III BCA - ODD Semester - Lecturer Notes- COMPUTER NETWORKS – USA20502J - Reg-2020

 COMPUTER NETWORKS – USA20502J

UNIT 1

Computer Networks

A computer network, also referred to as a data network, is a series of interconnected nodes that can transmit, receive and exchange data, voice and video traffic. Examples of nodes in a network include servers or modems. Computer networks commonly help endpoint users share resources and communicate.

Two or more computers connected that allows sharing their data, resources, and application is called a computer network. The vital computer network is divided into four types based on their size and functions. They are LAN, MAN, PAN, and WAN.

Evolution of Networks

Advancement of systems administration began path back in 1969’s by the improvement of first system called ARPANET, which prompted the improvement of web. At that point after constantly everyday upgradation occur in the system innovation. The system has gone through a few phases which are described below:

ARPANET (Advanced Research Agency Network):

          ARPANET was the network that became the basis for the Internet. It was the first network that came into existence in 1969, which was designed and named by the Advanced Research Projects Agency (ARPA) and US Department of Defence (DoD). It was where a bunch of PCs were associated at various colleges and US DoD for sharing of information and messages and playing long separation diversions and associating with individuals to share their perspectives.

 

NSFNET (National Science Federation Network):

         In mid 80’s another federal agency, NSFNET (National Science Federation Network) created a new network which was more capable than ARPANET and became the first backbone infrastructure for the commercial public Internet.

 

ARPANET + NSFNET + PRIVATE NETWORKS = INTERNET  

 

The Internet:

The Internet provides different online services. Some examples include: Web – a collection of billions of webpages that you can view with a web browser. Email – the most common method of sending and receiving messages online. Social media – websites and apps that allow people to share comments, photos, and videos.

In the Internet, which is a network of networks, came into existence. The internet has evolved from ARPANET. The internet is a globally connected network system that utilizes TCP/IP to transmit information. It allows computers of different types to exchange information and is known as internet. The Internet is the financially communications method on the planet, in which the following services are instantly available

·         Email

·         Web-enabled audio/video conferencing services

·         Online movies and gaming

·         Data transfer/file-sharing, often through File Transfer Protocol (FTP)

·         Instant messaging

·         Internet forums

·         Social networking

·         Online shopping

·         Financial services

 

Interspace:
Interspaces is a software that allows multiple users in a client-server environment to communicate with each other to send and receive data of various types such as data files, video, audio and textual data. Interspaces give the most exceptional type form of communication available on the Internet today.

 

Data Communication:  

The term telecommunication means communication at a distance. The word data refers to information presented in whatever form is agreed upon by the parties creating and using the data. Data communications are the exchange of data between two devices via some form of transmission medium such as a wire cable

Components of a data communications system

5 components of data communication network

Ø  Data.

Ø  Sender.

Ø  Receiver.

Ø  Transmission Medium.

Ø  Protocol.

1. Message:

·         The message is the information (data) to be communicated. Popular forms of information include text, numbers, pictures, audio, and video.

2. Sender:

·         The sender is the device that sends the data message. It can be a computer, workstation, telephone handset, video camera, and so on. 

3. Receiver:

·         The receiver is the device that receives the message. It can be a computer, workstation, telephone handset, television, and so on.

 

4. Transmission medium:

·         The transmission medium is the physical path by which a message travels from sender to receiver. Some examples of transmission media include twisted-pair wire, coaxial cable, fiber-optic cable, and radio waves.

5. Protocol:

·         A protocol is a set of rules that govern data communications. It represents an agreement between the communicating devices. Without a protocol, two devices may be connected but not communicating, just as a person speaking French cannot be understood by a person who speaks only Japanese.

 Data flow (simplex, half-duplex, and full-duplex)

Networks

A network is a set of devices (often referred to as nodes) connected by communication links. A node can be a computer, printer, or any other device capable of sending and/or receiving data generated by other nodes on the network. A link can be a cable, air, optical fiber, or any medium which can transport a signal carrying information.

A network is a collection of computers, servers, mainframes, network devices, peripherals, or other devices connected to allow data sharing. An example of a network is the Internet, which connects millions of people all over the world.

Network Criteria

Ø  Performance

                                                                    I.            Depends on Network Elements

                                                                 II.            Measured in terms of Delay and Throughput

Ø  Reliability

                                                                    I.            Failure rate of network components

                                                                 II.            Measured in terms of availability/robustness

Ø  Security

                                                                    I.            Data protection against corruption/loss of data due to:

                                                                 II.            Errors

 Physical Structures

Ø   Type of Connection

                                            I.            Point to Point - single transmitter and receiver

                                         II.            Multipoint - multiple recipients of single transmission

Ø  Physical Topology

                                            I.            Connection of devices

                                         II.            Type of transmission - unicast, mulitcast, broadcast

Categories of Networks

PERSONAL AREA NETWORK (PAN) ...

This is the smallest and most basic network that you’ll find. It’s meant to cover a very small area (typically a single room or building). 

LOCAL AREA NETWORK (LAN) ...

This is an extremely common and well-known type of network. Just as the name suggests, a LAN connects a group of computers or devices together across a local area. This type of network can be utilized to connect devices throughout one building or even 2-3 buildings depending on the proximity to each other

WIRELESS LOCAL AREA NETWORK (WLAN) .

A WLAN is simply a LAN that does not rely on cables to connect to the network. So, when you’re using WiFi, you’re using a WLAN. WLANs are typically used in the same scenario as LANs, it just depends on whether you’d prefer an on premises or remote cloud solution (wires or wireless)...

METROPOLITAN AREA NETWORK (MAN) ...

Larger than a LAN but smaller than a WAN, a MAN incorporates elements of both types of networks. It connects multiple LANs together and spans an entire geographical area such as a city or town (or sometimes a campus). Ownership and management can be handled by a single person, but it’s more likely done by a larger company or organization.

WIDE AREA NETWORK (WAN) ...

Like LANs, you very well may recognize the term “WAN.” WANs do the same thing as LANs but across a larger area while connecting more devices. Even when miles apart, a WAN can connect devices together remotely. In fact, the most basic example of a WAN is the Internet which connects computers and devices worldwide.

STORAGE AREA NETWORK (SAN) ...

A SAN is another type of LAN that’s designed to handle large data transfers and storage. This purpose of this network is to move larger, more complex storage resources away from the network into a separate, high-performance atmosphere. Doing this not only allows for easy retrieval and storage of the data but it also frees up space and improves overall performance of the original network.

VIRTUAL PRIVATE NETWORK (VPN)

The point of a VPN is to increase security and privacy while accessing a network. The VPN acts as a middleman between you and the network by encrypting your data and hiding your identity. This is a great option for sending and receiving sensitive information, however, using a VPN is ideal anytime you connect to the Internet. 

Types of connections: point-to-point and multipoint

Types of topologies

Ø  Bus Topology.

Ø  Ring Topology.

Ø  Star Topology.

Ø  Mesh Topology.

Ø  Tree Topology.

Ø  Hybrid Topology.

 

Protocols

A protocol is synonymous with rule. It consists of a set of rules that govern data communications. It determines what is communicated, how it is communicated and when it is communicated. The key elements of a protocol are syntax, semantics and timing

Elements of a Protocol

n  Syntax

n  Structure or format of the data

n  Indicates how to read the bits - field delineation

n  Semantics

n  Interprets the meaning of the bits

n  Knows which fields define what action

n  Timing

n  When data should be sent and what

n  Speed at which data should be sent or speed at which it is being received.

Network Models

 

 

OSI NETWORK MODEL (Open Systems Interconnection)

Layers of OSI Model

OSI stands for Open Systems Interconnection. It has been developed by ISO – ‘International Organization for Standardization‘, in the year 1984. It is a 7 layer architecture with each layer having specific functionality to perform. All these 7 layers work collaboratively to transmit the data from one person to another across the globe

LAYER Wise Protocols

1. Physical Layer (Layer 1) :

The lowest layer of the OSI reference model is the physical layer. It is responsible for the actual physical connection between the devices. The physical layer contains information in the form of bits. It is responsible for transmitting individual bits from one node to the next. When receiving data, this layer will get the signal received and convert it into 0s and 1s and send them to the Data Link layer, which will put the frame back together.

The functions of the physical

1.      Bit synchronization: The physical layer provides the synchronization of the bits by providing a clock. This clock controls both sender and receiver thus providing synchronization at bit level.

2.      Bit rate control: The Physical layer also defines the transmission rate i.e. the number of bits sent per second.

3.      Physical topologies: Physical layer specifies the way in which the different, devices/nodes are arranged in a network i.e. bus, star, or mesh topology.

4.      Transmission mode: Physical layer also defines the way in which the data flows between the two connected devices. The various transmission modes possible are Simplex, half-duplex and full-duplex.

 

Data Link Layer (DLL) (Layer 2) :

The data link layer is responsible for the node-to-node delivery of the message. The main function of this layer is to make sure data transfer is error-free from one node to another, over the physical layer. When a packet arrives in a network, it is the responsibility of DLL to transmit it to the Host using its MAC address. 
Data Link Layer is divided into two sublayers:  

1.      Logical Link Control (LLC)

2.      Media Access Control (MAC)

 

The functions of the Data Link layer

1.      Framing: Framing is a function of the data link layer. It provides a way for a sender to transmit a set of bits that are meaningful to the receiver. This can be accomplished by attaching special bit patterns to the beginning and end of the frame.

2.      Physical addressing: After creating frames, the Data link layer adds physical addresses (MAC address) of the sender and/or receiver in the header of each frame.

3.      Error control: Data link layer provides the mechanism of error control in which it detects and retransmits damaged or lost frames.

4.      Flow Control: The data rate must be constant on both sides else the data may get corrupted thus, flow control coordinates the amount of data that can be sent before receiving acknowledgement.

5.      Access control: When a single communication channel is shared by multiple devices, the MAC sub-layer of the data link layer helps to determine which device has control over the channel at a given time.

 

3. Network Layer (Layer 3) :

The network layer works for the transmission of data from one host to the other located in different networks. It also takes care of packet routing i.e. selection of the shortest path to transmit the packet, from the number of routes available. The sender & receiver’s IP addresses are placed in the header by the network layer. 

The functions of the Network layer

1.      Routing: The network layer protocols determine which route is suitable from source to destination. This function of the network layer is known as routing.

2.      Logical Addressing: In order to identify each device on internetwork uniquely, the network layer defines an addressing scheme. The sender & receiver’s IP addresses are placed in the header by the network layer. Such an address distinguishes each device uniquely and universally.

3.      Internetworking: An internetworking is the main responsibility of the network layer. It provides a logical connection between different devices.

4.      Packetizing: A Network Layer receives the packets from the upper layer and converts them into packets. This process is known as Packetizing. It is achieved by internet protocol (IP)

4. Transport Layer (Layer 4) :

The transport layer provides services to the application layer and takes services from the network layer. The data in the transport layer is referred to as Segments. It is responsible for the End to End Delivery of the complete message.

The transport layer also provides the acknowledgement of the successful data transmission and re-transmits the data if an error is found.

At sender’s side: Transport layer receives the formatted data from the upper layers, performs Segmentation, and also implements Flow & Error control to ensure proper data transmission. It also adds Source and Destination port numbers in its header and forwards the segmented data to the Network Layer. 

At receiver’s side: Transport Layer reads the port number from its header and forwards the Data which it has received to the respective application. It also performs sequencing and reassembling of the segmented data. 

 

The functions of the transport layer

1.      Segmentation and Reassembly: This layer accepts the message from the (session) layer, and breaks the message into smaller units. Each of the segments produced has a header associated with it. The transport layer at the destination station reassembles the message.

2.      Service Point Addressing: In order to deliver the message to the correct process, the transport layer header includes a type of address called service point address or port address. Thus by specifying this address, the transport layer makes sure that the message is delivered to the correct process.

 

The services provided by the transport layer

A. Connection-Oriented Service: It is a three-phase process that includes 

–Connection Establishment 
–Data Transfer 
– Termination / disconnection 

In this type of transmission, the receiving device sends an acknowledgement, back to the source after a packet or group of packets is received. This type of transmission is reliable and secure.

B. Connectionless service: It is a one-phase process and includes Data Transfer. In this type of transmission, the receiver does not acknowledge receipt of a packet. This approach allows for much faster communication between devices. Connection-oriented service is more reliable than connectionless Service.

 

5. Session Layer (Layer 5) :

This layer is responsible for the establishment of connection, maintenance of sessions, authentication, and also ensures security

. 
The functions of the session layer :

1.      Session establishment, maintenance, and termination: The layer allows the two processes to establish, use and terminate a connection.

2.      Synchronization: This layer allows a process to add checkpoints which are considered synchronization points into the data. These synchronization points help to identify the error so that the data is re-synchronized properly, and ends of the messages are not cut prematurely and data loss is avoided.

3.      Dialog Controller: The session layer allows two systems to start communication with each other in half-duplex or full-duplex.

6. Presentation Layer (Layer 6):

 

The presentation layer is also called the Translation layer. The data from the application layer is extracted here and manipulated as per the required format to transmit over the network. 

