Topics: The Internet Network Edge Network Core Internet Backbone Loss & Delay Internet Protocol Stack CHAPTER 1: OVERVIEW OF COMPUTER NETWORKS

Transcription

1 Topics: The Internet Network Edge Network Core Internet Backbone Loss & Delay Internet Protocol Stack CHAPTER 1: OVERVIEW OF COMPUTER NETWORKS 1

2 THE INTERNET The Internet is a computer network that connects hosts or end systems throughout the world. These hosts could be PCs, mobiles or fridges that run applications. Each end system is connected together by a network of communication links and packet switches. Each of these communication links can have different transmission rates, which may be physically limited by the medium of transmission. The Internet is radically different to the traditional telephony network and bears some remarkable resemblance to the postal system- It provides Best Effort Service Data transmitted across the network is sent in segments of data called packets, much like envelopes or packages sent in the postal system. This type of transfer of data is known to be packet switched whereas a telephone call runs on a circuit switched network. This idea is based on minimalism. The network equipment (such as a router or a switch) is mostly stateless It is decentralised. The old telephony system use to work based on setting up a physical connection from the source to the destination. Using an old rotary telephone, the number of clicks back would determine how far a lever would turn at a particular exchange. For example, the first number 9, may rotate the lever 9 places in the first exchange and then connect it to another telephone exchange. The next number 3 may rotate the lever in the second exchange 3 places and so on, until all the numbers have been entered and a circuit has been established. This setup has one fundamental problem if one of the exchanges fail, then the telephone call cannot take place. A postal system however does not experience this problem. The failure of one sorting facility does not inhibit the flow of other messages. They merely get rerouted to another facility. The rationale for the Internet is due significantly to this reason. Historically, the Internet was used by the U.S military to ensure that communication systems would not be fail miserably in the case of a nuclear attack by the Russians. 2

3 NUTS & BOLTS VIEW VS SERVICES VIEW OF THE INTERNET Any means of communication requires a set of rules. The Internet, which is a network of networks is no different. These rules are known as protocols: 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. There are many types of network protocols used to handle different type of data transmissions including TCP, HTTP and UDP. Given the various number of protocols, it is important that standards are set. This is done so by the IETF standards documents called requests for comments (RFCs). The internet can also be thought of as an infrastructure that provides services to (distributed) applications through the use of various access mediums and protocols. These distributed applications could include web surfing, instant messaging, VoIP, P2P or video streaming. 3

4 NETWORK EDGE The network edge consists of hosts. Hosts can be loosely categorised as clients and servers: A client contains a client program that requests and receives a service from a server program running on another end system. A server handles the requests for a service. Applications which operate on hosts that use clients/servers are referred to as a client-server model based network. and newsgroups are typical based on the client-server model. There has also been an increasing use of peer-to-peer applications in which end systems interact and run programs that perform both client and server functions. This is seen by the likes of BitTorrent and KaZaa. ACCESS NETWORKS There are different ways of the network edge or the hosts to connect to the Internet. Different access networks have different transmission speeds and characteristics such as being shared or dedicated. DIAL-UP Dial-up uses a traditional telephone line to access the Internet at speeds up to 56 kbps. Because this method of access uses the whole telephone line, it is not possible to surf the internet and be on the telephone simultaneously. Dial up requires a modem which physically dials the ISP s number over a traditional telephone line. Since the telephone line is built for analog signals, the modems on both ends must form analog to digital and digital to analog conversions. DSL The most prevalent broadband residential access, is the digital subscriber line. DSL works by hooking unto the existing telephone line using a modem. The telephone exchange is terminated with a filter which filters the phone signals and the other signals are fed into a digital subscriber line access multiplexer (DSLAM). Using this mechanism (frequency division multiplexing), it is possible to hold both telephone and data signals simultaneously. ADSL- Asynchronous DSL is commonly deployed to residential users who generally prefer faster download speed rather than fast upload speeds (hence the asymmetric nature of the Internet link). This is done by assigning the downstream channel with a larger frequency band: A high speed downstream channel: 50 khz-1mhz A upstream channel: 4 khz-50 khz Ordinary voice channel: 0-4 khz 4

5 Since the channels are side-by-side in the frequency spectrum, a splitter is often used on the customer s end to separate the data and the telephone signals properly. The actual downstream and upstream transmission rate available to the residence is a function of the distance between the home and the exchange, the gauge of the twisted pair line and the degree of electrical interference. To boost the data rates, DSL relies on advanced signal processing technology and error correction algorithms, which can lead to high packet delays. CABLE While DSL and dialup make use of the existing telephone infrastructure, cable Internet makes use of the cable television infrastructure. Hybrid fiber coax (HFC) is often use fiber optics connect the cable head end to neighbourhood-level junctions, from which traditional coaxial cable is then used to reach individual houses and apartments. Cable modems divide the HFC network into two asymmetric channels, a downstream and upstream channel. The downstream rate is usually faster than the upstream rate. 5

