What is presentation layer?

The presentation layer is located at the sixth level of the OSI model, it is responsible for the delivery and
formatting of information to the application layer for further processing or display. This type of service is needed
because different computer architectures use different data representations. In contrast to providing
transparent data transport at the fifth level, the presentation layer handles all issues related to data
presentation and transport, including translation, encryption, and compression.

Presentation layer functions

The actual functions of the presentation layer include the following aspects:

1. Network security and confidentiality management, text compression and packaging, virtual terminal
protocol (VTP).
2. Syntax conversion - The abstract syntax is converted to the transfer syntax, and the other side to achieve
the opposite conversion (transfer syntax will be converted to abstract syntax). Involved in the contents of
the code conversion, character conversion, data format modification, as well as data structure operation
adaptation, data compression, encryption and so on.
3. Grammar negotiation - According to the requirements of the application layer to negotiate the appropriate
choice of context, that is, to determine the transmission syntax and transmission.
4. Connection management - Including the use of the session layer service to establish a connection,
manage data transport and synchronization control over this connection (using the corresponding services
at the session level), and terminate the connection either normally or absently.

Presentation layer of OSI model

It has developed a series of standards such as DP8822, DP8823, DIS6937 / 2 for service, agreement, text
communication symbol. The presentation layer acts as a translator between the application and the network,
mainly addressing the syntax representation of user information, ie, providing formatted representations and
translation data services. Data compression, decompression, encryption, decryption are completed in this
layer.

Presentation layer protocols

Other protocols sometimes considered at this level (though perhaps not strictly adhering to the OSI
model) include:
 Apple Filing Protocol (AFP)
 Independent Computing Architecture (ICA), the Citrix system core protocol
 Lightweight Presentation Protocol (LPP)
 NetWare Core Protocol (NCP)
 Network Data Representation (NDR)
 Telnet (a remote terminal access protocol)
 Tox, The Tox protocol is sometimes regarded as part of both the presentation and application layer
 eXternal Data Representation (XDR)
 25 Packet Assembler/Disassembler Protocol (PAD)

Presentation Layer - OSI Model

The primary goal of this layer is to take care of the syntax and semantics of the information
exchanged between two communicating systems. Presentation layer takes care that the data is
sent in such a way that the receiver will understand the information(data) and will be able to use
the data. Languages(syntax) can be different of the two communicating systems. Under this
condition presentation layer plays a role translator.
In order to make it possible for computers with different data representations to communicate,
the data structures to be exchanged can be defined in an abstract way. The presentation layer
manages these abstract data structures and allows higher-level data structures(eg: banking
records), to be defined and exchanged.

Functions of Presentation Layer

1. Translation: Before being transmitted, information in the form of characters and
numbers should be changed to bit streams. The presentation layer is responsible for
interoperability between encoding methods as different computers use different
encoding methods. It translates data between the formats the network requires and the
format the computer.
2. Encryption: It carries out encryption at the transmitter and decryption at the receiver.
3. Compression: It carries out data compression to reduce the bandwidth of the data to be
transmitted. The primary role of Data compression is to reduce the number of bits to be
0transmitted. It is important in transmitting multimedia such as audio, video, text etc.

Question no 2
Data compression is the function of presentation layer in OSI reference model.
Compression is often used to maximize the use of bandwidth across a network or to
optimize disk space when saving data.
There are two general types of compression algorithms:

1. Lossless compression
2. Lossy compression

Lossless Compression
Lossless compression compresses the data in such0 a way that when data is
decompressed it is exactly the same as it was before compression i.e. there is no loss of
data.
A lossless compression is used to compress file data such as executable code, text files,
and numeric data, because programs that process such file data cannot tolerate mistakes
in the data.
Lossless compression will typically not compress file as much as lossy compression
techniques and may take more processing power to accomplish the compression.