The functions of the presentation layer

●        Translation: For example, ASCII to EBCDIC.

●        Encryption/ Decryption: Data encryption translates the data into another form or code. The encrypted data is known as the ciphertext and the decrypted data is known as plain text. A key value is used for encrypting as well as decrypting data.

●        Compression: Reduces the number of bits that need to be transmitted on the network.

7. Application Layer (Layer 7) :

At the very top of the OSI Reference Model stack of layers, we find the Application layer which is implemented by the network applications. These applications produce the data, which has to be transferred over the network. This layer also serves as a window for the application services to access the network and for displaying the received information to the user. 

Example: Application – Browsers, Skype Messenger, etc. 

The functions of the Application layer are :  

1.      Network Virtual Terminal

2.      FTAM-File transfer access and management

3.      Mail Services

4.      Directory Services

 

TCP/IP model

• The TCP/IP model was developed prior to the OSI model.
• The TCP/IP model is not exactly similar to the OSI model.
• The TCP/IP model consists of five layers: the application layer, transport layer, network layer, data link layer and physical layer.
• The first four layers provide physical standards, network interface, internetworking, and transport functions that correspond to the first four layers of the OSI model and these four layers are represented in TCP/IP model by a single layer called the application layer.
• TCP/IP is a hierarchical protocol made up of interactive modules, and each of them provides specific functionality.

Here, hierarchical means that each upper-layer protocol is supported by two or more lower-level protocols.

Functions of TCP/IP layers:

 

Network Access Layer

• A network layer is the lowest layer of the TCP/IP model.
• A network layer is the combination of the Physical layer and Data Link layer defined in the OSI reference model.
• It defines how the data should be sent physically through the network.
• This layer is mainly responsible for the transmission of the data between two devices on the same network.
• The functions carried out by this layer are encapsulating the IP datagram into frames transmitted by the network and mapping of IP addresses into physical addresses.
• The protocols used by this layer are ethernet, token ring, FDDI, X.25, frame relay.

Internet Layer

• An internet layer is the second layer of the TCP/IP model.
• An internet layer is also known as the network layer.
• The main responsibility of the internet layer is to send the packets from any network, and they arrive at the destination irrespective of the route they take.

Following are the protocols used in this layer are:

IP Protocol: IP protocol is used in this layer, and it is the most significant part of the entire TCP/IP suite.

Following are the responsibilities of this protocol:

• IP Addressing: This protocol implements logical host addresses known as IP addresses. The IP addresses are used by the internet and higher layers to identify the device and to provide internetwork routing.
• Host-to-host communication: It determines the path through which the data is to be transmitted.
• Data Encapsulation and Formatting: An IP protocol accepts the data from the transport layer protocol. An IP protocol ensures that the data is sent and received securely, it encapsulates the data into message known as IP datagram.
• Fragmentation and Reassembly: The limit imposed on the size of the IP datagram by data link layer protocol is known as Maximum Transmission unit (MTU). If the size of IP datagram is greater than the MTU unit, then the IP protocol splits the datagram into smaller units so that they can travel over the local network. Fragmentation can be done by the sender or intermediate router. At the receiver side, all the fragments are reassembled to form an original message.
• Routing: When IP datagram is sent over the same local network such as LAN, MAN, WAN, it is known as direct delivery. When source and destination are on the distant network, then the IP datagram is sent indirectly. This can be accomplished by routing the IP datagram through various devices such as routers.

ARP and RARP Protocols Java Program

Stay

• ARP stands for Address Resolution Protocol.
• ARP is a network layer protocol which is used to find the physical address from the IP address.
• The two terms are mainly associated with the ARP Protocol:
• ARP request: When a sender wants to know the physical address of the device, it broadcasts the ARP request to the network.
• ARP reply: Every device attached to the network will accept the ARP request and process the request, but only recipient recognize the IP address and sends back its physical address in the form of ARP reply. The recipient adds the physical address both to its cache memory and to the datagram header.

Reverse Address Resolution Protocol (RARP) is a protocol a physical machine in a local area network (LAN) can use to request its IP address. It does this by sending the device's physical address to a specialized RARP server that is on the same LAN and is actively listening for RARP requests.

ICMP Protocol

• ICMP stands for Internet Control Message Protocol.
• It is a mechanism used by the hosts or routers to send notifications regarding datagram problems back to the sender.
• A datagram travels from router-to-router until it reaches its destination. If a router is unable to route the data because of some unusual conditions such as disabled links, a device is on fire or network congestion, then the ICMP protocol is used to inform the sender that the datagram is undeliverable.
• An ICMP protocol mainly uses two terms:
• ICMP Test: ICMP Test is used to test whether the destination is reachable or not.
• ICMP Reply: ICMP Reply is used to check whether the destination device is responding or not.
• The core responsibility of the ICMP protocol is to report the problems, not correct them. The responsibility of the correction lies with the sender.
• ICMP can send the messages only to the source, but not to the intermediate routers because the IP datagram carries the addresses of the source and destination but not of the router that it is passed to.

IGMP

• The Internet Group Management Protocol (IGMP) manages the membership of hosts and routing devices in multicast groups. IP hosts use IGMP to report their multicast group memberships to any immediately neighboring multicast routing devices.

Transport Layer

The transport layer is responsible for the reliability, flow control, and correction of data which is being sent over the network.

The two protocols used in the transport layer are User Datagram protocol and Transmission control protocol.

• User Datagram Protocol (UDP)
• It provides connectionless service and end-to-end delivery of transmission.
• It is an unreliable protocol as it discovers the errors but not specify the error.
• User Datagram Protocol discovers the error, and ICMP protocol reports the error to the sender that user datagram has been damaged.

UDP consists of the following fields:
Source port address: The source port address is the address of the application program that has created the message.
Destination port address: The destination port address is the address of the application program that receives the message.
Total length: It defines the total number of bytes of the user datagram in bytes.
Checksum: The checksum is a 16-bit field used in error detection.

UDP does not specify which packet is lost. UDP contains only checksum; it does not contain any ID of a data segment.

 

Transmission Control Protocol (TCP)

o It provides a full transport layer services to applications.

o It creates a virtual circuit between the sender and receiver, and it is active for the duration of the transmission.

o TCP is a reliable protocol as it detects the error and retransmits the damaged frames. Therefore, it ensures all the segments must be received and acknowledged before the transmission is considered to be completed and a virtual circuit is discarded.

o At the sending end, TCP divides the whole message into smaller units known as segment, and each segment contains a sequence number which is required for reordering the frames to form an original message.

o At the receiving end, TCP collects all the segments and reorders them based on sequence numbers.

Application Layer

• An application layer is the topmost layer in the TCP/IP model.
• It is responsible for handling high-level protocols, issues of representation.
• This layer allows the user to interact with the application.
• When one application layer protocol wants to communicate with another application layer, it forwards its data to the transport layer.
• There is an ambiguity occurs in the application layer. Every application cannot be placed inside the application layer except those who interact with the communication system. For example: text editor cannot be considered in application layer while web browser using HTTP protocol to interact with the network where HTTP protocol is an application layer protocol.

Following are the main protocols used in the application layer:

• HTTP: HTTP stands for Hypertext transfer protocol. This protocol allows us to access the data over the world wide web. It transfers the data in the form of plain text, audio, video. It is known as a Hypertext transfer protocol as it has the efficiency to use in a hypertext environment where there are rapid jumps from one document to another.
• SNMP: SNMP stands for Simple Network Management Protocol. It is a framework used for managing the devices on the internet by using the TCP/IP protocol suite.
• SMTP: SMTP stands for Simple mail transfer protocol. The TCP/IP protocol that supports the e-mail is known as a Simple mail transfer protocol. This protocol is used to send the data to another e-mail address.
• DNS: DNS stands for Domain Name System. An IP address is used to identify the connection of a host to the internet uniquely. But, people prefer to use the names instead of addresses. Therefore, the system that maps the name to the address is known as Domain Name System.
• TELNET: It is an abbreviation for Terminal Network. It establishes the connection between the local computer and remote computer in such a way that the local terminal appears to be a terminal at the remote system.
• FTP: FTP stands for File Transfer Protocol. FTP is a standard internet protocol used for transmitting the files from one computer to another computer.

Peer to Peer Approach

A peer-to-peer network is designed around the notion of equal peer nodes simultaneously functioning as both "clients" and "servers" to the other nodes on the network. This model of network arrangement differs from the client–server model where communication is usually to and from a central server.

Types of P2P networks:

 

1.      Unstructured P2P networks –
In this type of P2P network, each device is able to make an equal contribution. This network is easy to build as devices can be connected randomly in the network. But being unstructured, it becomes difficult to find content.

2.      Structured P2P networks –
It is designed using the software which creates a virtual layer in order to put the nodes in a specific structure. These are not easy to set-up but can give easy access to users to the content.

3.      Hybrid P2P networks –
It combines the features of both P2P network and client-server architecture. An example of such a network is to find a node using the central server.

 

 

Advantages of P2P Network :

·         Network is easy to maintain because each node is independent of each other.

·         Since each node acts as a server, therefore the cost of the central server is saved.

·         Adding, deleting and repairing nodes in this network is easy.

 

Disadvantages of P2P Network :

·         Because of no central server, data is always vulnerable to get lost because of no backup.

·         It becomes difficult to secure the complete network because each node is independent.

 

UNIT -2

 Network Addressing

A computer network is a group of some interconnected computers that are sharing a common or different resources provided on or by network nodes. These sharing or communication between the machines is governed by some set of rules or network protocols. These computers or machines are identified by network addresses, and may have hostnames.

A Network Address is a logical or physical address that uniquely identifies a host or a machine in a telecommunication network. A network may also not be unique and can contain some structural and hierarchical information of the node in the network. Internet protocol (IP) address, media access control (MAC) address and telephone numbers are some basic examples of network addresses. It can be of numeric type or symbolic or both in some cases.

Four types of addressing methods used are the following:

§  Physical address

§  Logical address (IP)

§  Port address and

§  Specific address

 

Four levels of addresses are used in the TCP/IP protocol: physical address, logical address, port address, and application-specific address as shown in Figure.

MAC Addresses (Physical Address)

•  The physical address, also known as the link address, is the address of a node as defined by its LAN or WAN.
• The size and format of these addresses vary depending on the network. For example, Ethernet uses a 6-byte (48-bit) physical address.
• Physical addresses can be either unicast (one single recipient), multicast (a group of recipients), or broadcast (to be received by all systems in the network.
• Example: Most local area networks use a 48-bit (6-byte) physical address written as 12 hexadecimal digits; every byte (2 hexadecimal digits) is separated by a colon, as shown below: A 6-byte (12 hexadecimal digits) physical address 07:01:02:01:2C:4B

Logical Addresses ((IP Addresses)

• Logical addresses are used by networking software to allow packets to be independent of the physical connection of the network, that is, to work with different network topologies and types of media.
• A logical address in the Internet is currently a 32-bit address that can uniquely define a host connected to the Internet. An internet address in IPv4 in decimal numbers 132.24.75.9
• No two publicly addressed and visible hosts on the Internet can have the same IP address.
• The physical addresses will change from hop to hop, but the logical addresses remain the same.
• The logical addresses can be either unicast (one single recipient), multicast (a group of recipients), or broadcast (all systems in the network). There are limitations on broadcast addresses.

Port Addresses

• There are many application running on the computer. Each application run with a port no.(logically) on the computer.
• A port number is part of the addressing information used to identify the senders and receivers of messages.
• Port numbers are most commonly used with TCP/IP connections.
• These port numbers allow different applications on the same computer to share network resources simultaneously.
• The physical addresses change from hop to hop, but the logical and port addresses usually remain the same.
• Example: a port address is a 16-bit address represented by one decimal number 753

Application-Specific Addresses

• Some applications have user-friendly addresses that are designed for that specific application.
• Examples include the e-mail address (for example, forouzan@fhda.edu) and the Universal Resource Locator (URL) (for example, www.mhhe.com). The first defines the recipient of an e-mail; the second is used to find a document on the World Wide Web.

IPv4 addresses

IPv4 addresses are 32-bit numbers that are typically displayed in dotted decimal notation. A 32-bit address contains two primary parts: the network prefix and the host number. All hosts within a single network share the same network address. Each host also has an address that uniquely identifies it.

IP (version 4) addresses are 32-bit integers that can be expressed in hexadecimal notation. The more common format, known as dotted quad or dotted decimal, is x.x.x.x, where each x can be any value between 0 and 255. For example, 192.0. 2.146 is a valid IPv4 address.

  IP stands for Internet Protocol and v4 stands for Version Four (IPv4). IPv4 was the primary version brought into action for production within the ARPANET in 1983. 
    IP version four addresses are 32-bit integers which will be expressed in decimal notation. 
Example- 192.0.2.126 could be an IPv4 address. 

 

 

Parts of IPv4

·         Networkpart: 
The network part indicates the distinctive variety that’s appointed to the network. The network part conjointly identifies the category of the network that’s assigned.

·         HostPart: 
The host part uniquely identifies the machine on your network. This part of the IPv4 address is assigned to every host. 
For each host on the network, the network part is the same, however, the host half must vary.