6 As shown in the figure, cable Internet is a shared broadcast medium (whereas the DSL is a point to point connection and hence is a dedicated rather than shared transmission link). Every packet sent by a home travels on the upstream channel to the head end. For this reason, if several users are simultaneously downloading a video file on the downstream channel, the actual rate at which each user receives its video file will be significantly lower than the aggregate cable downstream. The converse is true for a few simultaneous active users. ETHERNET Ethernet users use twisted-pair copper wire to connect to an Ethernet switch. It is possible to achieve data transmission rates of 10 Gbps. These connections are typically found in small home networks, corporate and university campuses. WIRELESS ACCESS NETWORKS This access method is a shared network which connects hosts to a access point. When using wireless LAN technology, the n provides Internet access speeds of up to 150 Mbps over a distance of a few tens of metres. 3G networks have also been deployed to provide packet switched wide-area wireless Internet access speeds. WiMAX is a long distance cousin of the WiFi protocol. It operates independently of the cellular network and delivers speeds of up to 10 Mbps over tens of kilometres. 6

7 PHYSICAL MEDIA All physical media (the link that lies between the transmitter and receiver) have bandwidth limits. If we are to transmit digital signals, we must use Fourier series or transforms an infinite number of sinusoidal waves with an infinite number of harmonics to generate the square wave signals across a physical media. Given that we cannot make infinite number of sinusoids, we can make approximations. But with approximations comes uncertainties; an undesirable property for equipment dealing with digital signals. This is the reason why different physical media have different bandwidth limitations, and some are more error prone than others! TWISTED PAIR The least expensive and most commonly used guided transmission medium. Twisted pair consists of two insulated copper wires, each about 1 mm thick, arranged in a regular spiral pattern to reduce the electrical interference from similar pairs close by. Typically, a number of pairs are bundled together in a cable by wrapping the pairs in a protective shield. Note that each pair constitutes a single communication link, since we must have a complete circuit and a reference voltage for which we can setup a potential difference and measure the signal. The typical phone line is a category 3 twisted pair, whereas Ethernet cables are usually unshielded twisted pair category 5 or 6, and allow data rates of 1 Gbps for distances up to a hundred metres. The difference between the categories is dependent on the twist per unit length- Cat 5 cable typically has three twists per inch of each twisted pair of 24 gauge (AWG) copper wires within the cables. This will affect the data rate and issues to do with cross talk. COAXIAL CABLE A coaxial cable consists of two concentric copper conductors. With this construction and special insulation and shielding, these cables can have high bit rates. In cable television and Internet access, the transmitter shifts the digital signal to a specific frequency band, and the resulting analog signal is sent from the transmitter to one or more receivers. Coaxial cable can be used as a guided bi-directional shared medium. 7

8 FIBER OPTICS Fiber optics is a thin, flexible medium that conducts pulses of light, with each pulse representing a bit. Each fiber can support up to 50 Gbps and are immune to electromagnetic interference. Moreover, they have low signal attenuation up to 100 kilometres and can be amplified using an optical amplifier. Fiber optics are based upon the use of total internal reflection to direct the light through the link. This requires the pulses of light to be incident on the fiber at a critical angle. There are different grades of fibers; each of which will require different transmission wavelengths. WIRELESS Wireless media encodes a signal using the electromagnetic spectrum. It is bidirectional and its propagation is affected by obstructions, interferences and reflections. There are several types of wireless links: Terrestrial microwave Wifi Wide-Area Satellite 8

9 RADIO TRANSMISSION This is an attractive medium because it requires no physical medium, is capable of penetrating walls and provides connectivity to mobile users over a potentially long distance. The characteristics of a radio signal depends significantly on the propagation environment and the distance over which a signal is to be carried. Environmental considerations determine path loss and shadow fading, multipath fading due to signal reflection and interference. In the VLF, LF and MF bands, radio waves follow the curvature of the earth whereas in the HF band, they bounce off the ionosphere. FREE-SPACE OPTICS Free-space optics involves the use of laser communication systems in free air space such as across building roof tops. Again the characteristics of the transmission is dependent on environmental considerations such as convection currents that can interfere with the laser beam. SATELLITE Satellites receive transmissions on one frequency band, regenerates the signal using a repeater and transmits the signal on another frequency. Depending on the type of communication, satellites are located at different altitudes. This will affect the number of satellites needed for global coverage and the round trip delay time. Satellite communication has largely been unsuccessful commercially due to the significant upfront costs in building and sending satellites into space. 9

10 MODULATION & DEMODULATION All signals must be modulated and demodulated especially if we are to represent digital data using analog signals. A binary signal may be frequency modulated, amplitude modulated or phase modulated. Depending on the modulation, its characteristics will be demodulated to obtain the original binary signal again. There are more complex methods of modulation/demodulation that increase the bit rate (baud-rate x bits/symbol) despite a low baud rate (symbols/sec). This is done by encoding multiple bits in one particular symbol. For example in figure below, each point represents one possible combination of amplitude and phase that could stand for a particular binary combination. These methods of modulations were first widely implemented on 56 kbps dialup modems has QPSK, QAM-16 and QAM-64. FREQUENCY DIVISION MULTIPLEXING Each of the channels operate at the same frequency, but by multiplexing the frequencies, we can shift the frequency band for each channel such that the channels operate a different frequency band. TIME DIVISION MULTIPLEXING With TDM, each channel is assigned a time slot in which transmission or receipt can be performance. 10

