data communications with voip

Data Communication w/ Emphasis of VOIP

(COMP 22)

Encoded by: arfel c. arcabal

Prepared by:

marl t. gonzalez

PHASE I: DATA COMMUNICATION 1F.C. Ledesma Avenue, San Carlos City, Negros Occidental Tel. #: (034) 312-6189 / (034) 729-4327

What is Data Communication? The distance over which data moves within a computer may vary from a few thousandths of an inch, as is the case within a single IC chip, to as much as several feet along the backplane of the main circuit board. Over such small distances, digital data may be transmitted as direct, two-level electrical signals over simple copper conductors. Except for the fastest computers, circuit designers are not very concerned about the shape of the conductor or the analog characteristics of signal transmission. Frequently, however, data must be sent beyond the local circuitry that constitutes a computer. In many cases, the distances involved may be enormous. Unfortunately, as the distance between the source of a message and its destination increases, accurate transmission becomes increasingly difficult. This results from the electrical distortion of signals traveling through long conductors, and from noise added to the signal as it propagates through a transmission medium. Although some precautions must be taken for data exchange within a computer, the biggest problems occur when data is transferred to devices outside the computer's circuitry. In this case, distortion and noise can become so severe that information is lost. Data Communications concerns the transmission of digital messages to devices external to the message source. "External" devices are generally thought of as being independently powered circuitry that exists beyond the chassis of a computer or other digital message source. As a rule, the maximum permissible transmission rate of a message is directly proportional to signal power and inversely proportional to channel noise. It is the aim of any communications system to provide the highest possible transmission rate at the lowest possible power and with the least possible noise.

Lesson I: 2F.C. Ledesma Avenue, San Carlos City, Negros Occidental Tel. #: (034) 312-6189 / (034) 729-4327

Reference ModelsISO OSI reference model A set of protocol is open if: Protocol details are publicly available Changes are managed by an organization whose membership and transactions are open to the public. A system that implements open protocols is called an open system. International Organization for Standards (ISO) prescribes a standard to connect open systems Open system interconnect (OSI)

Figure 6: The (OSI) Seven Layer Model Physical Layer Specification of voltage levels, cables, connectors, timing of bots, electrical access and maintenance of circuit (i.e. corresponds to the basic hardware). Data Link Layer Transforms basic physical services to enable the transmission of units of data called frames. Frames carry data between two points on the same type of physical network, and maybe relayed if the network is extended. They normally contain low level addressing information and

3F.C. Ledesma Avenue, San Carlos City, Negros Occidental Tel. #: (034) 312-6189 / (034) 729-4327

some error checking. This layer may be involved in arbitrating access to the physical network. The Data Link Layer detects, and possibly corrects errors in the physical layer. Network Controls routing of data by providing an address domain, and in consequence the routing of messages. This addressing is separate from the hardware which implements the network connections, i.e. specifies how addresses are assigned and who packets are forwarded from one end of the network to another.

Transport Provides an interface for the upper layers to communication facilities. The presence of this layer obscures the underlying network hardware and topology from the applications. A very complex set of protocols are required for this layer! Session The protocols for this year specify how to establish a communication session with a remote system (e.g., How to login to a remote timesharing computer). Specifications for security details such as authentication using passwords are described in this layer. Presentation Layer 6 protocols specify how to represent data. Such protocols are needed because different brands of computer use different internal representation for integer and characters. Thus layer 6 protocols are needed to translate from the representation on one computer to the representation on another. Application Layer This is where the application using the network resides. Common network applications include remote login, file transfer, e-mail, and web page browsing.

Internet Protocol Suite The internet protocol suite, commonly referred to as TCP/IP, was developed about 25 years ago by DARPA for the ARPANET. The goal of the TCP/IP is to interconnect existing, often dissimilar, networks. Fundamental structure is a packet switched system in which distinct networks are connected by store-and-forward routers. The Internet Protocols are used in the Internet. The Table below compares the TCP/IP protocol with the OSI.


Transport Control Protocol (TCP) This layer creates a connection between sender receiver using port numbers. This layer can ensure that the receiver is not overrun with data (end-to-end flow control). TCP can multiplex multiple connections (using port numbers) over a single IP line. TCP can perform end-to-end error correction. Internet Protocol (IP) Allows for the sending of high priority data. IP prepares a packet for transmission across the Internet. The IP header is encapsulated onto a transport data packet. The IP packet is then passed to the next layer where further network information is encapsulated onto it. IP addresses are represented by 32-bit unsigned binary values. Normally expressed in a dotted decimal format: 168.167.8.3 is a valid IP address. The numeric form is used by IP software. The mapping between numeric IP address and easy-to-read symbolic name (mopipi.ub.bw) is done by the Domain Name System (DNS) The Application Layer The purpose of the application layer is to allow two application programs on different hosts to work together. The Transport Layer The purpose of the transport layer is to allow two host computers to talk to one another even if they have very different internal designs, such as a PC and a workstation server. The Internet Layer The purpose of the internet layer is to route packets from the source host to the destination host across one or more networks connected by routers. TCP required the use of the Internet Protocol (IP) at the internet layer. The Network Interface Layer The purpose of this layer is to govern the movement of messages from a source station to a destination station or router across a single network containing switches. And to govern the transmission of bits one at a time over a wire, radio, or other connection between station and a switch, between pairs of switches, or between a switch and a router.


Below is a diagram of Internet protocol examples. The first column shows the TCP/IP layers. The other two columns indicate example protocol stacks that are commonly used in market.

Organization of the Internet A handful of network service provider (NSPs) (e.g. BT) maintain a series of nationwide links Links are like pipes- data flows through the pipes. NSPs are continually adding links with extra capacity to cater for increased Internet use Individually, we connect to the Internet via an ISP (Internet Service Provider) which in turn connects to the backbone. The setup below shows a typical Internet. Users (PCs or Terminals) connect to an (Internet Service Provider)ISP. The ISP in turn connects to the Network Service Provider (NSP).

Figure 9: Typical Set-up of the Internet Internet Service Providers (ISPs) Some are free although many charge a monthly fee Requirements Computer Modem


Phone line A normal phone line does not provide particularly fast access to the internet 56K bps

World Wide WebThis is a particular part of the internet which allows users to view information stored on participating computers Information is stored on pages which can be accessed directly, or via hypertext links

Who controls the Internet?Although there is no overall governing body to issue regulations and directives for the internet, The Internet Society (ISOC) serves as the standardizing body for the internet community. ISOC is organized and managed by the Internet Architecture Board (IAB). The IAB on the one hand relies on the Internet Engineering Task Force (IETF) for issuing new standards, and the Internet Assigned Numbers Authority (IANA) for coordinating values shared among multiple protocols. The Request For Comment (RFC) editor is responsible reviewing and publishing new standards documents. The IETF is itself governed by the Internet Engineering Steering Group (IESG), and it is further divided into areas and working groups where new specs are discussed and new standards proposed. The Internet Standards Process (in RFC 2026) is concerned with all protocols, procedures and conventions that are used by the Internet.

Standardization ProcessTo have new standard approved: Applicants submit the spec to IESG where it will de discussed and reviewed On positive conclusion by IESG: They issue a last call notification to allow spec to be reviewed by Internet community Final approval by IESG Internet draft is recommended to IETF for publication as RFCVoice Over IP (VoIP)

VoIP can simply be defined as the transmission of voice over IP networks. Originating and Terminating devices can be traditional telephones, fax machine and multimedia PCs, etc. Generally based on the following technology. VoIP gateways that provide enterprise-based dial


tone solutions (i.e., VoIP gateways seek to save toll charges by routing long distance calls over dedicated data lines between company offices). The following routes are possible with VoIP Computer to Computer Computer to Handset Handset to Handset

Figure 10: Handset-to-Handset IP Technology Above, Figure 10 shows a typical VoIP call using two handsets at either terminating endpoints. Below, Figure 11 shows a different VoIP scenario, where a call is between two computers at terminating ends.

