CSC 501: COMPUTER NETWORKS II

Course Content
CSC 501(2 Units) – Computer Networks II
Architecture of computer networks and network protocols, protocol layering, reliable transmission,
congestion control, flow control, naming and addressing, unicast and multicast routing, network security,
network performance, widely used protocols such as Ethernet, wireless LANs, IP, and HTTP. Bus structures
and loop systems computer network. Examples and design consideration, data switching principles,
broadcast techniques, and network structure for packet switching, protocols, and description of network e.g.
ARPANET, DSC etc. Prerequisite CSC 206 30h (L) 0h(T) 45h(P).

Part 1: Computer Network- Brief Recap

 A computer network is a system in which multiple computers are connected to each other to
share information and resources.
 The physical connection between networked computing devices is established using either
cable media or wireless media.
 The best-known computer network is the Internet.

Figure 1: Computer Network

Advantages of Computer Networks

 Filesharing
The major advantage of a computer network is that is allows file sharing and remote file access.
A person sitting at one workstation that is connected to a network can easily see files present on
another workstation, provided he is authorized to do so.
 Resource sharing
All computers in the network can share resources such as printers, fax machines, modems, and
scanners.
 Better connectivity and communications
It allows users to connect and communicate with each other easily. Various communication
applications included e-mail and groupware are used. Through e-mail, members of a network can send
message and ensure safe delivery of data to other members, even in their absence.
 Internet access
Computer networks provide internet service over the entire network. Every single computer attached
to the network can experience the high speed internet.
 Entertainment

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 Many games and other means of entertainment are easily available on the internet.
Furthermore, Local Area Networks (LANs) offers and facilitates other ways of enjoyments, such as
many players are connected through LAN and play a particular game with each other from remote
location.
 Inexpensive system
Sharedresources mean reduction in hardware costs. Shared files mean reduction in
memoryrequirement, which indirectly means reduction in file storage expenses. A particular
softwarecan be installed only once on the server and made available across all connectedcomputers
atonce. This saves the expense of buying and installing the same software as many times for asmany
users.
 Flexible access
A user can log on to a computer anywhere on the network and access his files. This offers flexibility to
the user as to where he should be during the course of his routine.
 Instant and multiple access
Computer networks are multiply processed . many of the users can access the same information at
the same time. Immediate commands such as printing commands can be made with the help of
computer networks.

Disadvantages of Computer Networks

 Lack of data security and privacy
Because there would be a huge number of people who would be using a computer network toget and
share some of their files and resources, a certain user’s securitywould be always at risk.
There might even be illegal activities that would occur, which you need to be careful about andaware
of.
 Presence of computer virusesand malwares
If even one computer on a network gets affected by a virus, there is a possible threat for theother
systems getting affected too. Viruses can spread on a network easily, because of the interconnectivity
of workstations. Moreover, multiple systems with common resources are theperfect breeding ground
for viruses that multiply.
 Lack of Independence
Since most networks have a centralized server and dependent clients, the client users lack any
freedom whatsoever. Centralized decision making can sometimes hinder how a client user wants to
use his own computer.
 Lack of Robustness
As previously stated, if a computer network’s main server breaks down, the entire system would
become useless. Also, if it has a bridging device or a central linking server that fails, the entirenetwork
would also come to a standstill.
 Need an efficient handler or technical personnel
For a computer network to work efficiently and optimally, it requires high technical skills and know-
how of its operations and administration. A person just having basic skills cannot do thisjob. Take note
that the responsibility to handle such a system is high, as allotting permissionsand passwords can be
daunting. Similarly, network configuration and connection is very tediousand cannot be done by an
average technician who does not have advanced knowledge.

Use(Applications)of Computer Networks

 Financial services
Nowadays, almost all the financial services depend on the computer network. You can accessthe
financial services across the world. For example, a user can transfer money from one placeto another
by using the electronic fund transfer feature. You can use networking in variousfinancial areas such as
ATM, foreign exchange and credit history search.
 Business
Nowadays, most of the works of businesses are done over the computers. To exchange the dataand
ideas, you need an effective data and resources sharing features. To do this, you need toconnect

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 the computer with each other through a network. For example, a person of onedepartment of an
organization can share or access the electronic data of other departmentthrough network.
 Email services
A computer network provides you the facility to send or receive mails across the globe in
fewseconds.
 Mobile applications
By using the mobile applications, such as cellular or wireless phones, you can
communicate(exchange your views and ideas) with one other.
 Directory services
It provides you the facility to store files on a centralized location to increase the speed of
searchoperation worldwide.
 Teleconferencing
It contains voice conferencing and video conferencing which are based in networking.
Inteleconferencing the participants need not to be presented at the same location.

Topologies (Network Topologies) – LAN

 Network Topology is the schematic description of a network arrangement, connecting various nodes
(sender and receiver) throughlines of connection.
 A Network Topology is the arrangement with which computer systems or network devices
areconnected to each other.

Types of LAN network topologies:

1. Bus 2. Ring 3. Star. 4. Mesh 5. Tree 6. Hybrid

Bus Topology
 Bus topology is anetwork type in whichevery computer and network device is connected to single
cable.

Features:
 It transmits data only in one direction.
 Every device is connected to a single cable.

Advantages:
 It is cost effective (cheaper).
 Cable required is least compared to other network topology.
 Used in small networks.
 It is easy to understand.
 Easy to expand joining two cables together.

Disadvantages:
 Cables fails then whole network fails.
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  If network traffic is heavy or nodes are more,
 The performance of the network decreases.
 Cable has a limited length.

Ring Topology
 It is called ring topology because it forms a ring as each computer is connected to another computer,
with the last one connected to the first. Exactly two neighbours for each device.

Features:
 Anumber of repeaters are used and the transmission is unidirectional.
 Date is transferred in a sequential manner that is bit by bit.
Advantages:
 Transmitting network is not affected by high traffic or by adding more nodes, as only the nodes having
tokens can transmit data.
 Cheap to install and expand.
Disadvantages:
 Troubleshooting is difficult in ring topology.
 Adding or deleting the computers disturbs the network activity.
 Failure of one computer disturbs the whole network.

Star Topology
 In this type of topology all the computers are connected to a single hub through a cable. Thishub is the
central node and all others nodes are connected to the central node.

Features:

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  Every node has its own dedicated connection to the hub.
 Acts as a repeater for dataflow.
 Can be used with twisted pair, Optical Fibre or coaxial cable.
Advantages:
 Fast performance with few nodes and low network traffic.
 Hub can be upgraded easily.
 Easy to troubleshoot.
 Easy to setup and modify.
 Only that node is affected which hasfailed rest of the nodes can work smoothly.
Disadvantages:
 Cost of installation is high.
 Expensive to use.
 If the hub is affected then the whole network is stopped because all the nodes depend on thehub.
 Performance is based on the.

Mesh Topology
 It is apoint-to-point connection to other nodes or devices.
 Traffic is carried only between two devices or nodes to which it is connected.

Features:
 Fully connected.
 Robust.
 Not flexible.
Advantages:
 Each connection can carry its own data load.
 It is robust.
 Fault is diagnosed easily.
 Provides security and privacy.
Disadvantages:
 Installation and configuration is difficult.
 Cabling cost is more.
 Bulk wiring is required.

Tree Topology
 It has a root node and all other nodes are connected to it forming a hierarchy.
 It is also called hierarchical topology.

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  It should at least have three levels to the hierarchy.

Features:

 Ideal if workstations are located in groups.

 Used in Wide AreaNetwork.

Advantages:

 Extension of bus and star topologies.

 Expansion of nodes is possible and easy.
 Easily managed and maintained.
 Error detection is easily done.
Disadvantages:
 Heavily cabled.
 Costly.
 If more nodes are added maintenance is difficult.
 Central hub failsthennetwork fails.

Hybrid Topology
Features:
 A network structure whose design contains more than one topology is said to be hybrid topology
 For example if in an office in one department ring topology is used and in another star topology is
used, connecting these topologies will result in Hybrid Topology (ring topology and star topology).
 It is a combination of two ormoretopologies
 Inherits the advantages and disadvantages of the topologies included
Advantages:
 Reliable aserror detecting and trouble shooting is easy.
 Scalable as size can be increased easily.
 Flexible.

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 Disadvantages:
 Complex in design.
 Costly.

Types of Computer Networks

LAN (Local Area Network)
 It isprivately-owned networks within a single building or campus of up to a few kilometers insize.
 They are widely usedto connect personal computers and workstations in company offices andfactories
to share resources (e.g., printers) and exchange information.
 LANs are easy to design and troubleshoot
 In LAN, all the machines are connected to a single cable.
 Different types of topologies such as Bus, Ring, Star and Tree are used.
 The datatransferrates for LANis up to 10Gbits/s.
 They transfer data at high speeds.High transmission rate are possible in LAN because of theshort
distance between variouscomputer networks.
 They exist in a limited geographical area.
Advantages
 LAN transfers data at high speed.
 LAN technology is generally less expensive.

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 Figure 2: Local Area Network

MAN (Metropolitan Area Network)

 MAN is a larger version of LAN whichcovers an area that is larger than the covered by LAN butsmaller
than the area covered by WAN.
 Ametropolitan area networkorMANcovers a city. The best-known example of a MAN is thecable
television network available in many cities.
 MAN connects two ormore LANs.
 At first, the companies began jumping into the business, getting contracts from city governments to
wire up an entire city.
 The next step was television programming and even entire channels designed for cable only.

Figure3: Metropolitan Area Network

WAN (Wide Area Network)

 WANspans a large geographical area, often a country orregion.
 WAN links differentmetropolitan’s countries and national boundaries there by enabling easy
communication.
 t may be located entirely with in a state or a country or it may be interconnected around the world.
 It contains a collection of machines intended for running user (i.e. application) programs.
 We will follow traditional usage and call these machineshosts.
 The communication between different users of WAN is established using leased telephone lines or
satellite links and similar channels.

Figure4: Wide Area Network

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Internet
 The internet is a type of world-wide computer network.
 he internet is the collection of infinite numbers of connected computers that are spread across the
world.
 We can also say that, the Internet is a computer network that interconnects hundreds of millions of
computing devices throughout the world.
 It is established as the largest network and sometimes called network of networks that consists of
numerous academic, business and government networks, which together carry various information.
 Internet is a global computer network providing a variety of information and communication facilities,
consisting of interconnected networks using standardized communication protocols.
 When two computers are connected over the Internet, they can send and receive all kinds of
information such as text, graphics, voice, video, and computer programs.

F
i
F
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Figure 5: Some pieces of the Internet

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 PART 2: NETWORK ARCHITECTURE, PROTOCOLS, &
PROTOCOL LAYERING

Network Architecture & The Network Edge

The Network Edge
 It defines those computers of the network used at the edge (end) of the network. These computers
are known as hosts or end system.
 Host can be classified into the following two types: Clients and Servers.
Clients:
 Refer to the computer systems that request servers for the completion of a task.
 The clients are generally called desktop PCs or workstations.

Servers:
 Refer to the computer systems that receive requests from the clients and process them. After the
processing is complete, the servers send a reply to the clients who sent the request.
 The concept of clients and servers is essentialin the network design, i.e. Network Architecture

Network Architecture
1. It is basically the physical and logical design which refers to the software, hardware, protocols and the
media of transmission of data. It refers to how computers are organized and how tasks are allocated
among these computers.
2. It is the design of a communication network. It is a framework for the specification of a network's physical
components and their functional organization and configuration, its operational principles and procedures,
as well as data formats used.
3. Network Architecture is the complete framework of an organization's computer network. The diagram of
the network architecture provides a full picture of the established network with detailed view of all the
resources accessible. It includes hardware components used for communication, cabling and device types,
network layout and topologies, physical and wireless connections, implemented areas and future plans. In
addition, the software rules and protocols also constitute to the network architecture. This architecture is
always designed by a network manager/administrator with coordination of network engineers and
other design engineers.