11 WAVELENGTH DIVISION MULTIPLEXING This is much like FDM, but is given a different name since it is implemented mostly in fibers. By using different wavelengths of light, it is possible to send multiple signals simultaneously down a fiber. This allows the fiber to be utilised to its full capacity since no electronic device is capable of utilising the entire link capacity. Unlike the other types of multiplexing, this type of multiplexing requires no electrical power and can be done by using prisms. This is what makes fiber so attractive. 11

12 NETWORK CORE CIRCUIT SWITCHING VS PACKET SWITCHING In circuit-switched networks, the resources needed along a path, to provide for communication between the end systems are reserved for the duration of the communication session between the hosts. 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 for access to the communication link. Since resources are assigned as needed and hence someone who is active may be assigned resources which may otherwise be hogged by an idle user. This also allows for the opportunity for a user to be able to use the link s bandwidth to its full capacity. Previously we used the postal service and telephone analogy. The telephone service was described as a circuit switched network because A bona fide connection for which the switches on the path between the sender and receiver maintain a stateful connection state for that connection. It has a circuit-like guaranteed performance It reserves a constant transmission rate in the network s links for the duration of the connection. The diagram shows the principle behind circuit switching in which data can only take one path once the circuit is setup. Since the bandwidth has been reserved for this sender-to-receiver connection, the sender can transfer the data to the receiver at the guaranteed constant rate. If we are to have n circuits on one link, then we must divide the network resources by either frequency or time. This requires that the n circuits reserve 1/n of the link s resources and hence 1/n of the link s bandwidth for the duration of the connection. Given this situation, an idle user would be wasting resources, and saturation of the link would not be possible. 12

13 Example Consider a file that is bits. If it is sent over a circuit switched network with a TDM of 24 slots and the link bit rate is Mbps. If it takes 500 msec to establish an end-to-end circuit, how long will the transmission take? Each circuit has a transmission rate of = 64 kkkkkkkk. This means that it takes = 10 seconds to transmit the file. The whole transmission including establishing the connection takes 10.5 seconds. 24 For packet switched networks, each end to end data stream is divided into packets. Packet switches tend to use a store-and-forward transmission; and packets must be received entirely before it can be transmitted on the outbound link. If there are Q links between two hosts, each of rate R bps, then to send L bits the total delay due to the store-and-forward transmission mechanism would take tt = QQQQ RR. Since each output consists of a buffer or queue, then it may also experience queuing delays due to a busy output link. These delays are variable and depend on the level of congestion in the network. In the case of a full buffer, packet loss will occur either the arriving packet or one of the already-queued packets will be dropped. Let us see why packet switched networks are more efficient when it comes to statistical multiplexing. Consider a 1 Mb/s link. With a circuit switched network, where each active user uses 100 kb/s and is active 10% of the time, it is only possible to service 1MMMMMMMM 100kkkkkkkk = 10 users, since 100 kbps must be reserved for each user at all times. For a packet switched network of similar properties, the probability that there will be more than 10 active users out of a possible 35 users at any one time is less than 0.004: PP(XX > 10) = (0.1)10 (0.9) (0.1)9 (0.9)

14 For more than 99% of the case, there are 10 or fewer simultaneous active users, and the aggregate arrival rate of data is less than 1 Mbps, the output rate link. This essentially means that packets flow through the link essentially without delay. If there are more than 10 simultaneous users, than the aggregate arrival rate of packets exceed the output capacity of the link, and the output queue will begin to grow until the input rate falls back below 1 Mbps. This indicates that packet switched networks are more efficient for situations in which there is bursty data. But for applications requiring a steady stream of data, such as audio and video applications, this could be an issue, since we no longer have the circuit-like reliability. 14

15 INTERNET BACKBONE The Internet is roughly hierarchical. This means that the general life of a packet is routed through many Internet Service Providers before reaching its destination. On the outer edge of the Internet, access ISPs connect to the rest of the Internet through a tiered hierarchy of ISPs, and it itself is at the bottom. At the very top, there are Tier 1 ISPs. Tier 1 ISPs Are characterised as peering with other tier-1 ISPs, connected to a large number of tier-2 ISPs and other customer networks and have either national or international coverage. The likes of Sprint, Verizon, MCI and Level 3 are tier-1 ISPs. Tier 2 ISPs Typically have regional or national coverage. They may be multi-homed such that they are connected to a few tier-1 ISPs. To connect to a large portion of the public Internet, a tier-2 ISP must route a majority of its traffic through one of its tier-1 ISPs hence the name tier-2. To this end, a tier-2 ISP is said to be a customer of a tier-1 ISP provider. Tier 2 ISPs may also choose to connect (peer) with other tier-2 ISPs. This usually is a business strategy. Tier 3 & Local ISPs Are at the edge of the Internet and provide access to end users. They a generally customers of either Tier-2 or Tier-1 ISPs. Within an ISP s network, the points at which the ISP connects to other ISPs is known as Points of Presence (POPs). A POP is simply a group of routers in the ISP s network at which routers in other ISPs or in the networks belonging to the ISP s customers can connect. 15