·         Subnet number: 

This is the nonobligatory part of IPv4. Local networks that have massive numbers of hosts are divided into subnets and subnet numbers are appointed to that.

 

Characteristics of IPv4

·         IPv4 could be a 32-Bit IP Address.

·         IPv4 could be a numeric address, and its bits are separated by a dot.

·         The number of header fields is twelve and the length of the header field is twenty.

·         It has Unicast, broadcast, and multicast style of addresses.

·         IPv4 supports VLSM (Virtual Length Subnet Mask).

·         IPv4 uses the Post Address Resolution Protocol to map to the MAC address.

·         RIP may be a routing protocol supported by the routed daemon.

·         Networks ought to be designed either manually or with DHCP.

·         Packet fragmentation permits from routers and causing host.

 

Advantages of IPv4

·         IPv4 security permits encryption to keep up privacy and security.

·         IPV4 network allocation is significant and presently has quite 85000 practical routers.

·         It becomes easy to attach multiple devices across an outsized network while not NAT.

·         This is a model of communication so provides quality service also as economical knowledge transfer.

·         IPV4 addresses are redefined and permit flawless encoding.

·         Limits net growth for existing users and hinders the use of the net for brand new users.

·         Internet Routing is inefficient in IPv4.

·         IPv4 has high System Management prices and it’s labor-intensive, complex, slow & frequent to errors.

·         Security features are nonobligatory.

·         Difficult to feature support for future desires as a result of adding it on is extremely high overhead since it hinders the flexibility to attach everything over IP.

There are four different types of IP addresses: public, private, static, and dynamic.

Internet Protocol hierarchy contains several classes of IP Addresses to be used efficiently in various situations as per the requirement of hosts per network. Broadly, the      IPv4 Addressing system is divided into five classes of IP Addresses. All the five classes are identified by the first octet of IP Address.

Internet Corporation for Assigned Names and Numbers is responsible for assigning IP addresses.

The first octet referred here is the left most of all. The octets numbered as follows depicting dotted decimal notation of IP Address −

The number of networks and the number of hosts per class can be derived by this formula −

When calculating hosts' IP addresses, 2 IP addresses are decreased because they cannot be assigned to hosts, i.e. the first IP of a network is network number and the last IP is reserved for Broadcast IP.

Network Addressing:

It is the prime responsibility of the network layer to assign unique addresses to different nodes in a network. As mentioned earlier they can be physical or logical but primarily they are logical addresses i.e. software-based addresses.  The most widely used network address is an IP address. It uniquely identifies a node in an IP network.

An IP address is a 32-bit long numeric address represented in a form of dot-decimal notation where each byte is written in a decimal form separated by a period. For example 196.32.216.9 is an IP address where 196 represents first 8 bits, 32 next 8 bits and so on. The first three bytes of an IP address represents the network and the last byte specifies the host in the network. An IP address is further divided into sub classes

Classful Addressing:

Classful addressing is an IPv4 addressing architecture that divides addresses into five groups.

Prior to classful addressing, the first eight bits of an IP address defined the network a given host was a part of. This would have had the effect of limiting the internet to just 254 networks. Each of those networks contained 16,777,216 different IP addresses. As the internet grew, the inefficiency of allocating IP addresses this way became a problem. After all, there are a lot more than 254 organizations that need IP addresses, and a lot fewer networks that need 16.7 million IP addresses to themselves.

·         Class A : An IP address is assigned to those networks that include large number of hosts.

·         Class B : An IP address is assigned to networks range from small sized to large sized.

·         Class C : An IP address is assigned to networks that are small sized.

·         Class D : IP address are reserved for multicast address and does not possess subnetting.

·         Class E :  An IP address is used for the future use and for the research and development purposes and does not possess any subnetting.

 

An IP address is divided into two parts:

1.      Network ID :  represents the number of networks.

2.      Host ID : represents the number of hosts

.

In the above diagram, we observe that each class have a specific range of IP addresses. The class of IP address is used to determine the number of bits used in a class and number of networks and hosts available in the class.

Class A

In Class A, an IP address is assigned to those networks that contain a large number of hosts.

• The network ID is 8 bits long.
• The host ID is 24 bits long.

In Class A, the first bit in higher order bits of the first octet is always set to 0 and the remaining 7 bits determine the network ID. The 24 bits determine the host ID in any network.

The total number of networks in Class A = 2⁷ = 128 network address

The total number of hosts in Class A = 2²⁴ - 2 = 16,777,214 host address

Class B

In Class B, an IP address is assigned to those networks that range from small-sized to large-sized networks.

• The Network ID is 16 bits long.
• The Host ID is 16 bits long.

In Class B, the higher order bits of the first octet is always set to 10, and the remaining14 bits determine the network ID. The other 16 bits determine the Host ID.

The total number of networks in Class B = 2¹⁴ = 16384 network address

The total number of hosts in Class B = 2¹⁶ - 2 = 65534 host address

Class C

In Class C, an IP address is assigned to only small-sized networks.

• The Network ID is 24 bits long.
• The host ID is 8 bits long.

In Class C, the higher order bits of the first octet is always set to 110, and the remaining 21 bits determine the network ID. The 8 bits of the host ID determine the host in a network.

The total number of networks = 2²¹ = 2097152 network address

The total number of hosts = 2⁸ - 2 = 254 host address

Class D

In Class D, an IP address is reserved for multicast addresses. It does not possess subnetting. The higher order bits of the first octet are always set to 1110, and the remaining bits determine the host ID in any network.

Class E

In Class E, an IP address is used for the future use or for the research and development purposes. It does not possess any subnetting. The higher order bits of the first octet is always set to 1111, and the remaining bits determines the host ID in any network.

Rules for assigning Host ID:

The Host ID is used to determine the host within any network. The Host ID is assigned based on the following rules:

• The Host ID must be unique within any network.
• The Host ID in which all the bits are set to 0 cannot be assigned as it is used to represent the network ID of the IP address.
• The Host ID in which all the bits are set to 1 cannot be assigned as it is reserved for the multicast address.

Rules for assigning Network ID:

If the hosts are located within the same local network, then they are assigned with the same network ID. The following are the rules for assigning Network ID:

• The network ID cannot start with 127 as 127 is used by Class A.
• The Network ID in which all the bits are set to 0 cannot be assigned as it is used to specify a particular host on the local network.
• The Network ID in which all the bits are set to 1 cannot be assigned as it is reserved for the multicast address.

Classful addressing works

Classful addressing divides the IPv4 address space (0.0.0.0-255.255.255.255) into 5 classes: A, B, C, D, and E. However, only A, B, and C are used for network hosts. Class D, which covers the 224.0.0.0-239.255.255.255 IP address range, is reserved for multicasting, and class E (240.0.0.0-255.255.255.255) is reserved for “future use.”

 

IPv4 address class	Network Mask	Number of IPv4 Networks	Number of IPv4 addresses per network	IPv4 address range
A	255.0.0.0	128	16,777,214	0.0.0.0 – 127.255.255.255
B	255.255.0.0	16,384	65,534	128.0.0.0 – 191.255.255.255
C	255.255.255.0	2,097,152	254	192.0.0.0 – 223.255.255.255

 

Limitations of classful IP addressing

o    As you can probably guess, the internet is hungry for IP addresses. While classful IP addressing was much more efficient than the older “first 8-bits” method of chopping up the IPv4 address space, it still wasn’t enough to keep up with growth.

o    As internet popularity continued to surge past 1981, it became clear that allocating blocks of 16,777,216, 65,536, or 256 addresses simply wasn’t sustainable. Addresses were being wasted in too-large blocks, and it was clear there’d be a tipping point where we ran out of IP address space altogether.

o    One of the best ways to understand why this was a problem is to consider an organization that needed a network just slightly bigger than a Class C.

Classless addressing:

 

Classless addressing is an IPv4 addressing architecture that uses variable-length subnet masking.

Classless addressing is an IPv4 addressing architecture that uses variable-length subnet masking. The solution would come in 1993, as Classless Inter-Domain Routing (CIDR) introducing the concept of classless addressing. You see, with classful addressing, the size of networks is fixed.

Using classless addressing and VLSM, addresses can be allocated much more efficiently. This is because network admins get to pick network masks, and in turn, blocks of IP addresses that are the right size for any purpose.

Classless Inter-Domain Routing (CIDR) is another name for classless addressing. This addressing type aids in the more efficient allocation of IP addresses. This technique assigns a block of IP addresses based on specified conditions when the user demands a specific amount of IP addresses. This block is known as a "CIDR block", and it contains the necessary number of IP addresses.

Ø  Classless Addressing is an improved IP Addressing system.

Ø  It makes the allocation of IP Addresses more efficient.

Ø  It replaces the older classful addressing system based on classes.

Ø  It is also known as Classless Inter Domain Routing (CIDR).

CIDR Block-

When a user asks for specific number of IP Addresses,

·         CIDR dynamically assigns a block of IP Addresses based on certain rules.

·         This block contains the required number of IP Addresses as demanded by the user.

·         This block of IP Addresses is called as a CIDR block.

When allocating a block, classless addressing is concerned with the following three rules.

Rules For Creating CIDR Block-

·         Rule 1 − The CIDR block's IP addresses must all be contiguous.

·         Rule 2 − The block size must be a power of two to be attractive. Furthermore, the block's size is equal to the number of IP addresses in the block.

·         Rule 3 − The block's first IP address must be divisible by the block size

Network Address and Mask

Network address – It identifies a network on internet.  Using this, we can find range of addresses in the network and total possible number of hosts in the network.

Mask – It is a 32-bit binary number that gives the network address in the address block when AND operation is bitwise applied on the mask and any IP address of the block.

The default mask in different classes are :

Class A – 255.0.0.0

Class B – 255.255.0.0

Class C – 255.255.255.0

Example : Given IP address 132.6.17.85 and default class B mask, find the beginning address (network address).

Solution : The default mask is 255.255.0.0, which means that the only the first 2 bytes are preserved and the other 2 bytes are set to 0. Therefore, the network address is 132.6.0.0.

Subnetting:  Dividing a large block of addresses into several contiguous sub-blocks and assigning these sub-blocks to different smaller networks is called subnetting. It is a practice that is widely used when classless addressing is done.

 

Some values calculated in subnetting :

 

1. Number of subnets : Given bits for mask – No. of bits in default mask

2. Subnet address : AND result of subnet mask and the given IP address

3. Broadcast address : By putting the host bits as 1 and retaining the network bits as in the IP address

4. Number of hosts per subnet : 2^{(32 – Given bits for mask)} – 2

5. First Host ID : Subnet address + 1 (adding one to the binary representation of the subnet address)

6. Last Host ID : Subnet address + Number of Hosts 

Example : Given IP Address – 172.16.0.0/25, find the number of subnets and the number of hosts per subnet. Also, for the first subnet block, find the subnet address, first host ID, last host ID and broadcast address.

Solution : This is a class B address. So, no. of subnets = 2^(25-16) = 2⁹ = 512.

No. of hosts per subnet = 2^(32-25) – 2 = 2⁷ – 2 = 128 – 2 = 126

For the first subnet block, we have subnet address = 0.0, first host id = 0.1, last host id = 0.126 and broadcast address = 0.127

Classless addressing work

Classless addressing works by allowing IP addresses to be assigned arbitrary network masks without respect to “class.” That means /8 (255.0.0.0), /16 (255.255.0.0), and /24 (255.255.255.0) network masks can be assigned to any address that would have traditionally been in the Class A, B, or C range. Additionally, that means that we’re no longer tied down to /8, /16, and /24 as our only options, and that’s where classless addressing gets very interesting.

Advantages of classless addressing

1.      More IP address allocations. Today, we know IPv6 is our long-term IP address solution to the IP address exhaustion problem. However, IPv6 is not yet widely used. In the early 1990s, it was clear we would rapidly exhaust the IPv4 address space if nothing changed. As a result, classless addressing was used as a medium-term solution to help us stretch the life of IPv4.

2.      More balanced use of IP address ranges. Classless addressing decoupled the relationship between network size and IP address and allowed for balanced use across what used to be the Class A, B, and C ranges. Far less wasted addresses.

3.      More efficient routing. VLSM and subnetting make route aggregation and classless routing protocols possible. With route aggregation (sometimes called route summarization or supernetting), routing tables can be smaller, reducing resource consumption on routers, and saving bandwidth.

Difference Between Classful and Classless Addressing

 

Ø Classful addressing is a technique of allocating IP addresses that divides them into five categories. Classless addressing is a technique of allocating IP addresses that is intended to replace classful addressing in order to reduce IP address depletion.

Ø The utility of classful and classless addressing is another distinction. Addressing without a class is more practical and helpful than addressing with a class.

Ø The network ID and host ID change based on the classes in classful addressing. In classless addressing, however, there is no distinction between network ID and host ID. As a result, another distinction between classful and classless addressing may be made.

