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Internet: the basic hardware and software components that make up the Internet, Another way is to describe the

Internet in terms of a networking infrastructure that provides services to distributed applications.

The public Internet is a world-wide computer network, i.e., a network that interconnects millions of computing devices throughout the world. Most of these computing devices are traditional desktop PCs, Unix-based workstations, and so called “servers" that store and transmit information such as WWW pages and e-mail messages In the Internet jargon, all of these devices are called hosts or end systems. End systems, as well as most other "pieces" of the Internet, run protocols that control the sending and receiving of information within the Internet. TCP (the Transmission Control Protocol) and IP (the Internet Protocol) are two of the most important protocols in the Internet. End systems are connected together by communication links. Links are made up of different types of physical media: coaxial cable, copper wire, fiber optics, and radio spectrum. Different links can transmit data at different rates. Usually, end systems are not directly attached to each other via a single communication link. Instead, they are indirectly connected to each other through intermediate switching devices known as routers. A router takes information arriving on one of its incoming communication links and then forwards that information on one of its outgoing communication links the Internet uses a technique known as packet switching that allows multiple communicating end systems to share a path, or parts of a path, at the same time The Internet is really a network of networks. That is, the Internet is an interconnected set of privately and publicly owned and managed networks. Any network connected to the Internet must run the IP protocol and conform to certain naming and addressing conventions. the structure of the interconnection among the various pieces of the Internet, is loosely hierarchical. A Service Description: The Internet allows distributed applications running on its end systems to exchange data with each other. These applications include remote login, file transfer, electronic mail, audio and video streaming, real-time audio and video conferencing, distributed games, the World Wide Web, and much much more

The Internet provides two services to its distributed applications: a connection-oriented service and a connectionless service. Loosely speaking, connection-oriented service guarantees that data transmitted from a sender to a receiver will eventually be delivered to the receiver inorder and in its entirety. Connectionless service does not make any guarantees about eventual delivery 1.3 The Network Edge: In computer networking jargon, the computers that we use on a daily basis are often referred to as or hosts or end systems. They are referred to as "hosts" because they host (run) application-level programs such as a Web browser or server program, or an e-mail program. They are also referred to as "end systems" because they sit at the "edge" of the Internet. Hosts are sometimes further divided into two categories: clients and servers. Informally, clients often tend to be desktop PC's or workstations, while servers are more powerful machines. In the so-called client-server model, a client program running on one end system requests and receives information from a server running on another end system. The client and the server interact with each other by communicating (i.e., sending each other messages) over the Internet.

Connection less and connection oriented services: We have seen that end systems exchange messages with each other according to an application-level protocol in order to accomplish some task. The Internet, and more generally TCP/IP networks, provide two types of services to its applications: connectionless service and connection-oriented service.

In particular. Indeed.Connection oriendted services: When an application uses the connection-oriented service. it sends an acknowledgment. so a source never knows for sure which packets arrive at the destination. we mean that an application can rely on the connection to deliver all of its data without error and in the proper order. But the two end systems are connected in a very loose manner. only the end systems themselves are aware of this connection. This so-called handshaking procedure alerts the client and server. hence the terminology "connection-oriented". if every pair of communicating end systems continues to pump packets into the network as fast as they can. its buffers can overflow and packet loss can occur In such circumstances. But there are no acknowledgments either. When a router becomes congested. Flow control makes sure that neither side of a connection overwhelms the other side by sending too many packets too fast. 1. flow control and congestion control. there is a risk of overwhelming either side of an application. the resources needed along a path (buffers. When one side of an application wants to send packets to another side of an application. it assumes that the packet it sent was not received by B. the client and the server (residing in different end systems) send control packets to each other before sending packets with real data (such as e-mail messages). link . The Internet's connection oriented service comes bundled with several other services. including reliable data transfer.4 Network core: There are two fundamental approaches towards building a network core: circuit switching and packet switching. Reliability in the Internet is achieved through the use of acknowledgments and retransmissions. By reliable data transfer. To get a preliminary idea about how the Internet implements the reliable transport service. The Internet's congestion control service helps prevent the Internet from entering a state of grid lock. When end system B receives a packet from A. In circuit-switched networks. When end system A doesn't receive an acknowledgment. The flow-control service forces the sending end system to reduce its rate whenever there is such a risk. it therefore retransmits the packet. consider an application that has established a connection between end systems A and B. the application at one one side of the connection may not be able to process information as quickly as it receives the information. Once the handshaking procedure is finished. it knows that the corresponding packet has definitely been received. gridlock sets in and few packets are delivered to their destinations Connection less services: There is no handshaking with the Internet's connectionless service. the sending application simply sends the packets. allowing them to prepare for an onslaught of packets. a "connection" is said to be established between the two end systems. data can be delivered faster. Therefore. when end system A receives the acknowledgment. Since there is no handshaking procedure prior to the transmission of the packets.