16 LOSS & DELAY As a packet travels to its destination, it suffers from several types of delays at each node along its path: Nodal processing delay Queuing Delay Transmission Delay Propagation Delay These delays accumulate to give a total nodal delay. For the purpose of explanations, we shall consider the router as the node of interest. NODAL PROCESSING Nodal processing is the time required to examine each packet s header and determine where to direct the packet next. Nodal processing may also involve bit level error checking. Typical processing delays are of the order of microseconds or less. Once the router has finished its processing, it will forward the packet to an output queue that precedes the outgoing link. QUEUEING DELAY Sometimes the transmission link is busy and packets experience a queuing delay. 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 across the link. In the absence of any queue, there will be no queuing delay, whereas a congested network will result in a long queuing delay. TRANSMISSION DELAY The transmission delay (store-and-forward delay) is the time to push a complete packet on a link (a function of packet s length and transmission rate). Normally, nodes tend to only transmit the packet after the packet has been received. 16

17 Assuming a FIFO manner then, our packet can only be transmitted only if all the previous packets have been transmitted. If the packet length is L bits and the transmission rate is R bits/sec, then the transmission delay is given by: dd tttttttttt = LL RR PROPAGATION DELAY Once the packet is on the link, it needs to propagate down the link. The propagation delay is the time required to travel from one end of the link to the other, and is dependent on the physical medium of the link. It is given by: dd pppppppp = dd ss Where d is the length of the link and s is the propagation speed of the link mmss mmss 1. Once the last bit of the packet propagates to the adjacent node, it and all its preceding bits of the packet are stored at the node for nodal processing. We thus see that the total nodal delay is given by: dd nnnnnnnnnn = dd pppppppp + dd qqqqqqqqqq + dd tttttttttt + dd pppppppp PACKET LOSS In reality, a queue preceding a link has finite capacity. This means that queuing delays do not approach infinity as the traffic intensity approaches 1. Instead, a packet can arrive to find a full queue, and either it or another packet in the queue will be dropped a packet will have been transmitted into the network core, but will never emerge from the network at the destination. To rectify this issue, a lost packet may be retransmitted. 17

18 INTERNET PROTOCOL STACK To organise the anarchy of network design, network designers have chosen the organise the protocols in layers. Each protocol belongs to one of the layers which roughly describes the services it offers. The aggregate of these layers form a protocol stack, and in the case of the Internet, there are five layers: The Physical Layer: consists of protocols that define how individual bits within the frame move from one node to the next. Data link Layer: consists of protocols that define how packets should be transferred between neighbouring nodes (movement of a packet from one node to the next node in the route). o PPP, Ethernet Network Layer: consists of protocols responsible for moving network layer packets known as datagrams from the source to the destination. o IP, routing protocols such as OSPF Transport Layer: consists of protocols that transport application-layer messages between application endpoints. o TCP, UDP Application Layer: The layer in which the network applications and their application layer protocols reside. o HTTP, FTP, SMTP, DNS 18

19 Topics: CHAPTER 2: Link Layer Functionality Framing Error Control Multiple Access Control Flow Control Reliable Transmission Link Layer Technologies Wireless Ethernet DATALINK LAYER 19

20 THE LINK LAYER The link-layer protocol is used to move a datagram over an individual link. Hence it is mostly implemented on the network interface card (NIC) using a link layer controller. A link layer protocol must define the format of the packets exchanged between the nodes at the ends of the links, as well as the actions taken by these nodes when the packets are sent and received: Framing: Data bits are encapsulated at each level of the Internet Protocol Stack and contain headers and a data payload which is in line with the packet based nature of the Internet. In the link layer, almost all the linklayer protocols encapsulate each network layer datagram within a link layer framer prior to transmission. Error Control: Due to signal attenuation and electromagnetic noise, it is possible for bit errors to exist, leaving the receiver node to incorrectly decipher the frames. Since it is pointless to forward erroneous datagrams, link-layer protocols provide mechanisms to detect (and possibly correct) such bit errors. Multiple Access Control: In a shared medium, it is important for the medium access control (MAC) protocol to define how this single broadcast link can be shared between a number of nodes in essence coordinate the frame transmissions of the many nodes. Flow Control: The nodes on each side of a link have a limited amount of frame buffering capacity. This is a concern when a receiving node may receive frames at a rate faster than it can process them. Without flow control, the receiver s buffer can overflow and frames can get lost. Note that flow control is often performed in the transport layer. Reliable Delivery: When a link layer protocol provides reliable delivery service, it guarantees to move each network layer datagram across the link without error. This is often done with acknowledgements and retransmissions. Reliable delivery service is often used for error prone mediums such as wireless, with a goal of correcting an error locally rather than forcing a retransmission which itself may be erroneous. This feature may be deemed to introduce unnecessary overheads for low bit error-links and hence may not be provided in mediums such as wired coax. These are just some of the actions that can be taken by a link-layer protocol such as Ethernet, PPP or WAN. Since a link layer protocol is only responsible for transporting datagrams over an individual link, it is not unusual for a datagram to be carried by different link layer protocols before it reaches its destination. 20