 Designing a computer network to meet various network requirements involves a lot of

complexities.The Network Architecture shows the general blueprints; it guides the design and
implementation of the Computer network.
 The network architecture of the Internet is predominantly expressed by its use of the Internet Protocol
Suite, rather than a specific model for interconnecting networks or nodes in the network, or the usage of
specific types of hardware links

Classification of Network Architecture

Network architectures are often classified into two broad categories:
1. Client-server architectures
2. Peer-to-Peer architectures

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 Figure 7: Network Edge-Client/Server Network and Peer to Peer

PEER-TO-PEER ARCHITECTURE
 In Peer-to-Peer network (architecture), a group of computers is connected together so that users can
share resources and information.
 There is no central location (server) for authenticating users, storing files, or accessing resources.
 This means that users must remember which computers in the workgroup have the sharedresource or
information that they want to access.
 In Peer to Peer architecture task is allocated to every device on a network.
 All of them are considered equal as there is no hierarchy in this network.
 All have the same abilities to use the resources available on this network;
 It is mostly used in file sharing
 Each computer has its running software that allows communication between all the other computers.
 First P2P network was Napster.
 An example of a peer-to-peer architecture is a system of intelligent agents that collaborate to collect,
filter, and correlate information.

Advantages of a peer-to-peer network

i. The network implementation is quite cheap, since it does not require a dedicated server.
ii. If one computer stops working, the other computers connected to the network will continue working.
iii. Installation and setup is quite painless because of the built-in support in modern operating systems,
i.e. It is easy to setup.
iv. The resources of a peer can be shared with other peers very easily in the network.

Disadvantages of a peer-to-peer network

i. There is no central backup of files and folders. Data backups are to be done to each individual
computer.
ii. As the numbers of computers increases on a P2P network... performance, security, and access
becomes a major headache. The speed of the network decreases due to heavy usage.
iii. It is not easy to keep track of information on each computer.
iv. Network and data security are weak.
v. Lack of central device posses a problem in this network.

CLIENT/SERVER ARCHITECTURE
 A client/server network is a system where one or more computers called clients connect to a central
computer named as server to share or use resources.
 The client requests a service from server, which may include running an application, querying database,
printing a document, performing a backup or recovery procedure. The request made by the client is
handled by server.
 A client/server network is that in which the files and resources are centralized. This means that the server
can hold them and other computers (Client) can access them.
 In figure 7 above, we have several clients that wants access from sever.
 Server is a central device that provides services and data to the clients requesting such from the server.
 Servers are basically computer with fast speed and large memory.
 Clients provides services to the end user by taking or consuming services from server.

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Advantages of a Network client/server Architecture
i. Resources and data security are controlled through the server.
ii. The client/server network is not restricted to a small number of computers, i.e. it can be scaled up to
accomdate hundeds and thousands of computers.
iii. Server can be accessed anywhere and across multiple platforms.
iv. The server system holds the shared files.
v. The server system can be scheduled to take the file backups automatically.
vi. Network access is provided only to authorize users through user security at the server.
vii. The server system is a kind of central repository for sharing printer with clients.
viii. Internet access, e-mail routing and such other networking tasks are quite easily managed by the
server.
ix. The software applications shared by the server are accessible to the clients

Disadvantages of a client/server network

i. The implementation of the network is quite expensive, due to the need of a server as well as
networking devices such as hubs, routers, and switches.
ii. A NOS (Network Operating System) is essential and contributes to the cost.
iii. If and when the server goes down, the entire network crashes.
iv. Technical staff is needed to maintain and ensure the network functions efficiently.
v. There may be congestion if more than one client requests for a service at the sametime.

Sometimes layered (“tiered”) and peer-to-peer architectures are combined, where the nodes in particular
layers are in peer-to-peer relationships. For example, a multi-tiered architecture might include an enterprise
management layer, consisting of peer nodes for such things as network management, event management,
database management, Web sever management, and workload balancing.

Network Applications OR Principles of Network Applications

 A Network application is any application running on one host and provides acommunication
to another applicationrunning on a different host.
 At the core of network application development is writing programs that run on different end
systems and communicate with each other over the network.
 In the Web application there are two distinct programs that communicate with each other:
the browser program running in the user's host; and the Web server program running in the
Web server host.
 Examples of network applications are: (i) e-mail (ii) the web (iii) text messaging (iv) remote
login (v) P2P file sharing (vi) multi-user network games (vii) streaming stored video (YouTube)
(viii) voice over IP (e.g., Skype) (ix) real-time video conferencing (x) social networking

Network Application Architectures

 There are two possible structure of applications
1. Client-Server architecture
2. P2P (Peer to Peer) architecture

Client-Server architecture
 In client-server architecture, there is an always a host, called the server, which provides services
requested from many other hosts, called clients. There is usually a 1:M relationship between servers
and clients. It is also possible to have a 1:M relationship between one or more clients and servers. In
this model, a system is decomposed into client and server processors or processes.
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  A classic example is the Web application for which an always-on Web server services requests from
browsers running on client hosts. When a Web server receives a request for an object from a client
host, it responds by sending the requested object to the client host.
 Note that with the client-server architecture, clients do not directly communicate with each other; for
example, in the Web application, two browsers do not directly communicate with each other.
 Another characteristic of the client-server architecture is that the server has a fixed, well-known
address, called an IP address. Becausethe server is always on, a client can always contact the server by
sending a packet to the server’s IP address.
 Some of the better-known applications with client-server architecture include the Web, FTP, Telnet, and e-
mail.
 Client-server architectures are commonly organized into layers referred to as “tiers”.

Tiered Architectures
 Two-tier architecture. The system architecture consists of a data server layer and an application client
layer. Data access computation is associated with the data server layer, and the user interface is
associated with the client application layer. If most of the application logic is associated with the client
application logic, it is sometimes referred to as a “fat client.” If it is associated with the data access server,
the application client layer is sometimes referred to as a “thin client.”
 Three-tier architecture. The system architecture consists of data server layer, an application server layer
and a client application layer. The application server layer facilitates the separation of application logic
from presentation, and promotes distributed processing.
 Multi-tier (n-tier) architecture. The system architecture is a superset of a three-tier architecture, and
includes additional layers for data and/or application servers.

P2P architecture
 In P2P architecture, there is no dedicated server.
 Pairs of hosts, called peers communicate directly with each other.
 Because the peers communicate without passing through a dedicated server, the architecture is called
peer-to-peer.

Client-Server architecture P2P (Peer to Peer) architecture

Many of today’s most popular and traffic-intensive applications are based on P2P architectures.
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PROTOCOLS AND PROTOCOL LAYERING

Protocol
 A protocol is a set of rules that governs (manages) data communications.
 Protocols defines methods of communication, how to communicate, when to communicate etc.
 A protocol is an agreement between the communicating parties on how communication is toproceed.
 Important elements of protocols are
1. Syntax 2. Semantics 3. Timing
Syntax:-Syntax means format of data orthe structure of how it is presented e.g. first eight bits are for sender
address, next eight bits are for receiver address and rest of the bits for message data.
Semantics:-Semantics is the meaning of each section of bits e.g. the address bit means the route of
transmission or final destination of message.
Timing:-Timing means, at what time data can be sent and how fast data can be sent.
Some protocols also support message acknowledgement and data compression designed for reliable
and/or high-performance network communication.
Example: HTTP, IP, FTP etc…

Figure6: A human protocol and a computer network protocol

PROTOCOLS LAYERS AND THEIR SERVICE MODELS: (OSI & TCP/IP)

OSI Layer Architecture

 OSI model is based on a proposal developed by the International Standards Organization (ISO) as a first
step toward international standardization of the protocols used in the various layers.
 It was revised in 1995.
 The model is called the OSI (Open Systems Interconnection) Reference Model because it deals with
connecting open systems—that is, systems that are open for communication with other systems.
 The OSI model has seven layers.
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Figure 2.2 Seven layers of the OSI model

 Each Layer has a “Peer-to-Peer” protocol that seems to represent (and carry out) the rest of
the network task, yet it does only a specific part and delegate the rest to the layer beneath it,
via its interface, which that defines the services that is provided to the layer above (and
beneath) it.

Figure 2.3: The interaction between layers in the OSI model

 “Encapsulation”: Each layer has its own PDU (Protocol Data Unit) that’s passes (as a
parameter) to the layer beneath, which in turn adds a “ header ”(at layer 2 also adds trailer”)
before assign it to the next layer (except the physical layer).
 Why “header” and “trailer”?
 Physical movement of information PDU is “vertical” yet the user thinks (At each peer –to-
peer) layer that info moves” horizontal” (pipe)

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 Figure 2.4 An exchange using the OSI model

Physical Layer

Figure 2.5: Physical layer

 The physical layer, the lowest layer of the OSI model, is concerned with the transmission and reception
of the unstructured raw bit stream over a physical medium/communication channel.
 It describes the electrical/optical, mechanical, and functional interfaces to the physical medium, and
carries the signals for all of the higher layers. The design issues have to do with making sure that when
one side sends a 1 bit, it is received by the other side as a 1 bit, not as a 0 bit.
 Typical questions here are
o how many volts should be used to represent a 1 and how many for a 0,
o how many nanoseconds a bit lasts,
o whether transmission may proceed simultaneously in both directions,
o how the initial connection is established and how it is torn down when both sides are finished,
o How many pins the network connector has and what each pin is used for.
 The design issues here largely deal with mechanical, electrical, and timing interfaces, and the physical
transmission medium, which lies below the physical layer.

To summarize, the physical layer provides the following services or take into account the following
factors:
 Line Coding /Data (Signal) encoding: How are the bits 0 and 1 to be represented? Data
enconding modifies the simple digital signal pattern (1s and 0s) used by the PC to better
accommodate the characteristics of the physical medium, and to aid in bit and frame
synchronization. It allows data to be sent by hardware devices that are optimized for digital
communications that may have discreet timing on the transmission link
 Transmission technique (Signal type): determines whether the encoded bits will be

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 transmitted by baseband (digital) or broadband (analog) signaling.
 Physical medium transmission: transmits bits as electrical or optical signals appropriate for the
physical medium. What is the medium used, and what are its properties?
 Modulates the process of converting a signal from one form to another so that it can be physically
transmitted over a communication channel
 Start-stop signaling and flow control in asynchronous serial communication
 Circuit switching and multiplexing hardware control of multiplexed digital signals
 Carrier sensing and collision detection, whereby the physical layer detects carrier availability and
avoids the congestion problems caused by undeliverable packets
 Signal equalization to ensure reliable connections and facilitate multiplexing
 Forward error correction/channel coding such as error correction code
 Bit interleaving to improve error correction
 Bit synchronization: Is the transmission asynchronous or synchronous serial
communication?
 Transmission type: Is the transmission serial or parallel?
 Transmission mode control: Is the transmission simplex, half-duplex or full duplex?
 Topology: What is the topology (mesh, star ring, bus or hybrid) used?
 Multiplexing: Is multiplexing used, and if so, what is its type (FDM, TDM)?
 Interface: How are the two closely linked devices connected?
 Bandwidth: Which of the two baseband or broadband communication is being used?
 Auto-negotiation
 Bit-by-bit delivery

Examples of protocols that use physical layers include:

 Digital Subscriber Line (DSL)
 Integrated Services Digital Network (ISDN)
 Infrared Data Association
 Universal Serial Bus (USB)
 Bluetooth
 Controller Area Network
 Ethernet

Data linkLayer

Figure 2.6 Data link layer

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  The data link layer provides error-free transfer of data frames from one node to another over the
physical layer, allowing layers above it to assume virtually error-free transmission over the link. That is,
The data link layer is responsible for moving frames from one hop (node) to the next.
 To do this, the data link layer provides:
1. Media Access Control or management: Control the access to the physical medium among all
connected devices. It determines when the node "has the right" to use the physical medium.
2. Link establishment and termination: establishes and terminates the logical link between two
nodes.
3. Framing (Frame delimiting): creates and recognizes frame boundaries.
4. Frame sequencing: transmits/receives frames sequentially.
5. Frame acknowledgment: provides/expects frame acknowledgments. Detects and recovers from
errors that occur in the physical layer by retransmitting non-acknowledged frames and handling
duplicate frame receipt.
6. Physical Addressing: incorporate sender/receiver addresses in the frame header.
7. Flow Control (Frame Traffic Control): To prevent fast sender from flooding a slower receiver with
frames. Tells the transmitting node to "back-off"(stop) when no frame buffers are available.
8. Error Control (Frame error checking): To Increase physical layer reliability by adding mechanism to
detect and ReTx damages and lost frames using the Trailer info. Checks received frames for
integrity.

