Packet Networks Packet Networks: A network packet is a formatted unit of data carried by a packetswitched network. Computer communications links that do not support packets, such as traditional point-to-point telecommunications links, simply transmit data as a bit stream. OR, A network packet is a formatted unit of data carried by a packet-switched network. Computer communications links that do not support packets, such as traditionalpoint-to-point telecommunications links, simply transmit data as a bit stream. When data is formatted into packets, the bandwidth of the communication medium can be better shared among users than if the network were circuit switched. A packet consists of control information and user data, which is also known as the payload. Control information provides data for delivering the payload, for example: source and destination network addresses, error detection codes, and sequencing information. Typically, control information is found in packet headers and trailers. Example: IP packets 1. 4 bits that contain the version, that specifies if it's an IPv4 or IPv6 packet, 2. 4 bits that contain the Internet Header Length, which is the length of the header in multiples of 4 bytes (e.g., 5 means 20 bytes). 3. 8 bits that contain the Type of Service, also referred to as Quality of Service (QoS), which describes what priority the packet should have, 4. 16 bits that contain the length of the packet in bytes, 5. 16 bits that contain an identification tag to help reconstruct the packet from several fragments, 6. 3 bits. The first contains a zero, followed by a flag that says whether the packet is allowed to be fragmented or not (DF: Don't fragment), and a flag to state whether more fragments of a packet follow (MF: More Fragments) 7. 13 bits that contain the fragment offset, a field to identify position of fragment within original packet 8. 8 bits that contain the Time to live (TTL), which is the number of hops (router, computer or device along a network) the packet is allowed to pass before it dies (for example, a packet with a TTL of 16 will be allowed to go across 16 routers to get to its destination before it is discarded), 9. 8 bits that contain the protocol (TCP, UDP, ICMP, etc.) 10. 16 bits that contain the Header Checksum, a number used in error detection, 11. 32 bits that contain the source IP address, 12. 32 bits that contain the destination address. NICAM packet transmission The NICAM packet (except for the header) is scrambled with a nine-bit pseudo-random bit-generator before transmission. The topology of this pseudo-random generator yields a bitstream with a repetition period of 511 bits. The pseudo-random generator's polynomial is: x^9 + x^4 + 1. The pseudo-random generator is initialized with: 111111111. Making the NICAM bitstream look more like white noise is important because this reduces signal patterning on adjacent TV channels. The NICAM header is not subject to scrambling. This is necessary so as to aid in locking on to the NICAM data stream and resynchronisation of the data stream at the receiver. At the start of each NICAM packet the pseudo-random bit generator's shift-register is reset to allones. NICAM NICAM offers the following possibilities. The mode is auto-selected by the inclusion of a 3-bit type field in the data-stream One digital stereo sound channel. Two completely different digital mono sound channels. One digital mono sound channel and a 352 kbit/s data channel. One 704 kbit/s data channel. Example: Radio and TV broadcasting[edit] MPEG packetized stream[edit] Packetized Elementary Stream (PES) is a specification defined by the MPEG communication protocol (see the MPEG-2standard) that allows an elementary stream to be divided into packets. The elementary stream is packetized by encapsulating sequential data bytes from the elementary stream inside PES packet headers. A typical method of transmitting elementary stream data from a video or audio encoder is to first create PES packets from the elementary stream data and then to encapsulate these PES packets inside an MPEG transport stream (TS) packets or an MPEG program stream (PS). The TS packets can then be multiplexed and transmitted using broadcasting techniques, such as those used in an ATSC and DVB. Example: the NASA Deep Space Network[edit] The Consultative Committee for Space Data Systems (CCSDS) packet telemetry standard defines the protocol used for the transmission of spacecraft instrument data over the deep-space channel. Under this standard, an image or other data sent from a spacecraft instrument is transmitted using one or more packets. CCSDS packet definition[edit] A packet is a block of data with length that can vary between successive packets, ranging from 7 to 65,542 bytes, including the packet header. Packetized data is transmitted via frames, which are fixed-length data blocks. The size of a frame, including frame header and control information, can range up to 2048 bytes. Packet sizes are fixed during the development phase. Because packet lengths are variable but frame lengths are fixed, packet boundaries usually do not coincide