Computer Networks-UNIT 1-CO1.pdf

Computer Networks 19ECE304 Unit-I 1  Syllabus Unit 1 Introduction to the Internet - Services and Protocols, Edge and Core, Packet Switching vs. Circuit Switching - Performance Metrics Delay - Loss – Throughput - Protocol Layers and...

Computer Networks 19ECE304 Unit-I 1  Syllabus Unit 1 Introduction to the Internet - Services and Protocols, Edge and Core, Packet Switching vs. Circuit Switching - Performance Metrics Delay - Loss – Throughput - Protocol Layers and Service Models OSI and TCP/IP models - Application Layer: Client-Server and Peer-to-Peer architectures - Application Layer protocols - Transport Layer - Unreliable Connectionless vs. Reliable Connection-Oriented Services - Multiplexing; Stop-and Wait - Go-Back-N and Selective-Repeat - UDP vs. TCP - Flow and Congestion Control. Unit 2 Network Layer - Data plane forwarding vs. Control plane routing -Software Defined Networking (SDN) approach - Network Services - Router architecture - Switching fabrics - Input and output queueing IPv4 and IPv6 addressing DHCP -NAT - IPv4 and IPv6 fragmentation - SDN based generalized forwarding - Routing and Supporting Algorithms - Link State vs. Distance Vector - RIP - OSPF – BGP – ICMP - SNMP - SDN Control Plane. Unit 3 Link Layer – Services - Error Detection and Correction; Multiple Access protocols Channel partitioning - Random access - Taking-Turns protocols - Switched LANs ARP - Ethernet - Link layer switching – VLANs – MPLS - Introduction to Wireless and Mobile Networks - Link characteristics - CDMA - 802.11 WiFi - Bluetooth and Zigbee - Cellular Networks - GSM – UMTS – LTE - Mobility management and handoff - Mobile IP. Text Book(s) James Kurose and Keith Ross, “Computer Networking: A Top-Down Approach”, Seventh (Global) Edition, Pearson Education Ltd., 2017. Larry L. Peterson and Bruce S. Davie, “Computer Networks - A Systems Approach”, Morgan Kaufmann, Fifth Edition, 2011 2  Introduction to the Internet The Internet is a computer network that interconnects hundreds of millions of computing devices throughout the world Computing devices were primarily traditional desktop PCs, Linux workstations, and so-called servers that store and transmit information such as Web pages and e-mail messages. Laptops, smartphones, tablets, TVs, gaming consoles, Web cams, automobiles, environmental sensing devices, picture frames, and home electrical and security systems are being connected to the Internet. In Internet jargon, all of these devices are called hosts or end systems. 3 4  Contd., End systems are connected together by a network of communication links and packet switches. Communication links- made up of different types of physical media, including coaxial cable, copper wire, optical fiber, and radio spectrum. Different links can transmit data at different rates, with the transmission rate of a link measured in bits/second. When one end system has data to send to another end system, the sending end system segments the data and adds header bytes to each segment. The resulting packages of information-packets, in the internet, are then sent through the network to the destination end system, where they are reassembled into the original data A packet switch takes a packet arriving on one of its incoming communication links and forwards that packet on one of its outgoing communication links. Packet switch types-routers and link-layer switches. 5  Contd., Link-layer switches are typically used in access networks, while routers are typically used in the network core. The sequence of communication links and packet switches traversed by a packet from the sending end system to the receiving end system is known as a route or path through the network. End systems access the Internet through Internet Service Providers (ISPs), including residential ISPs such as local cable or telephone companies; corporate ISPs; university ISPs; and ISPs that provide WiFi access in airports, hotels, coffee shops, and other public places. The Transmission Control Protocol (TCP) and the Internet Protocol (IP) are two of the most important protocols in the Internet. The IP protocol specifies the format of the packets that are sent and received among routers and end systems. The Internet’s principal protocols are collectively known as TCP/IP 6  Services Internet is an infrastructure that provides services to applications. Distributed applications - messaging, Voiceover- IP (VoIP), video streaming, distributed games, peer-to-peer (P2P) file sharing, television over the Internet, remote login. End systems attached to the Internet provide an Application Programming Interface (API) that specifies how a program running on one end system asks the Internet infrastructure to deliver data to a specific destination program running on another end system Internet API is a set of rules that the sending program must follow so that the Internet can deliver the data to the destination program 7 Protocol 8  Network Protocols A network protocol is similar to a human protocol, except that the entities exchanging messages and taking actions are hardware or software components of some device. All activity in the Internet that involves two or more communicating remote entities is governed by a protocol. A protocol defines the format and the order of messages exchanged between two or more communicating entities, as well as the actions taken on the transmission and/or receipt of a message or other event. 