PCs and workstations) are each directly connected to one of the switches. For TDM.000 Hertz or 4. a session's messages use the resource on demand. The transmission rate of the frame is equal to the frame rate multiplied by the number of bits in a slot. 4.. so that each link can support n simultaneous connections. In this network the three circuit switches are interconnected by two links. In packet-switched networks. For FDM.bandwidth) to provide for communication between the endsystems are reserved for the duration of the session.g. the time domain is segmented into four circuits. illustrates FDM and TDM for a specific network link. each of these links has n circuits. In the below diagram illustrates a circuit-switched network. The endsystems (e. A circuit in a link is implemented with either frequency division multiplexing (FDM) or timedivision multiplexing (TDM). these resources are not reserved. Specifically. may have to wait (i.e. and as a consequence. The trend in modern telephony is to replace FDM with TDM. the network dedicates one time slot in every frame to the connection.. each circuit is assigned the same dedicated slot in the revolving TDM frames. queue) for access to a communication link Circute switching: it is important to understand why the Internet and other computer networks use packet switching rather than the more traditional circuit-switching technology used in the telephone networks. . time is divided into frames of fixed duration and each frame is divided into a fixed number of time slots. the link dedicates a frequency band to each connection for the duration of the connection. With FDM.000 cycles per second).. the frequency spectrum of a link is shared among the connections established across the link. the frequency domain is segmented into a number of circuits. each of bandwidth 4 KHz (i. When the network establish a connection across a link. For a TDM link.e.

000 frames per second and each slot consists of 8 bits. if the link transmits 8. an arriving packet may find that the buffer is completely filled with other packets waiting for transmission. Also suppose that it takes 500 msec to establish an end-to-end circuit before A can begin to transmit the file.536 Mbps)/24 = 64 Kbps. If an arriving packet needs to be transmitted across a link but finds the link busy with the transmission of another packet. Most packet switches use store and forward transmission at the inputs to the links. How long does it take to send the file? Each circuit has a transmission rate of (1. Between source and destination.either the arriving packet or one of the alreadyqueued packets will be dropped. For example. Suppose that all links in the network use TDM with 24 slots and have bit rate 1. . packets suffer output buffer queueing delays. packet loss will occur . then the transmission rate is 64 Kbps Let us consider how long it takes to send a file of 640 Kbits from host A to host B over a circuitswitched network. each of these packets traverse communication links and packet switches (also known as routers).The transmission rate of the frame is equal to the frame rate multiplied by the number of bits in a slot. the source breaks long messages into smaller packets. Packets are transmitted over each communication link at a rate equal to the full transmission rate of the link. so it takes (640 Kbits)/(64 Kbps) = 10 seconds to transmit the file Packet switching: In modern packet-switched networks. the arriving packet must wait in the output buffer addition to the store-and-forward delays. In this case. Storeand-forward transmission means that the switch must receive the entire packet before it can begin to transmit the first bit of the packet onto the outbound link The output buffers play a key role in packet switching. Since the amount of buffer space is finite.536 Mbps. These delays are variable and depend on the level of congestion in the network.

it supports more than 10 simulatanious users. Packet switching versus Circute switching: Let us compare the packet switching and circute switching. Let us look at a simple example. 100 Kbps must be reserved for each user at all times. they send a message into the network as a whole. Consider now how Host A and Host B packets are transmitted onto this link.544 Mbps link. The above diagram illustrates message switching in a route consisting of two packet switches (PSs) and three links.With circuit-switching. more efficient.e. With packet switching. Hosts A and B first send their packets along 28. Suppose users share a 1 Mbps link and each user alternates between periods of activity (when it generates data at a constant rate of 100Kbits/sec) and periods of inactivity (when it generates no data).illustrates a simple packet-switched network. With message switching.8 Kbps links to the first packet switch.. switches are store-and-forward packet switches. packet switching has more advantages i. the link can support only ten simultaneous users. Message switching network: a packet-switched network performs message switching if the sources do not segment messages. The packet switch directs these packets to the 1. Suppose Hosts A and B are sending packets to Host E. If there is congestion at this link. it offers better sharing of bandwidth than circuit switching and (2) it is simpler. Thus. and less costly to implement than circuit-switching. i. If the user is active only 10% of the time .e. the packets queue in the link's output buffer before they can be transmitted onto the link. .

it takes 5 seconds to move the message from the first switch to the second switch.5 Mbits long. Message switching performance: Consider a message that is 7. the first packet has arrived at the destination. Assuming there is no congestion in the network. Thus it takes ten seconds to move the message from the source to the second switch. Again assuming that there is no congestion in the network. And it takes the first switch 1 msec to move this first packet from the first to the second . Because the switches use store-and forward.5 Kbits long.5Mbps. It takes the source 1 msec to move the first packet from the source to the first switch. and the last two packets are still in the source. One major advantage of packet switching (with segmented messages) is that it achieves end-toend delays that are typically much smaller than the delays associated with message-switching. It takes the source 5 seconds to move the message from the source to the first switch. Once the first switch has received the entire message. with each packet being 1. the second and third packets are in transit in the network. Following this logic we see that a total of 15 seconds is needed to move the message from source to destination Packet switching performance: Continuing with the same example. and that each link has a transmission rate of 1. now suppose that the source breaks the message into 5000 packets.a packet switch must receive the entire message before it can begin to forward the message on an outbound link Packet Switching network: In the above diagram. Suppose that between source and destination there are two packet switches and three links. the first switch cannot begin to transmit any bits in the message onto the link until this first switch has received the entire message. the original message has been divided into five distinct packets.