with frame boundaries. Telecom processing notes[edit] Data in a frame is typically protected from channel errors by error-correcting codes. Even when the channel errors exceed the correction capability of the error-correcting code, the presence of errors is nearly always detected by the error-correcting code or by a separate errordetecting code. Frames for which uncorrectable errors are detected are marked as undecodable and typically are deleted. Handling data loss[edit] Deleted undecodable whole frames are the principal type of data loss that affects compressed data sets. In general, there would be little to gain from attempting to use compressed data from a frame marked as undecodable. When errors are present in a frame, the bits of the subband pixels are already decoded before the first bit error will remain intact, but all subsequent decoded bits in the segment usually will be completely corrupted; a single bit error is often just as disruptive as many bit errors. Furthermore, compressed data usually are protected by powerful, long-blocklength errorcorrecting codes, which are the types of codes most likely to yield substantial fractions of bit errors throughout those frames that are undecodable. Thus, frames with detected errors would be essentially unusable even if they were not deleted by the frame processor. This data loss can be compensated for with the following mechanisms. If an erroneous frame escapes detection, the decompressor will blindly use the frame data as if they were reliable, whereas in the case of detected erroneous frames, the decompressor can base its reconstruction on incomplete, but not misleading, data. However, it is extremely rare for an erroneous frame to go undetected. For frames coded by the CCSDS Reed–Solomon code, fewer than 1 in 40,000 erroneous frames can escape detection. All frames not employing the Reed–Solomon code use a cyclic redundancy check (CRC) errordetecting code, which has an undetected frame-error rate of less than 1 in 32,000. Mixed Network Traffic Traffic mix is a traffic model in telecommunication engineering and teletraffic theory. Definitions A traffic mix is a modelisation of user behaviour. In telecommunications, user behaviour activities may be described by a number of systems, ranging from simple to complex. For example, for plain old telephone service (POTS), a sequence of connection requests to an exchange can be modelled by fitting negative exponential distributions to the average time between requests and the average duration of a connection. This in turn can be used to work out the utilisation of the line for the purposes of network planning and dimensioning. Objectives Traffic mix has two goals: Network links dimensioning Network equipment dimensioning Both these functions are extremely important to network operators. If insufficient capability is deployed at a node (for example, if a backbone router has 1 gigabit/sec of switching capacity and more than this is offered) then the risk of equipment failure increases, and customers experience poor service. However, if the network is overprovisioned the cost in equipment can be high. Most providers therefore seek to maximise the effect of their spending by maintaining an unused overhead capacity for growth, and expanding key nodes to relieve problem areas. Identification of these areas is accomplished by network dimensioning. Traffic mix type Telephony traffic mix Call attempts per day Call holding time Mean holding time Mobile telephony traffic mix Call attempts Call holding time Mean holding time Mean number of SMS send Mean number of SMS received User mobility Internet traffic mix UL/DL acknowledged Kbs Packet Data Channel allocation successes User throughput Session/Packet interarrival time/Latency CELL Networks A cellular network or mobile network is a communications network where the last link is wireless. The network is distributed over land areas called cells, each served by at least one fixed-location transceiver, known as a cell site or base station. Cellular Network Definition - What does Cellular Network mean? A cellular network is a radio network distributed over land through cells where each cell includes a fixed location transceiver known as base station. These cells together provide radio coverage over larger geographical areas. User equipment (UE), such as mobile phones, is therefore able to communicate even if the equipment is moving through cells during transmission. Cellular networks give subscribers advanced features over alternative solutions, including increased capacity, small battery power usage, a larger geographical coverage area and reduced interference from other signals. Popular cellular technologies include the Global System for Mobile Communication, general packet radio service, 3GSM and code division multiple access. Techopedia explains Cellular Network Cellular network technology supports a hierarchical structure formed by the base transceiver station (BTS), mobile switching center (MSC), location registers and public switched telephone network (PSTN). The BTS enables cellular devices to make direct