9  Network structure Network Edge – Servers and Client- – Host= End system Access network – Wired or wireless Network Core – International Router, networks of network 10  Access Network Two most prevalent types of broadband residential access are digital subscriber line (DSL) and cable. DSL Line – A high-speed downstream channel, in the 50 kHz to 1 MHz band – A medium-speed upstream channel, in the 4 kHz to 50 kHz band – An ordinary two-way telephone channel, in the 0 to 4 kHz band. 11  Access Network While DSL makes use of the telco’s existing local telephone infrastructure, cable Internet access makes use of the cable television company’s existing cable television infrastructure. A residence obtains cable Internet access from the same company that provides its cable television. At the cable head end, the cable modem termination system (CMTS) serves a similar function as the DSL network’s DSLAM—turning the analog signal sent from the cable modems in many downstream homes back into digital format. Cable modems divide the HFC network into two channels, a downstream and an upstream channel. As with DSL, access is typically asymmetric, with the downstream channel typically allocated a higher transmission rate than the upstream channel 12  Contd., FTTH using the PON distribution architecture. Each home has an optical network terminator (ONT), which is connected by dedicated optical fiber to a neighborhood splitter. On corporate and university campuses, and increasingly in home settings, a local area network (LAN) is used to connect an end system to the edge router. Although there are many types of LAN technologies, Ethernet is by far the most prevalent access technology in corporate, university, and home networks. 13  Physical medium The physical medium can take many shapes and forms and does not have to be of the same type for each transmitter-receiver pair along the path. Examples of physical media include twisted-pair copper wire, coaxial cable, multimode fiber-optic cable, terrestrial radio spectrum, and satellite radio spectrum. Physical media fall into two categories: guided media and unguided media. With guided media, the waves are guided along a solid medium, such as a fiber-optic cable, a twisted-pair copper wire, or a coaxial cable, Satellite Radio channels. With unguided media, the waves propagate in the atmosphere and in outer space, such as in a wireless LAN or a digital satellite channel. 14 Contd., Wide-Area Wireless Access: 3G and LTE 15  Network Core The mesh of packet switches and links that interconnects the Internet’s end systems There are two fundamental approaches to moving data through a network of links and switches: packet switching and circuit switching. Packet Switiching: In a network application, end systems exchange messages with each other. Messages can contain anything the application designer wants. Messages may perform a control function (for example, the “Hi”messages in our handshaking) or can contain data, such as an e-mail message, a JPEG image, or an MP3 audio file. To send a message from a source end system to a destination end system, the source breaks long messages into smaller chunks of data known as packets. Between source and destination, each packet travels through communication links and packet switches (for which there are two predominant types, routers and link-layer switches). Packets are transmitted over each communication link at a rate equal to the full transmission rate of the link. So, if a source end system or a packet switch is sending a packet of L bits over a link with transmission rate R bits/sec, then the time to transmit the packet is L / R seconds. 16 store-and-forward transmission Most packet switches use store-and-forward transmission at the inputs to the links. Store-and-forward transmission means that the packet switch must receive the entire packet before it can begin to transmit the first bit of the packet onto the outbound link. To explore store-and-forward transmission in more detail, consider a simple network consisting of two end systems connected by a single router. A router will typically have many incident links, since its job is to switch an incoming packet onto an outgoing link; in this simple example, the router has the rather simple task of transferring a packet from one (input) link to the only other attached link. 17 Queuing Delays and Packet Loss Each packet switch has multiple links attached to it. For each attached link, the packet switch has an output buffer (also called an output queue), which stores packets that the router is about to send into that link. The output buffers play a key role in packet switching. If an arriving packet needs to be transmitted onto a link but finds the link busy with the transmission of another packet, the arriving packet must wait in the output buffer. Thus, in addition to the store-and-forward delays, packets suffer output buffer queuing delays. These delays are variable and depend on the level of congestion in the network arriving packet may find that the buffer is completely full with other packets waiting for transmission. In this case, packet loss will occur—either the arriving packet or one of the already- queued packets will be dropped 18  Circuit Switching In circuit-switched networks, the resources needed along a path (buffers, link transmission rate) to provide for communication between the end systems