21 FRAMING Just like we must form words using the alphabet, computers must encapsulate segments of data bits into frames. One of the fundamental issues with framing is determining when a frame ends and when the next one starts. With words, more often than not, spaces acts as some sort of a flag that separates one word from another. In a similar fashion, frames can be separated by flags. What happens when the data contains the same sequence that represents the flag? As an analogy, what happens when you wish to print in C, using the printf statement? To do so in C, we use the back slash \ as an escape (ESC) sequence. With framing, we do exactly the same thing, and append an escape sequence before the supposed flag. The example above shows how the flag is part of the original payload, and that when the data is encapsulated into a frame, an escape sequence is used to indicate that the flag sequence in the payload should not be treated as a flag. FLOW CONTROL Recall that it is possible for the sender and receiver to be operating at different speeds. When this occurs, it is possible that the receiver will simply be unable to handle the frames as they arrive, since its buffers are full and it is unable to process the frames quick enough. This results in a loss of frames. To rectify this issue we can implement: Feedback-based flow control: the receiver sends back information to the sender giving it permission to send more data or at least telling the sender how the receiver is doing Rate based flow control: the protocol has a built in mechanism that limits the rate at which senders may transmit data, without the use of feedback. 21

22 ERROR CONTROL With error detection and correction, we find redundant information is added to the original information such that we can either detect that an error has occurred (but not be able to correct that error) or at the cost of more overhead, be able to correct that error as well using forward error correction techniques. Before we continue, we must define the hamming distance the number of bit positions in which two codewords differ. Hence, with a hamming distance of d, it will require d single bit errors to convert one codeword into another codeword. Consider: and This has a hamming distance of 3. Consider a message size of m bits, with r redundant bits to bring a total of n=m+r bits. This indicates that 2 mm possible combinations of bits are valid, and that 2 nn 2 mm combinations are invalid. ERROR DETECTION To detect d errors, you need a distance dd + 11 code because with such a code there is no way that d single bit errors can change from one valid codeword to another. Consider the situation in which you wish to detect a 1 bit error. In this case, around every valid codeword, all the other codewords which are only 1 bit different (hamming distance of 1) should not be valid. Hence, only codewords which are at a minimum of 2 hamming distances apart are valid. This shows that a distance dd + 1 code is the minimum to detect errors ERROR CORRECTION To correct d error, you need a distance 2d+1 code because that way the legal codewords are far enough apart that even with d changes, the original codeword is still closer than any other codeword, so it can be uniquely determined. Let us derive the minimum number of redundant bits required to detect k bit errors from a total message size of n=m+r bits. Each of the 2 mm legal messages has n illegal codewords which are a distance 1 from it formed by systematically inverting each of the n bits. Thus to correct 1 bit errors, then for each of the 2 mm legal messages, its n neighbours cannot be valid. We can represent this with a shell - the centre of which is one of the 2 mm valid messages. Its radius is a hamming distance of 1, since its n neighbours cannot be valid. Hence for 1 bit errors the condition imposed would be: (nn + 1)2 mm 2 nn (mm + rr + 1) 2 rr Now for k bits then, a shell must have a radius equal to a hamming distance, k: 1 + nn + nn nn kk 2mm 2 nn 22

23 ERROR DETECTING CODES PARITY CHECKING In an even parity scheme, the sender includes one additional bit and chooses its value such that the total number of 1s in the entire message is even. This can also be applied with an odd parity scheme. The receiver need only count the number of 1s in the received messages. If there is an odd number of 1s, then an even parity scheme would flag an error. If however, an even number of bit errors occurs, then an error would go undetected. This method of error checking is rather ad-hoc, given that errors are not independent of each, but rather clustered together in bursts. Under burst error conditions, the probability of undetected errors in a frame protected by single-bit parity can approach 50%. The following figure shows a two-dimensional generalisation (i message rows and j message columns) of the single bit parity scheme. A parity value is computed for each row and for each column, resulting in ii + jj + 1 parity bits. This scheme allows the detection of single and two bit errors, and by isolating the row and column of a one bit error, it is possible to correct the error. Since each of the j columns will have the correct parity by accident is 0.5, the probability of a bad block being accepted when it should not be is 2 jj CHECKSUM In checksumming techniques, the d bits of the data are treated as a sequence of k- bit integers. The simplest approach is to sum these k-bit integers and use the resulting sum as the error detection bits. The Internet checked is based on this approach bytes of data are treated as 16 bit integers and summed. The 1s complement of this sum then forms the Internet checksum that is carried in the segment header. The receiver will then take the 1s complement of the sum of the received data (including the checksum). If the result is all 1 bits then there is no error detected, while a 0 bit will indicate an error has occurred. CYCLIC REDUNDANCY CHECK (CRC) CRC codes are known as polynomial codes, since it is possible to view the bit string to be sent as a polynomial whose coefficients are the 0 and 1 values in the bit strong, with operations on the bit string interpreted as polynomial arithmetic. Example We can represent as a six term polynomial: 1xx 5 + 1xx 4 + 0xx 3 + 0xx 2 + 0xx 1 + xx 0 23