Figure 11: Computer-to-Computer IP Technology


Voice QoS Problems with IP

IP was designated for carrying data, so it does not provide real time guarantees but only provides best effort service. For voice communications over IP to become acceptable to the users, the delay needs to be less than a threshold value. To ensure good quality of voice, we can use either Echo Cancellation, Packet Prioritization (giving higher priority to voice packets) or Forward Error Correction. Interoperability with PSTN In a public network environment, products from different vendors need to operate with each other if voice over IP is to become common among users. To achieve interoperability, standards are being devised and the most common standard for VoIP is the H.323 standard, or SIP (Session Initiation Protocol). SIP seems to be the latest fashionable protocol in VoIP. Security Security problems exist because in the Internet anyone can capture the packets meant for someone else. Some security can be provided by using encryption and tunneling. The common tunneling protocol used is Layer 2 Tunneling protocol and the common encryption mechanism used is Secured Sockets Layer (SSL). H.323 H.323 is the ITU-T standard that vendors may use to provide Voice over IP service. H.323 provides the technical requirements for voice communication over IP networks. It was originally developed for video teleconferencing on IP networks, from H.320 Video Telephony over Narrowband ISDN. The first version was released in 1996 while the second version of H.323 came into effect in January 1988. The standard encompasses both point to point communications and multipoint conferences. What is wrong with H.323 At the top of the list is call setup time. Since H.323 first establishes a session and only then negotiates the features and capabilities of the session, call setup can take significantly longer than an average PSTN call: H.323 doesnt scale well. A case in point is H.323 addressing. Creating separate phone-numbering schemes complicates interconnecting carrier networks. Critics also charge that the H.323 standard itself is too large and complex to make deployment easy. H.323 is built in a telecom manner SIP


Session Initiation Protocol (SIP) = a change from telephonys calls between handsets controlled by the network to sessions which can be between processes on any platform anywhere in the Internet and with both control and media content in form and hence can be easily manipulated. Thus a separate voice network is not necessary. Open and distributed nature enables lots of innovation (since both control and media can be manipulated and events are no longer restricted to start and end of calls). Advantages of SIP The intelligence is pushed to the network edge where processing capability is available in desktop computers. SIP allows multiparty calls to be setup using IP multicast capabilities. With SIP, one can fork calls, i.e. call two different extension from a single line. The extension that gets picked up first gets the call. This is useful if the receiver has two different offices. How SIP works SIP is a simple, ASCII-based protocol that uses requests and responses to establish communication among the various components in the network and to ultimately establish a conference between two or more end points. Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail address and is in the format of sip:[email protected]. The user ID can be either a user name or an E. 164 address. Users register with a registrar server using their assigned SIP address. The registrar server provides this information to the location server upon request. When a user initiates a call, a SIP request is sent to a SIP server. The request includes the address of the caller and the address of the intended callee Convergence ICT convergence involves the coming together of information distribution infrastructures; interactive information storage and processing capabilities; and widespread availability of consumer electronics products, publishing and IT content. One of the first practical examples of convergence was the coming together of certain technical elements of IT and telecommunications, which manifested itself in the digitization of telecommunications switching and the application of IT to telecommunications terminal equipment. The OSI Reference Model


Modern computer networks are designed in a highly structured way. To reduce their design complexity, most networks are organized as a series of layers, each one built upon its predecessor. The OSI Reference Model is based on a proposal developed by the International Organization for Standardization (ISO). The model is called ISO OSI (Open Systems Interconnection) Reference Model because it deals with connecting open systems - that is, systems that are open for communication with other systems. The OSI model has seven layers. The principles that were applied to arrive at the seven layers are as follows: 1. A layer should be created where a different level of abstraction is needed. 2. Each layer should perform a well defined function. 3. The function of each layer should be chosen with an eye toward defining internationally standardized protocols. 4. The layer boundaries should be chosen to minimize the information flow across the interfaces. 5. The number of layers should be large enough that distinct functions need not be thrown together in the same layer out of necessity, and small enough that the architecture does not become unwieldy. THE OPEN SYSTEMS INTERCONNECTION MODEL The International Standards Organization (ISO) has developed a universal architecture for computer communications. This standard, known as the Open Systems Interconnection Model, or OSI model, breaks down the task of communications into seven independent layers, each with its own tasks. OSIs purpose is to permit communications among devices made by many manufacturers. The exact methods for performing these tasks, including the protocols we discuss later in this Section, are still evolving. Almost all of the major host computer manufacturers have supported the concept of OSI in principle, even though their current product offerings may not all comply with OSI. The Corporation for Open Systems, or COS, is a non-profit corporation formed in 1985 consisting of representatives of major host computer manufacturers of that era, including Control Data, DEC, Hewlett-Packard, Honeywell, IBM, NCR, Tandem, Unisys, Wang, Xerox, and others. The corporations purpose is to facilitate the evolution of intervendor compatibility from a model to a reality. Perhaps the most significant contribution of the OSI model is that it provides all of us with a common language for describing communications tasks and functions. The seven layers of OSI are shown in Fig. 7-1. Each layer represents a particular function. Sometimes, each function is performed by a separate piece of hardware or software. Other times, a single program may perform the functions of several layers. All of the layers are


necessary for communications to occur. The different layer classifications are somewhat arbitrary, and a different standards committee might have chosen to break the communications functions into more or fewer layers. For example, we might describe the process of driving to work as (1) Open the car door. (2) Sit down. (3) Close the door. (4) Insert the key. (5) Turn the key, and so on. Another person might describe the same process as (1) Get in the car. (2) Start the car. (3) Put the car in gear, and so on. We are all describing the same task, and both descriptions are correct and accurate; however, each description chooses to break up the process of driving to work into different tasks. Similarly, the ISO-OSI model chooses to divide the function of computer communications into seven layers, though more or fewer layers could easily have been chosen. Rather than examine each layers functions in detail, we merely highlight its most important functions. The lowest layers, known as the Physical Layer, or Layer 1, are responsible for the transmission of bits. The Physical Layer is always implemented using hardware; this layer encompasses the mechanical, electrical, and functional interface. This layer is the interface to the outside world, where ones and zeroes leave and enter the device, usually using electronic signals as specified by interface standards. Examples of Physical Layer standards are RS-232-C, RS-449, RS-422-A, and RS-423-A. HOST COMPUTERApplication Layer (7) Presentation Layer (6) Session Layer Transport Layer Network Layer Data Link Layer Physical Layer (5) (4) (3) (2) (1)

Higher

layers

Lower layers

FIGURE 7-1 Layers of the Open Systems Interconnection Model

The Data Link Layer or Layer 2, assembles the data bits into a block, or frame, which is then sent to the Physical Layer for transmission. It is often also responsible for ensuring error-free, reliable transmission of data. The Data Link Layer typically scrutinizes the bits received to determine if errors occurred during transmission. This layer is often able to request retransmission or correction of any errors using protocols such as BSC, SDLC, HDLC, and PPP, presented later in this Section. The Network Layer, or Layer 3, is responsible for setting up the appropriate routing of messages throughout a network. This layer is the only layer concerned with the types of switching networks used to route the data. The routing of data between networks, and through packet switching networks, is also handled by the Network Layer. We discuss packet switching networks further in Section 8.