Network Layer

Figure 2.9 Source-to-destination delivery

 Responsible for the source to destination delivery of individual packets, possibly across multiple
networks.
 The network layer controls the operation of the subnet, deciding which physical path the data should
take based on network conditions, priority of service, and other factors.
 To do this, the Network Layer provides:
 Routing: route packets over the subnet cloud of routers and switches, make the optimal routing
decisions (src- destination)
 Packet Deliver: Source host system to destination host system delivery, utilizing the data link layer
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 for peer-to-peer delivery.
 Logical-physical address mapping: translates logical addresses or names, into physical addresses.
Physical addresses at the D.L (device level) are not enough; we need to add logical addressing in
the packet header, of the sender and receiver
 Internetworking: resolve any Net protocols conflicts while moving in the subnet.
 Subnet traffic control: routers (network layer intermediate systems) can instruct a sending station
to "throttle back" its frame transmission when the router's buffer fills up.
 Packet fragmentation: if it determines that a downstream router's maximum transmission unit
(MTU) size is less than the packet size, a router can fragment a frame for transmission and
reassembly at the destination station.
 Subnet usage accounting: has accounting functions to keep track of frames forwarded by subnet
intermediate systems, to produce billing information.

Transport Layer

Figure 2.10: Transport layer

 The most important layer since it abstracts the complete details of the subnet to the user.
 The transport layer is responsible for the delivery of a message from one process to another.
 The transport layer ensures that messages are delivered error-free, in sequence, and with no losses or
duplications. It relieves (release) the higher layer protocols from any concern with the transfer of data
between them and their peers.
 The size and complexity of a transport protocol depends on the type of service it can get from the
network layer. For a reliable network layer with virtual circuit capability, a minimal transport layer is
required. If the network layer is unreliable and/or only supports datagrams, the transport protocol
should include extensive error detection and recovery.
 The transport layer provides:
 Service Access Point Addressing (SAP): The network logical address (i.e. IP addresses) is for
src_sys to destination not src_user_process to destination_user_process, hence we need another
address mechanism => SAP addresses within the same system for user message delivery.
 Message segmentation and reassembly: of segments => packets
Accepts a Process (user) message from the (session) layer above it, splits the message into smaller
units called segments (if not already small enough or if needed) each with n seg. Sequence number
to aid in assembly (incorrect order) related segments into the original message at the destination’s
transport layer. It then passes the smaller units down to the network layer. Note: The maximum
Transmittion Unit (MTU) size is 1500 bytes, i.e. the max size a data packet can have.
 Typically, the transport layer can accept relatively large messages, but there are strict message
size limits imposed by the network (or lower) layer. Consequently, the transport layer must
break up the messages into smaller units, or packets, prepending a header to each packet.
 The transport layer header information must then include control information, such as
message start and message end flags, to enable the transport layer on the other end to
recognize message boundaries.
 In addition, if the lower layers do not maintain sequence, the transport header must contain
sequence information to enable the transport layer on the receiving end to get the pieces back
together in the right order before handing the received message up to the layer above.
19
  Message acknowledgment (Connection control): provides reliable end-to-end message delivery
with acknowledgments.
1. Connection reliable service (-no ACK, -no guarantee). Connection-oriented TCP,
guarantees delivery in order with ACK of segments.
2. Connectionless unreliable service
 Message traffic control (Flow Control): tells the transmitting station to "back-off" when no
message buffers are available. This works as in the data Link Layer but all the message level “end-
users”.
 Error Control: Like the DLL, but process_to_process delivery of messages. Errors (damaged, loss or
duplicate) cause reTransmission of messages.

Figure 2.11: Reliable process-to-process delivery of a message

Session Layer
 The session layer is responsible for dialog control and synchronization. It serves as the network
dialog controller.

Figure 2.12: Session layer

 It provides:
 Session establishment, maintenance and termination: allows two application processes on
different machines (or stations) to establish, use and terminate a connection, called a session.
 Session support: performs the functions that allow these processes to communicate over the
network, performing security, name recognition, logging, and so on.
20
  It established, maintain and synchronizes the interaction among communicating (processes on
different) system
o Dialog Controls H/Duplex or F/Duplex
o Synchronization: Checkpoints are added to data streams for dividing into units of
independent ACK. Communication robustness in case of crashes.

Presentation Layer
 The presentation layer is responsible for translation, compression, and encryption.
 Viewed as the translator for the network, the presentation layer formats the data to be presented to
the application layer. It may translate data from a format used by the application layer into a common
format at the sending station, and then translate the common format to a format known to the
application layer at the receiving station.
 The presentation layer provides:
 Translation (Character code translation): for example, ASCII to EBCDIC. Abstract syntax notation
(ASN). ASN is a standard interface description language (IDL) for defining data structures that can
be serialized and deserialized in a cross-platform way.
 Data conversion: bit order, CR-CR/LF, integer-floating point, and so on.
 Data compression: reduces the number of bits that need to be transmitted on the network, so as
to ensure efficient utilization of bandwidth
 Data encryption: encrypt data so as to secure information transmitted for privacy. For example,
password encryption.

Application Layer
 The application layer is responsible for providing services to the user.
 The application layer serves as the window for users and application processes to access network
services.

Figure 2.14 Application layer

 This layer contains a variety of commonly needed functions:

1. Resource sharing and device redirection
2. Remote file access, file transfer, and management
3. Remote printer access
4. Inter-process communication
5. Network management
6. Directory services
7. Mail Service, i.e. electronic messaging (such as mail)
8. Network virtual terminals, ”putty”to allow remote logins (emulations)

21
 9. SMTP, FTP, HTTP, DNS, SNMP, TELNET.

Brief Summary of OSI Layers

Figure 2.15: Summary of OSI layers

TCP/IP Reference Model (Internet Protocol Stack layers)

 Transmission Control Protocol/Internet Protocol (TCP/IP) protocol suite is the engine for the Internet
and networks worldwide. It is a set of protocols developed to allow cooperating computers to share
resources across the network.
 The layers in the TCP/IP protocol suite do not exactly match those in the OSI model. TCP/IP either
combines several OSI layers into a single layer, or does not use certain layers at all.
 The original TCP/IP protocol suite was defined as having four layers: host-to-network, internet,
transport, and application. However, when TCP/IP is compared to OSI, we can say that the TCP/IP
protocol suite is made of five layers: physical, data link, network, transport, and application. Figure _
shows the relationship between the TCP/IP protocol suite and the OSI model.

 Whent compareted to the ISO model, theTCP/IP model is show to have five layers, see Figure __

22
  As we can see from the above figure, presentation and session layers are not there in TCP/IP model.
Also note that the Network Access Layer in TCP/IP model combines the functions of Datalink Layer and
Physical Layer.

TCP/IP Application Layer

 Application layer is the topmost of the four layer TCP/IP model; it is on the top of the Transport layer.
 Application layer defines TCP/IP application protocols and how host programs interface withTransport
layer services to use the network.
 Application layer includes all the higher-level protocols like DNS (Domain Naming System),
HTTP(Hypertext Transfer Protocol), Telnet, SSH, FTP (File Transfer Protocol), TFTP (Trivial File
TransferProtocol), SNMP (Simple Network Management Protocol), SMTP (Simple Mail Transfer
Protocol),DHCP (Dynamic Host Configuration Protocol), X Windows, RDP (Remote Desktop Protocol)
etc.

TCP/IP Transport Layer

 The purpose of Transport layer is to permit devices on the source and destination hosts to carry on a
conversation.
 Transport layer defines the level of service and status of the connection used when transporting data.
 The transport layer provides the end-to-end data transfer by delivering data from an application to its
remote peer.
 Transport layer packets are called a segment or a user datagram, depending on the protocol in use. TCP or UDP.
 TCP: Transmission Control Protocol, the most-used transport layer protocol.
---Establishes source to destination connection before data transmission.
---Ordered and ACK segment transmission with segment numbers.
 TCP which provides:
 Reliable data delivery Congestion control
 Duplicate data suppression Flow control
 Reliable connection oriented (stream) transport protocol.
 UDP - User Datagram Protocol, provides:
 Connectionless Best-effort service
 Unreliable
 UDP is used by applications that need a fast transport mechanism and can tolerate the loss of some
data. It does very limited error checking (checksum) and provides no flow control.
 SCTP: Stream Control Transmission Protocol (SCTP), a new protocol designed to respond to new
applications that are emerging in the multimedia
 SCTP Connection oriented reliable transport protocol to supports voice IP (Internet telephony)

23
 combining the best of UDP and TCP

TCP/IP Network Layer (Internet Layer)

 The internet layer also called the network layer.
 Internet layer pack data into data packets known as IP datagrams, which contain source and
destination address (logical address or IP address) information that is used to forward the datagrams
between hosts and across networks.
 The Internet layer is also responsible for routing of IP datagrams.
 Internet Protocol (IP) is the most important protocol in this layer.
 It is a connectionless protocol that does not assume reliability from lower layers. IP does not provide
reliability, flow control or error recovery. Is a best-effort-delivery unreliable connectionless datagram
protocol
 IP provides a routing function that attempts to deliver transmitted messages to their destination.
 These message units in an IP network are called an IP datagram.
 At the network layer there are other protocol to help the poor IP:
a) Address Resolution Protocol (ARP): used to discover the physical (MAC) or hardware address
of a network host, knowing its software (or IP) address, i.e. IP to MAC address translation.
b) Reverse Address Resolution Protocol (RARP): Does the reverse of ARP. Used by old diskless
workstations that could not store their IP address as a result. Knowing the MAC address, a
machine requests its IP address froma specialized RARP server on te same LAN. The server
returns the Machine’s IP address, i.e. MAC to IP address translation.
c) Internet Control Msg Protocol (ICMP): Help in reporting any failure/congestion of a subnet
part(s), aiding in the subnet robustness.
d) Internet Group Msg Protocol (IGMP): Single source broadcasts to multi group destinations.

TCP/IP Network Interface Layer (Network Access Layer)- Datalink + Physical Layer
 Network Access Layer defines details of how data is physically sent through the network, including
how bits are electrically or optically signalled by hardware devices that interface directly with a
network medium, such as coaxial cable, optical fiber, or twisted pair copper wire.
 The protocols included in Network Access Layer are Ethernet, Token Ring, FDDI, X.25, Frame Relay etc.
 Physical Layer: Very vague, it can be LAN, MAN, WAN.

A mapping of the protocols and services to both TCP/IP and the OSI model is given in Figure __ and Figure__.