communication with mobile phones. The unit acts as a base station to route calls to the destination base center controller. The base station controller (BSC) coordinates with the MSC to interface with the landline-based PSTN, visitor location register (VLR), and home location register (HLR) to route the calls toward different base center controllers. Cellular networks maintain information for tracking the location of their subscribers' mobile devices. In response, cellular devices are also equipped with the details of appropriate channels for signals from the cellular network systems. These channels are categorized into two fields: Strong Dedicated Control Channel: Used to transmit digital information to a cellular mobile phone from the base station and vice versa. Strong Paging Channel: Used for tracking the mobile phone by MSC when a call is routed to it. A typical cell site offers geographical coverage of between nine and 21 miles. The base station is responsible for monitoring the level of the signals when a call is made from a mobile phone. When the user moves away from the geographical coverage area of the base station, the signal level may fall. This can cause a base station to make a request to the MSC to transfer the control to another base station that is receiving the strongest signals without notifying the subscriber; this phenomenon is called handover. Cellular networks often encounter environmental interruptions like a moving tower crane, overhead power cables, or the frequencies of other devices. Cellular network A cellular network or mobile network is a communications network where the last link is wireless. The network is distributed over land areas called cells, each served by at least one fixedlocation transceiver, known as a cell site or base station. In a cellular network, each cell uses a different set of frequencies from neighboring cells, to avoid interference and provide guaranteed bandwidth within each cell. When joined together these cells provide radio coverage over a wide geographic area. This enables a large number of portable transceivers (e.g., mobile phones,pagers, etc.) to communicate with each other and with fixed transceivers and telephones anywhere in the network, via base stations, even if some of the transceivers are moving through more than one cell during transmission. Cellular networks offer a number of desirable features: More capacity than a single large transmitter, since the same frequency can be used for multiple links as long as they are in different cells Mobile devices use less power than with a single transmitter or satellite since the cell towers are closer Larger coverage area than a single terrestrial transmitter, since additional cell towers can be added indefinitely and are not limited by the horizon Major telecommunications providers have deployed voice and data cellular networks over most of the inhabited land area of the Earth. This allows mobile phones and mobile computing devices to be connected to the public switched telephone network and public Internet. Private cellular networks can be used for research[1] or for large organizations and fleets, such as dispatch for local public safety agencies or a taxicab company.[2] Syncronous TDM Difference Between Synchronous TDM And Asynchronous TDM Synchronous TDM is the standard operation mode of TDM operation where each multiplexed channel has a specific size slot allowed for it, whether data exist to transmit or not. This is very wasteful of bandwidth space if you have channels that only occasionally transmit. Asynchronous TDM is actually called Statistical TDM. STDM does not reserve a time slot for each channel, rather it assigns a slot when the channel is requiring data to be sent or received. This allows for variable bandwidth per channel as required. Time-division multiplexing Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a common signal path by means of synchronized switches at each end of the transmission line so that each signal appears on the line only a fraction of time in an alternating pattern. This form of signalmultiplexing was developed in telecommunications for telegraphy systems in the late 1800s, but found its most common application in digital telephony in the second half of the 20th century. Application examples The plesiochronous digital hierarchy (PDH) system, also known as the PCM system, for digital transmission of several telephone calls over the same four-wire copper cable (T-carrier or E-carrier) or fiber cable in the circuit switched digital telephone network The synchronous digital hierarchy (SDH)/synchronous optical networking (SONET) network transmission standards that have replaced PDH. The Basic Rate Interface and Primary Rate Interface for the Integrated Services Digital Network (ISDN). The RIFF (WAV) audio standard interleaves left and right stereo signals on a per-sample basis TDM can be further extended into the time division multiple access (TDMA) scheme, where several stations connected to the same physical medium, for example sharing the same frequency channel, can communicate. Application examples include: The GSM telephone system The Tactical Data Links Link 16 and Link 22 Multiplexed digital transmission In circuit-switched networks, such as the public switched telephone network (PSTN), it is desirable to transmit multiple subscriber calls over the same transmission medium to effectively utilize the bandwidth of the medium.