INTERNET PROTOCOL SUITE
The Internet protocol suite is the set of communications protocols that implement the protocol stack on which the Internet and most commercial networks run. It has also been referred to as the TCP/IP protocol suite, which is named after two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were also the first two networking protocols defined. Today's IP networking represents a synthesis of two developments that began in the 1970s, namely LANs (Local Area Networks) and the Internet, both of which have revolutionized computing.
The Internet Protocol suite—like many protocol suites—can be viewed as a set of layers. Each layer solves a set of problems involving the transmission of data, and provides a well-defined service to the upper layer protocols based on using services from some lower layers. Upper layers are logically closer to the user and deal with more abstract data, relying on lower layer protocols to translate data into forms that can eventually be physically transmitted. The TCP/IP reference model consists of four layers.
INTERNET CONTROL MESSAGE PROTOCOL (ICMP)
The Internet Control Message Protocol (ICMP) is one of the core protocols of the Internet protocol suite operates in the Network/Internet layer. It is chiefly used by networked computers' operating systems to send error messages—indicating, for instance, that a requested service is not available or that a host or router could not be reached.
ICMP differs in purpose from TCP and UDP in that it is usually not used directly by user network applications. One exception is the ping tool, which sends ICMP Echo Request messages (and receives Echo Response messages) to determine whether a host is reachable and how long packets take to get to and from that host.
Technical Details
Internet control message protocol is part of the Internet protocol suite as defined in RFC 792. ICMP messages are typically generated in response to errors in IP datagrams (as specified in RFC 1122) or for diagnostic or routing purposes.
The version of ICMP for Internet Protocol version 4 is also known as ICMPv4, as it is part of IPv4. IPv6 has an equivalent protocol, ICMPv6.
ICMP messages are constructed at the IP layer, usually from a normal IP datagram that has generated an ICMP response. IP encapsulates the appropriate ICMP message with a new IP header (to get the ICMP message back to the original sending host) and transmits the resulting datagram in the usual manner.
For example, every machine (such as intermediate routers) that forwards an IP datagram has to decrement the time to live (TTL) field of the IP header by one; if the TTL reaches 0, an ICMP Time to live exceeded in transit message is sent to the source of the datagram.
Each ICMP message is encapsulated directly within a single IP datagram, and thus, like UDP, ICMP does not guarantee delivery.
Although ICMP messages are contained within standard IP datagrams, ICMP messages are usually processed as a special case, distinguished from normal IP processing, rather than processed as a normal sub-protocol of IP. In many cases, it is necessary to inspect the contents of the ICMP message and deliver the appropriate error message to the application that generated the original IP packet, the one that prompted the sending of the ICMP message.
Many commonly-used network utilities are based on ICMP messages. The traceroute command is implemented by transmitting UDP datagrams with specially set IP TTL header fields, and looking for ICMP Time to live exceeded in transit (above) and "Destination unreachable" messages generated in response. The related ping utility is implemented using the ICMP "Echo request" and "Echo reply" messages.
INTERNET GROUP MANAGEMENT PROTOCOL (IGMP)
The Internet Group Management Protocol (IGMP) is a communications protocol used to manage the membership of Internet Protocol multicast groups (operates in Transport Layer). IGMP is used by IP hosts and adjacent multicast routers to establish multicast group memberships. It is an integral part of the IP multicast specification, like ICMP for unicast connections. IGMP can be used for online video and gaming, and allows more efficient use of resources when supporting these uses. IGMP does allow some attacks[1] [2] [3] [4], and firewalls commonly allow the user to disable it if it will not be needed.
ADDRESS RESOLUTION PROTOCOL (ARP)
In computer networking, the Address Resolution Protocol (ARP) is the standard method for finding a host's hardware address when only its network layer address is known. It operates in Network/Internet layer.
ARP is not an IP-only or Ethernet-only protocol; it can be used to resolve many different network-layer protocol addresses to hardware addresses, although, due to the overwhelming prevalence of IPv4 and Ethernet, ARP is primarily used to translate IP addresses to Ethernet MAC addresses. It is also used for IP over other LAN technologies, such as Token Ring, FDDI, or IEEE 802.11, and for IP over ATM.