When a switch detects an error in a packet. If. while one node (the source or one of the switches) is transmitting. if the entire message is a packet and one bit in the message gets corrupted. a series of links and packet switches) between the source and destination hosts. one number for each link along the path. Virtual circuit network: A virtual circuit (VC) consists of (1) a path (i. message switching is performing sequential transmission whereas packet switching is performing parallel transmission. We shall call any network that routes packets according to virtual-circuit numbers a virtual-circuit network. But while the first packet is being moved from the first switch to the second switch. We shall call any network that routes packets according to host destination addresses a datagram network. Routing in Data networks: There are two broad classes of packet-switched networks: datagram networks and virtualcircuit networks. Observe that with message switching. . three nodes transmit at the same time. (2) virtual circuit numbers. once the first packet reaches the last switch.. the entire message is discarded. The IP protocol of the Internet routes packets according to the destination addresses. hence the Internet is a datagram network.switch. the remaining nodes are idle. the message is segmented into many packets and one bit in one of the packets is corrupted. So. it typically discards the entire packet. on the other hand. With packet switching. the second packet is simultaneously moved from the source to the first switch. and (3) entries in VC-number translation tables in each packet switch along the path. then only that one packet is discarded.e.

.the physical link(s) that connect an end system to its edge router. narrowband ISDN technology (Integrated Services Digital Network) allows for all-digital transmission of data from a home end system over ISDN "telephone" lines to a phone company central office. Ethernet technology is currently by far the most prevalent access technology in enterprise networks. that provides higher speed access (e. packets can be sent with the appropriate VC numbers. In this case. l mobile access networks. The home modem converts the digital output of the PC into analog format for transmission over the analog phone line. while simultaneously surfing the Web HFC access networks are extensions of the current cable network used for broadcasting cable television. 128 Kbps) from the home into a data network such as the Internet.Once a VC is established between source and destination. due to the poor quality of twisted-pair line between many homes and ISPs. One important characteristic of the HFC is that it is a shared broadcast medium. from the home end system to the central office router. a local area network (LAN) is used to connect an end system to an edge router there are many different types of LAN technology. many users get an effective rate significantly less than 56 Kbps. Asymmetric Digital Subscriber Line (ADSL) and hybrid fiber coaxial cable (HFC) are currently being deployed. each packet contains the destination address in the hader .g. One of the features of ADSL is that the service allows the user to make an ordinary telephone call. In particular. using the POTs channel. When a packet arrives at a packet switch in the network. Probably the most common form of home access is using a modem over a POTS (plain old telephone system) dialup line to an Internet service provider (ISP). ADSL is conceptually similar to dialup modems. l institutional access networks. but can transmit at rates of up to about 8 Mbps from the ISP router to a home end system. Cable modems divide the HFC network into two channels. this address has a hierarchical structure. A modem in the ISP converts the analog signal back into digital form for input to the ISP router. connecting a home end system into the network. Access networks can be loosely divided into three categories: l residential access networks. Enterprise Access network: In enterprise access networks. connecting an end system in a business or educational institution into the network. Ethernet operates 10 Mbps or 100Mbps (and now even at 1 Gbps). However. and every packet sent by a home travels on the upstream channel to the headend. 1. a downstream and an upstream channel.5 Access Networks and Physical Media: access network . Datagram Networks: In a datagram network. but perhaps a Web TV or other residential system) to an edge router. connecting a mobile end system into the network Residential access network: A residential access network connects a home end system (typically a PC. every packet sent by the headend travels downstream on every link to every home. is less than 1 Mbps. the "access network" is simply a point-to-point dialup link into an edge router However. It uses either twisted-pair . each packet switch has a routing table which maps destination addresses to an outbound link. The data rate in the reverse direction.

. in fact.copper wire are coaxial cable to connect a number of end systems with each other and with an edge router. With this construction and a special insulation and shielding. It is commonly used in LANs. is about a centimeter thick. let us reflect on the brief life of a bit The source endsystem first transmits the bit and shortly thereafter the first router in the series receives the bit. and stiffer than the baseband variety. for local area networks (LANs). the first router then transmits the bit and shortly afterwards the second router receives the bit. Broadband coaxial cable. lightweight.g. a number of pairs are bundled together in a cable by wrapping the pairs in a protective shield. each about 1 mm thick. is quite common in cable television systems. Baseband coaxial cable. Physical media: In order to define "physical medium. Both baseband and broadband coaxial cable can be used as a guided shared medium. Twisted pair consists of two insulated copper wires. Mobile Access Networks Mobile access networks use the radio spectrum to connect a mobile end system (e. however. the bit is sent by propagating electromagnetic waves across a physical medium TwistedPair Copper Wire The least-expensive and most commonly-used transmission medium is twisted-pair copper wire. Unshielded twisted pair Unshielded twisted pair (UTP) is commonly used for computer networks within a building. Data rates for LANs using twisted pair today range from 10 Mbps to 100 Mbps. that is. heavier. A wire pair constitutes a single communication link. Coaxial cable comes in two varieties: baseband coaxial cable and broadband coaxial cable. a laptop PC or a PDA with a wireless modem) to a base station. etc. the computer you use at work or at school is probably connected to a LAN with either baseband coaxial cable or with UTP. Broadband cable. also called 75-ohm cable. The wires are twisted together to reduce the electrical interference from similar pairs close by. arranged in a regular spiral pattern. is quite a bit thicker. and easy to bend. The data rates that can be achieved depend on the thickness of the wire and the distance between transmitter and receiver Coaxial-Cable coaxial cable consists of two copper conductors. also called 50-ohm cable. but the two conductors are concentric rather than parallel. . coaxial cable can have higher bit rates than twisted pair. An emerging standard for wireless data networking is Cellular Digital Packet Data (CDPD). Typically.