[3] TDM allows transmitting and receiving telephone switches to create channels (tributaries) within a transmission stream. A standard DS0voice signal has a data bit rate of 64 kbit/s.[3][4] A TDM circuit runs at a much higher signal bandwidth, permitting the bandwidth to be divided into time frames (time slots) for each voice signal which is multiplexed onto the line by the transmitter. If the TDM frame consists of n voice frames, the line bandwidth is n*64 kbit/s.[3] Each voice time slot in the TDM frame is called a channel. In European systems, standard TDM frames contain 30 digital voice channels (E1), and in American systems (T1), they contain 24 channels. Both standards also contain extra bits (or bit time slots) for signaling and synchronization bits.[3] Multiplexing more than 24 or 30 digital voice channels is called higher order multiplexing. Higher order multiplexing is accomplished by multiplexing the standard TDM frames. For example, a European 120 channel TDM frame is formed by multiplexing four standard 30 channel TDM frames. At each higher order multiplex, four TDM frames from the immediate lower order are combined, creating multiplexes with a bandwidth of n*64 kbit/s, where n = 120, 480, 1920, etc.[3] Telecommunications systems There are three types of synchronous TDM: T1, SONET/SDH, and ISDN.[5] Plesiochronous digital hierarchy (PDH) was developed as a standard for multiplexing higher order frames. PDH created larger numbers of channels by multiplexing the standard Europeans 30 channel TDM frames. This solution worked for a while; however PDH suffered from several inherent drawbacks which ultimately resulted in the development of theSynchronous Digital Hierarchy (SDH). The requirements which drove the development of SDH were these:[3][4] Be synchronous – All clocks in the system must align with a reference clock. Be service-oriented – SDH must route traffic from End Exchange to End Exchange without worrying about exchanges in between, where the bandwidth can be reserved at a fixed level for a fixed period of time. Allow frames of any size to be removed or inserted into an SDH frame of any size. Easily manageable with the capability of transferring management data across links. Provide high levels of recovery from faults. Provide high data rates by multiplexing any size frame, limited only by technology. Give reduced bit rate errors. SDH has become the primary transmission protocol in most PSTN networks. It was developed to allow streams 1.544 Mbit/s and above to be multiplexed, in order to create larger SDH frames known as Synchronous Transport Modules (STM). The STM-1 frame consists of smaller streams that are multiplexed to create a 155.52 Mbit/s frame. SDH can also multiplex packet based frames e.g. Ethernet, PPP and ATM.[3][4] While SDH is considered to be a transmission protocol (Layer 1 in the OSI Reference Model), it also performs some switching functions, as stated in the third bullet point requirement listed above.[3] The most common SDH Networking functions are these: SDH Crossconnect – The SDH Crossconnect is the SDH version of a Time-Space-Time crosspoint switch. It connects any channel on any of its inputs to any channel on any of its outputs. The SDH Crossconnect is used in Transit Exchanges, where all inputs and outputs are connected to other exchanges.[3] SDH Add-Drop Multiplexer – The SDH Add-Drop Multiplexer (ADM) can add or remove any multiplexed frame down to 1.544Mb. Below this level, standard TDM can be performed. SDH ADMs can also perform the task of an SDH Crossconnect and are used in End Exchanges where the channels from subscribers are connected to the core PSTN network.[3] SDH network functions are connected using high-speed optic fibre. Optic fibre uses light pulses to transmit data and is therefore extremely fast. Modern optic fibre transmission makes use of wavelength-division multiplexing (WDM) where signals transmitted across the fibre are transmitted at different wavelengths, creating additional channels for transmission. This increases the speed and capacity of the link, which in turn reduces both unit and total costs.