These layers of OSI (Physical, Data Link and Network) are usually referred to as the lower layers. Layers 4 through 7 (Transport, Session, Presentation, and Application) are usually referred to as the higher layers, or upper layers. The Transport Layer, or Layer 4, is responsible for isolating the function of the lower layers from the higher layers. This layer will accept messages from higher layers and break these messages down into messages that can be accepted by the lower layers. For example, a file being transferred may contain thousands of characters; the lower layers may be transmitting data 100 characters at a time, so the Transport Layer breaks the file into many blocks, each 100 character long. If communication technology changes and longer messages can be accepted in the future, the Transport Layer will need modification, but not either higher layers. The Transport Layer is also responsible for monitoring the quality of the communications channel and for selecting the most cost-efficient communication service based on the reliability required for a particular transmission. The Session Layer, or Layer 5, request that a logical connection be established based on the end users request. In this case, an end user might be the terminal operator using the computer. For example, if the user wants to transfer a file, the Session Layer is informed of the location of the file on the users system and the location of the destination file on the remote host computer. Any necessary log-on and password procedures are also usually handled by this layer. The Session Layer is also responsible for terminating the connection. The Presentation Layer, or Layer 6, provides format and code conversion services. For example, if the host computer is connected to many different types of printers, each printer may require different character sequences to invoke special features, such as boldface and italics. The Presentation Layer handles all of necessary formatting. In addition, if files are being transferred from the host computer of one manufacturer to the host computer of another, there may be different file formats, or even different character codes. The Presentation Layer would handle any necessary conversion (e.g., ASCII-to-EBCDIC conversion). The Application Layer, or Layer 7, provides access to the network for the end user. The users capabilities on the network are determined by the Application Layer software, which can be tailored to the needs of the user. Some Application Layer software might permit remote terminals only to access a host computer; other Application Layer software might also permit file transfers. Network management statistics, diagnostics, and other on-line monitoring capabilities can also be implemented in this layer. We have already mentioned that the Physical Layer must be implemented in hardware. Since this layer is the only part of the model where bits are actually transmitted, it is also the only part of the model requiring hardware implementation. The other layers all manipulate the data in some way, perhaps adding to it or modifying it, but all of these techniques can generally be performed using software. However, since functions can be performed more efficiently and inexpensively by hardware than by software, some functions of the Data Link and Network


Layers are sometimes implemented in hardware. The higher layers are almost always implemented in software.

Lesson II: Serial Networks & Protocols DTE and DCEThe terms DTE and DCE are very common in the data communication market. DTE is short for Data Terminal Equipment and DCE stands for Data Communications Equipments. But what do they really mean? As the full DTE name indicates this is a piece of device that ends a communication line, whereas the DCE provides a path for communication. Lets say we have a computer on which wants to communicate with the Internet through a modern and a dial-up connection. To get to the Internet you tell you modern to dial the number of your provider. After your modems has dialed the number, the modem of the provider will answer your call and your will hear a lot of noise. Then it becomes quiet and you see your login prompt or your dialing program tells you the connection is established. Now you have a connection with the server from your provider and you can wander the Internet. In this example you PC is a Data Terminal (DTE). The two modems (yours and that one of your provider) are DCEs, they make the communication between you and to provider possible. But now we have to look at the server of your provider. Is that a DTE or DCE? The answer is a DTE. It ends the communication line between you and the server. Although it gives you the possibility to surf around the glode. The reason why it is a DTE is that when you want to go from your provides server to another place it uses another interface. So DTE and DCE are interface dependent. It is e.g. possible that for your connection to the serve, the server is a DTE, but that same server is a DCE for the equipment that it is attached to on the rest of the Net. (Data Terminating Equipment) A communications device that is the source or destination of signals on a network. It is typically a terminal or computer. Contrast with DCE.


(Data Communications Equipment or Data Circuit-terminating Equipment) A device that establishes maintains and terminates a session on a network. It may also convert signals for transmission. It is typically the modem.

Data Rates A data transfer rate (or often just data rate) is the amount of digital data that is moved from one place to another in a given time, usually in a second's time. The data transfer rate can be viewed as the speed of travel of a given amount of data from one place to another. In general, the greater the bandwidth of a given path, the higher the data transfer rate. In telecommunications, data transfer is usually measured in bits per second. For example, a typical low-speed connection to the Internet may be 33.6 kilobits per second (Kbps). On Ethernet local area networks, data transfer can be as fast as 10 megabits per second. Network switches are planned that will transfer data in the terabit range. In earlier telecommunication systems, data transfer was sometimes measured in characters or blocks (of a certain size) per second. Data transfer time between the microprocessor or RAM and devices such as the hard disk and CD-ROM player is usually measured in milliseconds.


In computers, data transfer is often measured in bytes per second. The highest data transfer rate to date is 14 terabits per second over a single optical fiber, reported by Japan's Nippon Telegraph and Telephone (NTT DoComo) in 2006. (Or "data Transfer rate", "transmission rate") The amount of data transferred per second by a communications channel or a computing or storage device. Data rate is measured in units of bits per second (written "b/s" or "bps"), bytes per second (Bps), or baud. When applied to data rate, the multiplier prefixes "kilo-", "mega-", "giga-", etc. (and their abbreviations, "k", "M", "G", etc.) always denote powers of 1000. For example, 64 kbps is 64,000 bits per second. This contrasts with units of storage where they stand for powers of 1024, e.g. 1 KB = 1024 bytes. Flow Control In communications, the process of adjusting the flow of data from one device to another to ensure that the receiving device can handle all of the incoming data. This is particularly important where the sending device is capable of sending data much faster than the receiving device can receive it. There are many flow control mechanisms. One of the most common flow control protocols for asynchronous communication is called xon-xoff. In this case, the receiving device sends a an xoff message to the sending device when its buffer is full. The sending device then stops sending data. When the receiving device is ready to receive more data, it sends an xon signal. Flow control can be implemented in hardware or software, or a combination of both. TCP manages limited network bandwidth by performing flow control. Modern data networks are designed to support a diverse range of hosts and communication mediums. Consider a 200MHz Pentium-based host transmitting data to a 25MHz 80386/SX. Obviously, the Pentium will be able to drown the slower processor with data. Likewise, consider two hosts, each using an Ethernet LAN, but with the two Ethernets connected by a 28.8 Kbps modem link. If one host begins transmitting to the other at Ethernet speeds, the modem link will quickly become overwhelmed. In both cases, flow control is needed to pace the data transfer at an acceptable speed. Request/reply flow control requires each data packet to be acknowledge by the remote host before the next packet is sent. Sliding window algorithms, used by TCP, permit multiple data packets to be in simultaneous transit, making more efficient use of network bandwidth. Finally, Internet's Unreliable Delivery Model allows packets to be discarded if network resources are not available, and demands that protocols make provisions for retransmission. The collection of techniques used in serial communications to stop the sender sending data until the receiver can accept it. This may be either {software flow control} or {hardware flow control}. The receiver typically has a fixed size {buffer} into which received data is written as soon as it is received. When the amount of buffered data exceeds a "high water mark", the


receiver will signal to the transmitter to stop transmitting until the process reading the data has read sufficient data from the buffer that it has reached its "low water mark", at which point the receiver signals to the transmitter to resume transmission. (1995-03-22) Synchronous Communication Adapters for use on HP Alpha Systems Models 4-port Intelligent Synchronous Communications Adapter 2-port Intelligent Synchronous Communications Adapter 3X-PBXDD-AB 3X-PBXDD-AA