Prefix

A network prefix is an aggregation of IP addresses. Currently, the Internet runs two protocol versions of IP: version 4 and 6. An IP address version 4 (or short IPv4) consists of a 32-bit number. Whereas an IPv6 consists of a 128-bit number

Prefix usage

The network prefix determines the number of IP addresses within a particular host section of IP addresses

Example

In IPv4, the prefix (or network portion) of the address can be identified by a dotted-decimal netmask, commonly referred to as a subnet mask. For example, 255.255. 255.0 indicates that the network portion, or prefix length, of the IPv4 address is the leftmost 24 bits.

Network Address Translation (NAT)

 A Network Address Translation (NAT) is the process of mapping an internet protocol (IP) address to another by changing the header of IP packets while in transit via a router. This helps to improve security and decrease the number of IP addresses an organization needs.

To access the Internet, one public IP address is needed, but we can use a private IP address in our private network. The idea of NAT is to allow multiple devices to access the Internet through a single public address. To achieve this, the translation of a private IP address to a public IP address is required. Network Address Translation (NAT) is a process in which one or more local IP address is translated into one or more Global IP address and vice versa in order to provide Internet access to the local hosts.

Network Address Translation (NAT) working – 

     The border router is configured for NAT i.e the router which has one interface in the local (inside) network and one interface in the global (outside) network. When a packet traverse outside the local (inside) network, then NAT converts that local (private) IP address to a global (public) IP address. When a packet enters the local network, the global (public) IP address is converted to a local (private) IP address.

NAT Types

There are three different types of NATs. People use them for different reasons, but they all still work as a NAT.

1. Static NAT

A single unregistered (Private) IP address is mapped with a legally registered (Public) IP address i.e one-to-one mapping between local and global addresses. This is generally used for Web hosting. These are not used in organizations as there are many devices that will need Internet access and to provide Internet access, a public IP address is needed. .

2. Dynamic NAT

An unregistered IP address is translated into a registered (Public) IP address from a pool of public IP addresses. If the IP address of the pool is not free, then the packet will be dropped as only a fixed number of private IP addresses can be translated to public addresses. 

3. PAT

PAT stands for port address translation. It’s a type of dynamic NAT, but it bands several local IP addresses to a singular public one.

                               This is also known as NAT overload. In this, many local (private) IP addresses can be translated to a single registered IP address. Port numbers are used to distinguish the traffic i.e., which traffic belongs to which IP address. This is most frequently used as it is cost-effective as thousands of users can be connected to the Internet by using only one real global (public) IP address.

Advantages of NAT – 
 

·         NAT conserves legally registered IP addresses. 
 

·         It provides privacy as the device’s IP address, sending and receiving the traffic, will be hidden. 
 

·         Eliminates address renumbering when a network evolves. 
 

Disadvantage of NAT – 
 

·         Translation results in switching path delays. 
 

·         Certain applications will not function while NAT is enabled. 
 

·         Complicates tunneling protocols such as IPsec. 
 

·         Also, the router being a network layer device, should not tamper with port numbers(transport layer) but it has to do so because of NAT. 

NAT inside and outside addresses – 
            Inside refers to the addresses which must be translated. Outside refers to the addresses which are not in control of an organization. These are the network Addresses in which the translation of the addresses will be done.

·         Inside local address – An IP address that is assigned to a host on the Inside (local) network. The address is probably not an IP address assigned by the service provider i.e., these are private IP addresses. This is the inside host seen from the inside network. 
 

·         Inside global address – IP address that represents one or more inside local IP addresses to the outside world. This is the inside host as seen from the outside network. 
 

·         Outside local address – This is the actual IP address of the destination host in the local network after translation. 
 

·         Outside global address – This is the outside host as seen from the outside network. It is the IP address of the outside destination host before translation. 

Nat terminology

Network Address Translation, or NAT, is a process that involves translating Private IP addresses into Public IP addresses. There are different operations within NAT and understanding each of them requires understanding NAT terminology.

Translation Table in NAT

Network Address Translation (NAT) conserves IP addresses by enabling private IP networks using unregistered IP addresses to go online. Before NAT forwards packets between the networks it connects, it translates the private internal network addresses into legal, globally unique addresses.

Create a NAT Table Connection Object

1.      Go to CONFIGURATION > Configuration Tree > Box > Assigned Services > Firewall > Forwarding Rules.

2.      From the left menu, click Connections.

3.      Click Lock.

4.      Right-click the table and select New > NAT Table.

5.      Enter a Name.

6.      (optional) Select Use Same Port to disable port address translation.

IPV6 ADDRESS

An IPv6 address is a 128-bit alphanumeric value that identifies an endpoint device in an Internet Protocol Version 6 (IPv6) network. IPv6 is the successor to a previous addressing infrastructure, IPv4, which had limitations IPv6 was designed to overcome.

Internet Protocol version 6 (IPv6) is the most recent version of the Internet Protocol (IP), the communications protocol that provides an identification and location system for computers on networks and routes traffic across the Internet.

IP address is your digital identity. It’s a network address for your computer so the Internet knows where to send you emails, data, etc.

IP address determines who and where you are in the network of billions of digital devices that are connected to the Internet.

Uses

The primary function of IPv6 is to allow for more unique TCP/IP address identifiers to be created, now that we've run out of the 4.3 billion created with IPv4. This is one of the main reasons why IPv6 is such an important innovation for the Internet of Things (IoT).

 

Types of IPv6 Address

Now that we know about what is IPv6 address let’s take a look at its different types.

1. Unicast Address –
Unicast Address identifies a single network interface. A packet sent to a unicast address is delivered to the interface identified by that address. 

 

 

2. Multicast Address –
Multicast Address is used by multiple hosts, called as Group, acquires a multicast destination address. These hosts need not be geographically together. If any packet is sent to this multicast address, it will be distributed to all interfaces corresponding to that multicast address. 

 

3. Anycast Address –
Anycast Address is assigned to a group of interfaces. Any packet sent to an anycast address will be delivered to only one member interface (mostly nearest host possible). 

Note: Broadcast is not defined in IPv6. 

 

Advantages of IPv6

·         Reliability

·         Faster Speeds: IPv6 supports multicast rather than broadcast in IPv4.This feature allows bandwidth-intensive packet flows (like multimedia streams) to be sent to multiple destinations all at once.

·         Stronger Security: IPSecurity, which provides confidentiality, and data integrity, is embedded into IPv6.

·         Routing efficiency

·         Most importantly it’s the final solution for growing nodes in Global-network.

Disadvantages of IPv6

·         Conversion: Due to widespread present usage of IPv4 it will take a long period to completely shift to IPv6.

·         Communication: IPv4 and IPv6 machines cannot communicate directly with each other. They need an intermediate technology to make that possible.

·        Slow adaptation: It is based on the fact that IPv4 is still very popular, and a large part of users are using it. The transition to the newer IPv6 is a slow process.

 

IPV6 Notation

IPv6 addresses are represented in hexadecimal notation. The IPv6 address consists of 128 binary bits. These bits are divided into eight 16-bit segments and each 16-bit segment is converted into a 4-digit hexadecimal number and separated by a colon. The term nibble is used to represent a group of four hex digits or 16 binary bits; thus, in an IPv6 address, we have eight nibbles values separated by colons.

 

VLSM

Variable Length Subnet Mask (VLSM) is a subnet -- a segmented piece of a larger network -- design strategy where all subnet masks can have varying sizes. This process of "subnetting subnets" enables network engineers to use multiple masks for different subnets of a single class A, B or C network.

With VLSM, an IP address space can be divided into a well-defined hierarchy of subnets with different sizes. This helps enhance the usability of subnets because subnets can include masks of varying sizes.

A subnet mask helps define the size of the subnet and create subnets with very different host counts without wasting large numbers of addresses.

VLSM fundamentals

To fully understand VLSM, it's important to be familiar with several fundamental terms: subnet mask, subnetting and supernetting.

Subnet mask

Every device on a network has an IP address. A subnet mask splits this IP address into the host and network addresses. This helps define which part of the IP address belongs to the network, and which part belongs to the device.

Subnetting

In subnetting (or subnetworking), a large network is logically or physically divided into multiple small networks or "subnets." The reason for subnetting a large network is to address network congestion and its negative impact on speed and productivity.

Supernetting

In supernetting, multiple contiguous networks are combined into a single large network known as a supernet (or supernetwork). Supernetting advertises many routes in one summarized advertisement or routing entry, instead of individually. This routing entry encompasses all the networks in the supernet, and provides route updates very efficiently.

Implementing a VLSM subnet

In VLSM, each subnet chooses the block size based on its requirement. So, if requirements change, subnetting will be required multiple times.

In an organization with multiple departments, different departments may require a different number of IP addresses and subnets (some more and some less). To subnet the subnets in a way that minimizes IP address wastage, VLSM is preferable to FLSM.

 

Masking

Masking identifies the boundary between the host ID and the combination of net ID and subnet ID. Each subnet mask comprises 32 bits that correspond to the bits in an IP address. In a subnet mask, the consecutive ones represent the net ID and subnet ID, and consecutive zeros represent the host ID.

IP masking is the technique of concealing your IP address by adopting a false one. This is how hiding your IP address works — they're two ways to refer to the same thing. If you're interested in learning how to mask your IP address, you can apply the same techniques described in this article.

CIDR

CIDR (Classless Inter-Domain Routing) -- also known as supernetting -- is a method of assigning Internet Protocol (IP) addresses that improves the efficiency of address distribution and replaces the previous system based on Class A, Class B and Class C networks.

The CIDR number is typically preceded by a slash “/” and follows the IP address. For example, an IP address of 131.10. 55.70 with a subnet mask of 255.0. 0.0 (which has 8 network bits) would be represented as 131.10.

USES

The assignment of CIDR blocks is handled by the Internet Assigned Numbers Authority (IANA). One of the duties of the IANA is to issue large blocks of IP addresses to regional Internet registries (RIRs). These blocks are used for large geographical areas, such as Europe, North America, Africa and Australia.

CIDR ranges

These groups, commonly called CIDR blocks, share an initial sequence of bits in the binary representation of their IP addresses. IPv4 CIDR blocks are identified using a syntax similar to that of IPv4 addresses: a dotted-decimal address, followed by a slash, then a number from 0 to 32, i.e., a.b.c.d/n.

 

 

Address Aggregation

Aggregation is an address allocation goal for any network requiring high availability. Aggregation, or supernetting as it is described in Cisco and Microsoft textbooks, is a less specific way to refer to a collection of more specific routes.

Uses

This allows control of the size of the routing table inside the network and yields efficiency when advertising subnets outside the local domain.

Networking Devices

The network device is one kind of device used to connect devices or computers together to transfer resources or files like fax machines or printers. The examples are switch, hub, bridge, router, gateway, modem, repeater & access point.

Router

The router is a physical or virtual internetworking device that is designed to receive, analyze, and forward data packets between computer networks. A router examines a destination IP address of a given data packet, and it uses the headers and forwarding tables to decide the best way to transfer the packets. There are some popular companies that develop routers; such are Cisco, 3Com, HP, Juniper, D-Link, Nortel, etc. Some important points of routers are given below

o    A router is used in LAN (Local Area Network) and WAN (Wide Area Network) environments. For example, it is used in offices for connectivity, and you can also establish the connection between distant networks such as from Bhopal to

o    It shares information with other routers in networking.

o    It uses the routing protocol to transfer the data across a network.

o    Furthermore, it is more expensive than other networking devices like switches and hubs.

Features of Router

o    It allows the users to configure the port as per their requirements in the network.

o    Routers' main components are central processing unit (CPU), flash memory, RAM, Non-Volatile RAM, console, network, and interface card.

o    Routers are capable of routing the traffic in a large networking system by considering the sub-network as an intact network.

o    Routers filter out the unwanted interference, as well as carry out the data encapsulation and decapsulation process.

o    Routers provide the redundancy as it always works in master and slave mode.

o    It allows the users to connect several LAN and WAN.

o    Furthermore, a router creates various paths to forward the data.

Applications of Routers

• Routers are used to connect hardware equipment with remote location networks like BSC, MGW, IN, SGSN, and other servers.
• It provides support for a fast rate of data transmission because it uses high STM links for connectivity; that's why it is used in both wired or wireless communication.

Types

Wireless Router: Wireless routers are used to offer Wi-Fi connectivity to laptops, smartphones, and other devices with Wi-Fi network capabilities, and it can also provide standard ethernet routing for a small number of wired network systems.

 Brouter: A brouter is a combination of the bridge and a router. It allows transferring the data between networks like a bridge.

Core router: A core router is a type of router that can route the data within a network, but it is not able to route the data between the networks.

Edge router: An edge router is a lower-capacity device that is placed at the boundary of a network. It allows an internal network to connect with the external networks.

Broadband routers: Broadband routers are mainly used to provide high-speed internet access to computers. It is needed when you connect to the internet through phone and use voice over IP technology (VOIP).

Benefits of Router

Security: Router provides the security, as LANs work in broadcast mode.

 

Performance enhancement: It enhances the performance within the individual network.

Reliability: Routers provide reliability.

Networking Range: In networking, a cable is used to connect the devices, but its length cannot exceed 1000 meters.