[3][4] Statistical time-division multiplexing[edit] Statistical time division multiplexing (STDM) is an advanced version of TDM in which both the address of the terminal and the data itself are transmitted together for better routing. Using STDM allows bandwidth to be split over one line. Many college and corporate campuses use this type of TDM to distribute bandwidth. On a 10-Mbit line entering a network, STDM can be used to provide 178 terminals with a dedicated 56k connection (178 * 56k = 9.96Mb). A more common use however is to only grant the bandwidth when that much is needed. STDM does not reserve a time slot for each terminal, rather it assigns a slot when the terminal is requiring data to be sent or received. In its primary form, TDM is used for circuit mode communication with a fixed number of channels and constant bandwidth per channel. Bandwidth reservation distinguishes time-division multiplexing from statistical multiplexing such as statistical time division multiplexing. In pure TDM, the time slots are recurrent in a fixed order and pre-allocated to the channels, rather than scheduled on a packetby-packet basis. In dynamic TDMA, a scheduling algorithm dynamically reserves a variable number of time slots in each frame to variable bit-rate data streams, based on the traffic demand of each data stream. Dynamic TDMA is used in: HIPERLAN/2 Dynamic synchronous transfer mode IEEE 802.16a Asynchronous time-division multiplexing (ATDM),[5] is an alternative nomenclature in which STDM designates synchronous time-division multiplexing, the older method that uses fixed time slots. What is TDM? Time Division Multiplexing (TDM) Time Division Multiplexing processes information of different transmitters successively in defined time segments for transmission over one channel. Time Division Multiplexing is differentiated in synchronous and asynchronous multiplexing. Asynchronous Time Division Multiplexing Single data streams are classified in variable time segments and subsequently transmitted using the asynchronous time division multiplexing procedure. Thereby transmission occurs in no definite order. Each time segment receives a channel information number to separate them again in the demultiplexer procedure. Synchronous Time Division Multiplexing Using the synchronous method, single data streams are classified in defined time segments for subsequent transmission in predetermined order. . Using this TDM procedure for example 8 x Gigabit Ethernet applications can be multiplexed to one 10 Gigabit signal. In the following this independent 10 Gigabit signal can easily be connected to a DWDMCDWM multiplexer. Consequently more signals per wavelength can be transmitted and the entire xWDM system is utilized more efficiently. Theoretically up to 240 Gigabit Ethernet signals can be transmitted using one pair of fibers by adding TDM multiplexer to a xWDM system. Synchronous TDM works by the muliplexor giving exactly the same amount of time to each device connected to it. This time slice is allocated even if a device has nothing to transmit. This is wasteful in that there will be many times when allocated time slots are not being used. Therefore, the use of Synchronous TDM does not guarantee maximum line usage and efficiency. Asynchronous TDM is a more flexible method of TDM. With Asynchronous TDM the length of time allocated is not fixed for each device but time is given to devices that have data to transmit. This version of TDM works by tagging each frame with an identification number to note which device it belongs to. This may require more processing by the multiplexor and take longer, however, the time saved by efficient and effective bandwidth utilization makes it worthwhile. Asynchronous TDM allows more devices than there is physical bandwidth for. Virtual Connection: A virtual circuit (VC) is a means of transporting data over a packet switched computer network in such a way that it appears as though there is a dedicated physical layer link between the source and destination end systems of this data. virtual connection A temporary connection made between two nodes. Computer Desktop Encyclopedia copyright ©1981-2015 by The Computer Language Company Inc. All Right reserved. THIS DEFINITION IS FOR PERSONAL USE ONLY. All other reproduction is strictly prohibited without permission from the publisher. virtual connection (networking) 1. (VC) A connection or a path through an ATM network. The word "virtual" indicates that theconnection is logical rather than physical. Nothing to do with a virtual circuit on a packet switching network. [Fred Halsall, "Data Communications, Computer Networks and Open Systems", 1996,Addison Wesley]. 2. A communications link that appears to be a direct connection between sender andreceiver, although physically the link can be routed through a more circuitous path, running over virtual circuits instead of a private network built primarily with dedicated lines. A virtual connection can provide full-time connection among many sites, including thoseconfigured for SNA/SDLC protocol. A virtual connection can handle any trans mission protocol and is supported worldwide. It can provide high throughput and low delay for LAN andInternet applications, peer-to-peer connectivity, client-server computing, and other distributedprocessing applications. Identifiers: Identifier/Locator Network Protocol From Wikipedia, the free encyclopedia The Identifier/Locator Network Protocol (RFCs) is a network protocol designed to separate the two functions of network addresses, the identification of network endpoints, and assisting routing by separating topological information from node identity. ILNP is backwards-compatible with existing IP, and is incrementally-deployable. ILNP itself is an