Lossless Compression Algorithms

The various algorithms used to implement lossless data compression are :

1. Run length encoding

2. Differential pulse code modulation
3. Dictionary based encoding
1. Run length encoding
• This method replaces the consecutive occurrences of a given symbol with only one
copy of the symbol along with a count of how many times that symbol occurs. Hence the
names ‘run length'.
• For example, the string AAABBCDDDD would be encoded as 3A2BIC4D.
• A real life example where run-length encoding is quite effective is the fax machine.
Most faxes are white sheets with the occasional black text. So, a run-length encoding
scheme can take each line and transmit a code for while then the number of pixels, then
the code for black and the number of pixels and so on.
• This method of compression must be used carefully. If there is not a lot of repetition in
the data then it is possible the run length encoding scheme would actually increase the
size of a file.
2. Differential pulse code modulation
• In this method first a reference symbol is placed. Then for each symbol in the data, we
place the difference between that symbol and the reference symbol used.
• For example, using symbol A as reference symbol, the string AAABBC DDDD would be
encoded as AOOOl123333, since A is the same as reference symbol, B has a difference of
1 from the reference symbol and so on.
3. Dictionary based encoding
• One of the best known dictionary based encoding algorithms is Lempel-Ziv (LZ)
compression algorithm.
• This method is also known as substitution coder.
• In this method, a dictionary (table) of variable length strings (common phrases) is
built.
• This dictionary contains almost every string that is expected to occur in data.
• When any of these strings occur in the data, then they are replaced with the
corresponding index to the dictionary.
• In this method, instead of working with individual characters in text data, we treat
each word as a string and output the index in the dictionary for that word.
• For example, let us say that the word "compression" has the index 4978 in one
particular dictionary; it is the 4978th word is usr/share/dict/words. To compress a body
of text, each time the string "compression" appears, it would be replaced by 4978.

Lossy Compression
Lossy compression is the one that does not promise that the data received is exactly the
same as data send i.e. the data may be lost.
This is because a lossy algorithm removes information that it cannot later restore.
Lossy algorithms are used to compress still images, video and audio.
Lossy algorithms typically achieve much better compression ratios than the lossless
algorithms.
Audio Compression
• Audio compression is used for speech or music.
• For speech, we need to compress a 64-KHz digitized signal; For music, we need to
compress a 1.411.MHz signal

• Two types of techniques are used for audio compression:

1. Predictive encoding
2. Perceptual encoding

Predictive encoding
• In predictive encoding, the differences between the samples are encoded instead of
encoding all the sampled values.
• This type of compression is normally used for speech.
• Several standards have been defined such as GSM (13 kbps), G. 729 (8 kbps), and
G.723.3 (6.4 or 5.3 kbps).
Perceptual encoding
• Perceptual encoding scheme is used to create a CD-quality audio that requires a
transmission bandwidth of 1.411 Mbps.
• MP3 (MPEG audio layer 3), a part of MPEG standard uses this perceptual encoding.
• Perceptual encoding is based on the science of psychoacoustics, a study of how people
perceive sound.
• The perceptual encoding exploits certain flaws in the human auditory system to encode
a signal in such a way that it sounds the same to a human listener, even if it looks quite
different on an oscilloscope.
• The key property of perceptual coding is that some sounds can mask other sound. For
example, imagine that you are broadcasting a live flute concert and all of a sudden
someone starts striking a hammer on a metal sheet. You will not be able to hear the flute
any more. Its sound has been masked by the hammer.
• Such a technique explained above is called frequency masking-the ability of a loud
sound in one frequency band to hide a softer sound in another frequency band that
would have been audible in the absence of the loud sound.
• Masking can also be done on the basis of time. For example: Even if the hammer is not
striking on a metal sheet, the flute will be inaudible for a short period of time because
the ears turn down its gain when they start and take a finite time to turn up again.
• Thus, a loud sound can numb our ears for a short time even after the sound has
stopped. This effect is called temporal masking.