Introduction The Digi DataFire SYNC 2000 adapters available from HP provide remote WAN and SNA connectivity for PCI servers, which make them ideal for branch offices of central sites. X.25 is a proven packet switched technology that has been around for many years. X.25 provides 100 percent error correction and network-managed flow control. It guarantees that every packet will arrive at its destination without any errors. This is a slow, deliberate process that involves a great deal of overhead and is widely used internationally where leased lines are not readily available. High speed Synchronous WAN Communications Subscribers pay a variable rate based on connect time and packets transmitted. Frame Relay is designed as the successor to X.25 for transmitting data over the phone network. It is also a packet switching protocol, but is provides no guarantee of data integrity. Frame Relay links have more in common with dedicated lines than switched lines, but the cost can be substantially lower for an equivalent capacity, as subscribers pay a variable rate based on bandwidth and the committed information rate. Intelligent Synchronous Adapters The Digi DataFire SYNC 2000 is a family of intelligent synchronous communication adapters that provide advanced server-based Wide Area Network (WAN) solutions. Available in two-and four-port models. The DataFire SYNC 2000 2P and 4P models are mid-level, intelligent WAN adapters based on the Motorola MPC860 PowerQUICC processor running at 25 MHz and 40 MHz, respectively. All DataFire SYNC 2000 adapters run Frame Relay FRF.9 compression to boost throughout. All DataFire SYNC 2000 models work with PCI-based servers running at either 3.3or 5-volts. Four MB of on-board RAM supports T1/E1 speeds on all ports in full-duplex mode. Cables are available for the common interfaces-V.24 EIA-530, V.35, V.36, V11 and EIA-449. Each port uses an optional independent cable, allowing any combination of electrical interfaces to be used. The cable is automatically configured when plugged into the board, eliminating troublesome configuration options. Each port can report the status of all compliant modem signals and attached cables. Each interface also can measure and report the speed of modems and CSU/DSUs for faster troubleshooting. Async


Asynchrony is the state of not being synchronized. Contrast with plesiochronous systems. In terms of digital logic and data transfer, an asynchronous object does not require a clock signal. Examples: asynchronous circuit asynchronous communication Asynchronous Transfer Mode asynchronous serial interfaces packet switched systems such as Ethernet or internet protocol asynchronous computer APIs Collaborative editing systems Asynchronous Cellular Automaton Telecommunications - Asynchronous (stop/start) data transmission This is an extension to telegraph methods used by computer terminals from the model 33 teletype to VDUs/VDTs (Video Display Units/Terminals). When serial data is transmitted, timing information must be sent to allow the information to be correctly decoded at the distant end. Bit synchronization information is required to allow the receiver to sample each bit at the correct time. Character synchronization allows the receiver to divide the data stream into characters, ie to know where each character starts and stops. In asynchronous operation both bit and character synchronization are provided by the start and stop bits, when nothing is being transmitted a continuous mark (logic 1) is being sent to line, when a character is sent the start bit causes a 1 -> 0 transition, 1.5 bit lengths after that will be the middle of the first bit, each bit is then sampled in turn until the stop bit which is always 1 to ensure a 1 -> 0 transition at the start of the next character. Therefore no additional timing signals need to be provided by the modem but the terminal must know what speed is being transmitted to sample at the correct rate. Users have to sort out, baud rates, parity, number of stop bits/ data bits and any handshaking. this is how a CR is sent to line ---+ +---+ +-------+ +--------------1=mark ||||||||||| +---+ +---+ +-----------+ 0=space s1234567ps


tat almro rssip tddt y Stop/start is used when connecting to the Public Network pad. line rate 110 b/s line rate > 110 b/s number of stop bits PAD min 2 min 1 DTE-C min 1 min 1 coding of parity bit optional in all transmissions from the DTE-C, however in all user data transmitted or received by the DTE-C the coding of all 8 bits (7 character bits plus parity bit) will be passed transparently between the DTE-C and DTE-P. All characters generated by the PAD (eg PAD service signals) will be transmitted with even parity. Sync (Synchronization) Synchronization (or Sync) is a problem in timekeeping which requires the coordination of events to operate a system in unison. The familiar conductor of an orchestra serves to keep the orchestra in time. Systems operating with all their parts in synchrony are said to be synchronous or in sync. Some systems may be only approximately synchronized, or plesiochronous. For some applications relative offsets between events need to be determined, for others only the order of the event is important. Today, synchronization can occur on a global basis due to GPS-enabled timekeeping systems. Transport Apart from its use for navigation (see John Harrison), synchronization was not important in transportation until the nineteenth century, when the coming of the railways made travel fast enough for the differences in local time between adjacent towns to be noticeable. In some territories, sharing of single railroad tracks was controlled by the timetable. Thus strict timekeeping was a safety requirement. To this day, railroads can communicate and signal along their tracks, independently of other systems for safety. Communication The lessons of timekeeping are part of engineering technology. In electrical engineering terms, for digital logic and data transfer, a synchronous object requires a clock signal.


Timekeeping technologies such as the GPS satellites and Network time protocol (NTP) provide real-time access to a close approximation to the UTC timescale, and are used for many terrestrial synchronization applications. Synchronization is an important concept in the following fields: Computer science "In computer science, especially parallel computing, synchronization means the coordination of simultaneous threads or processes to complete a task in order to get correct runtime order and avoid unexpected race conditions." Telecommunication Physics The idea of simultaneity has many difficulties, both in practice and theory. Cryptography Multimedia Photography Music (rhythm) Synthesizers Synchronization has several subtly distinct sub-concepts: Rate synchronization Phase synchronization Time offset synchronization Time order synchronization Some uses of synchronization Whilst well-designed time synchronization is an important tool for creating reliable systems, excessive use of synchronization where it is not necessary can make systems less fault-tolerant, and hence less reliable. Film synchronization of image and sound in sound film. Synchronization is important in fields such as digital telephony, video and digital audio where streams of sampled data are manipulated. Arbiters are needed in digital electronic systems such as microprocessors to deal with asynchronous inputs. There are also electronic digital circuits called synchronizers that attempt to perform arbitration in one clock cycle. Synchronizers, unlike arbiters, are prone to failure. (See metastability in electronics).


Encryption systems usually require some synchronization mechanism to ensure that the receiving cipher is decoding the right bits at the right time. Automotive transmissions contain synchronizers which allow the toothed rotating parts (gears and splined shaft) to be brought to the same rotational velocity before engaging the teeth. Synchronization is also important in industrial automation applications. Time codes are often used as a means of synchronization in film, video, and audio applications. Flash photography, see Flash synchronization File synchronization is used to maintain the same version of files on multiple computing devices. For example, an address book on a telephone might need to by synchronized with an address book on a computer. Software applications must occasionally incorporate application-specific data synchronization in order to mirror changes over time among multiple data sources at a level more granular than File synchronization. An example use of this is the Data Synchronization specification of the Open Mobile Alliance, which continues the work previously done by the SyncML initiative. SyncML was initially proposed to synchronize changes in personal address book and calendar information from computers to mobile phones, but has subsequently been used in applications that synchronize other types of data changes among multiple sources, such as project status changes. The term synchronization is also sometimes used for the transfer of content from a computer to an MP3 player connected to it. High-Level Data Link Control High-Level Data Link Control (HDLC) is a bit-oriented synchronous data link layer protocol developed by the International Organization for Standardization (ISO). The original ISO standards for HDLC were: ISO 0009 Frame Structure ISO 4335 Elements of Procedure ISO 6159 Unbalanced Classes of Procedure ISO 6256 Balanced Classes of Procedure The current standard for HDLC is ISO 13239, which replaces all of those standards. HDLC provides both connection oriented and connectionless service.