architecture with two different instantiations at present. ILNPv4 is ILNP engineered to work as a set of IPv4 extensions, while ILNPv6 is ILNP engineered as a set of IPv6 extensions. At least 2 independent open-source implementations of ILNPv6 exist. U. St Andrews (Scotland) has a prototype in FreeBSD/x86, while Tsinghua U. (China) has a prototype in Linux/x86. In February 2011, IRTF Routing Research Group (RRG) Chairs recommended that the IETF standardise ILNP (RFC 6115) as the preferred evolutionary direction for IPv6. ILNP Specifications (RFCs) ILNP Architectural Description (RFC 6740) ILNP Engineering Considerations (RFC 6741) DNS Resource Records for ILNP (RFC 6742) ICMPv6 Locator Update Message for ILNPv6 (RFC 6743) IPv6 Nonce Destination Option for ILNPv6 (RFC 6744) ICMP Locator Update for IPv4 (RFC 6745) IPv4 Options for ILNPv4 (RFC 6746) Address Resolution Protocol (ARP) for ILNPv4 (RFC 6747) Optional Advanced Deployment Scenarios for ILNP (RFC 6748) The Network Access Identifier (NAI) is the user identity submitted by the client during network access authentication. It is used mainly for two purposes: 1) The NAI is used when roaming, to identify the user. 2) To assist in the routing of the authentication request to the user's authentication server. Here are a few examples of UIDs: A Uniform Resource Identifier (URI) is a unique identifier that makes content addressable on the Internet by uniquely targeting items, such as text, video, images and applications. A Uniform Resource Locator (URL) is a particular type of URI that targets Web pages so that when a browser requests them, they can be found and served to users. A Universal Unique Identifier (UUID) is a 128-bit number used to uniquely identify some object or entity on the Internet. A global unique identifier (GUID) is a number that Microsoft programming generates to create a unique identity for an entity such as a Word document. A bank identifier code (BIC) is a unique identifier for a specific financial institution. A unique device identifier (UDID) is a 40-character string assigned to certain Apple devices including the iPhone, iPad, and iPod Touch. An service set identifier (SSID) is a sequence of characters that uniquely names a wireless local area network (WLAN). A national provider identifier (NPI) is a unique ten-digit identification number required by HIPAAfor all health care providers in the United States. IP address - Internet Protocol (IP) address IP address is short for Internet Protocol (IP) address. IP specifies the format of packets, also called datagrams, and the addressing scheme. An IP address is an identifier for a computer or device on aTCP/IP network. Networks using the TCP/IP protocol route messages based on the IP address of the destination. The Format of an IP Address The format of an IP address is a 32-bit numeric address written as four numbers separated by periods. Each number can be zero to 255. For example, 1.160.10.240 could be an IP address. Within an isolated network, you can assign IP addresses at random as long as each one is unique. However, connecting a private network to the Internet requires using registered IP addresses (called Internet addresses) to avoid duplicates. Static Versus Dynamic IP Addresses An IP address can be static or dynamic. A static IP address will never change and it is a permanent Internet address. A dynamic IP address is a temporary address that is assigned each time a computer or device accesses the Internet. The four numbers in an IP address are used in different ways to identify a particular network and a host on that network. Four regional Internet registries -- ARIN, RIPE NCC, LACNIC and APNIC-- assign Internet addresses from the following three classes: Class A - supports 16 million hosts on each of 126 networks Class B - supports 65,000 hosts on each of 16,000 networks Class C - supports 254 hosts on each of 2 million networks The number of unassigned Internet addresses is running out, so a new classless scheme called CIDR is gradually replacing the system based on classes A, B, and C and is tied to adoption of IPv6. In IPv6 the IP address size is increased from 32 bits to 128 bits. What is My IP Address? To view your IP address you can use the ipconfig (IPCONFIG) command line tool. Ipconfig displays all current TCP/IP network configuration values and refreshes Dynamic Host Configuration Protocol (DHCP) and Domain Name System (DNS) settings. To launch the command prompt from a Windows-based computer click: Start > All Programs > Accessories >Command Prompt. Type ipconfig and press the Enter key. You can also use Google search to find your IP address. Type "what is my IP address" as a search query and Google will show the IP address of the computer from which the query was received as the top search result. IP blocking IP blocking is a form of security used on mail, Web or any otherInternet servers to block connections from a specific IP address or range of addresses that are considered undesirable or hostile. For example, a Web site forum administrator who sees spam or unwanted posts from a user may block that