Think of a router as an air traffic controller and data packets as aircraft headed to different airports (or networks). Just as each plane has a unique destination and follows a unique route, each packet needs to be guided to its destination as efficiently as possible. In the same way that an air traffic controller ensures that planes reach their destinations without getting lost or suffering a major disruption along the way, a router helps direct data packets to their destination IP address.

Switches

A network switch connects devices (such as computers, printers, wireless access points) in a network to each other, and allows them to 'talk' by exchanging data packets. Switches can be hardware devices that manage physical networks, as well as software-based virtual devices.

Types of Network Switches

·         KVM Switch.

·         Managed Switch.

·         Unmanaged Switch.

·         Smart Switch.

·         PoE Switch.

 

Uses

Switches allow you to connect dozens of devices. Switches keep traffic between two devices from getting in the way of your other devices on the same network. Switches allow you to control who has access to various parts of the network.

Hub

Hubs. A hub is a physical layer networking device which is used to connect multiple devices in a network. They are generally used to connect computers in a LAN. A hub has many ports in it. A computer which intends to be connected to the network is plugged in to one of these ports.

A hub refers to a hardware device that enables multiple devices or connections to connect to a computer. An example is a USB hub, which allows multiple USB devices to connect to one computer, even though that computer may only have a few USB connections. Pictured is an example of a USB hub.

Types

passive, active, and intelligent.

Uses

A hub is a physical layer networking device which is used to connect multiple devices in a network. They are generally used to connect computers in a LAN. A hub has many ports in it.

Features of Hub

• It acts with shared bandwidth and broadcasting.
• It includes only one collision domain and broadcast domain.
• It works at the physical layer of the OSI model and also offers support for half-duplex transmission mode.
• It cannot create a virtual LAN and does not support spanning tree protocol.
• Furthermore, mainly packet collisions occur inside the hub.
• It also has a feature of flexibility, which means it includes a high transmission rate to different devices.

Applications of Hub

• Hub is used to create small home networks.
• It is used for network monitoring.
• They are also used in organizations to provide connectivity.
• It can be used to create a device that is available thought out of the network.

Advantages of Hub

• It provides support for different types of Network Media.
• It can be used by anyone as it is very cheap.
• It can easily connect many different media types.
• The use of a hub does not impact on the network performance.
• Additionally, it can expand the total distance of the network.

Disadvantages of Hub

• It has no ability to choose the best path of the network.
• It does not include mechanisms such as collision detection.
• It does not operate in full-duplex mode and cannot be divided into the Segment.
• It cannot reduce the network traffic as it has no mechanism.
• It is not able to filter the information as it transmits packets to all the connected segments.
• Furthermore, it is not capable of connecting various network architectures like a ring, token, and ethernet, and more.

Bridges

A bridge is a network device that connects multiple LANs (local area networks) together to form a larger LAN. The process of aggregating networks is called network bridging. A bridge connects the different components so that they appear as parts of a single network.

A bridge is a networking device that works in both the physical and data link layer in a network. This devices can divide a large network into smaller segments and pass the frames between two originally separated LANs. A bridge maintains a MAC address of various stations attached to it. When a frames enters a bridge, it checks the address contained in the frame and compares it with a table of all the stations on both segments.

Use of Bridge in Computer Network

A bridge in a computer network connects with other bridge networks that utilize a similar protocol. These network devices work at the data link layer in an OSI model to connect two different networks and provide communication between them.

Types of Bridges

Transparent Bridge

As the name suggests, it is an invisible bridge in the computer network. The main function of this bridge is to block or forward the data depending on the MAC address

Translational Bridge

A translational bridge plays a key role in changing a networking system from one type to another. These bridges are used to connect two different networks like token ring & Ethernet.

Source-route Bridge

 Source-route Bridge is one type of technique used for Token Ring networks and it is designed by IBM. In this bridge, the total frame route is embedded in one frame
Functions of Bridges in Network

·         This networking device is used for dividing local area networks into several segments.

·         In the OSI model, it works under the data link layer.

·         It is used to store the address of MAC in PC used in a network and also used for diminishing the network traffic.

 

The advantages ;

·         It acts as a repeater to extend a network

·         Network traffic on a segment can be reduced by subdividing it into network communications

·         Collisions can be reduced.

·         Some types of bridges connect the networks with the help of architectures & types of media.

·         Bridges increase the available bandwidth to individual nodes because fewer network nodes share a collision domain

·         It avoids waste BW (bandwidth)

·         The length of the network can be increased.

·         Connects different segments of network transmission

 

The disadvantages :

·         It is unable to read specific IP addresses because they are more troubled with the MAC addresses.

·         They cannot help while building the network between the different architectures of networks.

·         It transfers all kinds of broadcast messages, so they are incapable to stop the scope of messages.

·         These are expensive as we compare with repeaters

·         It doesn’t handle more variable & complex data load which occurs from WAN.

UNIT 3

Network layer functionalities

The main function of the network layer or layer 3 of the OSI (Open Systems Interconnection) model is delivery of data packets from the source to the destination across multiple hops or links. It also controls the operation of the subnet.

The functions are elaborated as below –

 

·       When data is to be sent, the network layer accepts data from the transport layer above, divides and encapsulates it into packets and sends it to the data link layer. The reverse procedure is done during receiving data.

·         The network layer is responsible for routing packets from the source host to the destination host. The routes can be based upon static tables that are rarely changed; or they can be automatically updated depending upon network conditions.

·         Many networks are partitioned into sub-networks or subnets. The network layer controls the operations of the subnets. Network devices called routers operate in this layer to forward packets between the subnets or the different networks.

·         The lower layers assign the physical address locally. When the data packets are routed to remote locations, a logical addressing scheme is required to differentiate the source system and the destination system. This is provided by the network layer.

·         This layer also provides mechanisms for congestion control, in situations when too many packets overload the subnets.

·         The network layer tackles issues like transmission delays, transmission time, avoidance of jitters etc.

 

Some of the other services which are expected from the network layer are:  

1.      Error Control – 

Although it can be implemented in the network layer, but it is usually not preferred because the data packet in a network layer maybe fragmented at each router, which makes error checking inefficient in the network layer. 
 

2.      Flow Control – 

            It regulates the amount of data a source can send without overloading the receiver. If the source produces a data at a very faster rate than the receiver can consume it, the receiver will be overloaded with data. To control the flow of data, the receiver should send a feedback to the sender to inform the latter that it is overloaded with data. 

There is a lack of flow control in the design of the network layer. It does not directly provide any flow control. The datagrams are sent by the sender when they are ready, without any attention to the readiness of the receiver. 

3.      Congestion Control – 

Congestion occurs when the number of datagrams sent by source is beyond the capacity of network or routers. This is another issue in the network layer protocol. If congestion continues, sometimes a situation may arrive where the system collapses and no datagrams are delivered. Although congestion control is indirectly implemented in network layer, but still there is a lack of congestion control in the network layer. 

 

Delivery vs Forwarding

Network Layer is the third layer of the OSI Model. It's responsible for source-to-destination or host-to-host delivery of packets across multiple networks. This layer takes the data from the transport layer, adds its header, and forwards it to the data link layer.

Direct Delivery

In a direct delivery, the final destination of the packet is a host connected to the same physical network as the deliverer. Direct delivery occurs when the source and destination of the packet are located on the same physical network or when the delivery is between the last router and the destination host.

Indirect Delivery

When two devices are not on the same physical network, the delivery of datagrams from one to the other is indirect. Since the source device can't see the destination on its local network, it must send the datagram through one or more intermediate devices to deliver it.

Packetizing – 
             The process of encapsulating the data received from upper layers of the network(also called as payload) in a network layer packet at the source and decapsulating the payload from the network layer packet at the destination is known as packetizing. 

The source host adds a header that contains the source and destination address and some other relevant information required by the network layer protocol to the payload received from the upper layer protocol, and delivers the packet to the data link layer. 

The destination host receives the network layer packet from its data link layer, decapsulates the packet, and delivers the payload to the corresponding upper layer protocol. The routers in the path are not allowed to change either the source or the destination address. The routers in the path are not allowed to decapsulate the packets they receive unless they need to be fragmented. 

Routing andForwarding – 
                  These are two other services offered by the network layer. In a network, there are a number of routes available from the source to the destination. The network layer specifies has some strategies which find out the best possible route. This process is referred to as routing. There are a number of routing protocols which are used in this process and they should be run to help the routers coordinate with each other and help in establishing communication throughout the network. 

Forwarding is simply defined as the action applied by each router when a packet arrives at one of its interfaces. When a router receives a packet from one of its attached networks, it needs to forward the packet to another attached network (unicast routing) or to some attached networks(in case of multicast routing). 

 

Unicast Routing Protocol

Unicast – Unicast means the transmission from a single sender to a single receiver. It is a point-to-point communication between sender and receiver. There are various unicast protocols such as TCP, HTTP, etc.  

·         TCP is the most commonly used unicast protocol. It is a connection-oriented protocol that relies on acknowledgement from the receiver side. 

·         HTTP stands for HyperText Transfer Protocol. It is an object-oriented protocol for communication.  

There are three major protocols for unicast routing: 

1.      Distance Vector Routing

2.      Link State Routing

3.      Path-Vector Routing

 

Distance Vector Routing

 

A distance-vector routing (DVR) protocol requires that a router inform its neighbors of topology changes periodically. Historically known as the old ARPANET routing algorithm (or known as Bellman-Ford algorithm)

Distance Vector Algorithm –

1.      A router transmits its distance vector to each of its neighbors in a routing packet.

2.      Each router receives and saves the most recently received distance vector from each of its neighbors.

3.      A router recalculates its distance vector when:

·         It receives a distance vector from a neighbor containing different information than before.

·         It discovers that a link to a neighbor has gone down.

Features – 

Ø  Updates of the network are exchanged periodically. 

Ø  Updates (routing information) is not broadcasted but shared to neighbouring nodes only. 

Ø  Full routing tables are not sent in updates but only distance vector is shared. 

Ø  Routers always trust routing information received from neighbor routers. This is also known as routing on rumors. 

 

 

 

Advantages of Distance Vector routing –

·         It is simpler to configure and maintain than link state routing.

Disadvantages of Distance Vector routing –

·         It is slower to converge than link state.

·         It is at risk from the count-to-infinity problem.

·         It creates more traffic than link state since a hop count change must be propagated to all routers and processed on each router. Hop count updates take place on a periodic basis, even if there are no changes in the network topology, so bandwidth-wasting broadcasts still occur.

·         For larger networks, distance vector routing results in larger routing tables than link state since each router must know about all other routers. This can also lead to congestion on WAN links.

 

Link State Routing

link-state routing protocols, each router possesses information about the complete network topology. Each router then independently calculates the best next hop from it for every possible destination in the network using local information of the topology. The collection of best-next-hops forms the routing table.

Features of link state routing protocols – 

·         Link state packet – A small packet that contains routing information. 

·         Link state database – A collection of information gathered from the link-state packet.  

·         Shortest path first algorithm (Dijkstra algorithm) – A calculation performed on the database results in the shortest path 

·         Routing table – A list of known paths and interfaces. 

Calculation of shortest path – 
To find the shortest path, each node needs to run the famous Dijkstra algorithm. This famous algorithm uses the following steps: 

Output: 0 4 12 19 21 11 9 8 14

Explanation: The distance from 0 to 1 = 4.

The minimum distance from 0 to 2 = 12. 0->1->2

The minimum distance from 0 to 3 = 19. 0->1->2->3

The minimum distance from 0 to 4 = 21. 0->7->6->5->4

The minimum distance from 0 to 5 = 11. 0->7->6->5

The minimum distance from 0 to 6 = 9. 0->7->6

The minimum distance from 0 to 7 = 8. 0->7

The minimum distance from 0 to 8 = 14. 0->1->2->8

 

• Step-1: The node is taken and chosen as a root node of the tree, this creates the tree with a single node, and now set the total cost of each node to some value based on the information in Link State Database
• Step-2: Now the node selects one node, among all the nodes not in the tree-like structure, which is nearest to the root, and adds this to the tree. The shape of the tree gets changed.
• Step-3: After this node is added to the tree, the cost of all the nodes not in the tree needs to be updated because the paths may have been changed.
• Step-4: The node repeats Step 2. and Step 3. until all the nodes are added to the tree
  Link State protocols in comparison to Distance Vector protocols have: 

1.      It requires a large amount of memory. 
 

2.      Shortest path computations require many CPU circles. 
 

3.      If a network uses little bandwidth; it quickly reacts to topology changes 
 

4.      All items in the database must be sent to neighbors to form link-state packets. 
 

5.      All neighbors must be trusted in the topology. 
 

6.      Authentication mechanisms can be used to avoid undesired adjacency and problems.  

 

OSPF Messages – OSPF is a very complex protocol. It uses five different types of messages. These are as follows:

1.      Hello message (Type 1) - It is used by the routers to introduce themselves to the other routers.  

2.      Database description message (Type 2) - It is normally sent in response to the Hello message.  

3.      Link-state request message (Type 3) - It is used by the routers that need information about specific Link-State packets.  