are reserved for the duration of the communication session between the end systems. In packet-switched networks, these resources are not reserved; a session’s messages use the resources on demand and, as a consequence, may have to wait (that is, queue) for access to a communication link Traditional telephone networks are examples of circuit-switched networks. Consider when one person wants to send information (voice or facsimile) to another over a telephone network. Before the sender can send the information, the network must establish a connection between the sender and the receiver. This is a bona fide connection for which the switches on the path between the sender and receiver maintain connection state for that connection. In the jargon of telephony, this.connection is called a circuit When the network establishes the circuit, it also reserves a constant transmission rate in the network’s links (representing a fraction of each link’s transmission capacity) for the duration of the connection. Since a given transmission rate has been reserved for this sender-to receiver connection, the sender can transfer the data to the receiver at the 20 guaranteed constant rate.  Multiplexing in Circuit-Switched Networks A circuit in a link is implemented with either frequency-division multiplexing (FDM) or time-division multiplexing (TDM). With FDM, the frequency spectrum of a link is divided up among the connections established across the link. Specifically, the link dedicates a frequency band to each connection for the duration of the connection. In telephone networks, this frequency band typically has a width of 4 kHz (that is, 4,000 hertz or 4,000 cycles per second). The width of the band is called, not surprisingly, the bandwidth. FM radio stations also use FDM to share the frequency spectrum between 88 MHz and 108 MHz, with each station being allocated a specific frequency band. For a TDM link, time is divided into frames of fixed duration, and each frame is divided into a fixed number of time slots. When the network establishes a connection across a link, the network dedicates one time slot in every frame to this connection. These slots are dedicated for the sole use of that connection, with one time slot available for use (in every frame) to transmit the connection’s data. 21 22  Multiplexing in Circuit-Switched Networks FDM and TDM for a specific network link supporting up to four circuits. For FDM, the frequency domain is segmented into four bands, each of bandwidth 4 kHz. For TDM, the time domain is segmented into frames, with four time slots in each frame; each circuit is assigned the same dedicated slot in the revolving TDM frames. For TDM, the transmission rate of a circuit is equal to the frame rate multiplied by the number of bits in a slot. For example, if the link transmits 8,000 frames per second and each slot consists of 8 bits, then the transmission rate of each circuit is 64 kbps Proponents of packet switching have always argued that circuit switching is wasteful because the dedicated circuits are idle during silent periods. For example, when one person in a telephone call stops talking, the idle network resources (frequency bands or time slots in the links along the connection’s route) cannot be used by other ongoing connections Let us consider how long it takes to send a file of 640,000 bits from Host A to Host B over a circuit-switched network. Suppose that all links in the network use TDM with 24 slots and have a bit rate of 1.536 Mbps. Also suppose that it takes 500 msec to establish an end-to-end circuit before Host A can begin to transmit the file. How long does it take to send the file? Each circuit has a transmission rate of so it takes seconds to transmit the file. To this 10 seconds we add the circuit establishment time, giving 10.5 seconds to send the file. Note that the transmission time is independent of the number of links: The transmission time would be 10 seconds if the end-to-end circuit passed through one link or a hundred links 23  Delay, loss and Throughout The most important of these delays are the nodal processing delay, queuing delay, transmission delay, and propagation delay; together, these delays accumulate to give a total nodal delay. dnodal=dproc+dqueue+dtrans+dprop 24  As part of its end-to-end route between source and destination, a packet is sent from the upstream node through router A to router B. Our goal is to characterize the nodal delay at router A. Note that router A has an outbound link leading to router B. This link is preceded by a queue (also known as a buffer). When the packet arrives at router A from the upstream node, router A examines the packet’s header to determine the appropriate outbound link for the packet and then directs the packet to this link. The outbound link for the packet is the one that leads to router B. A packet can be transmitted on a link only if there is no other packet currently being transmitted on the link and if there are no other packets preceding it in the queue; if the link is currently busy or if there are other packets already queued for the link, the newly arriving packet will then join the queue 25  Processing Delay The time required to examine the packet’s header and determine where to direct the packet is part of the processing delay. The processing delay can also include other factors, such as the time needed to check for bit-level errors in the packet