user's IP address to prevent them from using the discussion board. FCB (File Control Block) is an internal file system structure used inDOS for accessing files on disk. The FCB block contains information about the drive name, filename, file type and other information that is required by the system when a file is accessed or created. CELL: A cell is the intersection between a row and a column on a spreadsheet that starts with cell A1. Below is an illustrated example of a highlighted cell in Microsoft Excel; the cell address,cell name, or cell pointer "D8" (column D, row 8) is the selected cell and the location of what is being modified. Each cell in a spreadsheet can contain any value that can be called using a relative cell reference or called upon using a formula. See our spreadsheet definition for further information on using spreadsheets. How many cells are in a spreadsheet? With Microsoft Excel 2010, 1,048,576 rows by 16,384 columns are support, which means there can be 17,179,869,184 cells. Each cell supports up to a maximum of 32,767 characters. 2. Like a spreadsheet cell, a cell is section within an HTML table that is created using the HTML <td> tag. 3. Cell is another name for a cell phone. 4. A cell refers to a unit of data that is transferred over an ATM network. 5. Cell is the geographical area that a cell tower covers, for cell phone reception. It is often measured in square miles and may overlap with another cell tower's cell. ATM ATM adaptation layer The use of Asynchronous Transfer Mode (ATM) technology and services creates the need for an adaptation layer in order to support information transfer protocols, which are not based on ATM. This adaptation layer defines how to segment and reassemble higher-layer packets into ATM cells, and how to handle various transmission aspects in the ATM layer. Examples of services that need adaptations are Gigabit Ethernet, IP, Frame Relay, SONET/SDH, UMTS/Wireless, etc. only optical fibers can be used. The main services provided by AAL (ATM Adaptation Layer) are: Segmentation and reassembly Handling of transmission errors Handling of lost and misinserted cell conditions Timing and flow control The following ATM Adaptation Layer protocols (AALs) have been defined by the ITU-T.[1] It is meant that these AALs will meet a variety of needs. The classification is based on whether a timing relationship must be maintained between source and destination, whether the application requires a constant bit rate, and whether the transfer is connection oriented or connectionless. AAL Type 0 (also referred as raw cells) consists of 48 bytes of payload without any reservation for special fields. AAL Type 1 supports constant bit rate (CBR), synchronous, connection oriented traffic. Examples include T1 (DS1), E1, and x64 kbit/s emulation. AAL Type 2 supports time-dependent Variable Bit Rate (VBR-RT) of connection-oriented, synchronous traffic. Examples include Voice over ATM. AAL2 is also widely used in wireless applications due to the capability of multiplexing voice packets from different users on a single ATM connection. AAL Type 3/4 supports VBR, data traffic, connection-oriented, asynchronous traffic (e.g. X.25 data) or connectionless packet data (e.g. SMDS traffic) with an additional 4-byte header in the information payload of the cell. Examples include Frame Relay and X.25. AAL Type 5 is similar to AAL 3/4 with a simplified information header scheme. This AAL assumes that the data is sequential from the end user and uses the Payload Type Indicator (PTI) bit to indicate the last cell in a transmission. Examples of services that use AAL 5 are classic IP over ATM, Ethernet Over ATM, SMDS, and LAN Emulation (LANE). AAL 5 is a widely used ATM adaptation layer protocol. This protocol was intended to provide a streamlined transport facility for higher-layer protocols that are connection oriented. AAL 5 was introduced to: reduce protocol processing overhead. reduce transmission overhead. ensure adaptability to existing transport protocols. The AAL 5 was designed to accommodate the same variable bit rate, connection-oriented asynchronous traffic or connectionless packet data supported by AAL 3/4, but without the segment tracking and error correction requirements. Class A Class B Class X Class C Class D Circuit emulation Compressed video Cell relay Bursty data Datagram service constant Bit Rate Variable Bit Rate VBR VBR VBR Timing Required Timing Required Timing Not Required Timing Not Required Timing Not Required Connection Oriented Connection Oriented Connection Oriented Connection Oriented Connection less AAL 1 AAL 2 AAL 0 AAL 3/4 AAL --3/4 & AAL 5 ATM Adaptation Layer (AAL) The ATM Adaption Layer, (AAL), makes the ATM layer services more adaptable to specific services. The specific services may include user services, control services and management services. The AAL is the layer above the ATM layer and it is responsible for converting the information from the higher layers into 48 byte lengths so that the ATM layer can add the 5 byte header to make the 53 byte cell. The two main functions of this AAL are to provide