24 1. Let r being the degree of GG(xx), the generator polynomial. This is agreed upon by the sender and receiver. 2. For a given piece of data, DD(xx), with d bits, the sender will choose r additional bits, R, and append them to DD(xx) such that the resulting dd + rr bits pattern is exactly divisible by GG(xx) using modulo 2 arithmetic with no carries in addition or borrows in subtraction. This indicates that addition and subtraction are identical and is just the bitwise XOR operation. To append the r bits, we must bit shift the d bits: DD 2 rr The resulting r bits at the end are zero padded, thus: DD 2 rr XXXXXX RR Will give us the d+r bit message we should send. 3. The receiver will divide the d+r received bits by G. Multiplication and division are the same in base-2 arithmetic, except any additions and subtractions are done without carries or borrows. If the remainder is nonzero, the receiver knows that an error has occurred. The crucial question is determining how to compute R. The condition is that we want to choose R such that G divides into DD 2 rr XXXXXX RR. This implies that: DD 2 rr XXXXXX RR = nnnn RR = rrrrrrrrrrrrrrrrrr DD 2rr GG Example Consider the case in which D=101110, d=6, G=1001. This indicates that r=3, since it is 1 less bit than the generator. We do long division and the remainder R, is what we append onto the message. Thus the message we send is: DD 2 rr XXXXXX RR =

25 MULTIPLE ACCESS CONTROL Multiple Access Control protocols are not very useful on a point-to-point link in which there is only a single sender and receiver at either end. If however, we have a broadcast link, then we can have multiple sending and receiving nodes all connected to the same, single, shared broadcast channel. Simultaneous transmission by nodes can result in collisions or interference. The coordination of the shared broadcast medium is thus a problem that must be resolved by Multiple Access Control (MAC) protocols rules that regulate the transmission of data by nodes on a shared broadcast channel. Regardless of the type of multiple access protocol whether it be channel partitioning, random access or taking turns, there are several desirable characteristics: When only one node sends data, the node throughput should be the maximum capable on the broadcast channel, say R bps. When M nodes have data to send, the average throughput for each node should be R/M bps. The protocol should be decentralised such that there won t be a single point of failure The protocol is simple and inexpensive to implement CHANNEL PARITIONING These types of multiple access mechanisms include FDM and TDM. For a channel that supports N nodes with a transmission rate of R bps on the channel, TDM will divide time into time frames and further divided into N time slots for each of the N nodes. TDM is appealing in the sense that it eliminates collisions and is perfectly fair but each node can only send at a maximum rate of R/N bps and must wait for its turn in the transmission sequence. For FDM, a single channel is divided into different frequencies, each with a bandwidth of R/N. The result is N smaller channels of R/N bps out of the single larger R bps channel. Again, it prevents collision and bandwidth is fairly distributed. This creates bandwidth limitation for any single node. CDMA code division multiple access assigns a different code to each node to encode the data bits it sends. Good CDMA protocols allow simultaneous transmission to occur in spite of interfering transmission by other nodes. 25

26 RANDOM ACCESS PROTOCOLS In a random access protocol, a transmitting node always transmits at the full rate of the channel. When there is a collision, each node involved in the collision will repeatedly retransmit (at random intervals) the frame until it is sent successfully. SLOTTED ALOHA We shall assume that all frames are L bits, and the time is divided into slots of size L/R seconds. For simplicity, we shall assume that nodes start to transmit frames only at the beginning of slots (at the full rate of the channel) and are synchronised to know when the slots begin. When a collision occurs (denoted C on diagram), the nodes detect a collision before the end of the slot. The probability that the nodes will retransmit on the next frame is p note that this value can be different for each node and that the nodes are independent of each other. We can easily see that the probability that any one of the N nodes successfully transmits its data in a slot is: NN pp(1 pp)nn 1 1 There are N possible nodes that can transmit successfully with a probability of p. The other N-1 have a probability of 1-p of not transmitting. If we let NN we see that the maximum efficiency the protocol is ee There are three inefficiencies that occur here: Synchronisation is required There are empty slots (denoted E on diagram) since all the active nodes refrain from transmitting after a collision as a result of the probabilistic transmission policy. There are waste slots due to the collisions In pure ALOHA, there is no synchronisation, so that it is fully decentralised. It is however only half as efficient as the slotted ALOHA since there is a higher chance of collision occurring since there are two possible situations in which collisions can occur a collision with the frame before or after it. 26

