transmitted data. A familiar form of
checksums in use can be seen in Z-modem transfers as part of communications or
terminal software. The wonderful part of z-modem transfers is that it becomes possible
for an interrupted z-modem download to be resumed where it left off with a minimal
amount of retransmitted data. This may not be a method used at this layer, but it shows
how using a system of synchronization with each part of the data being transferred can
allow for interruptions to limit the problems associated with having to start the whole
transmission over again.

 4.) Transport Layer:
This layer is responsible for many things that individually may not seem exceptionally
important but actually provide for some critical needs.
Just as you will read in some of the next layers below, this layer also looks to prevent a
fast sender from over-running a slow receiver. An analogy is made in the Networking
Layer section between data throughput rates and pipe sizes may better illustrate this and
keep the amount of reading smaller if it is examined later.
This layer, just as all layers, accept data from layer immediately above and below it
(except for layer 1, and layer 7) as well as provide services for the layers above them
(except for Layer 7.) In this case, the Transport Layer must create a connection of the
type needed by the Session Layer for each connection requested by the Session Layer.
In cases where data being pushed down this model towards this layer is larger than the
maximum allowed size of packets for this layer, it is up to this layer to re-size the
incoming data from above. It does this by breaking the larger sized data from the layer
above into smaller sizes that may fit within packets for this layer. The peer level Session
Layer then re-creates the larger sized data for its upper layer by connecting payloads of
separate packets together in a “stream”.
If a higher layer requires that a connection be created that is “reliable” (able to notice an
error and then correct for the error so that all data sent eventually arrives at its
destination) and the above layer cannot provide its own method of ensuring a “reliable”
connection, then it is up to this layer to create a “reliable connection” that ensures all
data sent eventually arrives.
If a “reliable” service is not required, but instead an “unreliable” connection is desired,
then it is up to this layer to ensure that the packets arriving are the same as the packets
that were sent, or else they are discarded. This can be performed by a number of ways,
but the most common is to use special checksums (explained in a lower layer below
this.)
(Described Later: Brief summary) Both reliable and unreliable connections may use
something called a checksum (explained later in this page) If a packets computed
checksum matches its carried checksum, then the packet may be considered valid, and
have its payload passed on up to a higher layer. If the checksums do not match, then the
packet may be discarded.
 3.) Networking Layer:
It is the opinion of one of my associates that this layer is potentially the most complex of
all the layers due to the issues that it must address. Most importantly: routing. This layer
is responsible primarily with routing of data from the layer above (Transport Layer) to a
remote location that may or may not share the same Physical Layer-direct-link, or even
Data Link Layer protocol. Issues in differences in the commonly referenced name
“bandwidth” to describe the size of an imaginary pipe for pushing data from one point to
another as if it were liquid in a real pipe, are also address here at this layer. Just as the
Data Link Layer below must ensure that a fast sender does not flood-out a slower
receiver and possibly lead to lost data, this layer also must address problems that may
exist when a stream of packets coming form a network with unused big pipes encounters