functions needed to support applications and to break up information into units that will fit into cells. The AAL layer is thus divided into two sublayers: the convergence sublayer (CS) and segmentation and reassembly sublayer (SAR). The convergence sublayer provides the functions needed to support specific applications, such as handling the cell delay variation and keeping a track of the clock. Each application accesses the AAL at a service access point (SAP), which is the address of the application. The SAR sublayer packs the information from the CS into cells and unpacks the information at the destination. The SAR maps SAR headers plus CS information into 48 byte cells. The AAL accommodates all services and in particular adapts both packet switched and circuit switched services. The CCITT service classification is based upon the timing relation, bit rate, and connection mode. Figure 3 depicts the CCITT service classification according to these parameters. There are five AAL types that correspond approximately to the CCITT service classes as shown in Figure 3. Figure 3: CCITT Service Classification Class A is a constant bit rate connection with a timing relationship between source and destination and is often called circuit emulation. This could be used to carry voice of 64 kb/s or constant bit rate video. This could also be used for intelligent multiplexing equipment that needs what is essentially a circuit. The adaption layer that deals with this type of traffic is called AAL 1. AAL 1 operates by placing a 1 byte header on 47 bytes of user data and then transferring the 48 bytes to the ATM layer. The SAR of the AAL 1 will be notified of the extistance of the CS sublayer by the CS indicator, (CSI). A sequence number, (SN), is passed from the CS sublayer to the SAR and this SN can be used to detect lost of missing SAR loads. Finally the header of the SAR is protected by a sequence number protection, (SNP), field which can inform the CS sublayer of bit errors. The layout of the SAR is shown in Figure 4. Figure 4: AAL 1 Cell Format Class A is most appropriate for voice transmission that does not incorporate time assignment speech interpolation (TASI). In TASI speech is only transmitted when the speaker is active. To incorporate efficiencies that can be achieved with coding and compression techniques on real time services there is a second class called Class B. Class B is used for services similar to Class A but which are not constant bit rate. Examples of these would be variable bit rate audio and video. AAL 2 is the AAL layer responsible for providing these type of services from the ATM layer to the higher layers. AAL 2 is not yet fully specified but there is some indication as to the format of the protocol. As the intended use is for compressed voice and video there will likely be strict bounds on the bit error rate. To help the system there is likely to be a CRC in the SAR to protect all the data being sent. As the user field may not be full it is likely that the user amount of information will be variable length and this will be indicated by the length indicator, (LI). The remaining CCITT classes of service and AAL's are used for services which have no relationship of timing between source and destination. These services are esentially variable bit rate data services and can be differentiated by whether they are connection oriented or not. Class C is connection oriented data transfer while Class D is connectionless. The distinction between the conectionless and connection orientated AAL's has been lessened to such an extent that they now share the same AAL called AAL 3/4. Initially AAL 3 was for Class C and AAL 4 was for Class B services. The AAL 3/4 takes information from the higher layer and after the CS sublayer operates on it the SAR breaks the data up into 44 byte sizes and adds 4 bytes of header fields to make a 48 byte information load for the ATM layer cell. The four bytes of header are made up of a 10 bit CRC, a LI of 6 bits and an SN of 4 bits. There is also a 10 bit field reserved for either multiplexing or else are reserved for future use. There is also a field called the segment type, (ST), which indicated whether the SAR is the start, middle or end of a message. The CS sublayer also adds a header and trailer to the data coming from the higher layers. As yet that is not fully defined. Because of the high overhead of the AAL 3/4, 4 bytes for every 48 bytes of ATM user information, and because of the complexity of the protocol there has been a simplified AAL proposed called AAL 5 for data transfer. The AAL 5 basically puts the headers and trailers onto the CS-PDU rather than the SAR-PDU. This has a large number of advantages like improved efficiency and better error correction and detection [11]. The format of the AAL 5 is shown in Figure 5. Figure 5: AAL 5 Cell Format PPT is a file extension for a presentation file format used by Microsoft PowerPoint, the popular presentation software commonly used for office and educational slide shows. All text images, sound and video used in the presentation are contained in the PPT file.