Terrestrial radio channels can be broadly classified into two groups: those that operate as local area networks (typically spanning 10's to a few hundred meters) and wide-area radio channels that are used for mobile data services (typically operating within a metropolitan region). the packet suffers from several different types of delays at each node along the path. can penetrate walls. transmission delay and propagation delay. up to tens or even hundreds of gigabits per second These characteristics have made fiber optics the preferred long-haul guided transmission media. The most important of these delays are the nodal processing delay. 1. flexible medium that conducts pulses of light. provide connectivity to a mobile user. A single optical fiber can support tremendous bit rates. and then directs the packet . When the packet arrives at router A (from the upstream node).Fiber Optics An optical fiber is a thin. these delays accumulate to give a total nodal delay. together. router A examines the packet's header to determine the appropriate outbound link for the packet. with each pulse representing a bit. They are an attractive media because require no physical "wire" to be installed. queuing delay. and can potentially carry a signal for long distances.6 Delay and Loss in packet Switched network: As a packet travels from one node (host or router) to the subsequent node (host or router) along this path. particularly for overseas links Terrestrial and Satellite Radio Channels Radio channels carry signals in the electromagnetic spectrum.

our packet can be transmitted once all the packets that have arrived before it have been transmitted. With no place to store such a packet. such as the time needed to check for bit-level errors in the packet that occurred in transmitting. This is the amount of time required to transmit all of the packet's bits into the link. The national and international ISPs are connected together at the highest tier in the hierarchy.) connected to local Internet Service Providers (ISPs). Roughly speaking. as is common in the the link. the router directs the packet to the queue that precedes the link to router B. For example. that is. At the queue. while the last packet transmitted will suffer a relatively large queuing delay Packet Loss: we have assumed that the queue is capable of holding an infinite number of packets a packet can arrive to find a full queue. This model is structured in five layers. the first packet transmitted will suffer no queuing delay. 1. the packet experiences a queuing delay as it waits to be transmitted onto the link. the newly arriving packet will then join the queue. Transport layer 5. the packet will be lost 1. workstations. 1. The local ISPs are in turn connected to regional ISPs. a router will drop that packet. Physical layer 2. Network layer 4. Denote the length of the packet by L bits and denote the transmission rate of the link (from router A to router B) by R bits/sec. The time required to examine the packet's header and determine where to direct the packet is part of the processing delay. transmission delay (also called the store-and-forward delay eg: Assuming that packets are transmitted in first-come-first-serve manner. etc.8 Internet Backbone and ISP: the topology of the Internet is loosely hierarchical.7 5-Layer TCP/IP Model The basic structure of communication networks is represented by the Transmission Control Protocol/Internet Protocol (TCP/IP) model. In this example. A packet can only be transmitted on a link if there is no other packet currently being transmitted on the link and if there are no other packets preceding it in the queue. which are in turn connected to national and international ISPs. from bottom-to-top the hierarchy consists of end systems (PCs. the outbound link for the packet is the one that leads to router B. if the link is currently busy or if there are other packets already queued for the link. L/R. if ten packets arrive to an empty queue at the same time. Link layer 3. Queue Delay: the queuing delay can vary from packet to packet. The processing delay can also include other factors. Application layer .

the link layer. does not involve any tasks in layers 4 and 5. such as switches and routers. Layer 2 specifies how packets access links and are attached to additional headers to form frames when entering a new networking environment. A message is transmitted from host 1 to host 2. Consequently. . the data is transmitted upward from the physical layer to the application layer. acting as a gateway to the operating regions of host 2. Among such applications are the Simple Mail Transfer Protocol (SMTP). The same scenario is applied at the other end: router R2.Layer 1. and the World Wide Web (WWW). such as a LAN. The data being transmitted from host 1 is passed down through all five layers to reach router R1. provides a reliable synchronization and transfer of information across the physical layer for accessing the transmission medium. Similarly. the application layer. This layer handles the way that addresses are assigned to packets and the way that packets are supposed to be forwarded from one end point to another. lies just above the network layer and handles the details of data transmission. Layer 2 also provides error detection and flow control. Layer 3. Router R1 is located as a gateway to the operating regions of host 1 and therefore does not involve any tasks in layers 4 and 5. File Transfer Protocol (FTP). The physical layer represents the basic network hardware. defines electrical aspects of activating and maintaining physical links in networks. and. the physical layer. the transport layer. Finally at host 2. the network layer (IP) specifies the networking aspects. determines how a specific user application should use a network. router R2. Layer 4 is implemented in the end-points but not in network routers and acts as an interface protocol between a communicating host and a network. Layer 4. all five layers of the protocol model participate in making this connection. this layer provides logical communication between processes running on different hosts Layer 5. Layer 2. as shown.