Figure16: Mapping of Protocols and Services - TCP/IP vs ISO Models

24
 OSI and TCP/IP – Comparing and Contrating

OSI (Open System Interconnection) TCP/IP(Transmission Control Protocol/

InternetProtocol)
OSI provides layer functioning and also defines TCP/IP model is more based on protocols and
functions of all the layers. protocols are not flexible with other layers.
In OSI model the transport layer guarantees the In TCP/IP model the transport layer does not
delivery of packets Guarantees delivery of packets.
Follows horizontal approach Follows vertical approach.
OSI model has a separate presentation layer TCP/IP does not have a separate presentation layer
OSI is a general model. TCP/IP model cannot be used in any other
application
Network layer of OSI model provide both The Network layer in TCP/IP model provides
connection oriented and connectionless service connectionless service.
OSI model has a problem of fitting the protocols in TCP/IP model does not fit any protocol
the model
Protocols are hidden in OSI model and are easily In TCP/IP replacing protocol is not easy.
replaced as the technology changes.
OSI model defines services, interfaces and protocols In TCP/IP it is notclearly separated its services,
very clearly and makes clear distinction between interfaces and protocols.
them.
It has 7 layers It has 4 layers

******************** FIRST PART OF NOTE STOPS HERE ***************************

TCP/IP ADDRESSING
 Four levels of addresses are used in an internet employing the TCP/IP protocols. These are (1) Physical
Addresses (2) Logical Addresses (3)Port Addresses (4) Specific Addresses

Figure 2.17 Addresses in TCP/IP

Figure __ captures the Relationship of Layers and Addresses in TCP/IP

Example 2.1
In Figure 2.19 a node with physical address 10 sends a frame to a node with physical address 87. The two nodes
are connected by a link (bus topology LAN). As the figure shows, the computer with physical address 10 is the
sender, and the computer with physical address 87 is the receiver.

25
 Figure 2.18 Relationship of layers and addresses in TCP/IP

Figure 2.19 Physical addresses

Example 2.2
As we will see in Chapter 13, most local-area
local networks use a 48-bit (6-byte)
byte) physical address written as 12
hexadecimal digits; every byte (2 hexadecimal digits) is separated by a colon, as shown below:

Example 2.3
Figure 2.20 shows a part of an internet with two routers connecting three LANs. Each device (computer or
router) has a pair of addresses (logical and physical) for each connection
connection.. In this case, each computer is
connected to only one link and therefore has only one pair of addresses. Each router, however, is connected
to three networks (only two are shown in the figure). So each router has three pairs of addresses, one for
each connection.

26
 Figure 2.20 IP addresses

Example 2.4
Figure 2.21 shows two computers communicating via the Internet. The sending computer is running three
processes at this time with port addresses a, b, and c. The receiving computer is running two processes at
this time with port addresses j and k. Process a in the sending computer needs to communicate with
process j in the receiving computer. Note that although physical addresses change from hop to hop,
logical and port addresses remain the same from the source to destination.

27
Figure 2.21 Port addresses

NOTE:
The physical addresses will change from hop to hop, but the logical addresses usually remain the same.

Example 2.5
As we will see in Chapter 23, a port address is a 16-bit
16 address represented by one decimal number as
shown.

NOTE:
The physical addresses change from hop to hop, but the logical and port addresses usually remain the same.

Understanding of Delay, Loss and Throughput in the Packet Switching

Network
Basics
 Recall that a packet starts in a host (the source), passes through a series of routers, and ends its
journey in anotherr host (the destination).
 As a packet travels from one node (host or router) to the subsequent node (host or router) along this
path, the packet suffers from several types of delays at each node along the path.
 The most important of these delays are the
Nodal processing delay
Nodal Transmission delay
Queuing delay Propagation delay
 Together, these delays accumulate to givea total nodal delay.
28
 The performance of many Internet applications—such as search, Web browsing, email, maps, instant
messaging, and voice-over-IP—are greatly affected by network delays.

Figure17: Delay in Packet Switched Network

Processing Delay
 The time required to examine the packet’s header and determine where to direct the packet is part
of the processing delay.
 The processing delay can also include other factors, such as the time needed to check for bit-level
errors in the packet that occurred in transmitting the packet’s bits from the upstream node to
router.
 It is typically on the order of microseconds or less.

Queuing Delay
 At the queue, the packet experiences a queuing delay as it waits to be transmitted onto the link.
 The length of the queuing delay of a specific packet will depend on the number of earlier arriving
packets that are queued and waiting for transmission onto the link.
 If the queue is empty and no other packet is currently being transmitted, then our packet’s queuing
delay will be zero.
 On the other hand, if the traffic is heavy and many other packets are also waiting to be transmitted,
the queuing delay will be long.
 Queuing delays can be on the order of microseconds to milliseconds in practice.

Transmission Delay
 Assuming that packets are transmitted in a first-come-first-served manner like packet-switched
networks.
 Now packet can be transmitted only after all the packets that have arrived before it have been
transmitted.
 Denote the length of the packet by L bits, and denote the transmission rate of the link from router to
router by R bits/sec.
 The transmission delay is L/R.
 Transmission delays are typically on the order of microseconds to milliseconds in practice.

Propagation Delay
 Once a bit is pushed into the link, it needs to propagate to router B. The time required to propagate
from the beginning of the link to router B is the propagation delay.
 The bit propagates at the propagation speed of the link.
 The propagation speed depends on the physical medium of the link.
 Propagation delays are on the order of milliseconds.
 Propagations delay = d (Length of Physical Link) /s (Propagation speed in medium).

29
Packet Loss
 Packet loss is the failure of one or more transmitted packets to arrive at their destination.
 This event can cause noticeable effects in all types of digital communications.
 The loss of data packets depends on the switch queue. The loss of data packets increases with the
increases in the traffic intensity.
 It affects the performance of the network.

Throughput
 Throughput or Network Throughput is the rate of successful message delivery over a communication
channel.
 The data these messages belong to may be delivered over a physical or logical link or it can pass
through a certain network node.
 Throughput is usually measured in bits per second (bit/s or bps), and sometimes in data packets per
second (p/s or pps) or data packets per time slot.

Example
In this problem, we consider sending real-time voice from Host A to Host B over a packet-switched
network (VoIP). Host A converts analog voice to a digital 64 kbps bit stream on the fly. Host A then
groups the bits into 56-byte packets. There is one link between Hosts A and B; its transmission rate is
2 Mbps and its propagation delay is 10 msec. As soon as Host A gathers a packet, it sends it to Host
B. As soon as Host B receives an entire packet, it converts the packet’s bits to an analog signal. How
much time elapses from the time a bit is created (from the original analog signal at Host A) until the
bit is decoded (as part of the analog signal at Host B)?

Given: Analog to Digital conversion rate = 64 kbps

Packet size = 56 bytes Convert into bits so 56 bytes = 56 * 8 = 448 bits).
Transmission rate = 2 Mbps
Propagation delay = 10 msec

 Since this is a packet switched network, the data will be transmitted packet by packet.
 A packet is 56 byte and the analog to digital conversation rate is 64 kbps, thus the preparing time
 PT for a packet is 448/(64*1000)= 0.007 sec=7 msec
 The transition delay TD for a packet is (Size or Length of packet)/(Speed or Transmission rate) So,TD =
(56*8)/(2*1000*1000) = 0.000224 s = 0.224msec.
 Propagation delay PD = 10msec (Given in sum)
 Finally, the total time elapses from the time a bit is create until the bit is decoded is PT+TD+PD =
7+0.224+10 = 17.224 msec.

Principles of Reliable Data Transfer

30
 Fig.7: Reliable data transfer commands

 The sending side of the data transfer protocol will be invoked from above by a call to rdt_send().
 On the receiving side, rdt_rcv() will be called when a packet arrives from the receiving side of the
channel.
 When the rdt protocol wants to deliver data to the upper layer, it will do so by calling
deliver_data().
 Both the send and receive sides of rdt send packets to the other side by a call to udt_send().

Building a Reliable Data Transfer Protocol

Reliable Data Transfer over a Perfectly Reliable Channel: rdt1.0

 We first consider the simplest case in which the underlying channel is completely reliable.
 The protocol itself, which we will call rdt1.0, is trivial.

 The sender and receiver FSMs (Finite State Machines) have only one state.
 The arrows in the FSM description indicate the transition of the protocol from one stateto
another. (Since each FSM has just one state, a transition is necessarily from the onestate
back to itself).

31
  The event causing the transition is shown above the horizontal line labeling the transition,
and the action(s) taken when the event occurs are shown below the horizontal line.
 The sending side of rdt simply accepts data from the upper-layer via the rdt_send(data)
event, puts the data into a packet (via the action make_pkt(packet,data)) and sends the
packet into the channel.
 On the receiving side, rdt receives a packet from the underlying channel via the
rdt_rcv(packet) event, removes the data from the packet using the action
extract(packet,data) and passes the data up to the upper-layer.
 Also, all packet flow is from the sender to receiver-with a perfectly reliable channel there is
no need for the receiver side to provide any feedback to the sender since nothing can go
wrong.
Reliable Data Transfer over a Channel with Bit Errors: rdt2.0
 A more realistic model of the underlying channel is one in which bits in a packet may be
corrupted.
 Such bit errors typically occur in the physical components of a network as a packet is
transmitted, propagates, or is buffered.
 We'll continue to assume for the moment that all transmitted packets are received (although
their bits may be corrupted) in the order in which they were sent.
 Before developing a protocol for reliably communicating over such a channel, first consider
how people might deal with such a situation.
 Consider how you yourself might dictate a long message over the phone. In a typical scenario,
the message taker might say ``OK'' after each sentence has been heard, understood, and
recorded. If the message taker hears a garbled sentence, you're asked to repeat the garbled
sentence. This message dictation protocol uses both positive acknowledgements (``OK'') and
negative acknowledgements (``Please repeat that'').
 These control messages allow the receiver to let the sender know what has been received
correctly, and what has been received in error and thus requires repeating. In a computer
network setting, reliable data transfer protocols based on such retransmission are known ARQ
(Automatic Repeat reQuest) protocols.
 Fundamentally, two additional protocol capabilities are required in ARQ protocols to handle
the presence of bit errors:
 Error detection: First, a mechanism is needed to allow the receiver to detect when bit
errors have occurred. UDP transport protocol uses the Internet checksum field for exactly
this purpose. Error detection and correction techniques allow the receiver to detect, and
possibly correct packet bit errors.
 Receiver feedback: Since the sender and receiver are typically executing on different
end systems, possibly separated by thousands of miles, the only way for the sender to
learn of the receiver's view of the world (in this case, whether or not a packet was received
correctly) is for the receiver to provide explicit feedback to the sender. The positive (ACK)
and negative acknowledgement (NAK) replies in the message dictation scenario are an
example of such feedback. Our rdt2.0 protocol will similarly send ACK and NAK packets
back from the receiver to the sender

 The send side of rdt2.0 has two states.

 In one state, the send-side protocol is waiting for data to be passed down from the upper
layer.
 In the other state, the sender protocol is waiting for an ACK or a NAK packet from the
receiver.

32
  If an ACK packet is received (the notation rdt_rcv(rcvpkt) && is ACK (rcvpkt), the sender
knows the most recently transmitted packet has been received correctly and thus the
protocol returns to the state of waiting for data from the upper layer.
 If a NAK is received, the protocol retransmits the last packet and waits for an ACK or NAK to
be returned by the receiver in response to the retransmitted data packet.
 It is important to note that when the receiver is in the wait-for-ACK-or-NAK state, it cannot
get more data from the upper layer; that will only happen after the sender receives an ACK
and leaves this state.
 Thus, the sender will not send a new piece of data until it is sure that the receiver has
correctly received the current packet.