MP3
• MP3 uses these two phenomena, i.e. frequency masking and temporal masking to
compress audio signals.
• In such a system, the technique analyzes and divides the spectrum into several groups.
Zero bits are allocated to the frequency ranges that are totally masked.
• A small number of bits are allocated to the frequency ranges that are partially masked.
• A larger number. of bits are allocated to the frequency ranges that are not masked.
• Based on the range of frequencies in the original analog audio, MP3 produces three
data rates: 96kbps, 128 kbps and 160 kbps.

Question Number 3
Session Layer - OSI Model
The Session Layer allows users on different machines to establish active communication sessions
between them.
It's main aim is to establish, maintain and synchronize the interaction between communicating
systems. Session layer manages and synchronize the conversation between two different
applications. In Session layer, streams of data are marked and are resynchronized properly, so
that the ends of the messages are not cut prematurely and data loss is avoided.

Functions of Session Layer

1. Dialog Control : This layer allows two systems to start communication with each other in half-
duplex or full-duplex.
2. Token Management: This layer prevents two parties from attempting the same critical
operation at the same time.
3. Synchronization : This layer allows a process to add checkpoints which are considered as
synchronization points into stream of data. Example: If a system is sending a file of 800 pages,
adding checkpoints after every 50 pages is recommended. This ensures that 50 page unit is
successfully received and acknowledged. This is beneficial at the time of crash as if a crash
happens at page number 110; there is no need to retransmit 1 to100 pages.
Design Issues with Session Layer

 To allow machines to establish sessions between them in a seamless

fashion.
 Provide enhanced services to the user.
 To manage dialog control.
 To provide services such as Token management and Synchronization.

Remote Procedure Call (RPC)

Computer ScienceMCAOperating System

A remote procedure call is an interprocess communication technique that is used for

client-server based applications. It is also known as a subroutine call or a function call.
A client has a request message that the RPC translates and sends to the server. This
request may be a procedure or a function call to a remote server. When the server
receives the request, it sends the required response back to the client. The client is
blocked while the server is processing the call and only -resumed execution after the
server is finished.
The sequence of events in a remote procedure call are given as follows:

 The client stub is called by the client.

 The client stub makes a system call to send the message to the server and puts the parameters in the
message.
 The message is sent from the client to the server by the client’s operating system.
 The message is passed to the server stub by the server operating system.
 The parameters are removed from the message by the server stub.
 Then, the server procedure is called by the server stub.

A diagram that demonstrates this is as follows:

Advantages of Remote Procedure Call

Some of the advantages of RPC are as follows:

 Remote procedure calls support process oriented and thread oriented models.
 The internal message passing mechanism of RPC is hidden from the user.
 The effort to re-write and re-develop the code is minimum in remote procedure calls.
 Remote procedure calls can be used in distributed environment as well as the local environment.
 Many of the protocol layers are omitted by RPC to improve performance.

Disadvantages of Remote Procedure Call

Some of the disadvantages of RPC are as follows:
  The remote procedure call is a concept that can be implemented in different ways. It is not a standard.
 There is no flexibility in RPC for hardware architecture. It is only interaction based.
 There is an increase in costs because of remote procedure call.

Question number 4
Multimedia over Internet

There are other ways to transmit multimedia data, like dedicated links, cables and
ATM. However, the idea of running multimedia over Internet is extremely attractive.

Dedicated links and cables are not practical because they require special installation
and new software. Without an existing technology like LAN,WAN, the software
development will be extremely expensive. ATM was said to be the ultimate solution
for multimedia because it supports very high bandwidth, is connection-oriented and
can tailor different level of quality of service to different type of applications. But at
this moment, very few users have ATM networks reaching their organization, even
fewer have ATM connections to their desktops.

On the other hand, the Internet is growing exponentially. The well established LAN
and WAN technologies based on IP protocol suite connect bigger and bigger networks
all over the world to the Internet. In fact, Internet has become the platform of most
networking activities. This is the primary reason to develop multimedia protocols over
Internet. Another benefit of running multimedia over IP is that users can have
integrated data and multimedia service over one single network, without investing on
another network hardware and building the interface between two networks.