HDLC can be used for point to multipoint connections, but is now used almost exclusively to connect one device to another, using what is known as Asynchronous Balanced Mode (ABM). The other modes are Normal Response Mode and Asynchronous Response Mode. Framing HDLC frames can be transmitted over synchronous or asynchronous links. Those links have no mechanism to mark the beginning or end of a frame, so the beginning and end of each frame has to be identified. This is done by using a frame delimiter, or flag, which is a unique sequence of bits that is guaranteed not to be seen inside a frame. This sequence is '01111110', or, in hexadecimal notation, 7E. Each frame begins and ends with a frame delimiter. When no frames are being transmitted on a synchronous link, a frame delimiter is continuously transmitted on the link. Using the standard NRZI encoding from bits to line levels (0 bit = transition, 1 bit = no transition), this generates a continuous bit pattern: 01111110011111100111111001111110 _____________ _____________ _____________ _____________ _/ \_/ \_/ \_/ \ This is used by modems to train and synchronize their clocks via phase-locked loops. Actual binary data could easily have a sequence of bits that is the same as the flag sequence. So the data's bit sequence must be transmitted so that it doesn't appear to be a frame delimiter. On synchronous links, this is done with bit stuffing. The sending device ensures that any sequence of 5 contiguous 1-bits is automatically followed by a 0-bit. A simple digital circuit inserts a 0-bit after 5 1-bits. The receiving device knows this is being done, and will automatically strip out the extra 0-bits. So if a flag is received, it will have 6 contiguous 1-bits. The receiving device see 6 1-bits and knows it is a flag otherwise the 6th bit would have been a 0-bit. This also (again, assuming NRZI encoding of the output) provides a minimum of one transition per 6 bit times, so the receiver can stay in sync with the transmitter. Asynchronous links using serial ports or UARTs just send bits in groups of 8. They lack the special bit-stuffing digital circuits. Instead they use "control-octet transparency", also called "byte stuffing" or "octet stuffing". The frame boundary octet is 01111110, (7E in hexadecimal notation). A "control escape octet", has the bit sequence '01111101', (7D hexadecimal). The escape octet is sent before a data byte with the same value as either an escape or frame octet. Then, the following data has bit 5 (counting from right to left and starting at zero) inverted. For example, the data sequence "01111110" (7E hex) would be transmitted as "01111101 01011110" ("7D 5E" hex). Any octet value can be escaped in the same fashion. Structure The contents of an HDLC frame, including the flag, are


Flag

Add r es Control s

Information

FCS

(Optional Flag)

8 bits

8 bits

8 or bits

16 Variable length, multiples of 8

0

or

more

bits,

in 16 or bits

32

8 bits

Note that the end flag of one frame can be (but does not have to be) the beginning (start) flag of the next frame. Note that the data comes in groups of 8 bits. The telephone and teletype systems arranged most long-haul digital transmission media to send bits eight at a time, and HDLC simply adapts that standard to send bulk binary data. Voice is encoded by A-law or u-law into 8-bit samples. Teletypes send 8-bit codes to represent each character. The FCS is the Frame Check Sequence, and is a more sophisticated version of the parity bit. The field contains the result of a binary calculation that uses the bit sequences that make up the 'Address', 'Control' and 'Information' fields. The calculation is designed to detect errors in the transmission of the frame lost bits, flipped bits, extraneous bits so that the frame can be dropped by the receiver if an error is detected. It is this method of detecting errors that can set an upper bound on the size of the data portion of the frame. Essentially, the longer the length of the data portion of the frame becomes, the harder it is to guarantee that certain types of transmission errors will be found. There are multiple types of Frame Check Sequence, and the most commonly used in this context will be CRC-16 or CRC-CCITT. The FCS is needed to detect transmission errors. When HDLC was designed, long-haul digital media were designed for telephone systems, which only need a bit error rate of 1105 errors per bit. Digital data for computers normally requires a bit error rate better than 11012 errors per bit. By checking the FCS, the receiver can discover bad data. If the data is ok, it sends an "acknowledge" packet back to the sender. The sender can then send the next frame. If the receiver sends a "negative acknowledge" or simply drops the bad frame, the sender either receives the negative acknowledge, or runs into its time limit while waiting for the acknowledge. It then retransmits the failed frame. Modern optical networks have reliability substantially better than 1105 errors per bit, but that simply makes HDLC even more reliable. Types of Stations (Computers), and Data Transfer Modes Primary terminal is responsible for operation control over the link. It issues the frames which are called commands. Secondary terminal operates under the control of the primary. Frames issues, are responses only. Primary is linked with secondaries by multiple logical links.


Combined terminal, has the features of both primary and secondary terminals. It issues both commands and responses. HDLC Operations, and Frame Types I-Frames (user data) Contain user data, sequence number of the transmitted frame, piggybacking acknowledgment number of received I-Frame. Their maximum window size is 7 or 127. I-Frames also contain poll/final (P/F) bit. Depending on response mode, In NRM the primary terminal sets the P-bit to poll. The secondary sets the F-bit in last I-frame to a response. IN ARM and ABM, the P/F bits are used to force response. S-Frames (control) Used both for flow and error control. Receive Ready (RR) used as positive acknowledgement (thruN(r)-1) and a request that no more I-frames be sent until a subsequent RR is in use. Primary terminal can issue a POLL by P-bit setting Secondary terminal responds with F-bit set, if it has no data to send. Receive Not Ready (RNR) Used as positive ACK and a request that no more i-frames should be sent till the subsequent RR is received. Either Primary or Combined station can set P-bit to solicit the receive status of a secondary/combined station. Secondary/Combined station response to Poll with F-bit set if the station is busy. Reject (REJ) Uses Go-Back-N technique (Retransmit from N(r)) Selective Reject Uses Selective Repeat Technique ((Repeat N(r))


U-Frames Mode settings (SNRM, SNRME, SARM, SARME, SABM, SABME, UA, DM, RIM, SIM, RD, DISC) Information Transfer(UP, UI) Recovery (FRMR, RSET) Invalid Control Field Data Field Too Long Data field not allowed with received Frame Type Invalid Receive Count Miscellaneous (XID, TEST) Link Configurations Link configurations can be categorized as being either: Unbalanced, which consists of one primary terminal, and one or more secondary terminals. Balanced, which consists of two peer terminals.

HDLC Data Transfer Modes illustrated The three link configurations are: Normal Response Mode (NRM) is an unbalanced configuration in which only the primary terminal may initiate data transfer. The secondary terminal transmits data only in response to commands from the primary terminal. The primary terminal polls the secondary terminal(s) to determine whether they have data to transmit, and then selects one to transmit. Asynchronous Response Mode (ARM) is an unbalanced configuration in which secondary terminals may transmit without permission from the primary terminal. However, the primary terminal still retains responsibility for line initialization, error recovery, and logical disconnect. Asynchronous Balanced Mode (ABM) is a balanced configuration in which either station may initiate the transmission.


HDLC Command and response repertoire Commands (I, RR, RNR, (SNRM or SARM or SABM) DISC Responses (I, RR, RNR, UA, DM, FRMR) Basic Operations Initialization can be requested by either side. When the six-mode set-command is issued. This command: Signals the other side that initialization is requested Specifies the mode, NRM, ABM, ARM Specifies whether 3 or 7 bit sequence numbers are in use. The HDLC module on the other end transmits (UA) frame when the request is accepted. And if the request is rejected it sends (DM) disconnect mode frame. Functional Extensions (Options) For Switched Circuits Commands: ADD - XID Responses: ADD - XID, RD For 2-way Simultaneous commands & responses are ADD - REJ For Single Frame Retransmission commands & responses: ADD - SREJ For Information Commands & Responses: ADD - Ul For Initialization Commands: ADD - SIM Responses: ADD - RIM For Group Polling Commands: ADD - UP Extended Addressing Delete Response I Frames Delete Command I Frames


Extended Numbering For Mode Reset (ABM only) Commands are: ADD - RSET Data Link Test Commands & Responses are: ADD - TEST Request Disconnect. Responses are ADD - RD 32-bit FCS HDLC Command/Response Repertoire Command / Response Information(I) Supervisory (S) Receive Ready (RR) Receive Not Ready (RNR)Reject (REJ) Selective Reject (SREJ) Unnumbered Frames Command / Response Set normal response SNRM Set normal response extended mode SNRME C C Set mode; extended Set mode; extended = 7 bit sequence number = 7 bit sequence number ..1...0...0...P...1...1...0...1 ..1...1...0...P...1...1...1...1 C-Field Format 8...7...6...5...4...3...2...1.....