27 CSMA Carrier Sense Multiple Access (CSMA) is an improvement of ALOHA in which the nodes listen to the channel to determine whether it is busy or not. If it is busy, then we wait for a random amount of time before retransmission. Collisions can still occur, since it takes time for signals to propagate through the channel channel propagation delay. This can be seen by the space diagram. At time tt 0 B senses that the channel is idle, and transmission propagates in both directions along the medium. At time tt 1 D still senses an idle channel, since B s transmission hasn t reached it yet. It thus starts transmitting. A short time later, B s transmission begins to interfere with D s transmission at D. If collision detection is implemented, then B and D will not transmit the frame in its entirety. TAKING TURNS The polling protocol requires one node to be the master node which polls each the of the nodes in a round robin fashion. The master node will transmit a message to node 1, that it can begin transmission. After this, it will do the same for node 2. Although it prevents collisions and empty slots, it has polling inefficiencies and has a single point of failure. The token passing protocol is decentralised, by using a special purpose frame known as a token. This token is exchanged amongst the nodes in some fixed order and is indicates that it has the right to transmit. If it has nothing to transmit, it will pass it on immediately to the next node. Note that the failure of one node can cause the whole network to fail and that latency issues are present when passing the token. 27

28 LINK LAYER TECHNOLOGIES WIFI With WIFI, a wireless host connects to a base station or to another wireless host through a wireless communication link. A base station is responsible for sending and receiving data between wireless hosts and is often used to coordinate the transmission of multiple wireless hosts with which it is associated. It is also possible to have an Ad-Hoc Mode network in which the hosts can directly communicate with each other. This is done by discovering and communicating with hosts in range in a peer-to-peer fashion without the use of base stations. The WIFI has 4 standards: b, a, g and more recently, n. Standard Frequency Data Rate b Ghz 11 Mbps a Ghz 54 Mbps g Ghz 54 Mbps The above three standards all use CSMA/CA as its MAC protocol and have the ability to reduce their transmission rate in order to reach out over greater distances. WIRELESS CHARACTERISTICS There are other inherent problems when we use a wireless medium for communication. The most prominent issues include: Decreasing signal strength: electromagnetic radiation attenuates as it passes through patter according to the inverse square law. Interference from other sources: Many cordless phones also work on the same 2.4 Ghz range, which can cause interference. Multipath propagation: Multipath propagation occurs when portions of the electromagnetic wave reflect off objects and the ground, taking paths of different lengths between a sender and receiver. This results in the blurring of the received signal at the receiver. Hidden Terminal Problem: Suppose that host A and C wish to transmit to B but A and C can t hear each other s transmission. If both A and C transmit, they will be unaware of their interference at B. 28

29 MAC PROTOCOL The standards use carrier sense multiple access to prevent collisions. But instead of using collision detection, they use collision avoidance CSMA/CA. The reasoning for not implementing collision detection is due to the difficulties in: Dealing with the hidden terminal problem and attenuation Dealing with separating the weak received signals from your own transmission signals. The protocol follows a set of procedures: 1. If initially the sender senses the channel is idle, it transmits its frame after a short period of time known as the Distributed Inter-frame Space (DIFS). 2. Otherwise, the sender chooses a random backoff value and counts down this value when the channel is sensed idle. While the channel is sensed busy, the counter value remains frozen. 3. When the counter reaces zero, the station transmits the entire frame and then waits for an acknowledgement (ACK). 4. The sender will return an ACK after SIFS (Short Inter-frame Spacing) if the data has passed the CRC check. 5. If an acknowledgement is received by the source, the transmitting station knows that its frame has been correctly received at the destination. If the station has another frame to send, it begins the CSMA/CA protocol at step 2. If the acknowledgement isn t received, the transmitting station reenters the backoff phase in step 2, with the random value chosen from a larger interval. As we have just seen, collisions can still occur. We can further reduce the waste due to collisions by allowing the sender to reserve the channel. The sender will first transmit a small request to send (RTS) packet to base station using CSMA. If the RTSs collide, this will be detected but this is negligible since they re short. If the RTS packet is received by the base station, it will broadcast a clear to send CTS response addressed to the sender. Since this is heard by all other stations, all stations except the sender will refrain from transmission. In this way, collision of actual data packets is avoided. 29