a possibly busy network with small pipes. Even in the case where two networks both
have the same sized pipes, a network with a pipe that is almost full may have difficulties
in passing on incoming packets from a network with a pipe that is nearly empty.
Other issues that are resolved by this layer include dealing with packet sized in the case
of dissimilar settings, or protocols between networks forcing the size of a packet to
become smaller before being passed on (called fragmentation in IP of the TCP/IP suite
of protocols). (Part of another document goes to better describe this using the often used
“carrier pigeon” lesson to convey the problems of packet fragmentation by using
symbolism and allowing the reader to draw some parallels. It can be found here at
/networking/integrated.html#pigeon. I do not promote the slaughter of innocent birds
here. I use the often-used “carrier pigeon” scenario to describe TCP/IP based
transmissions of data but extend it to include packet fragmentation. The idea is to use a
pre-exiting model for teaching that the user may know, and extend it to also include the
special case of fragmentation.)
Often some sort of accounting mechanisms are included at this layer to allow a network
administrator to see how many packets, bytes, and various numbers of different sized
packets may have been transmitted. Though it is not an absolute necessity to the
function of this layer, it does often provide statistical data for making charges to parties,
optimizing links and arguing for bigger more expensive pipes, or smaller cheaper pipes,
or bigger slower pipes, or smaller faster pipes. It can be one thing to tell your boss you
need a bigger pipe, and it is entirely another to show your boss you need a bigger pipe.
 2.) Data Link Layer: This layer is responsible for creating what appears to
the layer above (Network Layer) as a channel that is free of detected errors.
Often this is done by packaging bits into cells, or frames, or generically
“packets” with a predictable beginning and end and special calculations
performed on the data known as checksums.
It is necessary for the sender and receiver to agree upon the beginning and ends of
packets so their transmission may be synchronized. The beginning of the packet may be
known by both the sender and receiver based on a shared timing. For example (this one
is not a real-case), a packet should be expected every second with a 1 second pause
after each packet (synchronous). Another example may include using a special
sequence of bits that act as a signature for the beginning of a packet and another
special signature that acts as the end of a packet (asynchronous). With either system it
is possible for the sender and receiver to both know when a packet starts, and stops.
However the bits are packaged, a system is devised and used by the sender and
receiver to allow the receiver to detect a bad cell, frame or generically packet. Often this
is a checksum. A checksum is a special mathematical check performed on the data
being transmitted by the sender’s Data Link Layer. The sender examines the payload it
will be encapsulating in a packet and performs a special mathematical equation on the
payload (or complete packet depending on the defined Data Link Layer protocol). Then it
includes the results of that equation in a part of the packet that is not the payload, such
as the beginning, or header of the packet. When the receiving machine gets the packet,
it looks in the agreed upon location of the packet for the checksum value, and removes it

from the packet. Then the receiver performs the same mathematical equation on the
payload (or remainder of packet, depending upon the protocol’s agreed upon method)
and compares its result with that of the transmitted packet’s checksum value. If both are
different, then there is something wrong with the received packet, and it may be
discarded.
Some examples of some protocols that operate mostly in the Data Link Layer include
Ethernet, TokenRing, ATM, and PPP.
1) Physical Layer
This layer is responsible for moving bits across a shared media between two points.
Agreed upon specification by both parties involved (or all parties involved) on how (1) bit
and off (0) bit should be signaled. For what duration should an amperage, how a voltage
signature proceeds in order for the sender to the receiver foor it to “hear” the signal, and
decode the signal back into the bits transmitted by the sender. If not a wired physical
media, but instead a wireless system, then it would be this layer that specified what
frequency of light or sound would be used and if luminous intensity or amplitude
changed meanings of bits. This layer also specifies how the channel may be used: Full
Duplex, Half Duplex, or (possibly?) Simplex. This layer also deals with conductor
mapping in the case of wired media, and frequency/amplitude/cycle-offsets in the case
of wireless media for mapping Receiving and Transmission.
(b) Describe how to monitor the number of TCP connection failures in Linux and UNIX
Solution:
You can use auditd to monitor system call. It has the ability to log system call based on
function name and its return value. Therefore you can monitor things like socket creating
failure, read failure etc.
In Linux I would like know what is the retry mechanism (how many times and how far
apart). Asking because for a TCP client connect() call I am getting ETIMEDOUT error.
This socket has O_NONBLOCK option and monitored by epoll() for the events.
If someone can point to me where in the code this retry logic is implemented that would
be helpful too. I tried following a bit starting with tcp_v4_connect() from
net/ipv4/tcp_ipv4.c, but lost my way pretty soon..
Tcp:
1478133 active connections openings
121093 passive connection openings
906 failed connection attempts
76814 connection resets received
1 connections established
12674512 segments received
14727243 segments send out
14561 segments retransmited
0 bad segments received.
3603 resets sent

Bcs 052 solved assignment

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