Type Of Frame

C-Field Format Description Info 8...7...6...5...4...3...2...1.... .

Name

C/R C/R

User exchange data

.-N(R)-... P/F.....-N(S)-..0

Ready to Positive receive .-N(R)-... P/F...0...0...0...1 Acknowledgement I-Frame Not Positive Ready to .-N(R)-... P/F...0...1...0...1 Acknowledgement receiveNegative Acknowledgement Negative Acknowledgement go back N .-N(R)-... P/F...1...0...1...0 selective reject .-N(R)-... P/F...1...1...0...1

C/R

C/R C/R

Name

Description

Info


Set asynchronous response SARM Set asynchronous response extended mode SARME Set asynchronous balanced/extended mode SABM Set asynchronous balanced extended mode SABME Set initialization mode SIM Disconnect DISC Unnumbered Acknowledgement UA Disconnect Mode (DM) Requested Disconnect (RD) Request Initialization Mode (RIM) Unnumbered Information (UI) Unnumbered Poll (UP) Reset (RSET)

C C

Set mode; extended Set mode; extended Set mode; extended Set mode; extended Initialize link control function Terminate logical link connection Acknowledge acceptance Responder in Disconnect Mode Responder for Disc Command Initialization needed Used to exchange Used to solicit

= 7 bit sequence number = 7 bit sequence number = 7 bit sequence number = 7 bit sequence number in the addressed station

..0...0...0..P/F..1...1...0...1 ..0...1...0...P..1...1...1...1

C

..0...0...1..P/F..1...1...1...1

C C C R R R R C/R C C

..0...1...1...P...1...1...1...1 ..0...0...0..P/F..0...1...1...1 ..0...1...0..P/F..0...0...1...1

of one of hte set-mode commands.

..0...1...0....F..0...0...1...1

..0...1...0..P/F..0...0...1...1 Request for SIM command control information control information ..0...0...0..P/F..0...0...1...1 ..0...0...1..P....0...0...1...1

Used for recovery Resets N(R), N(S) ..1...0...0..P....1...1...1...1 Used to Request/ Report Status Exchange identical information Report receipt fields for testing of unacceptable frame ..1...0...1..P/F..1...1...1...1 ..1...1...1..P/F..0...0...1...1

Exchange Indication (XID) C/R Test (TEST) Frame Reject FRMR C/R C/R

SDLC: Synchronous Data Link Control by IBM


The Synchronous Data Link Control (SDLC) protocol, an IBM data link layer protocol for use in the Systems Network Architecture (SNA) environment. The data link control Layer provides the error-free movement of data between the Network Addressable Units (NAUs) within a given communication network via the Synchronous Data Link Control (SDLC) Protocol. The flow of information passes down from the higher layers through the data link control Layer and is passed into the physical control Layer. It then passes into the communication links through some type of interface. SDLC supports a variety of link types and topologies. It can be used with point-to-point and multipoint links, bounded and unbounded media, half-duplex and full-duplex transmission facilities, and circuit-switched and packet-switched networks. SDLC identifies two types of network nodes: primary and secondary. Primary nodes control the operation of other stations, called secondaries. The primary polls the secondaries in a predetermined order, and secondaries can then transmit if they have outgoing data. The primary also sets up and tears down links and manages the link while it is operational. Secondary nodes are controlled by a primary, which means that secondaries can send information to the primary only if the primary grants permission. SDLC primaries and secondaries can be connected in four basic configurations: Point-to-point- Involves only two nodes, one primary and one secondary. Multipoint- Involves one primary and multiple secondaries. Loop- Involves a loop topology, with the primary connected to the first and last secondaries. Intermediate secondaries pass messages through one another as they respond to the requests of the primary. Hub go-ahead- Involves an inbound and an outbound channel. The primary uses the outbound channel to communicate with the secondaries. The secondaries use the inbound channel to communicate with the primary. The inbound channel is daisy-chained back to the primary through each secondary. SDLC has a few derivatives which are adopted in different environment: HDLC, an ISO protocol for x.25 network LAPB, an ITU-T protocol used in the ISDN network LAPF, an ITU-T protocol used in the Frame Relay network IEEE 802.2, often referred to as LLC and has three types, used in the local area network QLLC, used to transport SNA data across X.25 networks Protocol Structure - SDLC: Synchronous Data Link Control by IBM


1 byte Flag

1-2 bytes Address field

1-2 bytes Control field

variable Data

2 byte FCS

1 byte Flag

Flag- Initiates and terminates error checking. Address- Contains the SDLC address of the secondary station, which indicates whether the frame comes from the primary or secondary. Control- Employs three different formats, depending on the type of SDLC frame used: Information (I) frame- Carries upper-layer information and some control information. Supervisory (S) frame- Provides control information. An S frame can request and suspend transmission, report on status, and acknowledge receipt of I frames. S frames do not have an information field. Unnumbered (U) frame- Supports control purposes and is not sequenced. A U frame can be used to initialize secondaries. Depending on the function of the U frame, its control field is 1 or 2 bytes. Some U frames have an information field. Data- Contains a path information unit (PIU) or exchange identification (XID) information. Frame check sequence (FCS)- Precedes the ending flag delimiter and is usually a cyclic redundancy check (CRC) calculation remainder. LAPB: Link Access Procedure Balanced Link Access Procedure, Balanced (LAPB) is a data link layer protocol used to manage communication and packet framing between data terminal equipment (DTE) and the data circuit-terminating equipment (DCE) devices in the X.25 protocol stack. LAPB, a bit-oriented protocol derived from HDLC, is actually the HDLC in BAC mode (Balanced Asynchronous Class). LAPB makes sure that frames are error free and properly sequenced. LAPB shares the same frame format, frame types, and field functions as SDLC and HDLC. Unlike either of these, however, LAPB is restricted to the Asynchronous Balanced Mode (ABM) transfer mode and is appropriate only for combined stations. Also, LAPB circuits can be established by either the DTE or DCE. The station initiating the call is determined to be the primary, and the responding station is the secondary. Finally, LAPB use of the P/F bit is somewhat different from that of the other protocols. In LAPB, since there is no master/slave relationship, the sender uses the Poll bit to insist on an immediate response. In the response frame this same bit becomes the receivers Final bit. The receiver always turns on the Final bit in its response to a command from the sender with the Poll bit set. The P/F bit is generally used when either end becomes unsure about proper frame


sequencing because of a possible missing acknowledgement, and it is necessary to re-establish a point of reference. LAPB's Frame Types: I-Frames (Information frames): Carries upper-layer information and some control information. I-frame functions include sequencing, flow control, and error detection and recovery. I-frames carry send and receive sequence numbers. S-Frames (Supervisory Frames): Carries control information. S-frame functions include requesting and suspending transmissions, reporting on status, and acknowledging the receipt of I-frames. S-frames carry only receive sequence numbers. U-Frames (Unnumbered Frames): carries control information. U-frame functions include link setup and disconnection, as well as error reporting. U-frames carry no sequence numbers Protocol Structure - LAPB: Link Access Procedure Balanced The format of LAPB frame is as follows:

1 byte Flag

1 byte Address field

1-2 bytes Control field

Variable Data/Information

2 bytes FCS

1 byte Flag

Flag - The value of the flag is always (0x7E). In order to ensure that the bit pattern of the frame delimiter flag does not appear in the data field of the frame (and therefore cause frame misalignment), a technique known as Bit Stuffing is used by both the transmitter and the receiver. Address field - In LAPB, the address field has no meaning since the protocol works in a point to point mode and the DTE network address is represented in the layer 3 packets. Control field - it serves to identify the type of the frame. In addition, it includes sequence numbers, control features and error tracking according to the frame type. Modes of operation - LAPB works in the Asynchronous Balanced Mode (ABM). This mode is totally balanced (i.e., no master/slave relationship) and is signified by the SABM(E) frame. Each station may initialize, supervise, recover from errors, and send frames at any time. The DTE and DCE are treated as equals. FCS - The Frame Check Sequence enables a high level of physical error control by allowing the integrity of the transmitted frame data to be checked.