 Because of this behaviour, protocols such as rdt2.0 are known as stop-and-wait protocols.
 The receiver-side FSM for rdt2.0 still has a single state.
 On packet arrival, the receiver replies with either an ACK or a NAK, depending on whether
or not the received packet is corrupted.
 In the above figure the notation rdt_rcv(rcvpkt) && corrupt(rcvpkt) corresponds to the
event where a packet is received and is found to be in error.

rdt2.1: sender, handles garbled ACK/NAKs

 Protocol rdt2.0 may look as if it works but unfortunately has a fatal flaw.
 In particular, we haven't accounted for the possibility that the ACK or NAK packet could be
corrupted.
 Minimally, we will need to add checksum bits to ACK/NAK packets in order to detect such errors.
 The more difficult question is how the protocol should recover from errors in ACK or NAK packets.
 The difficulty here is that if an ACK or NAK is corrupted, the sender has no way of knowing
whether or not the receiver has correctly received the last piece of transmitted data.
 An approach is for the sender to simply resend the current data packet when it receives a garbled
ACK or NAK packet.
 This, however, introduces duplicate packets into the sender-to-receiver channel.
 The fundamental difficulty with duplicate packets is that the receiver doesn't know whether the
ACK or NAK it last sent was received correctly at the sender.
 Thus, it cannot know a priori whether an arriving packet contains new data or is a duplicate.
 A simple solution to this new problem is to add a new field to the data packet and have the
sender number its data packets by putting a sequence number into this field.
 The receiver then need only check this sequence number to determine whether or not the
received packet is a retransmission.
 For this simple case of a stop-and-wait protocol, a 1-bit sequence number will suffice, since it will
allow the receiver to know whether the sender is resending the previously transmitted packet
(the sequence number of the received packet has the same sequence number as the most
recently received packet) or a new packet (the sequence number changes).
 Since we are currently assuming a channel that does not lose packets, ACK and NAK packets do
not themselves need to indicate the sequence number of the packet they are ACKing or NAKing,
since the sender knows that a received ACK or NAK packet(whether garbled or not) was
generated in response to its most recently transmitted data packet.

33
 Fig 10 rdt 2.1 Sender

Fig 11 rdt 2.1 Receiver

 The rdt2.1 sender and receiver FSM's each now have twice as many states as rdt2.0.
 This is because the protocol state must now reflect whether the packet currently being sent
(by the sender) or expected (at the receiver) should have a sequence number of 0 or 1

34
Note that the actions in those states where a 0-numbered packet is being sent or expected are mirror
images of those where a 1-numbered packet is being sent or expected; the only differences have to
do with the handling of the sequence number

 Protocol rdt2.1 uses both positive and negative acknowledgements from the receiver to the
sender
 A negative acknowledgement is sent whenever a corrupted packet, or an out of order packet,
is received
 We can accomplish the same effect as a NAK if instead of sending a NAK, we instead send an
ACK for the last correctly received packet
 A sender that receives two ACKs for the same packet (i.e., receives duplicate ACKs) knows that
the receiver did not correctly receive the packet following the packet that is being ACKed
twice. Our NAK-free reliable data transfer protocol for a channel with bit errors is rdt2.2.

Reliable Data Transfer over a Lossy Channel with Bit Errors: rdt3.0
Suppose now that in addition to corrupting bits, the underlying channel can lose packets as
well.
Two additional concerns must now be addressed by the protocol: how to detect packet loss
and what to do when this occurs.
The use of checksumming, sequence numbers, ACK packets, and retransmissions – the
techniques already developed in rdt 2.2-will allow us to answer the latter concern. Handling
the first concern will require adding a new protocol mechanism.
There are many possible approaches towards dealing with packet loss.
Suppose that the sender transmits a data packet and either that packet, or the
receiver's ACK of that packet, gets lost.
In either case, no reply is forthcoming at the sender from the receiver.
If the sender is willing to wait long enough so that it is certain that a packet has been lost, it
can simply retransmit the data packet.
But how long must the sender wait to be certain that something has been lost? It must clearly
wait at least as long as a round trip delay between the sender and receiver (which may include
buffering at intermediate routers or gateways) plus whatever amount of time is needed to
process a packet at the receiver.
If an ACKis not received within this time, the packet is retransmitted.
Note that if a packet experiences a particularly large delay, the sender may retransmit the
packet even though neither the data packet nor its ACK have been lost.
This introduces the possibility of duplicate data packets in the sender-to-receiver
channel.
Happily, protocol rdt2.2 already has enough functionality (i.e., sequence numbers) to handle
the case of duplicate packets.

 From the sender's viewpoint, retransmission is a solution. The sender does not know

35
 whether a data packet was lost, an ACK was lost, or if the packet or ACK was simply overly
delayed.
 In all cases, the action is the same: retransmit. In order to implement a time-based
retransmission mechanism, a countdown timer will be needed that can interrupt the
sender after a given amount of timer has expired.
 The sender will thus need to be able to (i) start the timer each time a packet (either a first
time packet, or a retransmission) is sent, (ii) respond to a timer interrupt (taking
appropriate actions), and (iii) stop the timer.
 The existence of sender-generated duplicate packets and packet (data, ACK) loss
alsocomplicates the sender's processing of anyACK packet it receives.
 If an ACK is received, how is the sender to know if it was sent by the receiver in responseto
its (sender's) own most recently transmitted packet, or is a delayed ACK sent inresponse to
an earlier transmission of a different data packet? The solution to this dilemma is to
augment the ACK packetwith an acknowledgement field. When thereceiver generates an
ACK, it will copy the sequence number of the data packet beingACK'ed into this
acknowledgement field. By examining the contents of the acknowledgment field, the
sender can determine the sequence number of the packet being positively acknowledged.
Fig 12 rdt 3.0 Sender

36
 Fig. 13 Operation of rdt3.0, the alternating-bit protocol

The Network Core & Switching

 Network core defines the connection of different network segments together and the process to
transmit the data packets across the network.
 The network core is implemented through the use of switching techniques.
 The classification of switching network is shown below:

37
Circuit Switching
 Circuit switching is used in public telephone networks and is the basis for private networks built on
leased-lines.
 Circuit switching was developed to handle voice traffic but also digital data (although inefficient)
 With circuit switching a dedicated path is established between two stations for communication.

 Switching and transmission resources within the network are reserved for the exclusive use of the
circuit for the duration of the connection.
 The connection is transparent once it is established; it appears to attach devices as if there were a
direct connection.
 Communication via circuit switching involves three phases:
1. Circuit Establishment
2. Data Transfer
3. Circuit Disconnect

 Connection path must be established before data transmission begins. Nodes must have switching

38
 capacity and channel capacity to establish connection.
 Circuit switching is inefficient
1. Channel capacity dedicated for duration of connection
2. If no data, capacity wasted
 Set up (connection) takes time
 Once connected, transfer is transparent to the users
1. Data is transmitted at a fixed data rate with no delay (except for the propagation delay)
 Developed for voice traffic (phone)
1. May also be used for data traffic via modem
 Interconnection of telephones within a building or office.
 In circuit switching, a direct physical connection between two devices is created by space division
switches, time-division switches, or both.

Space Division Switching

 Developed for analog environment.
 In a space-division switch, the path from one device to another is spatially separate from other paths.
 A crossbar is the most common space-division switch. It connects n inputs to m outputs via n × m cross
points.
 Crossbar switch.

Figure13: Space Division Switching

Time Division Switching

 In a time-division switch, the inputs are divided in time, using Time Division Multiplexing (TDM). A control
unit sends the input to the correct output device.
 Use digital time division techniques to set up and maintain virtual circuits.

39
 Figure14: Time Division Switching

Packet Switching
 Packet switching was designed to provide a more efficient facility than circuit-switching for bursty data
traffic.
 With packet switching, a station transmits data in small blocks, called packets.
 At each node packets are received, stored briefly (buffered) and passed on to the next node.
1. Store and 2. forward mechanism
 Each packet contains some portion of the user data plus control info needed for proper functioning of
the network.
 A key element of packet-switching networks is whether the internal operation is datagram or virtual
circuit (VC).
1. With internal VCs, a route is defined between two endpoints and all packets for that VC
follow the same route.
2. Without internal VCs, each packet is treated independently, and packets intended for the
same destination may follow different routes.
 Examples of packet switching networks are X.25, Frame Relay, ATM and IP.
 Station breaks long message into packets. Packets sent one at a time to the network.
 Packets handled in two ways:
1. Datagram
 Each packet treated independently
 Packets can take any practical route
 Packets may arrive out of order
 Packets may go missing
 Up to receiver to re-order packets and recover from missing packets
2. Virtual Circuit
 Preplanned route established before any packets sent.
 Once route is established, all the packets between the two communicating parties
follow the same route through the network
 Call request and call accept packets establish connection (handshake)
 Each packet contains a Virtual Circuit Identifier (VCI) instead of destination address
 No routing decisions required for each packet
 Clear request to drop circuit
 Not adedicated path

MessageSwitching
 This technique was somewhere in middle of circuit switching and packet switching.
 In message switching, the whole message is treated as a data unit and is transferred in its entirety.
 A switch working on message switching, first receives the whole message and buffers it until there are
resources available to transfer it to the next hop.
 If the next hop is not having enough resource to accommodate large size message, the message is
stored and switch waits.

Techniques used in data communications to transfer data

1. Connection-oriented method 2. Connectionless method

Connection-oriented method
40
  Connection-oriented communication includes the steps of setting up a call from one computer to
another, transmitting/receiving data, and then releasing the call, just like a voice phone call.
 However, the network connecting the computers is a packet switched network, unlike the phone
system's circuit switched network.
 Connection-oriented communication is done in one of two ways over a packet switched network:
1. Without virtual circuits. 2. With virtual circuits.

Without virtual circuits:

 This is what TCP does in the Internet.
 The only two machines in the Internet are aware about connectionwhich is established between the
two computers at the endpoints.
 The Internet itself, its routers and links have no information about the presence of a connection
between the two computers.
 This means that all of the packets flowing between the two computers can follow different routes.
 One benefit of establishing the connection is that the flow of packets from the source to the
destination can be slowed down if the Internet is congested and speeded up when
congestiondisappears.
 Another benefit is that the endpoints can anticipate traffic between them, and agree to cooperate to
ensure the integrity and continuity of the data transfers. This allows the network to be treated as a
"stream" of data.
With virtual circuit:
 This is not used in the Internet, but is used in other types of networks (eg. the "X.25" protocol, still
popular in Europe).
 The routers within the network route all packets in one connection over the same route. The
advantage is that video and voice traffic are easier to carry, because routers can reserve memory
space to buffer the transmission.

Connectionless method
 Connectionless communication is just packet switching where no call establishment and release occur.
 A message is broken into packets, and each packet is transferred separately. Moreover, the packets
can travel different route to the destination since there is no connection.
 Connectionless service is typically provided by the UDP (User Datagram Protocol). The packets
transferred using UDP are also called datagrams.

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 PROCESSES COMMUNICATING

Process
 A process is an instance of a program running in a computer or we can say that process is program
under execution.
 When processes are running on the same end system, they can communicate with each other with
interprocess communication, using rules that are governed by the end system’s operating system.
 Processes on two different end systems communicate with each other by exchanging messages across
the computer network.
 A sending process creates and sends messages into the network; a receiving process receives these
messages and possibly responds by sending messages back.
 In the context of a communication session between a pair of processes, the process that initiates the
communication is called the client. The process that waits to be contacted to begin the session is the
server.
 The client is a process (program) that sends a message to a server process (program), requesting that
the server perform a task (service).
 A server is a process (program) that fulfils the client request by performing the task requested by
client. Server programs generally receive requests from client programs, execute it and dispatch
responses to client requests

Sockets: Interface Between The Process And The Computer Network

 Any message sent from one process to another must go through the underlying network.
 A process sends messages into, and receives messages from, the network through a software interface called

42
 a socket.
 A process is similar to a house and its socket is similar to its door. When a process wants to send a message
to another process on another host, it shoves (passes) the message out its door (socket). This sending
process assumes that there is a transportation infrastructure on the otherside of its door that will transport
the message to the door of the destination process. Once the message arrives at the destination host, the
message passes through the receiving process’s door (socket), and the receiving process then acts on the
message.