At current time, IP and Ethernet seem to be more favored in the desktops and LANs,
with ATM in wide area networks.

As a shared datagram network, Internet is not naturally suitable for real-time traffic.
To run multimedia over Internet, several issues must be solved. First, multimedia
means extremely dense data and heavy traffic. The hardware has to provide enough
bandwidth.

Second, multimedia applications are usually related to multicast, i.e., the same data
stream, not multiple copies, is sent a group of receivers. For example, in video
conference, the video data need to be sent to all participants at the same time. Live
video can be sent to thousands of recipients. The protocols designed for multimedia
applications must take into account multicast in order to reduce the traffic.
Third, the price tag attached shared network resources is unpredictable availability.
But real-time applications require guaranteed bandwidth when the transmission takes
place. So there must be some mechanisms for real-time applications to reserve
resources along the transmission path.

Fourth, Internet is a packet-switching datagram network where packets are routed

independently across shared networks. The current technologies cannot guarantee that
real-time data will reach the destination without being jumbled and jerky. Some new
transport protocols must be used to take care of the timing issues so that audio and
video data can be played back continuously with correct timing and synchronization.

Fifth, there should be some standard operations for applications to manage the
delivery and present the multimedia data.

The answers to the above issues are the protocols that will be discussed in this paper.

The solution

The Internet carries all types of traffic, each type has different characteristics and
requirements. For example, a file transfer application requires that some quantity of
data is transferred in an acceptable amount of time, while Internet telephony requires
that most packets get to the receiver in less than 0.3 seconds. If enough bandwidth is
available, best-effort service fulfills all of these requirements. When resources are
scarce, however, real-time traffic will suffer from the congestion.

The solution for multimedia over IP is to classify all traffic, allocate priority for
different applications and make reservations. The Integrated Services working group
in the IETF (Internet Engineering Task Force) developed an enhanced Internet service
model called Integrated Services that includes best-effort service and real-time
service, see RFC 1633. The real-time service will enable IP networks to provide
quality of service to multimedia applications. Resource ReServation Protocol (RSVP),
together with Real-time Transport Protocol (RTP), Real-Time Control Protocol
(RTCP), Real-Time Streaming Protocol (RTSP), provides a working foundation for
real-time services. Integrated Services allows applications to configure and manage a
single infrastructure for multimedia applications and traditional applications. It is a
comprehensive approach to provide applications with the type of service they need
and in the quality they choose.

This paper, which takes many materials from corresponding Internet Drafts and RFCs,
is a detailed review of the four protocols.
Question Number 5
Network security
Network security consists of the policies and practices adopted to prevent and
monitor unauthorized access, misuse, modification, or denial of a computer network and network-
accessible resources. Network security involves the authorization of access to data in a network,
which is controlled by the network administrator. Users choose or are assigned an ID and password
or other authenticating information that allows them access to information and programs within their
authority. Network security covers a variety of computer networks, both public and private, that are
used in everyday jobs; conducting transactions and communications among businesses,
government agencies and individuals. Networks can be private, such as within a company, and
others which might be open to public access. Network security is involved in organizations,
enterprises, and other types of institutions. It does as its title explains: it secures the network, as well
as protecting and overseeing operations being done. The most common and simple way of
protecting a network resource is by assigning it a unique name and a corresponding password.

DES Algorithm
The DES algorithm is the most popular security algorithm. It's a symmetric
algorithm, which means that the same keys are used to encrypt/decrypt sensitive
data. Key length is 8 byte (64 bit). So, to encrypt/decrypt data, the DES algorithm
uses an 8-byte key, but 1 byte (8 bit) for parity checking. It's a block cipher
algorithm — that's why the data block size of DES algorithm is 64 bit. To
encrypt/decrypt data, the DES algorithm uses the Feistel structure. So, it uses some
round to encrypt/decrypt data. Though data block size is 64 bit, the number of
rounds will be 16 rounds. So, it will use different subkeys for each round. so the
number of subkeys will be 16 subkeys. For more info on the process of finding
subkeys, you can learn more here. However, for this tutorial, we will be skipping this
part.