ARP is used in four cases of two hosts communicating:
When two hosts are on the same network and one desires to send a packet to the other
When two hosts are on different networks and must use a gateway/router to reach the other host
When a router needs to forward a packet for one host through another router
When a router needs to forward a packet from one host to the destination host on the same network
The first case is used when two hosts are on the same physical network (that is, they can directly communicate without going through a router). The last three cases are the most used over the Internet as two computers on the internet are typically separated by more than 3 hops.
Imagine computer A sends a packet to computer D and there are two routers, B & C, between them. Case 2 covers A sending to B; case 3 covers B sending to C; and case 4 covers C sending to D.
ARP is defined in RFC 826. It is a current Internet Standard, STD 37.
Reverse ARP (RARP), like InARP, also translates Layer 2 addresses to Layer 3 addresses. However, RARP is used to obtain the Layer 3 address of the requesting station itself, while in InARP the requesting station already knows its own Layer 2 and Layer 3 addresses, and it is querying the Layer 3 address of another station. RARP has since been abandoned in favor of BOOTP which was subsequently replaced by DHCP.
REVERSE ADDRESS RESOLUTION PROTOCOL (RARP)
Reverse Address Resolution Protocol (RARP) is a network layer protocol used to obtain an IP address for a given hardware address (such as an Ethernet address). It has been rendered obsolete by BOOTP and the more modern DHCP, which both support a much greater feature set than RARP.
The primary limitations of RARP are that each MAC must be manually configured on a central server, and that the protocol only conveys an IP address. This leaves configuration of subnetting, gateways, and other information to other protocols or the user.
Another limitation of RARP compared to BOOTP or DHCP is that it is a non-IP protocol. This means that like ARP it can't be handled by the TCP/IP stack on the client, but is instead implemented separately.
RARP is the complement of ARP.
RARP differs from Inverse Address Resolution Protocol (InARP), which is designed to locate the IP address associated with another station's MAC address.
RARP is described in RFC 903.
OPEN SYSTEMS INTERCONNECTION BASIC REFERENCE MODEL
The Open Systems Interconnection Basic Reference Model (OSI Reference Model or OSI Model for short) is a layered, abstract description for communications and computer network protocol design, developed as part of the Open Systems Interconnection (OSI) initiative. It is also called the OSI seven layer model. The layers, described below, are, from top to bottom, Application, Presentation, Session, Transport, Network, Data Link, and Physical. A layer is a collection of related functions that provides services to the layer above it and receives service from the layer below it. For example, a layer that provides error-free communications across a network provides the path needed by applications above it, while it calls the next lower layer to send and receive packets that make up the contents of the path.
Even though newer IETF and IEEE protocols, and indeed OSI protocol work subsequent to the publication of the original architectural standards that have largely superseded it, the OSI model is an excellent place to begin the study of network architecture. Not understanding that the pure seven-layer model is more historic than current, many beginners make the mistake of trying to fit every protocol they study into one of the seven basic layers. This is not always easy to do as many of the protocols in use on the Internet today were designed as part of the TCP/IP model, and may not fit cleanly into the OSI model.
Description of OSI layers
OSI Model
Data unit Layer Function
Host layers
-Data
7. Application- Network process to application
6. Presentation- Data representation and encryption
5. Session- Interhost communication
-Segments
4. Transport End-to-end connections and reliability (TCP)
Media layers
-Packets
3. Network Path determination and logical addressing (IP)
-Frames
2. Data link Physical addressing (MAC & LLC)
-Bits
1. Physical Media, signal and binary transmission
Layer 7: Application layer
The application layer interfaces directly to and performs common application services for the application processes; it also issues requests to the presentation layer. Note carefully that this layer provides services to user-defined application processes, and not to the end user. For example, it defines a file transfer protocol, but the end user must go through an application process to invoke file transfer. The OSI model does not include human interfaces. The common application services sublayer provides functional elements including the Remote Operations Service Element (comparable to Internet Remote Procedure Call), Association Control, and Transaction Processing (according to the ACID requirements).