30 ETHERNET Ethernet is the most prominent LAN technology due to its wide deployment, simplicity and continuous improvement in technology. First developed by Metcalfe, the Ethernet LAN consisted of a coaxial bus in which the nodes would tap into the cable. Hence, the Ethernet was said to have a bus topology and is a broadcast LAN. Improvements however made it possible to introduce hub-based star topology. A hub is a physical layer repeater which relays bits coming in from one interface on every other interface at the same rate (no frame buffering) and with no ability to perform collision detection. Hence, this topology is also a broadcast LAN. At the turn of the millennium, switches began to replace hubs a layer 2 device which stores and forwards packets. ETHERNET FRAME STRUCTURE Any information from higher layered protocols is encapsulated by a lower layered protocol. An Ethernet frame encapsulates network layer datagrams usually IP. PREAMBLE Each frame starts with an 8 bytes preamble: 7 bytes of byte of This is used to synchronise the clock rates between the sender and receiver. DESTINATION & SOURCE ADDRESS The destination and source address are 6 bytes long (giving 281 trillion combinations) and unique to each network adapter. A typical MAC address could be 08-0b-db-e4-b1-02 with the upper 24 bits interpreted as the organisationally unique identifier and the lower 24 bits being organisation assigned portion. Hence, a manufacturer such as Cisco or TP-LINK will have all their products with the first 24 bits being the same. The lower 24 bits is then individually assigned to each adapter. There are also special conditions with these MAC address: Unicast transmission have addresses with the lowest bit of the first byte being 0 Multicast transmissions have addresses with the lowest bit of the first byte being 1 Broadcast transmissions use a MAC address of ff-ff-ff-ff-ff-ff 30

31 As mentioned above, Ethernet was broadcast based. Hence the Ethernet bus can contain frames which are meant for many devices. Obviously, it would not be very useful to accept every Ethernet Frame and to pass it up to the network layer. Instead, the adapter only accepts the frame if the destination address matches the adapter s address, or if the destination is broadcast or if the destination address is a multicast in which the adapter has been configured to accept. TYPE FIELD The type field is 2 bytes long and permits Ethernet to multiplex network-layer protocols. This allows the adapter to forward the data to the associated network layer protocol such as IP, ARP, Novell IPX or AppleTalk. CRC The purpose of the CRC field is to allow the receiving adapter to detect bit errors in frame. If an error is detected, the frame is simply dropped. AN UNREALIABLE CONNECTIONLESS SERVICE Ethernet technologies provide connectionless service to the network layer since no handshaking is done between the sending and receiving adapter. It is also an unreliable technology since it does not send acknowledgements or NACKS. This allows Ethernet to be simple and cheap, at the expense of introducing gaps in the streams of datagrams passed to the network layer (unless we use TCP in which retransmissions will fill the gaps). ETHERNET S MAC PROTOCOL : CSMA/CD As we have seen already CSMA/CD will: 1. Allow adapters to begin transmission at any time (no slots like ALOHA) after it senses (the voltage level) that the channel is idle. 2. Abort transmission if it detects that another adapter is also transmitting, that is, collision detection. 3. Before reattempting retransmission, an adapter waits a random time that is typically small compared with the time to transmit a frame If many nodes have frames to transmit, the effective transmission rate of the channel can be much less than if only one node has a frame to send. The efficiency of Ethernet is defined to be the long run fraction of time during which frames are being transmitted on the channel without collisions when there is a large number of active nodes that have large frames in their buffers: EEEEEEEEEEEEEEEEEEEE = dd pppppppp dd tttttttttt 31

32 CSMA/CD ALGORITHM 1. The adapter obtains a datagram from the network layer, prepares an Ethernet frame and puts the frame in an adapter buffer 2. If the adapter senses the channel is idle ( there is no signal energy entering the adapter from the channel for 96 bit times i.e the time to transmit 96 bits), it starts to transmit the frame. If the adapter senses that the channel is busy, it waits until it senses no signal energy and then starts to transmit the frame. 3. While transmitting, the adapter monitors for the presence of signal energy coming from other adapters. If the adapter transmits the entire frame without detecting signal energy from other adapters, the adapter is finished with the frame. 4. If the adapter detects signal energy from other adapters while transmitting, it stops transmitting its frame and instead transmits a 48 bit jam signal. This ensures all other transmitting adapters become aware of the collision. 5. After transmitting the jam signal and aborting transmission, the adapter enters an exponential backoff phase. After the nth collision in a row for a particular frame, the adapter chooses a value for K at random from {0, 1, 2., 2 mm 1} where mm = min (nn, 10). The adapter then waits KK 512 bit times and returns to step 2. The goal of the exponential backoff is to adapt retransmission attempts to estimated current load. If there are only a small number of colliding adapters, it makes sense to choose K from a small set of values, whereas a heavy load demands for K to be chosen from a larger, more dispersed set of values. ETHERNET TECHNOLOGY Despite the changes in speed and media, and as we shall see, the MAC protocol is unnecessary in a switch based LAN, the enduring constant of Ethernet has been its frame format. It is thus with no surprise that even though the huge differences between Metcalfe s original Ethernet and today s Ethernets, we still call this link layer protocol Ethernet. 10BASE2 The name given to this technology comes from the fact that this technology has specification: 10Mbps, 200 metres max cable length. It uses a thin coaxial cable in a bus topology with repeaters connecting multiple segments together. 10BASET/100BASET FAST ETHERNET This technology uses the twisted pair and can provide speeds of 10 or 100 Mbps. The nodes are connected to either a hub or a switch in a star topology. Modern switches are full duplex - a switch and a node can each send frames to each other at the same time without interference. In essence then, a switch based LAN will have no collisions (if we have point to point links), and removes the need for a MAC protocol. 32