Modes of Operation
There are different modes of operation when using the DES algorithm. If each 64 bit
is encrypted or decrypted independently, then this mode is ECB.
 If each 64-bit data is dependent on the previous one, then this mode is called CBC
or CFB mode.

Here, in CBC mode, we can see that there is a new term called the Initial Vector (IV),
which will be the same as the data block size. The data block will be XOR with IV and
then encrypted with the key. Then, the output ciphertext uses the IV of the next
block. The reverse process is used during decryption. So, we can say that the
encryption of the current block is dependent on the encryption of the previous data
block. Therefore, it's more secure than that of ECB.
In CFB mode, the intial vector is encrypted with a key, and then, the data block will
XOR with encrypted output. Ciphertext, again, goes to it as an input for encryption
function and again XOR with next plaintext block and so on.

Terms to Memorize
 The DES algorithm is a block cipher algorithm
 The data block size of the DES algorithm is 64 bit (8 bytes)
 Key size is 64 bit (8 bytes), but 1 byte is used for parity, so the actual key size
is 56 bit
 Without ECB, for others, IV is mandatory
 There are different types of padding algorithms. If you use NoPadding, the
data size should be a multiple of 8.

DNS - Domain Name System

(1) Short for Domain Name System (or Service or Server), an Internet service that translates domain names into IP addresses.
Because domain names are alphabetic, they're easier to remember. The Internet however, is really based on IP addresses. Every
time you use a domain name, therefore, a DNS service must translate the name into the corresponding IP address. For example,
the domain name www.example.com might translate to 198.105.232.4.
The DNS system is, in fact, its own network. If one DNS server doesn't know how to translate a particular domain name, it asks
another one, and so on, until the correct IP address is returned.
(2) Short for digital nervous system,a term coined by Bill Gates to describe a network of personal computers that make it easier to
obtain and understand information.
What Is A Domain Name Server (DNS) And How Does It Work
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Domain Name Servers (DNS) are the Internet's equivalent of a phone book. They
maintain a directory of domain names and translate them to Internet Protocol (IP)
addresses.

This is necessary because, although domain names are easy for people to remember,
computers or machines, access websites based on IP addresses.

Information from all the domain name servers across the Internet are gathered
together and housed at the Central Registry. Host companies and Internet Service
Providers interact with the Central Registry on a regular schedule to get updated DNS
information.

When you type in a web address, e.g., www.jimsbikes.com, your Internet Service
Provider views the DNS associated with the domain name, translates it into a machine
friendly IP address (for example 216.168.224.70 is the IP for jimsbikes.com) and
directs your Internet connection to the correct website.

After you register a new domain name or when you update the DNS servers on your
domain name, it usually takes about 12-36 hours for the domain name servers world-
wide to be updated and able to access the information. This 36-hour period is referred
to as propagation.

Domain name resolution, how it works?

In the world of the Internet, the addresses don’t contain streets and cities, they have
a numbers and symbols. There are two types of addresses: IPv4 and IPv6. In order to
enter into a particular web site, the user needs to get its IP. Instead of remembering
all of the IPs of every website, we simply need to remember the domain name. The
domain name is usually not hard to remember, and this makes it easier. When the
user types the name of the website, the process of the domain name resolution
starts. The browser needs to get the IP and sends queries to the name servers. This
process is involving domain name resolvers. The first answer that your browser will
get is the root server, then the TLD (top level domain). The servers with the TLD of
the website you want to visit (com, net or another) will refer your queries to the next
step in searching authoritative servers who know the complete IP. The will resolve
the domain name.