and cost. The . The network layer. timestamp. An IP header contains the IP addresses of a source node and a destination node. such as priority level. Version specifies the IP version. To make the system scalable. Options is a rarely used variable-length field to specify security level. flags. with 20 bytes of fixed-length header and an options field whose size is variable up to 40 bytes.1. the address structure is subdivided into the network ID and the host ID. throughput. The size of the header is variable. respectively. IP Addressing Scheme An IP address is a unique identifier used to locate a device on the IP network. handles the method of assigning addresses to packets and determines how they should be forwarded from one end point to another. in fact. including the header and data. Header checksum is a method of error detection Source address and destination address are 32-bit fields specifying the source address and the destination address. respectively. Header length (HL) specifies the length of the header. The Internet Protocol produces a header for packets. reliability. and type of route. Type of service specifies the quality-of-service (QoS) requirements of the packet.8 Internet Protocols and Addressing The third layer of communication protocol hierarchy is the network layer. A brief description of the fields follows. and fragment offset are used for packet fragmentation and reassembly. An IP packet can be encapsulated in the layer 2 frames when the packet enters a LAN IP Packet Each packet comprises the header and data. Identification. Total length specifies the total length of the packet in bytes. Protocol specifies the protocol used at the destination. Padding is used to ensure that the header is a multiple of 32 bits. A total of 16 bits are assigned to this field. delay. Time to live specifies the maximum number of hops after which a packet must be discarded.

and E (reserved). or subnet scheme. Depending on the network size. Class A starts with a 0 and supports 126 networks and 16 million hosts per network. Classless Interdomain Routing (CIDR) . For example.. D (multicast). In this scheme.Based on the bit positioning assigned to the network ID and the host ID. Class E always starts with 1111 reserved for network experiments. Subnet Addressing and Masking The concept of subnetting was introduced to overcome the shortcomings of IP addressing. Class C addressing starts with 110 and supports 2 million networks and 254 hosts per ID identifies the network the device belongs to. Class D addressing starts with 1110 and is specifically designed for multicasting and broadcasting. a multiple-network address scheme. For ease of use. B. the IP address is further subdivided into classes A. the IP address is represented in dot-decimal notation. C. a company that uses a class B addressing scheme supports 65. If the company has more than one network.534 hosts per network.382 networks and 65. Class B addressing always starts with 10 and supports 16. the host ID of the original IP address is subdivided into subnet ID and host ID. the host ID identifies the device. different values of subnet ID and host ID can be chosen.534 hosts on one network.

whereas another organization may choose a 21-bit network ID with the first 20 bits of these two network IDs being identical.500 bytes.0/21. they are reassembled at the final destination to form the original packet. The offset field indicates the position of a fragment in the sequence of fragments making up the packet.0. owing to a longer match. Each fragment is routed independently through the network.8. L1 and L2. CIDR dictates that the longer prefix be the eligible match.0/20 and 205. Example. For example. the routing table may have two entries with the same prefix.101.0.1 is received by router R1. The packet has an IP header of 20 bytes plus another . Once all the fragments are received. Example. belonging to 205. link L1. Because of the use of a variable-length prefix. with its 21-bit prefix. In the entries of this router. must be divisible by 8. respectively.Classless interdomain routing (CIDR) is extremely flexible. are matched. Assume that a packet with destination IP address 205.101. As indicated at the bottom of this figure. one organization may choose a 20-bit network ID. a single routing entry is sufficient to represent a group of adjacent addresses. When the MF bit is set. The lengths of all the fragments. and offset fields of the IP header help with the fragmentation and reassembly process. flag. as shown in below figure. CIDR results in a significant increase in the speed of routers and has greatly reduced the size of routing tables.101. N3. This means that the address space of one organization contains that of another one. The identification. The maximum transmission unit (MTU) represents this restriction.The identification field is used to distinguish between various fragments of different packets. it implies that more fragments are on their way. with the exception of the last one. The physical layer has an MTU of 1. This link eventually routes the packet to the destination network. depending on the physical network being used. The fragments could in turn be split into smaller fragments. two routes.500 bytes. Suppose that a host application needs to transmit a packet of 3. is selected. allowing a variable-length prefix to represent the network ID and the remaining bits of the 32-field address to represent the hosts within the network. The flag field has a more-fragment (MF) bit. Packet Fragmentation and Reassembly The physical capacity of networks enforces an upper bound on the size of packets.