Above the common application service sublayer are functions meaningful to user application programs, such as messaging (X.400), directory (X.500), file transfer (FTAM), virtual terminal (VTAM), and batch job manipulation (JTAM). These contrast with user applications that use the services of the application layer, but are not part of the application layer itself.
File Transfer applications using FTAM (OSI protocol) or FTP (TCP/IP Protocol)
Mail Transfer clients using X.400 (OSI protocol) or SMTP/POP3/IMAP (TCP/IP protocols)
Web browsers using HTTP (TCP/IP protocol); no true OSI protocol for web applications
Layer 6: Presentation layer
The Presentation layer establishes a context between application layer entities, in which the higher-layer entities can use different syntax and semantics, as long as the Presentation Service understands both and the mapping between them. The presentation service data units are then encapsulated into Session Protocol Data Units, and moved down the stack.
The original presentation structure used the Basic Encoding Rules of Abstract Syntax Notation One (ASN.1), with capabilities such as converting an EBCDIC-coded text file to an ASCII-coded file, or serializing objects and other data structures into and out of XML. ASN.1 has a set of cryptographic encoding rules that allows end-to-end encryption between application entities.
Layer 5: Session layer
The Session layer controls the dialogues/connections (sessions) between computers. It establishes, manages and terminates the connections between the local and remote application. It provides for full-duplex, half-duplex, or simplex operation, and establishes checkpointing, adjournment, termination, and restart procedures. The OSI model made this layer responsible for "graceful close" of sessions, which is a property of TCP, and also for session checkpointing and recovery, which is not usually used in the Internet protocols suite. Session layers are commonly used in application environments that make use of remote procedure calls (RPCs).
iSCSI, which implements the Small Computer Systems Interface (SCSI) encapsulated into TCP/IP packets, is a session layer protocol increasingly used in Storage Area Networks and internally between processors and high-performance storage devices. iSCSI uses TCP for guaranteed delivery, and carries SCSI command descriptor blocks (CDB) as payload to create a virtual SCSI bus between iSCSI initiators and iSCSI targets.
Layer 4: Transport layer
The Transport layer provides transparent transfer of data between end users, providing reliable data transfer services to the upper layers. The transport layer controls the reliability of a given link through flow control, segmentation/desegmentation, and error control. Some protocols are state and connection oriented. This means that the transport layer can keep track of the segments and retransmit those that fail.
Although it was not developed under the OSI Reference Model and does not strictly conform to the OSI definition of the Transport Service, the best known example of a layer 4 protocol is the Transmission Control Protocol (TCP). The transport layer is the layer that converts messages into TCP segments or User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP), etc. packets.
Of the actual OSI protocols, not merely protocols developed under the model, there are five classes of transport protocols, ranging from class 0 (which is also known as TP0 and provides the least error recovery) to class 4 (which is also known as TP4 and is designed for less reliable networks, similar to the Internet). Class 4 is closest to TCP, although TCP contains functions, such as the graceful close, which OSI assigns to the Session Layer.
Perhaps an easy way to visualize the Transport Layer is to compare it with a Post Office, which deals with the dispatch and classification of mail and parcels sent. Do remember, however, that a post office manages the outer envelope of mail. Higher layers may have the equivalent of double envelopes, such as cryptographic presentation services that can be read by the addressee only. Roughly speaking, tunneling protocols operate at the transport layer, such as carrying non-IP protocols such as IBM's SNA or Novell's IPX over an IP network, or end-to-end encryption with IPsec. While Generic Routing Encapsulation (GRE) might seem to be a network layer protocol, if the encapsulation of the payload takes place only at endpoint, GRE becomes closer to a transport protocol that uses IP headers but contains complete frames or packets to deliver to an endpoint. L2TP carries PPP frames inside transport packets.
Layer 3: Network layer
The Network layer provides the functional and procedural means of transferring variable length data sequences from a source to a destination via one or more networks while maintaining the quality of service requested by the Transport layer. The Network layer performs network routing functions, and might also perform fragmentation and reassembly, and report delivery errors. Routers operate at this layer—sending data throughout the extended network and making the Internet possible. This is a logical addressing scheme – values are chosen by the network engineer. The addressing scheme is hierarchical.