Window size - LAPB supports an extended window size (modulo 128) where the number of possible outstanding frames for acknowledgement is raised from 8 to 128. LAPD The LAPD (Link Access Protocol - Channel D) is a layer 2 protocol which is defined in CCITT Q.920/921. LAPD works in the Asynchronous Balanced Mode (ABM). This mode is totally balanced (i.e., no master/slave relationship). Each station may initialize, supervise, recover from errors, and send frames at any time. The protocol treats the DTE and DCE as equals. The format of a standard LAPD frame is as follows: Fla g Address field Control field Information FCS LAPD frame structure F l a g The value of the flag is always (0x7E). In order to ensure that the bit pattern of the frame delimiter flag does not appear in the data field of the frame (and therefore cause frame misalignment), a technique known as Bit Stuffing is used by both the transmitter and the receiver. Address field The first two bytes of the frame after the header flag is known as the address field. The format of the address field is as follows: 8 SAPI TEI 7 6 5 4 3 2 C/R 1 EA1 EA2 Flag

LAPD address field EA1 First Address Extension bit which is always set to 0. C/R Command/Response bit. Frames from the user with this bit set to 0 are command frames, as are frames from the network with this bit set to 1. Other values indicate a response frame. EA2 Second Address Extension bit which is always set to 1. TEI Terminal Endpoint Identifier. Valid values are as follows: Used by non-automatic TEI assignment user 0-63 equipment. 64-126 Used by automatic TEI assignment equipment. 127 Used for a broadcast connection meant for all Terminal Endpoints.

Control field The field following the Address Field is called the Control Field and serves to identify the type of


the frame. In addition, it includes sequence numbers, control features and error tracking according to the frame type. F C S The Frame Check Sequence (FCS) enables a high level of physical error control by allowing the integrity of the transmitted frame data to be checked. The sequence is first calculated by the transmitter using an algorithm based on the values of all the bits in the frame. The receiver then performs the same calculation on the received frame and compares its value to the CRC. Window size LAPD supports an extended window size (modulo 128) where the number of possible outstanding frames for acknowledgement is raised from 8 to 128. This extension is generally used for satellite transmissions where the acknowledgement delay is significantly greater than the frame transmission times. The type of the link initialization frame determines the modulo of the session and an "E" is added to the basic frame type name (e.g., SABM becomes SABME). Frame The following are the Supervisory Frame Types in LAPD: RR REJ RNR Information frame acknowledgement and indication to receive more. Request for retransmission of all frames after a given sequence number. Indicates a state of temporary occupation of station (e.g., window full). types

The following are the Unnumbered Frame Types in LAPD: DISC UA DM FRMR SABM Request disconnection Acknowledgement frame. Response to DISC indicating disconnected mode. Frame reject. Initiator for asynchronous balanced mode. master/slave relationship. SABME SABM in extended mode. UI Unnumbered Information. XID Exchange Information. There is one Information Frame Type in LAPD: Info Information transfer frame.

No


LAPM Link Access Procedure for Modems, LAPM is an error control protocol defined in ITU-T recommendations V.42. Like the MNP protocols, LAPM uses cyclic redundancy checking (CRC) and retransmission of corrupted data (ARQ) to ensure data reliability. Lesson III: Analogue Networks, Modems and Multiplexers PSTN and Leased line (2 and 4 wire) PSTN (public switched telephone network) is the world's collection of interconnected voice-oriented public telephone networks, both commercial and government-owned. It's also referred to as the Plain Old Telephone Service (POTS). It's the aggregation of circuit-switching telephone networks that has evolved from the days of Alexander Graham Bell ("Doctor Watson, come here!"). Today, it is almost entirely digital in technology except for the final link from the central (local) telephone office to the user. In relation to the Internet, the PSTN actually furnishes much of the Internet's long-distance infrastructure. Because Internet service providers ISPs pay the long-distance providers for access to their infrastructure and share the circuits among many users through packet-switching, Internet users avoid having to pay usage tolls to anyone other than their ISPs. Analog Modems Analog modems use the existing telephone infrastructure to link sites together. The telephone cabling supports analogue frequencies in the range 300Hz to 3400KHz, and is primarily designed for speech. The available bandwidth of the speech circuits provided by telecommunication companies imposes limits on the available speed in bits per second that can be transmitted. The modems implement a dial-up connection. A connection is made between the two modems by dialing the number assigned to the other modem, using the existing dial up telephone network. Generally, connections are established for limited duration's. This suits remote access users who might want to dial into their network after hours, or small offices which dial into their


internet service provider at regular intervals during the day to exchange (upload and download) e m a i l . Current Modem standards are Standard Speed in bps V.21 V.22 V.22bis V.32 V.32bis V.FC V.34 V.34+ 300 1200 2400 9600 14400 19200 28800 33600

The speeds stated above are maximum speeds, and often, modems fail to achieve this. Errors caused by noise on the telecommunication lines often cause modems to fall back to a much lower speed, in order to reduce the number of errors. Thus a high speed modem rated at 33600bps often achieves a throughput of 9600bps due to the existing phone lines being too error prone to support the higher rate. Another problem that occurs is with modems that utilize compression techniques. Often, compression is measured on the transmission of uncompressed files like text files. When these same compression modems are asked to deal with the transfer of compressed files like .ZIP files, they do not perform well, and effectively either transfer at a much reduced rate or no compression at all. Some typical compression type modems are MNP4 and MNP5. In addition, modems utilizing the different compression schemes often fail to communicate properly with compression enabled. This is due to variances in manufacturers implementations of compression algorithms. Advantages Widely available Low Cost Most reliably Disadvantages Low speed Error Prone Common Usage Remote access Low bandwidth requirements like email changing Roving users

interoperate Technology rapidly


Portable

Dedicated Lines (Leased Line) Dedicated lines are fixed connections which do not involve dialing. They are permanent end to end connections. The telecommunications company provides a dedicated high speed connection between the two desired locations, at speeds ranging from as low as 9600bps to as high as 45Mbps. The higher the speed, the greater the cost, which is usually a fixed monthly rental charge (does not include data charges, only rental charges). The connection is available 24 hours a day, seven days a week, and is thus suited to companies who want permanent connections between their office branches, or perhaps to a company who wants a permanent connection to the Internet (they are providing a WWW server for people to access). The basic unit of measurement for dedicated lines is a T1 connection, which supports 1.544Mbps. A T3 connection supports 45Mbps. Fractional T1 circuits are available in units of 64Kbps, with connections of 384Kbps, 512Kbps and 768Kbps being common. The connection is implemented with two units Channel Service This provides the interface to the dedicated line Unit (CSU)

Data Service Unit (DSU) This interfaces between the CSU and the customers equipment, using RS232 for low speeds up to 56Kbps, and V.35 (RS-422/499) for higher speeds It is common to have the units as a single component. The CSU/DSU is normally the demarcation zone which defines where the customers responsibility ends the the telecommunications company begins. Most telecommunication companies provide the ability to perform real-time monitoring of the connection via the CSU/DSU. Advantages Private and secure Disadvantages Common Usage Locked into Connecting large sites Tele - c o mm uni c a tio ns pricing regime Establishment permanent presence of a internet