Instead. A brief description of the fields in the header follows. if in the middle of routing.540 bytes to be split into fragments of 1. routers in a network can report errors through the Internet Control Message Protocol (ICMP).500 . MF. as they do not know the address of R1. IP Version 6 (IPv6) 128-bit address spacing was introduced with Internet Protocol version 6 (IPv6). offset 182.20 = 1. Because 1. and fragment 3 = total length 628. the data to be transmitted is then 3. With TCP/IP. MF 1. R5 or R6 finds out about this error.456. IPv6 also supports real-time applications. it cannot issue an ICMP message to R1 to correct the routing.456 bytes. Solution. the allowable data length is limited to 1. IP may not be able to deliver a packet to its destination. fragment 2 = total length 1. they issue a redirect ICMP message to the source. owing to possible failures in the connectivity of a destination. But R1 incorrectly sends the message to a wrong path (R1-R3-R4-R5-R6) instead of to the short one (R1-R2-R6). offset 0. 1. and offset fields of all fragments. An ICMP message is encapsulated in the data portion of an IP datagram (packet). Fragment the packet. and offset 364.456. MF 1. When an error occurs. including those that require guaranteed QoS.460 bytes. Another issuerelated and equally importantis that a sender cannot know whether a delivery failure is a result of a local or a remote technical difficulty. Internet Control Message Protocol (ICMP) In large communication networks.456 and 628 bytes. MF 0. a source tries to send a message to a destination. and specify the ID.20 . The allowable data length = 1. Here fragment 1 = total length 1. Including the headers. ICMP reports it to the originating source of the connection.attached header of 20 bytes.456.460 is not divisible by 8. In this case. indicating the version number of the protocol Traffic class specifies the priority level assigned to a packet. .

If multiple extension headers are used. and header length fields. they are concatenated. eliminating the fragmentation. Source address and destination address are each identified by a 128-bit field address. must be delivered. For example. Extension Header: Extension headers are positioned between the header and the payload. The removal of the checksum field in IPv6 allows for faster processing at the routers without sacrificing functionality. excluding the header. IPv6 can provide built-in security features such as confidentiality and authentication Ipv6 Address formate: IPv6 network addressing is very flexible. such as real-time video. IPv6 has a simpler header format. The functionality of the option field in IPv4 is specified in the extension header.Flow label indicates the delay period within which application packets. [2FB4 : 10AB : 4123 : CEBF : 54CD : 3912 : AE7B : 0932] can be a source address.9 Equal size pocket model: . A colon separates each of the four hexadecimal digits. hexadecimal digits are used. the checksum. Hop limit is the same as the time-to-live field in IPv4. 1. In addition. the extension header is more flexible than the options field. Next header specifies the type of extension header used. Payload length is the 16-bit specification of the length of the data.

and bulk data regardless of traffic types and the speed of sources. As shown in below figure the identity of a "physical" link is identified by two "logical" links: virtual channel (VC) and virtual path (VP). The use of fixed-size cells can greatly reduce the overhead of processing ATM cells at the buffering and switching stages and hence increase the speed of routing. The three-dimensional model includes four layers in the vertical dimension The tightly linked layers consist of the physical layer. switching. The physical layer includes two sublayers: the physical medium and transmission convergence. and multiplexing functions The use of fixed-size cells can greatly reduce the overhead of processing ATM cells at the buffering and switching stages and hence increase the speed of routing. and multiplexing functions. the ATM layer. the ATM adaptation layer (AAL). ATM typically supports such bursty sources as FAX. VCI and VPI are combined to be used in a switch to route a cell. the traffic is converted into 53-byte ATM cells. and higher layers. which means that a connection must be preestablished between two systems in a network before any data can be transmitted. Each cell has a 48-byte data payload and a 5-byte header. ATM is a set of connection-oriented protocols. ATM Protocol structure: The ATM protocol structure is shown in below figure. coded video. ATM connections are identified by a virtual channel identifier (VCI) and a virtual path identifier (VPI). switching.The objective of Asynchronous Transfer Mode (ATM) technology is to provide a homogeneous backbone network in which all types of traffic are transported with the same small fixed-sized cells. The physical medium sublayer defines the physical and electrical/optical interfaces with the transmission media on both the transmitter and the receiver .

The higher layers incorporate some of the functionality of layers 3 through 5 of the TCP/IP model The control plane at the top of the cube involves all kinds of network signaling and control.. such as the flow-control and error-control mechanisms. An ATM network can support both a user-network interface (UNI) and a network-node interface (NNI). The AAL layer maps higher-layer service data units.The ATM layer provides services. A UNI is an interface connection between a terminal and an ATM switch. this layer collects and reassembles ATM cells into service data units for transporting to higher layers. generic flow control. The user plane involves the transfer of user information. In addition. header cell check generation and extraction. and most important.. which are fragmented into fixed-size cells to be delivered over the ATM interface. (b) NNI format is shown in the following figure. remapping of VPIs and VCIs.. whereas an NNI connects two ATM switches Overview of signaling: (a) UNI format. including cell multiplexing and demultiplexing. NNI format is shown in the following figure: . The management plane provides management function and an informationexchange function between the user plane and the control plane.