The best known example of a layer 3 protocol is the Internet Protocol (IP). Perhaps it's easier to visualize this layer as managing the sequence of human carriers taking a letter from the sender to the local post office, trucks that carry sacks of mail to other post offices or airports, airplanes that carry airmail between major cities, trucks that distribute mail sacks in a city, and carriers that take a letter to its destinations. Think of fragmentation as splitting a large document into smaller envelopes for shipping, or, in the case of the network layer, splitting an application or transport record into packets.
Layer 2: Data Link layer
The Data Link layer provides the functional and procedural means to transfer data between network entities and to detect and possibly correct errors that may occur in the Physical layer. Originally, this layer was intended for point-to-point and point-to-multipoint media, characteristic of wide area media in the telephone system. Local area network architecture, which included broadcast-capable multiaccess media, was developed independently of the ISO work, in IEEE Project 802. IEEE work assumed sublayering and management functions not required for WAN use. In modern practice, only error detection, not flow control using sliding window, is present in modern data link protocols such as Point-to-Point Protocol (PPP), and, on local area networks, the IEEE 802.2 LLC layer is not used for most protocols on Ethernet, and, on other local area networks, its flow control and acknowledgment mechanisms are rarely used. Sliding window flow control and acknowledgment is used at the transport layers by protocols such as TCP, but is still used in niches where X.25 offers performance advantages.
Both WAN and LAN services arrange bits, from the physical layer, into logical sequences called frames. Not all physical layer bits necessarily go into frames, as some of these bits are purely intended for physical layer functions. For example, every fifth bit of the FDDI bit stream is not used by the data link layer.
WAN Protocol Architecture
Connection-oriented WAN data link protocols, in addition to framing, detect and may correct errors. They also are capable of controlling the rate of transmission. A WAN data link layer might implement a sliding window flow control and acknowledgment mechanism to provide reliable delivery of frames; that is the case for SDLC and HDLC, and derivatives of HDLC such as LAPB and LAPD.
IEEE 802 LAN Architecture
Practical, connectionless LANs began with the pre-IEEE Ethernet specification, which is the ancestor of the IEEE 802.3. This layer manages the interaction of devices with a shared medium, which is the function of a Media Access Control sublayer. Above this MAC sublayer is the media-independent IEEE 802.2 Logical Link Control (LLC) sublayer, which deals with addressing and multiplexing on multiaccess media.
While IEEE 802.3 is the dominant wired LAN protocol and IEEE 802.11 the wireless LAN protocol, obsolescent MAC layers include Token Ring and FDDI. The MAC sublayer detects but does not correct errors.
Layer 1: Physical layer
The Physical layer defines all the electrical and physical specifications for devices. In particular, it defines the relationship between a device and a physical medium. This includes the layout of pins, voltages, cable specifications, Hubs, repeaters, network adapters, Host Bus Adapters (HBAs used in Storage Area Networks) and more.
To understand the function of the physical layer in contrast to the functions of the data link layer, think of the physical layer as concerned primarily with the interaction of a single device with a medium, where the data link layer is concerned more with the interactions of multiple devices (i.e., at least two) with a shared medium. The physical layer will tell one device how to transmit to the medium, and another device how to receive from it (in most cases it does not tell the device how to connect to the medium). Obsolescent physical layer standards such as RS-232 do use physical wires to control access to the medium.
The major functions and services performed by the physical layer are:
Establishment and termination of a connection to a communications medium.
Participation in the process whereby the communication resources are effectively shared among multiple users. For example, contention resolution and flow control.
Modulation, or conversion between the representation of digital data in user equipment and the corresponding signals transmitted over a communications channel. These are signals operating over the physical cabling (such as copper and optical fiber) or over a radio link.
Parallel SCSI buses operate in this layer, although it must be remembered that the logical SCSI protocol is a transport-layer protocol that runs over this bus. Various physical-layer Ethernet standards are also in this layer; Ethernet incorporates both this layer and the data-link layer. The same applies to other local-area networks, such as Token ring, FDDI, and IEEE 802.11, as well as personal area networks such as Bluetooth and IEEE 802.15.4.
CLASSFUL NETWORK
Classful networking is the name given to the first round of changes to the structure of the IP address in IPv4.