Fig.2 Application processes, sockets, and underlying transport protocol

Addressing Processes
 To identify the receiving process, two pieces of information need to be specified:
1. the address of the host and
2. an identifier that specifies the receiving process in the destination host.
 In the Internet, the host is identified by its IP address.
 An IPaddress is a 32-bit data/address that uniquely identifies the host.
 In addition to knowing the address of the host to which a message is destined, the sending process must
also identify the receiving process (more specifically, the receiving socket) runningin the host.
 This information is needed because in generala host could be running many network applications.

Transport Services available to Applications

Reliable Data Transfer

 Reliable data transfer is to guarantee that the data sent by one end of the application is delivered
correctly and completely to the other end of the application.
 If a protocol provides such a guaranteed data delivery service, it is said to provide reliable data
transfer.
 One important service that a transport-layer protocol can potentially provide to an application is
process-to-process reliable data transfer.
 When a transport protocol provides this service, the sending process can just pass its data into the
socket and know with complete confidence that the data will arrive without errors at the
receiving process.

Throughput
 Throughput is the rate at which the sending process can deliver bits to the receiving process.
 The transport protocol ensures that the available throughput is always at least r bits/sec.
43
 Timing
 A transport-layer protocol can also provide timing guarantees.
 An example guarantee might be that every bit that the sender pumps into the socket arrives at the
receiver’s socket no more than 100 msec later. Such a service would be appealing to interactive real-
time applications, such as Internet telephony, virtual environments,teleconferencing and multiplayer
games, all of which require tight timing constraintson datadelivery in order to be effective.

Security
 Finally, a transport protocol can provide an application with one or more security services.
 For example, in the sending host, a transport protocol can encrypt all data transmitted by the sending
process, and in the receiving host, the transport-layer protocol can decrypt the data before delivering
the data to the receiving process.

Transport-Layer services
 A transport-layer protocol provides logical communication between application processes
running on different hosts.
 By logical communication, application processes communicate with each other byusing the
logical communication provided by the transport layer to send messages to each other, free

44
 from the worry of the details of the physical infrastructure used to carry these messages.
 Transport-layer protocols are implemented in the end systems but not in network routers.
 On the sending side, the transport layer converts the application-layer messages it receives
from a sending application process into transport-layer packets, known as transport-layer
segments.
 On the receiving side, the transport layer reassembles segments into messages, passes to
application layer.
 A network-layer protocol provides logical communication between hosts.

Relationship between Transport and Network Layers

 The transport layer lies just above the network layer in the protocol stack.
 Whereas a transport-layer protocol provides logical communication between processes
running on different hosts, a network-layer protocol provides logical communication between
hosts.
 Let's examine this distinction with the aid of a household analogy.
 Consider two houses, one on the East Coast and the other on the West Coast, with each
house being home to a dozen kids.
 The kids in the East Coast household are cousins of the kidsin the West Coast household.
 The kids in the two households love to write to each other-each kid writes each cousin every
week, with each letter delivered by the traditional postal service in a separate envelope.
 Thus, each household sends 144 letters tothe other household every week.
 In each ofthe households there is one kidAnn in the West Coast house and Bill in the East
Coast houseresponsible for mail collection and mail distribution.
 Each week Ann visits all her brothers and sisters, collects themail, and gives the mail to a
postal-service mail person who makes daily visits to the house.
 When letters arrive at the West Coast house, Ann also has the job of distributing the mail to
her brothers and sisters. Bill has a similar job on the East Coast.
 In this example, the postal service provides logical communication between the two houses-
the postal service moves mail from house to house, not from person to person.
 On the other hand, Ann and Bill provide logical communication among the cousins-Ann and
Bill pick up mail from and deliver mail to their brothers and sisters.
 Note that from the cousins' perspective, Ann and Bill are the mail service, even though Ann
and Bill are only a part (the end system part) of the end-to-end delivery process.
 This household example serves as a nice analogy for explaining how the transport layer
relates to the network layer:
o hosts (also called end systems) = houses
o processes = cousins
o application messages = letters in envelopes
o network-layer protocol= postal service (including mail persons)
o transport-layer protocol = Ann and Bill
 Continuing with this analogy, observe that Ann and Bill do all their work within their
respective homes; they are not involved, for example, in sorting mail in any intermediate mail
centre or in moving mail from one mail centre to another.
 Similarly, transport-layer protocols live in the end systems. Within an end system, a transport
protocol moves messages from application processes to the network edge (that is, the
network layer) and vice versa; but it doesn't have any say about how the messages are moved
within the network core.
 In fact, intermediate routers neither act on, nor recognize, any information that the transport
45
 layer may have appended to the application messages.

Multiplexing and Demultiplexing

Fig. 1Transport-layer multiplexing and demultiplexing

 The job of gathering data chunks at the source host from different sockets, encapsulating each data chunk
with header information (that will later be used in demultiplexing) to create segments, and passing the
segments to the network layer is called multiplexing.
 At the receiving end, the transport layer examines these fields to identify the receivingsocket and
then directs the segment to that socket. This job of delivering the data in atransport-layer
segment to the correct socket is called demultiplexing
 Transport layer in the middle host in Figure 1 must demultiplex segments arriving fromthe
network layer below to either process P1 or P2 above; this isdone by directing thearriving
segment’s data to the corresponding process’s socket
 The transport layer in the middle host must also gather outgoing data from thesesockets, form
transport-layer segments, and pass these segments down to the networklayer

46
 Endpoint Identification
 Sockets must have unique identifiers.
 Each segment must include header fields identifying the socket, these header fields are the
source port number field and the destination port number field.
 Each port number is a 16-bit number:0 to 65535.

Fig. 2 Source and destination port-number fields in a transport-layer segment

Connectionless Multiplexing and Demultiplexing

 Supposea process on Host A, with port number 19157, wants to send data to a process with UDP
port 46428 on Host B.

Fig. 3: The inversion of source and destination port numbers

 Transport layer in Host A creates a segment containing source port, destination port,and dataand
thenpasses it to the network layer in Host A.
 Transport layer in Host B examines destinationport number and delivers segment tosocket
identified by port 46428
 Note: a UDP socket is fully identified by a two-tuple consisting of
1. a destination IP address
2. a destination port number
 Source port number from Host A is used at Host B as "return address".

47
 Connection-Oriented Multiplexing and Demultiplexing
 Each TCP connection has exactly two end-points.
 This means that two arriving TCP segments with different source IP addresses or source port
numbers will be directed to two different sockets, even if they have the same destination port
number.
 So a TCP socket is identified by four-tuple
1. source IP address 3. destination IP address
2. source port# 4. destination port #
 Whereas UDP is identified by only two-tuples
1. destination IP address 2. destination port #

Fig. 4: Two clients, using the same destination port number (80) tocommunicate with the same
Web server application

UDP Segment Structure

Fig. 5: UDP segment structure

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 The port numbers allow the destination host to pass the application data to the correct
process runningon the destination end system (that isusedto perform the demultiplexing
function).
 Thelength field specifies the number of bytes in the UDP segment (header plus data).
 The checksum is used by the receiving host to check whether errors have been introduced
into the segment.
How to calculate (find) checksum:
 The UDP checksum iscalculatedon the sending side by summing all the 16-bit words in the
segment, with any overflow being wrapped around and then the 1's complement is
performed and the result is added to the checksum field inside the segment.
 At the receiver side, all words inside the packet are added and the checksum is added upon
them if the result is 1111 1111 1111 1111 then the segment is valid else the segment has an
error.

WIDELY USED PROTOCOLS SUCH AS ETHERNET, WIRELESS LANS,

IP, AND HTTP

The Web and HTTP

Web
 A Web page consists of objects.
 An object is simply a file-such as an HTML file, a JPEG image, a Java applet, or a video clip, thatis
addressable by a single URL.
 Most Web pages consist of a base HTML file and several referenced objects.
 For example, if a Web page contains HTML textand five JPEG images, then the Web page has
sixobjects: the base HTML file plus the five images.
 The base HTML file references the other objects in the page with the objects’ URLs.
 Each URL has two components: the hostname of the server that houses theobject and theobject’s

49
 path name.
 For example, the
URLhttp://www.someSchool.edu/someDepartment/picture.gifhaswww.someSchool.eduis
hostname and“/someDepartment/picture.gif”is path name.

HTTP (HyperText Transfer Protocol)

 HTTPisWeb’s application layer protocolwhichdefines how Web clients request Web pagesfrom Web
servers and how servers transfer Web pages to clients.
F
i
g
.
3
H
T
T
P

r
e
q
u
e
s
t-responsebehaviour

 When a user requests a Web page (for example, clicks on a hyperlink), the browser sends
HTTPrequest messages for the objects in the page to the server.
 The server receives the requests and responds with HTTP response messages that contain
theobjects.
 HTTP uses TCP as its underlying transport protocol.
 The HTTP client first initiates aTCP connection with the server.
 Once the connection is established, the browser and the server processes access TCP
throughtheirsocket interfaces.
 HTTP follows client/server model
client: browser that requests, receives, (using HTTP protocol) and “displays” Web objects
server: Web server sends (using HTTP protocol) objects in response to requests
 HTTP connection types
1. Non-persistent HTTP
2. Persistent HTTP

Non-persistent HTTP
 A non-persistent connection is the one that is closed after the server sends the requested objectto
the client. In other words, the connection is used exactly for one request and one response.
 Fordownloading multiple objectsitrequired multiple connections.
 Non-persistent connections are the default mode for HTTP/1.0.
 suppose user enters URL: "www.someSchool.edu/someDepartment/home.index"
 Above link contains text and references to 10 jpeg images.

50
 1. The HTTP client process initiates a TCP connectionto the serverwww.someSchool.edu on port number 80,
which is the default port numberfor HTTP. Associated with the TCP connection, there will be a socket at
theclient and a socket at the server.
2. The HTTP client sends an HTTP request message to the servervia its socket. Therequest message includes
the path name /someDepartment/home.index.
3. The HTTP server process receives the request message via its socket, retrieves the
object /someDepartment/home.index from its storage (RAM ordisk), encapsulates the object in an HTTP
response message, and sends theresponse message to the client via its socket.
4. The HTTP server process tells TCP to close the TCP connection. (But TCPdoesn’t actually terminate the
connection until it knows for sure that the clienthas received the response message intact.).
5. The HTTP client receives the response message. The TCP connection terminates. The message indicates
that the encapsulated object is an HTML file. Theclient extracts the file from the response message,
examines the HTML file, and finds references to the 10 JPEG objects.
6. The first four steps are then repeated for each of the referenced JPEG objects.

 RTT(Round Trip Time)-which is the time it takes for a small packet to travel from client to
Serverand then back to theclient. See Figure 4 below.

 HTTP response time:

1. one RTT to initiate TCP connection
2. one RTT for HTTP request and first few bytes of HTTP response to return
3. file transmission time
 Non-persistent HTTP response time = 2RTT + file transmission time
 It overhead for each TCP connection.

51
 Fig.4 Calculation for the time needed to request and receive an HTML file

Persistent HTTP
 With persistent connections, the server leaves the TCP connection open after sending responses and hence
the subsequent requests and responses between the same client and server can be sent.
 The server closes the connection only when it is not used for acertain configurable amount of time.
 It requiresas little as one RTT for all the referenced objects
 With persistent connections, the performance is improved by 20%.
 Persistent connections are the default mode for HTTP/1.1.

HTTP messageformat
There aretwo types of HTTP messages:

1. Request 2. Response

HTTP request message

 HTTP request messageis inASCIIformat which means inhuman-readable format.