The header consists of several fields. The 1-bit cell-loss priority (CLP) field is used to prioritize cells. However. cells with CLP set to 1 (considered low priority) are discarded first. the cell gets higher priority and should be discarded only if it could not be delivered The 8-bit header error control (HEC) field is used for error checking. HEC functions include correcting single-bit errors and detecting multiple-bit errors . The details of the UNI 5-byte header are as follows: The 4-bit generic flow control (GFC) field is used in UNI only for controlling local flow control the virtual path identifier and the virtual channel identifier represent an ATM address. If the bit is set to 0.ATM cell structures: An ATM cell has two parts: a 48-byte payload and a 5-byte header. A VCI identifies a virtual channel within a virtual path. A VPI identifies a group of virtual channels with the same end point. When congestion occurs. the ATM cell header has two different formats: UNI and NNI.

Multiplexers are used in a network for maximum transmission capacity of a high-bandwidth line.Unit –II 2. wavelength-division multiplexing. Multiplexing schemes can be divided into three basic categories: frequency-division multiplexing. Wavelength Division multiplexing (WDM): Wavelength-division multiplexing (WDM) is fundamentally the same as FDM. WDM was invented as a variation of frequency-division multiplexing and is basically a multiplexing method of different wavelengths instead of frequencies. the following figure shows how n frequency channels are multiplexed using FDM. In the figure. in which each user can be assigned a band.1 Multiplexers: multiplexing is a technique that allows many communication sources to transmit data over a single physical line. or channels. n optical fibers come together at an optical multiplexer. This overlap normally creates spike noise at the edge of each channel. each with its energy present at a different wavelength. any two adjacent channels have some overlap because channel spectra do not have sharp edges. time-division multiplexing Frequency Division Multiplexing: (FDM) : In frequency-division multiplexing (FDM). The n optic lines are combined onto a single shared link for transmission to a distant destination. . FDM is normally used over copper wires or microwave channels and is suitable for analog circuitry. the frequency spectrum is divided into frequency bands.

This method removes all the empty slots on a frame and makes the multiplexer operate more efficiently. is greater than n channels. Synchronous Time Division Multiplex: In synchronous TDM. and the multiplexer scans them. m. based on demand. whereby unassigned sources are partially transmitted.2 Modems and Internet Access Devices: . the scanner should stay on that line. The scanning time for each line is Preallocated. the lack of data in any channel potentially creates changes to average bit rate on the ongoing link Statistical TDM: a frame's time slots are dynamically allocated. If the number of requesting input sources. time is divided into frames. the multiplexer scans all lines without exception. Packets arrive on n lines. or clipped. whether or not there is data for scanning within that time slot.n optical fibers come together at an optical multiplexer. or channels. each with its energy present at a different wavelength instead of frequencies Time Division Multiplex: With a time-division multiplexing (TDM). forming a frame with n channels on its outgoing link. the multiplexer typically reacts by clipping. 2. each one periodically getting the entire bandwidth for a portion of the total scanning time. users take turns in a predefined fashion. Once a synchronous multiplexer is programmed to produce same-sized frames. and each frame is further subdivided into time slots. Consider a multiplexer with n available channels. Each channel is allocated to one input. Given n inputs.

A modem is a device that converts the digital data to a modulated form of signal that takes less bandwidth. and a 0 is represented by a 0 voltage. a binary information sequence is converted into a digital code. A user can access the Internet by using either the existing telephone link infrastructure or the existing cable TV infrastructure. With line coding. The simplest form of line coding is the natural nonreturn-to-zero (NRZ) where a binary 1 is represented by a +V voltage level.Users access the Internet from residential areas primarily through modems. In this method. This process is required to maximize bit rate in digital transmission. a line coding process is performed on binary signals for digital transmission. A more powerefficient line coding method is known as polar NRZ. a binary 1 is mapped to +V/2 and a binary 0 is represented by -V/2. . Two commonly used modems are the digital subscriber line (DSL) modem and the cable modem Line Coding Methods: Before processing a raw signal for modulation. the cost and complexity are the main factor in the selection of encoder. Encoded signals are produced by the line codes for the binary sequence 1011 0100 1110 0010 are shown in the following figure.

the remaining bandwidth can become available to be allocated to data communications The standard modulation technique for ADSL is QAM. asymmetric DSL (ADSL) is popular and is designed for residential users. each using abandwidth of approximately 4. Voice communication uses channel 0. HDSL. The details of spectrum division for an ADSL modem are shown in below diagram . With the Manchester encoding method.312 KHz. only 4 KHz are used for a phone conversation. VDSL. the binary information is mapped into transitions at the beginning of each interval so that a binary 1 is converted to a transition at the beginning of a bit time and a 0 having no transition. and a binary 0 is represented by a 0 plus a transition to 1 and then a 1. The available bandwidth of 1. and SDSL. Channels 15 remain idle andtogether act as a guard band between voice and data communication. These links are capable of handling bandwidths up to 1. or. This technology offers various versions of DSL technology: ADSL. Because data communication bandwidth is split into two bandwidthsupstream for communications from the user to the Internet and downstream for communications from the Internet to the user the technique is said to be asymmetric. A modem is designed to be connected to telephone links. A great feature of the Manchester encoding is that it is self-clocking Digital Modulation Techniques: In order to reduce the bandwidth of digital signals.1 MHz is divided into 256 channels. the following types of modulation techniques: Amplitude shift keying (ASK) Frequency shift keying (FSK) Phase shift keying (PSK) Quadrature amplitude modulation (QAM) Digital Subscriber Line (DSL) Modems: Digital subscriber line (DSL) technology is a convenient option for home users to access the Internet. Out of this bandwidth.1 MHz. a binary 1 is represented by a 1 plus a transition to 0 and then a 0.NRZ-inverted coding. digital modulation technique is required before any transmission. in general. xDSL.