that occurred in transmitting the packet’s bits from the upstream node to router A. Processing delays in high-speed routers are typically on the order of microseconds or less. After this nodal processing, the router directs the packet to the queue that precedes the link to router B 26  Queuing Delay At the queue, the packet experiences a queuing delay as it waits to be transmitted onto the link. The length of the queuing delay of a specific packet will depend on the number of earlier-arriving packets that are queued and waiting for transmission onto the link. If the queue is empty and no other packet is currently being transmitted, then our packet’s queuing delay will be zero. On the other hand, if the traffic is heavy and many other packets are also waiting to be transmitted, the queuing delay will be long. We will see shortly that the number of packets that an arriving packet might expect to find is a function of the intensity and nature of the traffic arriving at the queue. Queuing delays can be on the order of microseconds to milliseconds in practice. 27  Transmission delay Assuming that packets are transmitted in a first-come-first-served manner, as is common in packet switched networks, our packet can be transmitted only after all the packets that have arrived before it have been transmitted. Denote the length of the packet by L bits, and denote the transmission rate of the link from router A to router B by R bits/sec. For example, for a 10 Mbps Ethernet link, the rate is R=100 Mbps for a 100 Mbps Ethernet link, the rate is The transmission delay is L/R. This is the amount of time required to push (that is, transmit) all of the packet’s bits into the link. Transmission delays are typically on the order of microseconds to milliseconds in practice Dr B Sakthi Abirami/Asst 28 Professor/ECE/ASE  Propagation delay Once a bit is pushed into the link, it needs to propagate to router B. The time required to propagate from the beginning of the link to router B is the propagation delay. The bit propagates at the propagation speed of the link. The propagation speed depends on the physical medium of the link (that is, fiber optics, twisted-pair copper wire, and so on) and is in the range of 2⋅108 meters/sec to 3⋅108 meters/sec which is equal to, or a little less than, the speed of light. The propagation delay is the distance between two routers divided by the propagation speed. That is, the propagation delay is d/s, where d is the distance between router A and router B and s is the propagation speed of the link. Once the last bit of the packet propagates to node B, it and all the preceding bits of the packet are stored in router B. The whole process then continues with router B now performing the forwarding. In wide-area networks, propagation delays are on the order of milliseconds. 29  Packet Loss we have assumed that the queue is capable of holding an infinite number of packets. In reality a queue preceding a link has finite capacity, although the queuing capacity greatly depends on the router design and cost. Because the queue capacity is finite, packet delays do not really approach infinity as the traffic intensity approaches 1. Instead, a packet can arrive to find a full queue. With no place to store such a packet, a router will drop that packet; that is, the packet will be lost. This overflow at a queue can again be seen in the Java applet for a queue when the traffic intensity is greater than 1. From an end-system viewpoint, a packet loss will look like a packet having been transmitted into the network core but never emerging from the network at the destination. The fraction of lost packets increases as the traffic intensity increases. Therefore, performance at a node is often measured not only in terms of delay, but also in terms of the probability of packet loss. As we’ll discuss in the subsequent chapters, a lost packet may be retransmitted on an end-to-end basis in order to ensure that all data are eventually transferred from source to destination. 30  Throughput To define throughput, consider transferring a large file from Host A to Host B across a computer network. This transfer might be, for example, a large video clip from one peer to another in aP2P file sharing system. The instantaneous throughput at any instant of time is the rate (in bits/sec) at which Host B is receiving the file. (Many applications, including many P2P file sharing systems, display the instantaneous throughput during downloads in the user interface—perhaps you have observed this before!) If the file consists of F bits and the transfer takes T seconds for Host B to receive all F bits, then the average throughput of the file transfer is F/T bits/sec. For some applications, such as Internet telephony, it is desirable to have a low delay and an instantaneous throughput consistently above some threshold (for example, over 24 kbps for some Internet telephony applications and over 256 kbps for some real-time video applications). For other applications, including those involving file transfers, delay is not critical, but it is desirable to have the highest possible throughput 31 Network latency is an expression of how much time it takes for a data packet to get from one designated point to another. In some environments, latency is measured by sending a packet that is returned to the sender -- the round-trip time is considered the latency. Ideally, latency is as close to zero as possible. 