52
  HTTP request messageconsist of three part
1. Request line 2. Header line 3. Carriage return
 The message is written in ordinary ASCII text, so that your ordinary computer-literate human being can
read it.
 Each line is followed by a carriage return and a line feed.
 The last line is followed by an additional carriage return and line feed.
 The first line of an HTTP request message is called the request line; the subsequent lines are called the
header lines.
 The request line has three fields: the method field, the URL field, and the HTTP version field.
 The method field can take on several different values, including GET, POST, HEAD, PUT, andDELETE.
 In this example, the browser is requesting the object/somedir/page.html. The version is self-
explanatory; in this example, thebrowser implements version HTTP/1.1.
 The header line Host: www.someschool.edu specifies the host on which the object resides.

HTTP response message

Fig.6HTTP request message

HTTP response message consist three part

1.Status line 2.Header line 3.Data (Entity body)
 The status line has three fields: the protocol version field, a status code and a corresponding status
message.
 In this example, the status line indicates thatthe server is using HTTP/1.1 and that everything is OK.
 The Date: header line indicates the time and date when the HTTP response was created and sent by
the server.
 The Server: header line indicates that the message was generated by an Apache Web server.
 The Last-Modified: header line indicates the time and date when the object was created or last
modified.
 The Content-Length: header line indicates the number of bytes in the object being sent.
 The Content-Type: header line indicates that the object in the entity body is HTML text.

User-Server Interaction OR Cookies

 A small text file created by a Web site that is stored in the user's computer either temporarily for that
53
 session only or permanently on the hard disk (persistent cookie).
 Cookies provide away for the Web site to recognize you and keep track of your preferences.
 Cookie technology has four components
1. a cookieheader line in the HTTP response message
2. a cookie header line in the HTTPrequest message
3. a cookie file kept on the user’s end system and managed by theuser’s browser
4. a back-end database at the Web site

 Suppose Susan, access Amazon.com for the first time.

 Let us suppose that in the past she has already visited the eBay site.
 When the request comes into the Amazon Web server, the server creates a unique identification number
and creates an entry in its back-end database by the identification number.
 The Amazon Web server then responds to Susan’s browser, including in the HTTP response a set-cookie:
header, which contains the identification number.
 For example, the header line might be: Set-cookie: 1678.
 When Susan’s browser receives the HTTP response message, it sees the Set cookie: header.
 The browser then appends a line to the special cookie file that it manages.
 This line includes the hostname of the server and the identification number in the Set-cookie: header.
 The cookie file already has an entry for eBay, since Susan has visited that site in the past.
 As Susan continues to browse the Amazon site, each time she requests a Web page, her browser consults
her cookie file, extracts her identification number for this site, and puts a cookie header line that includes
the identification number in the HTTP request.
 Specifically, each of her HTTP requests to the Amazon server includes the header line: Cookie: 1678.
 In this manner, the Amazon server is able to track Susan’s activity at the Amazon site.

Use of cookies
authorization recommendations
shopping carts user session state (Webe-mail)
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_____________________________________________________________________________________

Web Caching OR Proxy Server

A Web cacheORa proxy server is a network entity that satisfies HTTPrequests on the behalf of
an origin Web server.
The Web cachehas its own disk storageand keeps copies of recently requested objects in this

storage.
 Fig.8Clients requesting objects through a Web cache

A Web cacheORa proxy serveris a network entity that satisfies HTTPrequests on the behalf of
an origin Web server.
 The Web cache has its own disk storageand keeps copies of recently requested objects in
thisstorage.
 Auser’s browser can be configured so that all of the user’s HTTP requests are first directedtothe
Webcache.
 As an example, suppose a browser is requesting theobject
http://www.someschool.edu/campus.gif.
 Here is what happens
1.The browser establishes a TCP connection to the Web cache and sends anHTTP request
for the object to the Web cache.
2.The Web cache checks to see if it has a copy of the object stored locally. If itdoes, the
Web cache returns the object within an HTTP response message tothe client browser.
3.If the Web cache does not have the object, the Web cache opens a TCP connectionto the
Originserver, that is, to www.someschool.edu. The Web cachethen sends an HTTP
request for the object into the cache-to-server TCP connection.After receiving this
request, the origin server sends the object withinan HTTP response to the Web cache.
4.When the Web cache receives the object, it stores a copy in its local storage andsends a
copy, within an HTTP response message, to the client browser.
Note that a cache is both a server and a client at the same time.
When it receivesrequests from and sends responses to a browser, it is a server.
When it sends requeststo and receives responses from an origin server, it is a client.

Why Web cachingis needed(Required)?OR Advantages of Caching

To reduce response time for client request
To reduce traffic on aninstitution’s access link
To enables “poor” content providers to effectively deliver content

FTP (File Transfer Protocol)

55
File Transfer Protocol (FTP) is the commonly used protocol for exchanging files over theNetwork
OrInternet.
FTP uses the Internet's TCP/IP protocols to enable data transfer.
FTP usesclient-server architecture.
FTP promotes sharing of files via remote computers with reliable and efficient data transfer.

 Fig.9FTP moves files between local and remote file systems

In above figure user interacts with FTP through an FTP user agent.
 The user first provides the hostname of the remote host, causing the FTP client process in thelocal
host to establish a TCP connection with the FTP server process in the remote host.
 Theuser then provides the user identification and password, which are sent over the TCPconnection
as part of FTP commands.
 Once the server has authorized the user, the user copies one or more files stored in the local
filesystem into the remote file system (or vice versa).
 FTPuses two parallel TCP connections to transfer a file,
1.control connection
2.dataconnection

Fig.10Control and data connections

 The control connection is used for sending control information betweenthe two hosts such asuser
identification, password, commands tochange remote directory and commands to put andget files.
 The data connectionis used to actually send a file.
 Because FTP uses a separate control connection, FTP issaid to send its control information out-of-
band.
 When auser starts an FTP session with a remote host, the client side of FTP (user) first initiates
acontrol TCP connection with the server side (remote host) on server port number 21.
 The client side of FTP sends the user identification and password over thiscontrol connection.
 The client side of FTP also sends, over the control connection, commands to change the
remotedirectory.
 When the server side receives a command for a file transfer over the control connection (eitherto,
or from, the remote host),the server side initiates a TCP data connection to the client side.
 FTP sends exactly one file over the data connection and then closes the data connection.
 If, during the same session, the user wants to transfer another file, FTP opens another
dataconnection.
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  Thus, with FTP, the control connection remains open throughout the duration of the usersession,
but a new data connection is created for each file transferred within a session (that is,the data
connections are non-persistent).

Electronic mail in the Internet (Email)

As with ordinary postal mail, e-mail is an asynchronous communicationmediumin whichpeople use to
send and read messages when it is convenient for them, withouthaving to coordinate withother
people’s schedules.
In contrast with postal mail, electronicmail is fast, easy to distribute, and inexpensive.
Modern e-mail has many powerfulfeatures, including messages with attachments, hyperlinks,HTML-
formattedtext, and embedded photos.
Emailhas three major components
(i) user agents
(ii) mailservers
(iii) Simple Mail Transfer Protocol (SMTP)
User agents allow users to read, reply to, forward, save, and compose messages.
Microsoft Outlook and Apple Mail are examples of user agents for e-mail.

Fig.11A high-level view of the Internet e-mail system

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 Fig.12Alice sends a message to Bob

Inabove figure when Alice is finished composing her message, her user agent sends the
message to her mail server, where the message is placed in the mail server’s outgoing message
queue.

 When Bob wants to read amessage, his user agent retrieves the message from his mailbox in hismail
server.
 Mail servers form the core of the e-mail infrastructure.
 Each recipient, such asBob, has a mailbox located in one of the mail servers.
 Bob’s mailbox manages andmaintainsthe messages that have been sent to him.
 A typical message starts its journeyin the sender’s user agent, travels to the sender’s mail
serverandtravels to the recipient’s mail server, where it is deposited in the recipient’s mailbox.
 When Bob wants toaccess the messages in his mailbox, the mail server containing his mailbox
authenticates Bob (with usernames and passwords).
 Alice’s mail server must also deal with failures in Bob’s mail server.
 If Alice’s server cannot deliver mail to Bob’s server, Alice’s server holds the message in a
message queue and attempts to transfer the message later.
 Re-attempts are often done every 30 minutes or so; if there is no success after several days, theserver
removes the message and notifies the sender (Alice) with an e-mail message.
 SMTP is the principal application-layer protocol for Internet electronic mail. It uses the reliabledata
transfer service of TCP to transfer mail from the sender’s mail server to the recipient’s mailserver.
 SMTP has two sides: a client side, which executes on the sender’s mail server, and a server side,which
executes on the recipient’s mail server.
 Both the client and server sides of SMTP run on every mail server.
 When a mail server sends mail to other mail servers, it acts as an SMTP client. When a mailserver
receives mail from other mail servers, it acts as an SMTP server.

SMTP (Simple Mail Transfer Protocol)

 SMTP transfers messages from senders’ mail servers to the recipients’ mail servers.
 It restrictsthe body (not just the headers) of all mail messages to simple 7-bit ASCII.
 To illustrate the basic operation of SMTP, let’s take a common scenario. Suppose Alice wants tosend Bob a
simple ASCII message.

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 Fig.13Alice sends a message to Bob
 Alice invokesher user agent for e-mail, provides Bob’s e-mail address (for example,[email protected]),
composes a message and instructs the user agent to send themessage.
 Alice’s user agent sends the message to her mail server, where it is placed in a messagequeue.
 The client side of SMTP, running on Alice’s mail server, sees the message in the messagequeue. It
opens a TCP connection to an SMTP server, running on Bob’s mail server.
 After some initial SMTP handshaking, the SMTP client sends Alice’s message into theTCPconnection.
 At Bob’s mail server, the server side of SMTP receives the message. Bob’s mail server thenplaces the
message in Bob’s mailbox.
 Bob invokes his user agent to read the message at his convenience.
 SMTP does not normally use intermediate mail servers for sending mail, even when the two
mailservers are located at opposite ends of the world.
 If Bob’s mail server is down, the message remains in Alice’s mail server and waits for a
newattemptandthe message does not get placed in some intermediate mail server.

How SMTP transfers a message from a sending mail server to a receiving mail
server
First, the client SMTP (running on the sending mail server host) has TCP establish a connection
to port 25 at the server SMTP (running on the receiving mail server host).
If the server is down, the client tries again later.
Once this connection is established, the server and client perform some application-layer
handshaking, just as humans often introduce themselves before transferring information from
one to another.
During this SMTP handshaking phase, the SMTP client indicates the e-mail address of thesender
(the person who generated the message) and the e-mail address of the recipient.
Once the SMTP client and server have introduced themselves to each other, the client sends the
message.
SMTP can count on the reliable data transfer service of TCP to get the message to the server
without errors.
The client then repeats this process over the same TCP connection if it has other messages to
send to the server; otherwise, it instructs TCP to close the connection.

Comparison of SMTP with HTTP

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 HTTP is mainlya pull protocol-someone loads information on a web server and users use HTTP
to pull the information from the server. On the other hand, SMTP is primarily a push protocol-
the sending mail server pushes the file to receiving mail server.
SMTP requires each message, including the body of each message, to be in seven-bit ASCII
format. If the message contains binary data, then the message has to be encoded into seven-bit
ASCII format. HTTP does not have this restriction.
HTTP encapsulate each object of message in its own response message while SMTP places all of
the message's objects into one message.

Mail Access Protocols(POP3and IMAP)

POP3(Post Office Protocol)
POP3 is an extremely simple mail access protocol.
POP3 begins when the user agent ofthe client opens a TCP connectionto the mail server of the
Serveron port 110.