an encoding technique less susceptible to attenuationHDSL can achieve a data rate of 2 Mb/s without needing repeaters for up to 3. Very high bit-rate digital subscriber line (VDSL) is similar to ADSL but uses coaxial or fiber-optic cable for a bit rate of 50 Mb/s to 55 Mb/s downstream and 1. Video signals are transmitted downstream from headquarters to users.5 Mb/s upstream data transfer.Another type of DSL technique is symmetric digital subscriber line (SDSL). HDSL uses two twisted-pair wires and 2B1Q. depending on its demand and budget. High-bit-rate digital subscriber line (HDSL). each of which can then be connected to the optical infrastructure for TV. About 500 MHz of this bandwidth is assigned to TV channels. as another option. Cable Modems: A cable company lays out very high-speed backbone optical fiber cables all the way to the residential buildings. This network is called hybrid fiber-coaxial (HFC). The cable company divides the bandwidth into video/radio. and the Internet through either a coaxial (coax) cable or optical fiber. radio. This technique divides the available bandwidth equally between downstream and upstream data transfer. and upstream data. Some technical methods allow . downstream data.5 Mb/s to 2.6 km. Coaxial cables can carry signals up to 750 MHz. As the bandwidth of each TV channel is 6 MHz. was designed to compete with T-l lines. the assigned bandwidth can accommodate more than 80 channels. Communication in an HFC cable TV network is bidirectional.

3 Switching and Routing Devices: Switching devices are categorized by their complexity. as follows: Layers 1 and 2 switches are typically simple. About 200 MHz of the coax bandwidth. from 550 MHz to 750 MHz. are designed primarily to interconnect very small LAN units without any involvement in the complex routing processes . as layer 2 switches. repeaters and hubs are known as layer switches. The cable modem uses the 64-QAM or 256-QAM modulation technique for the downstream data transfer. for example. For example. is allocated to the downstream data transfer: from the Internet side to a user.this number to increase to 180 TV channels. the simplest switching devices. including 6 MHz-wide channels. Layer 3 or higher switches are complex. This bandwidth is also divided to about 33 channels. from 5 MHz to 42 MHz. each with 6 MHz bandwidth. are layer 3 switches Repeaters and hubs. The upstream data premises communicate to the Internet and occupy 37 MHz. The upstream data is modulated using the QPSK (Quadrature PSK) technique 2. bridges. Routers.

A bridge is a switch that connects two pieces of a LAN or two LANs. A hub is similar to the repeater but connects several pieces of a LAN. The system consists of four main parts: input port processors. a bridge can also be used in layer 1 for signal regeneration.Repeaters are used to connect two segments of a LAN. 2. Routers and Higher-Layer Switches A router is a layer 3 switch that connects other routing nodes. switch fabric (switching network). Layer 3 switches are of two types: Packet-by-packet switches.4 Router Structure: Routers are the building blocks of wide area networks. a bridge does not forward a frame to all LAN users and thus can isolate traffic between two LANs. A router has a routing look-up table for routing packets. Signal regeneration is needed when the LAN length is extended. As it operates at layer 2. as interfaces to switch fabric. A bridge checks the physical address of any destination user and enhances the efficiency of networks by facilitating simultaneous transmissions within multiple LANs. a bridge does more than simply extend the range of the network. router is dependent on protocols and establishes physical circuits for individual node-pair connection. Flow-based switches. and switch controller. packets of a message are sent individually but not in order. differentiates it from a piece of cable. by which packet forwarding is handled based on each individual packet. signal egeneration. Packets arrive at n input ports and are routed out from n output ports. The functionality . Input Port Processor (IPP) Input and output port processors. are commercially implemented together in router line cards. a hub is a multipoint repeater. This scheme speeds up the forwarding process. If a communication is connectionless. by which a number of packets having the same source and destination are identified and forwarded together. However. output port processors. A repeater's essential function. A hub is another simple device and is used to provide connections among multiple users in layer 1 of a protocol stack. which contain some of the task of the physical and data link layers. especially those operating in layer 2 of the protocol stack.

One solution to this problem is to partition packets into smaller fragments and then reassemble them at the output port processor (OPP) after processing them in the switching system Routing Table: The routing table is a look-up table containing all available destination addresses and the corresponding switch output port. main buffer. An input port processor (IPP) typically consists of several main modules. converts packets to smaller sizes. as shown in following figure. as buffer slots are usually only 512 bytes long. . packet encapsulator. routing table. multicast process. These modules are packet fragmentation. when large packets must be buffered at the input port interface of a router. which also provides a buffer to match the speed between the input and the switch fabric. and a comprehensive QoS Packet Fragmentation The packet fragmentation unit.of the data link layer is implemented as a separate chip in IPP. Large packets cause different issues at the network and link layers.