4.      Link-state update message (Type 4) - It is the main OSPF message for building Link-State Database. 

Features – 

Ø  Hello, messages, also known as keep-alive messages are used for neighbor discovery and recovery. 

Ø  Concept of triggered updates is used i.e updates are triggered only when there is a topology change. 

Ø  Only that many updates are exchanged which is requested by the neighbor router. 

Path Vector Routing

A path-vector routing protocol is a network routing protocol which maintains the path information that gets updated dynamically. Updates that have looped through the network and returned to the same node are easily detected and discarded.

It has three phases:

1.                   Initiation.

2.                   Sharing.

3.                   Updating.

Initialization

The tables in Figure 3.45 are stable; each node knows how to reach any other node and the cost. At the beginning, however, this is not the case. Each node can know only the distance between itself and its immediate neighbors, those directly connected to it. So for the moment, we assume that each node can send a message to the immediate neighbors and find the distance between itself and these neighbors. The distance for any entry that is not a neighbor is marked as infinite (unreachable).

Sharing

The whole idea of distance vector routing is the sharing of information between neighbors. Although node A does not know about node E, node C does. So if node C shares its routing table with A, node A can also know how to reach node E. On the other hand, node C does not know how to reach node D, but node A does. If node A shares its routing table with node C, node C also knows how to reach node D. In other words, nodes A and C, as immediate neighbors, can improve their routing tables if they help each other.

Updating

When a node receives a two-column table from a neighbor, it needs to update its routing table. Updating takes three steps: 

1. The receiving node needs to add the cost between itself and the sending node to each value in the second column. The logic is clear. If node C claims that its distance to a destination is x mi, and the distance between A and C is y mi, then the distance between A and that destination, via C, is x + y mi.

2. The receiving node needs to add the name of the sending node to each row as the third column if the receiving node uses information from any row. The sending node is the next node in the route.

3. The receiving node needs to compare each row of its old table with the corresponding row of the modified version of the received table.

Difference between Three Routing

 

Multicast Routing Protocols

A multicast routing protocol manages group membership and controls the path that multicast data takes over the network. Examples of multicast routing protocols include: Protocol Independent Multicast (PIM), Multicast Open Shortest Path First (MOSPF), and Distance Vector Multicast Routing Protocol (DVMRP).

Multicast IP Routing protocols are used to distribute data (for example, audio/video streaming broadcasts) to multiple recipients. Using multicast, a source can send a single copy of data to a single multicast address, which is then distributed to an entire group of recipients.

Multicast Protocols

·         Internet Group Management Protocol (IGMP) for IPv4 networks.

·         Multicast Listener Discovery (MLD) for IPv6 networks.

 

Multicast Listener Discovery (MLD) is a component of the Internet Protocol Version 6 (IPv6) suite. MLD is used by IPv6 routers for discovering multicast listeners on a directly attached link, much like Internet Group Management Protocol (IGMP) is used in IPv4.

Multicast Listener Discovery (MLD) is the IPv6 Multicast Group Membership Protocol. It works between the Multicast Routers and the Multicast hosts with Query,Report and Leave Messages. Multicast Listener Discovery works between MLD Querier router and the hosts. It controls Multicast member joins and leaves.

 

Applications :

Multicasting is used in many areas like:

• Internet protocol (IP)
• Streaming Media

 

1. Optimization

A router receives a packet from a network and passes it to another network. A router is usually attached to several networks. One approach is to assign a cost for passing through a network. We call this cost a metric. However, the metric assigned to each network depends on the type of protocol. Some simple protocols, such as the Routing Information Protocol (RIP), treat all networks as equals. The cost of passing through a network is the same; it is one hop

count. So if a packet passes through 10 networks to reach the destination, the total cost is 10 hop counts.

2. Intra- and Inter-domain Routing

An internet can be so large that one routing protocol cannot handle the task of updating the routing tables of all routers. For this reason, an internet is divided into autonomous systems. An autonomous system (AS) is a group of networks and routers under the authority of a single administration. Routing inside an autonomous system is referred to as intradomain routing. Routing between autonomous systems is referred to as interdomain routing

Several intradomain and interdomain routing protocols are in use.

O Two intradomain routing protocols: Distance vector and link state.

O One interdomain routing protocol: path vector.

Routing Information Protocol (RIP) is an implementation of the distance vector protocol. Open Shortest Path First (OSPF) is an implementation of the link state protocol. Border Gateway Protocol (BGP) is an implementation of the path vector protocol.

Two-Node Loop Instability

A problem with distance vector routing is instability, which means that a network using this protocol can become unstable. To understand the problem, let us look at the scenario depicted.

Defining Infinity 

The first obvious solution is to redefine infinity to a smaller number, such as100. For our previous scenario, the system will be stable in less than 20 update s. As a matter of fact, most implementations of the distance vector protocol define the distance between each node to be I and define 16 as infinity. However, this means that the distance vector routing cannot be used in large systems. The size of the network, in each direction, cannot exceed 15 hops.

Split Horizon 

Another solution is called split horizon. In this strategy, instead of flooding thetable through each interface, each node sends only part of its table through each interface. If, according to its table, node B thinks that the optimum route to reach X is via A, it does not need to advertise this piece of information to A; the information has corne from A (A already knows).

Taking information from node A, modifying it, and sending it back to node A creates the confusion. In our scenario, node B eliminates the last line of its routing table before it sends it to A. In this case, node A keeps the value of infinity as the distance to X.

Node Al is the speaker node for AS1, B1 for AS2, C1 for AS3, and Dl for AS4. Node Al creates an initial table that shows Al to A5 are located in ASI and can be reached through it. Node B1 advertises that Bl to B4 are located in AS2 and can be reached through Bl. And so on.

Routing Information Protocol (RIP)

Routing Information Protocol (RIP) is a distance-vector routing protocol. Routers running the distance-vector protocol send all or a portion of their routing tables in routing-update messages to their neighbors. You can use RIP to configure the hosts as part of a RIP network.

Routing Information Protocol

 (RIP) is a dynamic routing protocol that uses hop count as a routing metric to find the best path between the source and the destination network. It is a distance-vector routing protocol that has an AD value of 120 and works on the Network layer of the OSI model. RIP uses port number 520. 

Hop Count 

Hop count is the number of routers occurring in between the source and destination network. The path with the lowest hop count is considered as the best route to reach a network and therefore placed in the routing table. RIP prevents routing loops by limiting the number of hops allowed in a path from source and destination. The maximum hop count allowed for RIP is 15 and a hop count of 16 is considered as network unreachable. 

Features of RIP 

1. Updates of the network are exchanged periodically. 
2. Updates (routing information) are always broadcast. 
3. Full routing tables are sent in updates. 
4. Routers always trust routing information received from neighbor routers. This is also known as Routing on rumors. 

 

RIP versions : 

There are three versions of routing information protocol –

Ø  RIP Version1, 

Ø  RIP Version2,

Ø  RIPng. 

RIP Version-1:

It is an open standard protocol means it works on the various vendor's routers. It works on most of the routers, it is classful routing protocol. Updates are broadcasted. Its administrative distance value is 120, it means it is not reliable, The lesser the administrative distance value the reliability is much more. Its metric is hop count and max hop count is 15. There will be a total of 16 routers in the network. When there will be the same number of hop to reach the destination, Rip starts to perform load balancing. Load balancing means if there are three ways to reach the destination and each way has same number of routers then packets will be sent to each path to reach the destination. This reduces traffic and also the load is balanced. It is used in small companies, in this protocol routing tables are updated in each 30 sec. Whenever link breaks rip trace out another path to reach the destination. It is one of the slowest protocol.

Advantages of RIP ver1 -

1. Easy to configure, static router are complex.
2. Less overhead
3. No complexity.

Disadvantage of RIP ver1 -

1. Bandwidth utilization is very high as broadcast for every 30 seconds.
2. It works only on hop count.
3. It is not scalable as hop count is only 15. If there will be requirement of more routers in the network it would be a problem .
4. Convergence is very slow, wastes a lot of time in finding alternate path.

RIP Version-2:

Due to some deficiencies in the original RIP specification, RIP version 2 was developed in 1993. It supports classless Inter-Domain Routing (CIDR) and has the ability to carry subnet information, its metric is also hop count, and max hop count 15 is same as rip version 1. It supports authentication and does subnetting and multicasting. Auto summary can be done on every router. In RIPv2 Subnet masks are included in the routing update. RIPv2 multicasts the entire routing table to all adjacent routers at the address 224.0.0.9, as opposed to RIPv1 which uses broadcast (255.255.255.255).

Advantages of RIP ver2 -

1. It's a standardized protocol.
2. It's VLSM compliant.
3. Provides fast convergence.
4. It sends triggered updates when the network changes.
5. Works with snapshot routing - making it ideal for dial networks.

Disadvantage of RIP ver2 - There lies some disadvantages as well:

1. Max hopcount of 15, due to the 'count-to-infinity' vulnerability.
2. No concept of neighbours.
3. Exchanges entire table with all neighbours every 30 seconds (except in the case of a triggered update).

RIP ver1 versus RIP ver2:

RIP Ver1	RIP Ver2
RIP v1 uses what is known as classful routing	RIP v2 is a classless protocol and it supports variable-length subnet masking (VLSM), CIDR, and route summarization
RIPv1 routing updates are broadcasted	RIP v2 routing updates are multicasted
RIPv1 has no authentication	RIP v2 supports authentication
RIP v1 does not carry mask in updates	RIP v2 does carry mask in updates, so it supports for VLSM
RIP v1 is an older, no longer much used routing protocol	IP v2 can be useful in small, flat networks or at the edge of larger networks because of its simplicity in configuration and usage

 

Border Gateway Protocol (BGP) is used to Exchange routing information for the internet and is the protocol used between ISP which are different ASes. 

The protocol can connect together any internetwork of autonomous system using an arbitrary topology. The only requirement is that each AS have at least one router that is able to run BGP and that is router connect to at least one other AS's BGP router. BGP's main function is to exchange network reach-ability information with other BGP systems. Border Gateway Protocol constructs an autonomous systems' graph based on the information exchanged between BGP routers. 

Characteristics of Border Gateway Protocol (BGP):

• Inter-Autonomous System Configuration: The main role of BGP is to provide communication between two autonomous systems.
• BGP supports Next-Hop Paradigm.
• Coordination among multiple BGP speakers within the AS (Autonomous System).
• Path Information: BGP advertisement also include path information, along with the reachable destination and next destination pair.
• Policy Support: BGP can implement policies that can be configured by the administrator. For ex:- a router running BGP can be configured to distinguish between the routes that are known within the AS and that which are known from outside the AS.
• Runs Over TCP.
• BGP conserve network Bandwidth.
• BGP supports CIDR.
• BGP also supports Security.

Functionality of Border Gateway Protocol (BGP): 

BGP peers performs 3 functions, which are given below.

1. The first function consist of initial peer acquisition and authentication. both the peers established a TCP connection and perform message exchange that guarantees both sides have agreed to communicate.
2. The second function mainly focus on sending negative or positive reach-ability information.
3. The third function verifies that the peers and the network connection between them are functioning correctly.

BGP Route Information Management Functions:

• Route Storage: Each BGP stores information about how to reach other networks.
• Route Update: In this task, Special techniques are used to determine when and how to use the information received from peers to properly update the routes.
• Route Selection: Each BGP uses the information in its route databases to select good routes to each network on the internet network.
• Route advertisement: Each BGP speaker regularly tells its peer what is knows about various networks and methods to reach them.

Dijkstra algorithm

 

 

Application of Dijistkra Algorithm

1) It is used in Google Maps

2) It is used in finding Shortest Path.

3) It is used in geographical Maps

4) To find locations of Map which refers to vertices of graph.

5) Distance between the location refers to edges.

6) It is used in IP routing to find Open shortest Path First.

7) It is used in the telephone network.

 

Dynamic routing Protocol performs the same function as static routing Protocol does. In dynamic routing Protocol, if the destination is unreachable then another entry, in the routing table, to the same destination can be used. One of the routing protocols is EIGRP. 

EIGRP: 
Enhanced Interior Gateway Routing Protocol (EIGRP) is a dynamic routing protocol that is used to find the best path between any two-layer 3 devices to deliver the packet. EIGRP works on network layer Protocol of OSI model and uses protocol number 88. It uses metrics to find out the best path between two layer 3 devices (router or layer 3 switches) operating EIGRP.