32 Protocol layers and Services 33  Application Layers An application-layer protocol is distributed over multiple end systems, with the application in one end system using the protocol to exchange packets of information with the application in another end system. We’ll refer to this packet of information at the application layer as a message. HTTP SMTP FTP 34  Transport Layers The Internet’s transport layer transports application-layer messages between application endpoints. In the Internet there are two transport protocols, TCP and UDP, either of which can transport application layer messages. TCP provides a connection-oriented service to its applications. This service includes guaranteed delivery of application-layer messages to the destination and flow control (that is, sender/receiver speed matching). TCP also breaks long messages into shorter segments and provides a congestion-control mechanism, so that a source throttles its transmission rate when the network is congested. The UDP protocol provides a connectionless service to its applications. This is a no-frills service that provides no reliability, no flow control, and no congestion control. transport-layer packet as a segment. 35  Network Layers The Internet’s network layer is responsible for moving network-layer packets known as datagrams from one host to another. The Internet transport-layer protocol (TCP or UDP) in a source host passes a transport-layer segment and a destination address to the network layer, just as you would give the postal service a letter with a destination address. The network layer then provides the service of delivering the segment to the transport layer in the destination host. The Internet’s network layer includes the celebrated IP protocol, which defines the fields in the datagram as well as how the end systems and routers act on these fields. There is only one IP protocol, and all Internet components that have a network layer must run the IP protocol. The Internet’s network layer also contains routing protocols that determine the routes that datagrams take between sources and destinations. The Internet has many routing protocols. 36  Physical Layers While the job of the link layer is to move entire frames from one network element to an adjacent network element, the job of the physical layer is to move the individual bits within the frame from one node to the next. The protocols in this layer are again link dependent and further depend on the actual transmission medium of the link (for example, twisted-pair copper wire, single-mode fiber optics). 37  Link Layers The Internet’s network layer routes a datagram through a series of routers between the source and destination. To move a packet from one node (host or router) to the next node in the route, the network layer relies on the services of the link layer. In particular, at each node, the network layer passes the datagram down to the link layer, which delivers the datagram to the next node along the route. At this next node, the link layer passes the datagram up to the network layer. The services provided by the link layer depend on the specific link-layer protocol that is employed over the link. For example, some link-layer protocols provide reliable delivery, from transmitting node, over one link, to receiving node. Note that this reliable delivery service is different from the reliable delivery service of TCP, which provides reliable delivery from one end system to another the link-layer packets as frames. 38  OSI Layer Open Systems Interconnections (OSI) The role of the presentation layer is to provide services that allow communicating applications to interpret the meaning of data exchanged. These services include data compression and data encryption (which are self-explanatory) as well as data description (which frees the applications from having to worry about the internal format in which data are represented/stored—formats that may differ from one computer to another). The session layer provides for delimiting and synchronization of data exchange, including the means to build a checkpointing and recovery scheme 39 Encapsulation 40  Encapsulation At the sending host, an application-layer message (M) is passed to the transport layer. In the simplest case, the transport layer takes the message and appends additional information (so-called transport-layer header information, H) that will be used by the receiver-side transport layer. The application-layer message and the transport-layer header information together constitute the transport layer segment. The transport-layer segment thus encapsulates the application-layer message. The added information might include information allowing the receiver-side transport layer to deliver the message up to the appropriate application, and error-detection bits that allow the receiver to determine whether bits in the message have been changed in route. The transport layer then passes the segment to the network layer, which adds network- layer header information (H) such as source and destination end system addresses, creating a network-layer datagram. The datagram is then passed to the link layer, which (of course!) will add its own link- layer header information and create a link-layer frame. Thus, we see that at each layer, a packet has two types of fields: header fields and a payload field. The payload is typically a packet from the layer above 41