 With the TCP connection is established,POP3 progresses through three phases:

authorization,transaction and update.
 During the first phase, authorization, the user agent sends a username and a password
toauthenticate the user.
 During the second phase, transaction, the useragent retrieves messages; also during this phase,the
user agent can mark messagesfor deletion, remove deletion marks andobtain mailstatistics.
 The third phase,update, occurs after the client has issued the quit command, ending the
POP3session; at this time, the mail server deletes the messages that were marked fordeletion.
 POP3 is designed to delete mail on the serveras soon as the user has downloaded it.
 However, some implementations allow users or an administrator to specify that mail be savedfor
some period of time. POP can be thought of as a "store-and-forward" service.

IMAP(Internet Message Access Protocol)

With POP3 access, oncereceiverhas downloaded his messages to the local machine,he can
create mail folders and move the downloaded messages into the folders.
Receivercan then delete messages, move messages across folders, and search formessages (by
sender name or subject).
But this paradigm—namely, folders andmessages in the local machine—poses a problem for
theroaminguser, whowould prefer to maintain a folder hierarchy on a remote server that can
beaccessed from any computer.
This is not possible with POP3—the POP3 protocoldoes not provide any means for a user to
create remote folders and assign messagesto folders.
An IMAP server will associate each message with a folder; when a message first arrives at the
server, it is associated with the recipient’s INBOX folder.
The recipient can then move the message into a new, user-created folder, read the message,
delete the message, and so on.
The IMAP protocol provides commands to allow users to create folders and move messages
from one folder to another.
IMAP also provides commands that allow users to search remote folders for messages matching
specific criteria.
Another important feature of IMAP is that it has commands that permit a user agent to obtain
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 components of messages. Forexample, a user agent can obtain just the message header of a
message or just one part of a multipart MIME message.
This feature is useful when there is a low-bandwidth connection (for example, a slow-speed
modem link) between the user agent and its mail server.
With a low bandwidth connection, the user may not want to download all of the messages in its
mailbox, particularly avoiding long messages that might contain, for example, an audio or video
clip.

DNS (Domain Name System)

DNS (Domain Name System)isan internet service that translates domain names into IP
addresses.
Because domain names are alphabetic, they're easier to rememberfor human being butThe
Internet is really based on IP addresses.

 Every time you use a domain name, therefore, a DNSservice must translate thedomainnameinto the
corresponding IP address. For example, the domain name www.google.com mighttranslate to
198.105.232.4.
 The DNS system is, in fact, its own network. If one DNS server doesn't know how to translate
aparticular domain name, it asks another one, and so on, until the correct IP address is returned.

Distributed database design is more preferred over centralized design to

implement DNS in the Internet

1.A single point of failure:If the DNS server crashesthenthe entire Internetwill not stop.
2.Traffic volume:Today the Internet is so huge, with millions of device and users accessing
its services from all over the globe atthe same time. A Single DNS Server cannot handle
the huge global DNS trafficbutwith distributed system this trafficis distributedand
reduce overload on server.
3.Distant centralized database:A single DNS server cannot be “close to” all the querying
clients. If we put the single DNS server in New York City, then all queries from Australia
must travel to the other side of the globe, perhaps over slow and congested links. This
can lead to significant delays.
4.Maintenance:The single DNS server would have to keep records for all Internet hosts.
Not only would this centralized database be huge, but it would have to be updated
frequently to account for every new host.

A Distributed, Hierarchical Database

F
i
g
.
1
4
P
o
r
t
i
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 on of the hierarchy of DNS servers
suppose a DNS client wants to determine the IP address for the hostnamewww.amazon.com:
1.The clientfirst contacts one of the root servers, which returns IP addresses for TLD servers,
for the top-level domain com.
2.The client then contacts one of these TLD servers, which returns the IP address of an
authoritative server for amazon.com.
3.Finally, the clientcontacts one of the authoritative servers for amazon.com, which returns
the IP address for the hostnamewww.amazon.com.

Recursive queries and iterative queries

Fig.15Interaction of the various DNS servers

The example shown in abovefigure makes use ofboth recursive queries and iterative queries.
The query1sent from cis.poly.edu to dns.poly.edu is a recursive querybecausethe query asks
dns.poly.edu to obtain the mapping on its behalf.
But the subsequent three queries2, 4 and 6are iterative sinceall of the replies are directly
returned to dns.poly.edu.

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Protocol pipelining
Protocol pipelining is a technique in which multiple requests are written out to a single
socketwithout waiting for the corresponding responses(acknowledged).
Pipelining can be used in various application layer network protocols, like HTTP/1.1,
SMTP and FTP.
Range of sequence numbers must be increased.
 Data or Packet should bebufferedat sender and/or receiver.

 Two generic forms of pipelined protocolsare

1.Go-Back-N
2.Selective repeat
Go-Back-N
Go-Back-N ARQ is a specific instance of the automatic repeat request (ARQ) protocol, in
which the sending process continues to send a number of frames specified by a window
size even without receiving an acknowledgement (ACK) packet from the receiver.
Thereceiver process keeps track of the sequence number of the next frame it expects to
receive, and sends that number with every ACK it sends.
The receiver will discard any frame that does not have the exact sequence number it
expects (either a duplicate frame it already acknowledged or an out-of-order frame it
expects to receive later) and will resend an ACK for the last correct in-order frame.
Once the sender has sent all of the frames in its window, it will detect that all of the
frames since the first lost frame are outstanding, and will go back to the sequence
number of the last ACK it received from the receiver process and fill its window starting
with that frame and continue the process over again.
Go-Back-N ARQ is a more efficient use of a connectionthan Stop-and-wait ARQ, since
unlike waiting for an acknowledgement for each packet; the connection is still being
utilized as packets are being sent.
However, this method also results in sending frames multiple times–if any frame was
lost or damaged or the ACK acknowledging them was lost or damaged then that frame
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 and all following frames in the window (even if they were received without error) will be
re-sent. To avoid this, Selective Repeat ARQ can be used.
How does Go-Back-N ARQ protocol works:
Considerascenariogiven in fig 14there are four frames 0,1,2,3 respectively. Now as
sending window size is 3 itwill send frame 0,1and2 at a time.

Fig. 14 Go-Back-N Protocol

 At receiver side frame 0 arrives it sendsACKfor that.
 Now at sender sidewindow shift at frame 3 and it sends frame no 3.
 At receiver side frame 1 arrives it sendsACKfor that.
 Now at sender side window shift at frame4and it sends frame no4.
 Now receiver detects frame 2 is missing or lost. Itwill send NAK for that.
 Senderreceives NAK at this point sending window is pointing towards frame 2,3 and4so
send will resend frame 2, 3 and 4.
Selective repeat
Selective Repeat attempts to retransmit only thosepackets that are actually lostdueto
errors.
Receiver must be able toaccept packets out of order
Since receiver must release packets to higher layer in order, the receiver must be able to
buffer some packets
The receiver acknowledges every good packet, packets that are not ACKed before a
time-out areassumed lost or in error.
Notice that this approach must be used to be sure that every packet is eventually
received.
An explicit NAK (selective reject) can request retransmission of just one packet.

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 This approach can speed up the retransmission but is not strictly needed.

Fig.15 Selective Repeat

TCP segment structure

F
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1
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s
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  The unit of transmission in TCP is called segments.
 The header includes source and destination port numbers, which are used
formultiplexing/demultiplexing data from/toupper-layer applications.
 The 32-bit sequence number field and the 32-bit acknowledgment number field areused by
the TCP sender and receiver in implementing a reliable datatransfer service.
 The sequence number for a segment is the byte-stream number of the first byte in
thesegment.
 The acknowledgment number is the sequence number of the next byteaHostisexpecting
from anotherHost.
 The 4-bit header length field specifies the length of the TCP header in 32-bitwords. TheTCP
header can be of variablelength due to the TCP options field.
 The 16-bit receive window field is used for flow control.It is used to indicate thenumber of
bytes that a receiver is willing to accept.
 The 16-bit checksum field is used for errorchecking of the header and data.
 Unused6 bitsarereserved for future useand should besenttozero.
 Urgent Pointerisused in combining with the URG control bit for priority data transfer.This
field contains the sequence number of the last byte of urgent data.
 Data: The bytes of data being sent in the segment.
 URG (1 bit):indicates that the Urgent pointer field is significant.
 ACK (1 bit):indicates that the Acknowledgment field is significant.
 PSH (1 bit):Push function. Asks to push the buffered data to the receiving application.
 RST(1 bit):Reset the connection.
 SYN (1 bit):Synchronize sequence numbers. Only the first packet sent from each
endshould have this flag set. Some other flags and fields change meaning based on this
flag,and some are only valid for when it is set, and others when it is clear.
 FIN (1 bit): No more data from sender.

CONGESTION CONTROL, FLOW CONTROL

Flow Control
In data communications, flow control is the process of managing the rate of data
transmission between two nodes to prevent a fast sender from overwhelming a slow
receiver.
Itprevent receiver from becoming overloaded.
Receiver advertises a window rwnd witheach acknowledgement
Window
closed (by sender) when data is sent and ack’d
opened (by receiver) when data is read
The size of this window can be theperformance limit (e.g. on aLAN).

Congestion Control
When a connection is established, a suitable window size has to be chosen.
The receiver can specify a window based on its buffer size.
 If the sender sticks to this window size, problems will not occur due to buffer overflowat
the receiving end, but they may still occur due to internal congestion within thenetwork.
 In Figure17(a), we see a thick pipe leading to a small-capacity receiver.
 As long as the sender does not send more water than the bucket can contain, no waterwill
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 be lost.
 In Figure17(b), the limiting factor is not the bucket capacity, but the internal
carryingcapacity of the network.
Fig. 17TCP segment structure

 Figure 17: (a)afast networkfeeding a low capacity receiver.Figure 17:(b) A slownetwork

feeding a high-capacity receiver.
 If too much water comes in too fast, it will back up and some will be lost (in this case
byoverflowing the funnel).
 Each sender maintains two windows: the window the receiver has granted and a
secondwindow, the congestion window.
 Each reflects the number of bytes the sender may transmit.
 The number of bytes that may be sent is the minimum of the two windows.
 Thus, the effective window is the minimum of what the sender thinks is all right andwhat
the receiver thinks is all right.
 When a connection is established, the sender initializes the congestion window to thesize
of the maximum segment in use on theconnection.
 It then sends one maximum segment.

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  If this segment is acknowledged before the timer goes off, it adds one segment's worthof
bytes to the congestion window to make it two maximum size segments and sendstwo
segments.
 As each of these segments is acknowledged, the congestion window is increased by one
maximum segment size.
 When the congestion window is n segments, if all n are acknowledged on time,
thecongestion window is increased by the byte count corresponding to n segments.

TCP Slow Start

Slow-start is part of the congestion control strategy used by TCP, the data transmission
protocol used by many Internet applications.
Slow-start is used in conjunction withother algorithms to avoid sending more datathan
the network is capable oftransmittingto avoid causing network congestion.

Fig. 18 TCPSlow start

Slow-start begins initially with a congestion window Size (cwnd) of 1, 2 or 10.

The valueof the Congestion Window will be increased with each acknowledgement
(ACK) received, effectively doubling the window size each round trip time ("although it is
not exactly exponential because the receiver may delay its ACKs, typically sending one
ACK for every two segments that it receives).
The transmission rate will be increased with slow-start algorithm until either a loss is
detected, or the receiver's advertised window (rwnd) is the limiting factor, or the slow
start threshold (ssthresh) is reached.
If a loss event occurs, TCP assumes that it is due to network congestion and takes steps

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 to reduce the offered load on the network.
 Once ssthresh is reached, TCP changes from slow-start algorithm to the linear
growth(congestion avoidance) algorithm. At this point, the window is increased by 1
segmentfor each RTT.

********************** Part 2 Stops Here****************

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