Administrative Distance for EIGRP are:- 

EIGRP routes	AD values
Summary Routes	5
Internal Routes	90
external routes	170

It uses some messages to communicate with the neighbour devices that operate EIGRP. These are:- 

1. Hello message-These messages are kept alive messages which are exchanged between two devices operating EIGRP. These messages are used for neighbour discovery/recovery, if there is any device operating EIGRP or if any device(operating EIGRP) coming up again. 
   These messages are used for neighbor discovery if multicast at 224.0.0.10. It contains values like AS number, k values, etc. 
   These messages are used as acknowledgement when unicast. A hello with no data is used as the acknowledgement.
2. NULL update-It is used to calculate SRTT(Smooth Round Trip Timer) and RTO(Retransmission Time Out). 
   SRTT: The time is taken by a packet to reach the neighboring router and the acknowledgement of the packet to reach the local router. 
   RTO: If a multicast fails then unicast is being sent to that router. RTO is the time for which the local router waits for an acknowledgement of the packet.
3. Full Update - After exchanging hello messages or after the neighbourship is formed, these messages are exchanged. This message contains all the best routes.
4. Partial update-These messages are exchanged when there is a topology change and new links are added. It contains only the new routes, not all the routes. These messages are multicast.
5. Query message-These messages are multicast when the device is declared dead and it has no routes to it in its topology table.
6. Reply message - These messages are the acknowledgment of the query message sent to the originator of the query message stating the route to the network which has been asked in the query message.
7. Acknowledgement message 
   It is used to acknowledge EIGRP updates, queries, and replies. Acks are hello packets that contain no data. 
   
   Note:-Hello and acknowledgment packets do not require any acknowledgment. 
   Reply, query, update messages are reliable messages i.e require acknowledgement.

Composite matrix-The EIGRP composite metric calculation can use up to 5 variables, but only 2 are used by default (K1 and K3). The composite metric values are : 

K1 (bandwidth) 
K2 (load) 
K3 (delay) 
K4 (reliability) 
K5 (MTU) 

The lowest bandwidth, load, delay, reliability, MTU along the path between the source and the destination is considered in the composite matrix in order to calculate the cost. 
Note:- Generally, only k1 and k3 values are used for metric calculation by EIGRP. The values are 10100 for k1, k2, k3, k4, k5 respectively. 
criteria To form EIGRP neighbourship, these criteria should be fulfilled:- 

1. k values should match.
2. Autonomous system number should match. (AS is a group of networks running under a single administrative control) .
3. authentication should match (if applied). EIGRP supports MD5 authentication only.
4. subnet mask should be the same.

Timers:- 
Hello timer- The interval in which EIGRP sends a hello message on an interface. It is 5 seconds by default. 
Dead timer- The interval in which the neighbor will be declared dead if it is not able to send the hello packet. It is 15 seconds by default.
 

 UNIT-4

ERRORS AND ITS TYPES

            A network error is the error condition that caused a network request to fail. Each network error has a type , which is a string. Each network error has a phase , which describes which phase the error occurred in: dns. the error occurred during DNS resolution

Error A condition when the receiver's information does not match with the sender's information. During transmission, digital signals suffer from noise that can introduce errors in the binary bits travelling from sender to receiver. That means a 0 bit may change to 1 or a 1 bit may change to 0. 

There may be three types of errors:

·         Single bit error

In a frame, there is only one bit, anywhere though, which is corrupt.

·         Multiple bits error

Frame is received with more than one bits in corrupted state.

·         Burst error

Frame contains more than1 consecutive bits corrupted.

Error control mechanism may involve two possible ways:

·         Error detection

·         Error correction

Error Detecting Codes (Implemented either at Data link layer or Transport Layer of OSI Model) Whenever a message is transmitted, it may get scrambled by noise or data may get corrupted. To avoid this, we use error-detecting codes which are additional data added to a given digital message to help us detect if any error has occurred during transmission of the message. 
  Basic approach used for error detection is the use of redundancy bits, where additional bits are added to facilitate detection of errors. Some popular techniques for error detection are:

Types of Errors

Ø  Simple Parity check

Ø  Two-dimensional Parity check

Ø  Checksum

Ø  4.Cyclic redundancy check 
 

1.      Simple Parity check

Imagine a data transfer that looks like this: 1010001. This example has an odd number of 1s and and even number of 0s. When an even parity checking is used, a parity bit with value 1 could be added to the data's right side to make the number of 1s even -- and the transmission would look like this: 10100011.

A simple error detection method is based on the principle that if each bit pattern being manipulated as an odd numbers of 1s, and a pattern is detected that has an even number of 1s, then an error must have occurred. A parity bit is an extra bit that is associated with a word of storage.

Blocks of data from the source are subjected to a check bit or parity bit generator form, where a parity of :

• 1 is added to the block if it contains odd number of 1’s, and
• 0 is added if it contains even number of 1’s

This scheme makes the total number of 1’s even, that is why it is called even parity checking.

 

2.      Two-dimensional Parity check

Parity check bits are calculated for each row, which is equivalent to a simple parity check bit. Parity check bits are also calculated for all columns, then both are sent along with the data. At the receiving end these are compared with the parity bits calculated on the received data.

In Two-Dimensional Parity check, a block of bits is divided into rows, and the redundant row of bits is added to the whole block. At the receiving end, the parity bits are compared with the parity bits computed from the received data.

In an even parity check, parity bits ensure there are an even number of 1s and 0s in the transmission. In an odd parity check, there are an odd number of 1s and 0s in the transmission.

3.      Checksum

A checksum is a value that represents the number of bits in a transmission message and is used by IT professionals to detect high-level errors within data transmissions. Prior to transmission, every piece of data or file can be assigned a checksum value after running a cryptographic hash function.

A sum derived from the bits of a segment of computer data that is calculated before and after transmission or storage to assure that the data is free from errors or tampering

• In checksum error detection scheme, the data is divided into k segments each of m bits.
• In the sender’s end the segments are added using 1’s complement arithmetic to get the sum. The sum is complemented to get the checksum.
• The checksum segment is sent along with the data segments.
• At the receiver’s end, all received segments are added using 1’s complement arithmetic to get the sum. The sum is complemented.
• If the result is zero, the received data is accepted; otherwise discarded.

4.      Cyclic redundancy check (CRC)

An error detection technique using a polynomial to generate a series of two 8-bit block check characters that represent the entire block of data. These block check characters are incorporated into the transmission frame and then checked at the receiving end.

The Cyclic Redundancy Checks (CRC) is the most powerful method for Error-Detection and Correction. It is given as a kbit message and the transmitter creates an (n – k) bit sequence called frame check sequence. The out coming frame, including n bits, is precisely divisible by some fixed number

• Unlike checksum scheme, which is based on addition, CRC is based on binary division.
• In CRC, a sequence of redundant bits, called cyclic redundancy check bits, are appended to the end of data unit so that the resulting data unit becomes exactly divisible by a second, predetermined binary number.
• At the destination, the incoming data unit is divided by the same number. If at this step there is no remainder, the data unit is assumed to be correct and is therefore accepted.
• A remainder indicates that the data unit has been damaged in transit and therefore must be rejected.

 
 

Example :  
Error Correction

In the digital world, error correction can be done in two ways:

Backward Error Correction  When the receiver detects an error in the data received, it requests back the sender to retransmit the data unit.

Forward Error Correction  When the receiver detects some error in the data received, it executes error-correcting code, which helps it to auto-recover and to correct some kinds of errors.

The first one, Backward Error Correction, is simple and can only be efficiently used where retransmitting is not expensive. For example, fiber optics. But in case of wireless transmission retransmitting may cost too much. In the latter case, Forward Error Correction is used.

To correct the error in data frame, the receiver must know exactly which bit in the frame is corrupted. To locate the bit in error, redundant bits are used as parity bits for error detection.For example, we take ASCII words (7 bits data), then there could be 8 kind of information we need: first seven bits to tell us which bit is error and one more bit to tell that there is no error.

For m data bits, r redundant bits are used. r bits can provide 2r combinations of information. In m+r bit codeword, there is possibility that the r bits themselves may get corrupted. So the number of r bits used must inform about m+r bit locations plus no-error information, i.e. m+r+1.

Hamming code is a set of error-correction codes that can be used to detect and correct the errors that can occur when the data is moved or stored from the sender to the receiver. It is a technique developed by R.W. Hamming for error correction. Redundant bits – Redundant bits are extra binary bits that are generated and added to the information-carrying bits of data transfer to ensure that no bits were lost during the data transfer. The number of redundant bits can be calculated using the following formula:

 2^r ≥ m + r + 1

 where, r = redundant bit, m = data bit

Suppose the number of data bits is 7, then the number of redundant bits can be calculated using: = 2^4 ≥ 7 + 4 + 1 Thus, the number of redundant bits= 4 Parity bits.  A parity bit is a bit appended to a data of binary bits to ensure that the total number of 1’s in the data is even or odd. Parity bits are used for error detection.

 

There are two types of parity bits:

1.      Even parity bit: In the case of even parity, for a given set of bits, the number of 1’s are counted. If that count is odd, the parity bit value is set to 1, making the total count of occurrences of 1’s an even number. If the total number of 1’s in a given set of bits is already even, the parity bit’s value is 0.

2.      Odd Parity bit – In the case of odd parity, for a given set of bits, the number of 1’s are counted. If that count is even, the parity bit value is set to 1, making the total count of occurrences of 1’s an odd number. If the total number of 1’s in a given set of bits is already odd, the parity bit’s value is 0.

 

General Algorithm of Hamming code:

 Hamming Code is simply the use of extra parity bits to allow the identification of an error.

1.      Write the bit positions starting from 1 in binary form (1, 10, 11, 100, etc).

2.      All the bit positions that are a power of 2 are marked as parity bits (1, 2, 4, 8, etc).

3.      All the other bit positions are marked as data bits.

4.      Each data bit is included in a unique set of parity bits, as determined its bit position in binary form. a. Parity bit 1 covers all the bits positions whose binary representation includes a 1 in the least significant position (1, 3, 5, 7, 9, 11, etc). b. Parity bit 2 covers all the bits positions whose binary representation includes a 1 in the second position from the least significant bit (2, 3, 6,

5.      7, 10, 11, etc). c. Parity bit 4 covers all the bits positions whose binary representation includes a 1 in the third position from the least significant bit (4–7, 12–15, 20–23, etc). d. Parity bit 8 covers all the bits positions whose binary representation includes a 1 in the fourth position from the least significant bit bits (8–15, 24–31, 40–47, etc). e. In general, each parity bit covers all bits where the bitwise AND of the parity position and the bit position is non-zero.

6.      Since we check for even parity set a parity bit to 1 if the total number of ones in the positions it checks is odd.

7.      Set a parity bit to 0 if the total number of ones in the positions it checks is even.

Determining the position of redundant bits – These redundancy bits are placed at positions that correspond to the power of 2. 

As in the above example:

·         The number of data bits = 7

·         The number of redundant bits = 4

·         The total number of bits = 11

·         The redundant bits are placed at positions corresponding to power of 2- 1, 2, 4, and 8

·         Suppose the data to be transmitted is 1011001, the bits will be placed as follows: 

Determining the Parity bits:

·         R1 bit is calculated using parity check at all the bits positions whose binary representation includes a 1 in the least significant position. R1: bits 1, 3, 5, 7, 9, 11 

·          To find the redundant bit R1, we check for even parity. Since the total number of 1’s in all the bit positions corresponding to R1 is an even number the value of R1 (parity bit’s value) = 0

·         R2 bit is calculated using parity check at all the bits positions whose binary representationincludes a 1 in the second position from the least significant bit. R2: bits 2,3,6,7,10,11 

·         To find the redundant bit R2, we check for even parity. Since the total number of 1’s in all the bit positions corresponding to R2 is odd the value of R2(parity bit’s value)=1

·         R4 bit is calculated using parity check at all the bits positions whose binary representation includes a 1 in the third position from the least significant bit. R4: bits 4, 5, 6, 7 

 

1.       To find the redundant bit R4, we check for even parity. Since the total number of 1’s in all the bit positions corresponding to R4 is odd the value of R4(parity bit’s value) = 1

1.      R8 bit is calculated using parity check at all the bits positions whose binary representationincludes a 1 in the fourth position from the least significant bit. R8: bit 8,9,10,11  

·         To find the redundant bit R8, we check for even parity. Since the total number of 1’s in all the bit positions corresponding to R8 is an even number the value of R8(parity bit’s value)=0. Thus, the data transferred is:

 

Error detection and correction: Suppose in the above example the 6th bit is changed from 0 to 1 during data transmission, then it gives new parity values in the binary number: 

The bits give the binary number 0110 whose decimal representation is 6. Thus, bit 6 contains an error. To correct the error the 6th bit is changed from 1 to 0.

 

Framing:

Frames are the units of digital transmission, particularly in computer networks and telecommunications. Frames are comparable to the packets of energy called photons in the case of light energy. Frame is continuously used in Time Division Multiplexing process. 

Framing is a point-to-point connection between two computers or devices consists of a wire in which data is transmitted as a stream of bits. However, these bits must be framed into discernible blocks of information. Framing is a function of the data link layer. It provides a way for a sender to transmit a set of bits that are meaningful to the receiver. Ethernet, token ring, frame relay, and other data link layer technologies have their own frame structures. Frames have headers that contain information such as error-checking codes. 

At the data link layer, it extracts the message from the sender and provides it to the receiver by providing the sender’s and receiver’s addresses. The advantage of using frames is that data is broken up into recoverable chunks that can easily be checked for corruption. 

Problems in Framing –  

·         Detecting start of the frame: When a frame is transmitted, every station must be able to detect it. Station detects frames by looking out for a special sequence of bits that marks the beginning of the frame i.e. SFD (Starting Frame Delimiter).