Classful networking is obsolete on the modern Internet. There is no longer any such thing as a class A/B/C network. The correct modern representation for what would have been referred to as a "Class B" prior to 1993 would be "a set of /16 addresses", under the Classless Inter-Domain Routing (CIDR) system.
As a kludge, the definition of IP addresses was changed in 1981 by RFC 791 to allow three different sizes of the network number field (and the associated rest field), as specified in the table below:
Class | Leading Bits | Size of Network | Size of Rest |
Class A | 0 | 7 | 24 |
Class B | 10 | 14 | 16 |
Class C | 110 | 21 | 8 |
Class D (multicast) | 1110 | ||
Class E (reserved) | 1111 |
This allowed the following population of network numbers (excluding addresses consisting of all zeros or all ones, which are not allowed):
Class | Leading Bits | Number of Networks | Addresses Per Network |
Class A | 0 | 128 | 16,777,214 |
Class B | 10 | 16,384 | 65,534 |
Class C | 110 | 2,097,152 | 254 |
The number of valid networks and hosts available is always 2N - 2 (where N is the number of bits used, and the subtraction of 2 adjusts for the invalidity of the first and last addresses). Thus, for a class C address with 8 bits available for hosts, the number of hosts is 254.
The larger network number field allowed a larger number of networks, thereby accommodating the continued growth of the Internet.
The IP address netmask(which is so commonly associated with an IP address today) was not required because the mask length was part of the IP address itself. Any network device could inspect the first few bits of a 32-bit IP address to see which class the address belonged to.
The method of comparing two IP address's physical networks did not change, however. For each address, the network number field size and its subsequent value were determined (the rest field was ignored). The network numbers were then compared. If they matched, then the two addresses were on the same network
Class ranges
The address ranges used for each class are given in the following table, in the standard dotted decimal notation.
Class | Leading bits | Start | End | CIDR equivalent | Default subnet mask |
Class A | 0 | 0.0.0.0 | 127.255.255.255 | /8 | 255.0.0.0 |
Class B | 10 | 128.0.0.0 | 191.255.255.255 | /16 | 255.255.0.0 |
Class C | 110 | 192.0.0.0 | 223.255.255.255 | /24 | 255.255.255.0 |
Class D (multicast) | 1110 | 224.0.0.0 | 239.255.255.255 | /4 | |
Class E (reserved) | 1111 | 240.0.0.0 | 255.255.255.255 | /4 |
SIMPLE NETWORK MANAGEMENT PROTOCOL (SNMP)
The Simple Network Management Protocol (SNMP) forms part of the internet protocol suite as defined by the Internet Engineering Task Force (IETF). SNMP is used in network management systems to monitor network-attached devices for conditions that warrant administrative attention. It consists of a set of standards for network management, including an Application Layer protocol, a database schema, and a set of data objects.
SNMP exposes management data in the form of variables on the managed systems, which describe the system configuration. These variables can then be queried (and sometimes set) by managing applications.
Overview and basic concepts
In typical SNMP usage, there are a number of systems to be managed, and one or more systems managing them. A software component called an agent (see below) runs on each managed system and reports information via SNMP to the managing systems.
Essentially, SNMP agents expose management data on the managed systems as variables (such as "free memory", "system name", "number of running processes", "default route"). The managing system can retrieve the information through the GET, GETNEXT and GETBULK protocol operations or the agent will send data without being asked using TRAP or INFORM protocol operations. Management systems can also send configuration updates or controlling requests through the SET protocol operation to actively manage a system. Configuration and control operations are used only when changes are needed to the network infrastructure. The monitoring operations are usually performed on a regular basis.
The variables accessible via SNMP are organized in hierarchies. These hierarchies, and other metadata (such as type and description of the variable), are described by Management Information Bases (MIBs).
Management Information Bases (MIBs)
SNMP itself does not define which information (which variables) a managed system should offer. Rather, SNMP uses an extensible design, where the available information is defined by management information bases (MIBs). MIBs describe the structure of the management data of a device subsystem; they use a hierarchical namespace containing object identifiers (OID). Roughly speaking, each OID identifies a variable that can be read or set via SNMP. MIBs use the notation defined by ASN.1.