·         How does the station detect a frame: Every station listens to link for SFD pattern through a sequential circuit. If SFD is detected, sequential circuit alerts station. Station checks destination address to accept or reject frame.

·         Detecting end of frame: When to stop reading the frame.

 

Types of framing – 

There are two types of framing: 

1. Fixed size – The frame is of fixed size and there is no need to provide boundaries to the frame, the length of the frame itself acts as a delimiter.  

·                     Drawback: It suffers from internal fragmentation if the data size is less than the frame size

·                     Solution: Padding

 

2. Variable size – In this, there is a need to define the end of the frame as well as the beginning of the next frame to distinguish. This can be done in two ways: 
 

1.                  Length field – We can introduce a length field in the frame to indicate the length of the frame. Used in Ethernet(802.3). The problem with this is that sometimes the length field might get corrupted.

2.                  End Delimiter (ED) – We can introduce an ED(pattern) to indicate the end of the frame. Used in Token Ring. The problem with this is that ED can occur in the data. This can be solved by: 

 

Character/Byte Stuffing:

 Used when frames consist of characters. If data contains ED then, a byte is stuffed into data to differentiate it from ED. 

Let ED = “$” –> if data contains ‘$’ anywhere, it can be escaped using ‘\O’ character. 
–> if data contains ‘\O$’ then, use ‘\O\O\O$'($ is escaped using \O and \O is escaped using \O).

Disadvantage – It is very costly and obsolete method. 

 

Bit Stuffing: 

Let ED = 01111 and if data = 01111 

–> Sender stuffs a bit to break the pattern i.e. here appends a 0 in data = 011101. 
–> Receiver receives the frame. 
–> If data contains 011101, receiver removes the 0 and reads the data. 

Examples – 

·         If Data –> 011100011110 and ED –> 0111 then, find data after bit stuffing? 

–> 011010001101100 
 

·         If Data –> 110001001 and ED –> 1000 then, find data after bit stuffing? 

–> 11001010011 

 

1. FlowControl : 
                It is an important function of the Data Link Layer. It refers to a set of procedures that tells the sender how much data it can transmit before waiting for acknowledgement from the receiver. 

PurposeofFlowControl: 
               Any receiving device has a limited speed at which it can process incoming data and also a limited amount of memory to store incoming data. If the source is sending the data at a faster rate than the capacity of the receiver, there is a possibility of the receiver being swamped. The receiver will keep loosing some of the frames simply because they are arriving too quickly and the buffer is also getting filled up. 

This will generate waste frames on the network. Therefore, the receiving device must have some mechanism to inform the sender to send fewer frames or stop transmission temporarily. In this way, flow control will control the rate of frame transmission to a value that can be handled by the receiver. 

Example – Stop & Wait Protocol 

 

2. Error Control : 
The error control function of data link layer detects the errors in transmitted frames and re-transmit all the erroneous frames. 

 

Purpose of Error Control : 

The function of the error control function of the data link layer helps in dealing with data frames that are damaged in transit, data frames lost in transit, and the acknowledgement frames that are lost in transmission. The method used for error control is called Automatic Repeat Request which is used for the noisy channel. 

Example – Stop & Wait ARQ and Sliding Window ARQ 

 

Difference between Flow Control and Error Control : 

 

S.NO.	Flow control	Error control
1.	Flow control is meant only for the transmission of data from sender to receiver.	Error control is meant for the transmission of error free data from sender to receiver.
2.	For Flow control there are two approaches : Feedback-based Flow Control and Rate-based Flow Control.	To detect error in data, the approaches are : Checksum, Cyclic Redundancy Check and Parity Checking.  To correct error in data, the approaches are : Hamming code, Binary Convolution codes, Reed-Solomon code, Low-Density Parity Check codes.
3.	It prevents the loss of data and avoid over running of receive buffers.	It is used to detect and correct the error occurred in the code.
4.	Example of Flow Control techniques are : Stop&Wait Protocol and Sliding Window Protocol.	Example of Error Control techniques are : Stop&Wait ARQ and Sliding Window ARQ.

 

 

ARQ stands for Automatic Repeat Request

 Also known as Automatic Repeat Query. ARQ is an error-control strategy used in a two-way communication system. It is a group of error-control protocols to achieve reliable data transmission over an unreliable source or service. These protocols reside in Transport Layer and Data Link Layer of the OSI(Open System Interconnection) model . These protocols are responsible for automatic retransmission of packets that are found to be corrupted or lost during the transmission process. 

Working Principle of ARQ

The main function of these protocols is, the sender receives an acknowledgement from the receiver end implying that the frame or packet is received correctly before a timeout occurs, timeout is a specific time period within which the acknowledgement has to be sent by the receiver to the sender. If a timeout occurs: the sender does not receive the acknowledgement before the specified time, it is implied that the frame or packet has been corrupt or lost during the transmission. Accordingly, the sender retransmits the packet and these protocols ensure that this process is repeated until the correct packet is transmitted. 

 

Applications

ARQ protocols have a wide range of applications as they provide reliable transmissions over unreliable upper sources. These protocols are mainly functional on shortwave radio to ensure reliable delivery of signals. 
For the same function of ARQ, there are various applications: 

1.      Transmission Control Protocol (TCP)

2.      Specific Service Orientation Protocol: Error-correction of message signals in ATM networks.

3.      High-Level Data Link protocol.

4.      IBM Binary synchronous Communications Protocol.

5.      Xmodem : modem file transfer protocol.

Types
There are several types of ways in which these protocols function in the data link layer :  

 

Stop And Wait ARQ: 

       Stop and wait ARQ is also referred to as the alternating protocol is a method used in two-way communication systems to send information between two connected devices (sender and a receiver). It is referred to as stop and wait ARQ because the function of this protocol is to send one frame at a time.

After sending a frame or packet, the sender doesn't send any further packets until it receives an acknowledgement from the receiver. Moreover, the sender keeps a copy of the sent packet. After receiving the desired frame, the receiver sends an acknowledgement. If the acknowledgement does not reach the sender before the specified time, known as the timeout, the sender sends the same packet again. The timeout is reset after each frame transmission. The above scenario depicts a Stop and wait situation, so this control mechanism is termed as Stop and waitARQ. 
 

Go Back-N ARQ:

Go-Back-N ARQ is a type of the ARQ protocol, in which the sending process continues to send several frames or packets even without receiving an acknowledgement packet from the receiver. The receiver process keeps track of the sequence number of the next packet it expects to receive and sends that sequence number with every acknowledgement to the sender. The receiver will remove any packet that does not have the desired sequence number it expects and will resend an acknowledgement for the last correct frame.

There are only two possibilities that a frame won't match the sequence number: it is either a duplicated frame of an existing frame or an out-of-order frame that needs to be sent later, the receiver recognizes this scenario and sends an acknowledgement signal accordingly.

Once the sender has sent all of the frames in its window, it will identify that all of the frames since the first lost frame, and will go back to the sequence number of the last acknowledgement signal that it received from the receiver pr and continue the process over again. The only drawback of this type of system is that it results in sending packets multiple times: if any frame was lost or found to be corrupted, then that frame and all following frames in the send window will be re-transmitted. 
This protocol is more efficient than Stop and wait ARQ as there is no waiting time.

• Selective Repeat ARQ/Selective Reject ARQ: 

Selective Repeat ARQ/Selective Reject ARQ protocol mechanism is similar to the Go-Back-N protocol mechanism but in Selective Repeat ARQ the sending process continues even after a frame is found to be corrupt or lost. This is achieved: the receiver process keeps track of the sequence number of the earliest frame it has not received and sends the respective sequence number with the acknowledgement signal. If a frame is not received at the receiver end, the sender continues to send the succeeding frames until it has emptied its window. once this error-correction process has been done, the process continues where it left off. Unlike, Go back-N protocol this does not send a packet multiple times. 
 

Advantages of ARQ

• The Error-detection and correction mechanisms are quite simple compared to the other techniques.
• A much simpler decoding equipment can be put to use compared to the other techniques.

Disadvantages of ARQ

• A medium or a channel with a high error rate might cause too much transmission of the frames or packets of information.
• The high error rate in the channel might also lead to loss of information, therefore reducing the efficiency or the productivity of the system.

Multiple access protocol-

Random Access Control Protocol

Ø  ALOHA,

Ø  CSMA,

Ø  CSMA/CA

Ø  CSMA/CD

 ALOHA

 

ALOHA is a multiple access protocol for transmission of data via a shared network channel. It operates in the medium access control sublayer (MAC sublayer) of the open systems interconnection (OSI) model. Using this protocol, several data streams originating from multiple nodes are transferred through a multi-point transmission channel.

1.      Any station can transmit data to a channel at any time.

2.      It does not require any carrier sensing.

3.      Collision and data frames may be lost during the transmission of data through multiple stations.

4.      Acknowledgment of the frames exists in Aloha. Hence, there is no collision detection.

5.      It requires retransmission of data after some random amount of time.

Versions of ALOHA Protocols

Pure ALOHA

In pure ALOHA, the time of transmission is continuous.Whenever a station hasan available frame, it sends the frame. If there is collision and the frame is destroyed, the sender waits for a random amount of time before retransmitting it.

1. The total vulnerable time of pure Aloha is 2 * Tfr.
2. Maximum throughput occurs when G = 1/ 2 that is 18.4%.
3. Successful transmission of data frame is S = G * e ^ - 2 G.

Slotted ALOHA

Slotted ALOHA reduces the number of collisions and doubles the capacity of pure ALOHA. The shared channel is divided into a number of discrete time intervals called slots. A station can transmit only at the beginning of each slot. However,there can still be collisions if more than one station tries to transmit at the beginning of the same time slot.

1.      Maximum throughput occurs in the slotted Aloha when G = 1 that is 37%.

2.      The probability of successfully transmitting the data frame in the slotted Aloha is S = G * e ^ - 2 G.

3.      The total vulnerable time required in slotted Aloha is Tfr.

 

 CSMA

 

Carrier-sense multiple access with collision detection (CSMA/CD) is a medium access control (MAC) method used most notably in early Ethernet technology for local area networking. It uses carrier-sensing to defer transmissions until no other stations are transmitting.

t is a carrier sense multiple access based on media access protocol to sense the traffic on a channel (idle or busy) before transmitting the data. It means that if the channel is idle, the station can send data to the channel. Otherwise, it must wait until the channel becomes idle. Hence, it reduces the chances of a collision on a transmission medium.

CSMA Access Modes

1-Persistent: 

In the 1-Persistent mode of CSMA that defines each node, first sense the shared channel and if the channel is idle, it immediately sends the data. Else it must wait and keep track of the status of the channel to be idle and broadcast the frame unconditionally as soon as the channel is idle.

Non-Persistent:

 It is the access mode of CSMA that defines before transmitting the data, each node must sense the channel, and if the channel is inactive, it immediately sends the data. Otherwise, the station must wait for a random time (not continuously), and when the channel is found to be idle, it transmits the frames.

P-Persistent:

 It is the combination of 1-Persistent and Non-persistent modes. The P-Persistent mode defines that each node senses the channel, and if the channel is inactive, it sends a frame with a P probability. If the data is not transmitted, it waits for a (q = 1-p probability) random time and resumes the frame with the next time slot.

O- Persistent: 

It is an O-persistent method that defines the superiority of the station before the transmission of the frame on the shared channel. If it is found that the channel is inactive, each station waits for its turn to retransmit the data.

 

CSMA/ CD

It is a carrier sense multiple access/ collision detection network protocol to transmit data frames. The CSMA/CD protocol works with a medium access control layer.

Therefore, it first senses the shared channel before broadcasting the frames, and if the channel is idle, it transmits a frame to check whether the transmission was successful. If the frame is successfully received, the station sends another frame. If any collision is detected in the CSMA/CD, the station sends a jam/ stop signal to the shared channel to terminate data transmission. After that, it waits for a random time before sending a frame to a channel.

CSMA/ CA

It is a carrier sense multiple access/collision avoidance network protocol for carrier transmission of data frames. It is a protocol that works with a medium access control layer. When a data frame is sent to a channel, it receives an acknowledgment to check whether the channel is clear.

If the station receives only a single (own) acknowledgments, that means the data frame has been successfully transmitted to the receiver. But if it gets two signals (its own and one more in which the collision of frames), a collision of the frame occurs in the shared channel. Detects the collision of the frame when a sender receives an acknowledgment signal.

Following are the methods used in the CSMA/ CA to avoid the collision:

 

Interframe space:

 In this method, the station waits for the channel to become idle, and if it gets the channel is idle, it does not immediately send the data. Instead of this, it waits for some time, and this time period is called the Interframe space or IFS. However, the IFS time is often used to define the priority of the station.

Contention window:

In the Contention window, the total time is divided into different slots. When the station/ sender is ready to transmit the data frame, it chooses a random slot number of slots as wait time. If the channel is still busy, it does not restart the entire process, except that it restarts the timer only to send data packets when the channel is inactive.

Acknowledgment:

In the acknowledgment method, the sender station sends the data frame to the shared channel if the acknowledgment is not received ahead of time.