The MIB hierarchy can be depicted as a tree with a nameless root, the levels of which are assigned by different organizations. The top-level MIB OIDs belong to different standards organizations, while lower-level object IDs are allocated by associated organizations. This model permits management across all layers of the OSI reference model, extending into applications such as databases, email, and the Java EE reference model, as MIBs can be defined for all such area-specific information and operations.
A managed object (sometimes called an MIB object, an object, or a MIB) is one of any number of specific characteristics of a managed device. Managed objects comprise one or more object instances (identified by their OIDs), which are essentially variables.
Two types of managed objects exist:
Scalar objects define a single object instance.
Tabular objects define multiple related object instances that are grouped in MIB tables.
An example of a managed object is atInput, which is a scalar object that contains a single object instance, the integer value that indicates the total number of input AppleTalk packets on a router interface.
An object identifier (or object ID or OID) uniquely identifies a managed object in the MIB hierarchy.
SNMP basic components
An SNMP-managed network consists of three key components:
Managed devices
Agents
Network-management systems (NMSs)
A managed device is a network node that contains an SNMP agent and that resides on a managed network. Managed devices collect and store management information and make this information available to NMSs using SNMP. Managed devices, sometimes called network elements, can be any type of device including, but not limited to, routers and access servers, switches and bridges, hubs, IP telephones, computer hosts, or printers.
An agent is a network-management software module that resides in a managed device. An agent has local knowledge of management information and translates that information into a form compatible with SNMP.
An NMS executes applications that monitor and control managed devices. NMSs provide the bulk of the processing and memory resources required for network management. One or more NMSs may exist on any managed network.
TRANSMISSION CONTROL PROTOCOL (TCP)
The Transmission Control Protocol (TCP) is one of the core protocols of the Internet protocol suite. TCP provides reliable, in-order delivery of a stream of bytes, making it suitable for applications like file transfer and e-mail. It is so important in the Internet protocol suite that sometimes the entire suite is referred to as "the TCP/IP protocol suite."
Reason for TCP
The Internet Protocol (IP) works by exchanging groups of information called packets. Packets are short sequences of bytes consisting of a header and a body. The header describes the packet's destination, which routers on the Internet use to pass the packet along, in generally the right direction, until it arrives at its final destination. The body contains the application data.
In cases of congestion, the IP can discard packets, and, for efficiency reasons, two consecutive packets on the internet can take different routes to the destination. Then, the packets can arrive at the destination in the wrong order.
The TCP protocol's software libraries use the IP and provides a simpler interface to applications by hiding most of the underlying packet structures, rearranging out-of-order packets, minimizing network congestion, and re-transmitting discarded packets. Thus, TCP very significantly simplifies the task of writing networked applications.
Applicability of TCP
TCP is used extensively by many of the Internet's most popular application protocols and resulting applications, including the World Wide Web, E-mail, File Transfer Protocol, Secure Shell, and some streaming media applications.
However, because TCP is optimized for accurate delivery rather than timely delivery, TCP sometimes incurs long delays while waiting for out-of-order messages or retransmissions of lost messages, and it is not particularly suitable for real-time applications such as Voice over IP. For such applications, protocols like the Real-time Transport Protocol (RTP) running over the User Datagram Protocol (UDP) are usually recommended instead.
ERROR CONTROL
Network is responsible for transmission of data from one device to another device. The end to end transfer of data from a transmitting application to a receiving application involves many steps, each subject to error. With the error control process, we can be confident that the transmitted and received data are identical. Data can be corrupted during transmission. For reliable communication, error must be detected and corrected.
Error control is the process of detecting and correcting both the bit level and packet level errors.
Types of Errors
Single Bit Error
The term single bit error means that only one bit of the data unit was changed from 1 to 0 and 0 to 1.
Burst Error
In term burst error means that two or more bits in the data unit were changed. Burst error is also called packet level error, where errors like packet loss, duplication, reordering.
Error Detection
Error detection is the process of detecting the error during the transmission between the sender and the receiver.
Types of error detection
- Parity checking
- Cyclic Redundancy Check (CRC)
- Checksum
Parity checking
Parity adds a single bit that indicates whether the number of 1 bits in the preceding data is even or odd. If a single bit is changed in transmission, the message will change parity and the error can be detected at this point. Parity checking is not very robust, since if the number of bits changed is even, the check bit will be invalid and the error will not be detected.
[a] Single bit parity
[b] Two dimension parity
Moreover, parity does not indicate which bit contained the error, even when it can detect it. The data must be discarded entirely, and re-transmitted from scratch. On a noisy transmission medium a successful transmission could take a long time, or even never occur. Parity does have the advantage, however, that it's about the best possible code that uses only a single bit of space.
Cyclic Redundancy Check
CRC is a very efficient redundancy checking technique. It is based on binary division of the data unit, the remainder of which (CRC) is added to the data unit and sent to the receiver. The Receiver divides data unit by the same divisor. If the remainder is zero then data unit is accepted and passed up the protocol stack, otherwise it is considered as having been corrupted in transit, and the packet is dropped.
Sequential steps in CRC are as follows.
Sender follows following steps.
- Data unit is composite by number of 0s, which is one less than the divisor.
- Then it is divided by the predefined divisor using binary division technique. The remainder is called CRC. CRC is appended to the data unit and is sent to the receiver.
Receiver follows following steps.
- When data unit arrives followed by the CRC it is divided by the same divisor which was used to find the CRC (remainder).
- If the remainder result in this division process is zero then it is error free data, otherwise it is corrupted.
Checksum
Check sum is the third method for error detection mechanism. Checksum is used in the upper layers, while Parity checking and CRC is used in the physical layer. Checksum is also on the concept of redundancy.
In the checksum mechanism two operations to perform.
Checksum generator
Sender uses checksum generator mechanism. First data unit is divided into equal segments of n bits. Then all segments are added together using 1's complement. Then it complements ones again. It becomes Checksum and sends along with data unit.
Exp:
If 16 bits 10001010 00100011 is to be sent to receiver.
So the checksum is added to the data unit and sends to the receiver. Final data unit is 10001010 00100011 01010000.
Checksum checker
Receiver received the data unit and divided into equal size of segments. All segments are added using 1's complement. The result is completes once again. If the result is zero, data will be accept, otherwise rejects.
Exp: The final data is nonzero then it is rejected.
Error Correction
This type of error control allows a receiver to reconstruct the original information when it has been corrupted during transmission.
Hamming Code
It is a single bit error correction method using redundant bits.
In this method redundant bits are included with the original data those bits are arrange such that different incorrect bits produce different error results, that incorrect bits can be identified. Once the bit is identified, the receiver can reverse its value and correct the error. Hamming code can be applied to any length of data unit and uses the relationships between the data and the redundancy bits.
Algorithm:
1) Parity bits are positions at the power of two (2 r).
2) Rest of the positions is filled by original data.
3) Each parity bit will take care of its bits in the code.
4) Final code will sends to the receiver.
FLOW CONTROL
Flow Control is one important design issue for the Data Link Layer that controls the flow of data between sender and receiver.
In Communication, there is communication medium between sender and receiver. When Sender sends data to receiver than there can be problem in below case :
1) Sender sends data at higher rate and receive is too sluggish to support that data rate.
To solve the above problem, FLOW CONTROL is introduced in Data Link Layer. It also works on several higher layers. The main concept of Flow Control is to introduce EFFICIENCY in Computer Networks.
Approaches of Flow Control
I.Feed back based Flow Control
II.Rate based Flow Control
Feed back based Flow Control is used in Data Link Layer and Rate based Flow Control is used in Network Layer.
Feed back based Flow Control
In Feed back based Flow Control, Until sender receives feedback from the receiver, it will not send next data.
Types of Feedback based Flow Control
A. Stop-and-Wait Protocol
B. Sliding Window Protocol
A. A Simplex Stop-and-Wait Protocol
In this Protocol we have taken the following assumptions:
1.)It provides unidirectional flow of data from sender to receiver.
2.)The Communication channel is assumed to be error free.
In this Protocol the Sender simply sends data and waits for the acknowledgement from Receiver. That's why it is called Stop-and-Wait Protocol.
This type is not so much efficient, but it is simplest way of Flow Control.
In this scheme we take Communication Channel error free, but if the Channel has some errors than receiver is not able to get the correct data from sender so it will not possible for sender to send the next data (because it will not get acknowledge from receiver). So it will end the communication, to solve this problem there are two new concepts were introduced.
1)TIMER, if sender was not able to get acknowledgement in the particular time than, it sends the buffered data once again to receiver. When sender starts to send the data, it starts timer.
2)SEQUENCE NUMBER, from this the sender sends the data with the specific sequence number so after receiving the data, receiver sends the data with that sequence number, and here at sender side it also expect the acknowledgement of the same sequence number.
This type of scheme is called Positive Acknowledgement with Retransmission (PAR).
B. Sliding Window Protocol
Problems Stop –wait protocol - In the last protocols sender must wait for either positive acknowledgement from receiver or for time out to send the next frame to receiver. So if the sender is ready to send the new data, it can not send. Sender is dependent on the receiver. Previous protocols have only the flow of one sided, means only sender sends the data and receiver just acknowledge it, so the twice bandwidth is used.
To solve the above problems the Sliding Window Protocol was introduce.
In this, the sender and receiver both use buffer, it's of some size, so there is no necessary to wait for the sender to send the second data, it can send one after one without wait of the receiver's acknowledgement.
And it also solve the problem of uses of more bandwidth, because in this scheme both sender and receiver uses the channel to send the data and receiver just send the acknowledge with the data which it want to send to sender, so there is no special bandwidth is used for acknowledgement, so the bandwidth is saved, and this whole process is called PIGGYBACKING.
Types of Sliding Window Protocol
i. A One-Bit Sliding Window Protocol
ii. A Protocol Using Go Back N
iii. A Protocol Using Selective Repeat
i. A One-Bit Sliding Window Protocol
This protocol has buffer size of one bit, so only possibility for sender and receiver to send and receive packet is only 0 and 1. This protocol includes Sequence, Acknowledge, and Packet number.It uses full duplex channel so there is two possibilities:
1.) Sender first start sending the data and receiver start sending data after it receive the data.
2.) Receiver and sender both start sending packets simultaneously,
First case is simple and works perfectly, but there will be an error in the second one. That error can be like duplication of the packet, without any transmission error.
ii. A Protocol Using Go Back N
The problem with pipelining is if sender sending 10 packets, but the problem occurs in 8th one than it is needed to resend whole data. So the protocol called Go back N and Selective Repeat were introduced to solve this problem.In this protocol, there are two possibility at the receiver's end, it may be with large window size or it may be with window size one.
The window size at the receiver end may be large or only of one. In the case of window size is one at the receiver, as we can see in the figure (a), if sender wants to send the packet from one to ten but suppose it has error in 2nd packet, so sender will start from zero, one, two, etc. here we assume that sender has the time out interval with 8. So the time out will occur after the 8 packets, upto that it will not wait for the acknowledgment. In this case at the receiver side the 2nd packet come with error, and other up to 8 were discarded by receiver. So in this case the loss of data is more.
Whether in the other case with the large window size at receiver end as we can see in the figure (b) if the 2nd packet comes with error than the receiver will accept the 3rd packet but it sends NAK of 2 to the sender and buffer the 3rd packet. Receiver do the same thing in 4th and 5th packet. When the sender receiver the NAK of 2nd packet it immediately send the 2nd packet to the receiver. After receiving the 2nd packet, receiver send the ACK of 5th one as saying that it received upto 5 packet. So there is no need to resend 3rd , 4th and 5th packet again, they are buffered in the receiver side.
iii. A Protocol Using Selective Repeat
Protocol using Go back N is good when the errors are rare, but if the line is poor, it wastes a lot of bandwidth on retransmitted frames. So to provide reliability, Selective repeat protocol was introduced. In this protocol sender starts it's window size with 0 and grows to some predefined maximum number. Receiver's window size is fixed and equal to the maximum number of sender's window size. The receiver has a buffer reserved for each sequence number within its fixed window.
Whenever a frame arrives, its sequence number is checked by the function to see if it falls within the window, if so and if it has not already been received, it is accepted and stored. This action is taken whether it is not expected by the network layer.