Cost effective for regular High monthly rental transfer of large amounts of data Fixed costs easier to budget for than if you pay for data transferred

Packet Switching (X.25) Packet switching has been around for some time now. It is an established technology which


sends data across a packet switched network in small parcels called packets. If the data packets travel the same path to the destination, this is called virtual circuit, if packets can travel any path, not necessarily the same as each other, this is called datagram. Packet switched connections are normally in the speed of 19.2Kbps to 64Kbps, though some higher speed connections may be available in certain countries. It is a dial-up switched connection, in that the user pays connection charges, traffic charges and time charges. As such, its not suitable for permanent connections. X.25 was designed to be implemented over noisy analogue phone lines, thus has a lot of built in error control. With today's relatively low error links, this can result in an unnecessary overhead. An X.25 connection supports a number of virtual circuits which are each numbered. These represent a time division of the available bandwidth of the connection. This division into virtual circuits allows each VC to support a single device. X.25 uses the lower 3 levels of the OSI model. The virtual circuit is a full duplex connection which is established for the duration of the call. Devices which do not have built in packet switched support can be interfaced to a packet switched network using a Packet Assembly/Disassembly (PAD) unit. This allows existing computers or terminals to be connected. Integrated Services Digital Network (ISDN) ISDN was developed in order to provide the user with a single interface which supported a range of different devices simultaneously. The basic ISDN connection is a 2B + D connection, that is, 2 B channels each of 64Kbps, and a single D channel of 16Kbps. The B channels are designed to carry user data, whilst the D channel is meant to carry control and signaling information. This format is known as the Basic Rate Interface (BRI), which also provides for frame control and other heads, which gives an overall capacity of 192Kbps per BRI ISDN connection. Higher capacity circuits are available. ISDN uses the existing telecommunications dial-up infrastructure, though special ISDN connection interface boxes are required at the users premises. Each B channel can be used separately or combined with other B channels to achieve higher speeds. The Primary Rate Interface (PRI) offers 23B channels and one D channel at 64Kbps (North America and Japan) giving a total of 1.544Mbps. The PRI for Europe, Australia and some other parts of the world is 30B channels and one D channel at 64Kbps giving a total of 2.048Mbps.Advantages Low fixed cost Scalable (B circuits can be combined for greater speeds) Fast call set up times Disadvantages Not available in all centers or countries Not suited to mobile users (users dialing in via remote access) Common Usage Periodic Internet Access (for email etc) LAN-LAN remote connections which are not permanent

Line Drivers Device designed to increase the strength of a signal, which helps ensure that the signal reaches its destination.


Over half of the broadband modems and line cards shipping today depend on Analog Devices' high-performance line drivers. ADI's cable line drivers have been selected by the industry's leading manufacturers for DOCSIS 1.0 and 2.0 cable modems as well as the newest and most advanced cable set top boxes. ADI's xDSL line drivers are the most widely deployed in the world and are used in both Central Office (CO) DSLAM and DLC line cards as well as Customer Premise (CPE) modems. These high performance / low power dissipation drivers enable efficient high port count line cards and superior customer modem performance. Balanced Line Driver & Receiver Sometimes, you just can't get rid of that %$#*& hum, no matter what you do. Especially with long interconnects (such as to a powered sub-woofer), earth loops can be a real pain. For this reason, just about all professional equipment uses balanced lines, which, if properly executed, will eliminate the hum completely. With this simple project, you can have balanced lines too, simply adapting the unbalanced inputs and outputs of your hi-fi gear to become balanced, and then back to unbalanced at the other end. You can even be extra cunning, and power the remote converter from the cables carrying the signal. Professionally, this is called "Phantom Feed", and is used to power microphones and other low current equipment. The version I have shown is actually a differential feed. Whilst not as good as a true 48V phantom powering circuit, it does work, and makes an interesting experiment (if nothing else). Description Before we start, a brief description of the standard (unbalanced) and balanced line is in order. An unbalanced line is the type you have on the hi-fi, typically using an RCA connector, and feeding the signal through a coaxial cable. The inner cable carries the signal, and the outer shield is a screen, to prevent RF interference and general airborne noise from being picked up on the signal lead. This is fine, except for one small detail - the shield must also carry the signal! This is the return path, and is required in all electrical connections - otherwise there is no current flow and the system will probably just hum softly (or loudly) with none of the wanted signal. The problem with electricity (like water and most people) is that it always takes the path of least resistance, so when two pieces of equipment are connected, most likely there will be signal plus hum, because of the dreaded earth loop. This is formed when both items are connected to the mains earth, and also have their earth (zero Volt) points joined via the shields of the signal leads. In some cases it is possible to disconnect the earth at one end of the cable - some people have also disconnected the mains (safety) earth. Both achieve the same result, but disconnecting the mains earth is extremely dangerous. Unfortunately, the result is not always as one would hope.


RF interference can become much worse, and other noises become apparent that were absent before. In contrast, a balanced connection uses two wires for the signal (much like the telephone circuit), with the signal equal in amplitude in each wire, but opposite in phase. Only the out of phase signal is detected by the remote balanced receiver, and any in phase (common mode) signal is rejected. RF interference and other noise will be picked up equally by both wires in the cable and so will be in phase. It will therefore be rejected by the receiver. In this way, it is possible to have long interconnects, with the shield connected at one end only. This cuts the earth loop, and the balanced connection ensures that only the wanted signal is passed through to the amplifier(s). It is very important that the two signal leads are twisted together, and the tighter the twist, the better. The shield prevents RF and other interfering signals from causing too much trouble, and the final signal should be free from hum and noise. The shield serves the same function in an unbalanced circuit, but is less effective due to the fact that it usually serves as the signal return path, and any signal that does get through becomes part of the signal. The idea of this project is to give you some options, and to assist in creating a solution - it should not be seen as a complete solution in itself. There are many variables - far too many to be able to say with complete confidence that this WILL prevent all hum and other interference. It might, but it is likely that some experimentation will be needed to get the results you want. Note that for both transmitter and receiver, it is essential that 1% (or better) tolerance resistors are used. If the trimming option is implemented, then you could use 5% resistors, and you will be able to adjust the circuit to get maximum common mode rejection - however I recommend that you use the 1% metal film resistors. For the small extra cost you get much higher stability, and lower noise.

Figure 1 - Balanced Line Transmitter

The transmitter uses one opamp to buffer the signal, and the other to buffer and invert it. This creates a balanced signal, where as the signal swings positive on one lead, it swings exactly the same amount negative on the other. The 220 Ohm resistors at the output ensure stability with


any lead, and are also used to attenuate the signal slightly. The signal swing from the transmitter (across both wires) is double the voltage of the input signal.

Figure 2 - Balanced Line Receiver The receiver has an optional 3.3k resistor across the inputs (RO) to help balance the input against minor variations in cable impedance between the individual lines. The 220pF capacitor is for HF rolloff, and will attenuate any RF that might get picked up by the lead. Any common mode signal - where both leads provide a signal of the same polarity to the receiver circuit; typically noise - is rejected, leaving only the wanted signal. The rest of the circuit is a conventional balanced input stage. This particular configuration is somewhat notorious for having unequal input impedances referred to earth. The 3.3k resistor helps this (a little, anyway), and the 220pF capacitor also assists at higher frequencies. A more complex circuit could have been used, but that would require 3 opamps, and for the intended task would offer few real advantages. With the capacitor value chosen, there is about 0.1dB attenuation at 20kHz - if you don't like this idea, reduce the value to 100pF, however since 0.1dB is quite inaudible, there seems little point. With the values shown, there is a very slight overall gain of just over 0.3dB. This is unlikely to be a problem. The circuit is designed to send the maximum level possible across the balanced cable, and most of the attenuation is performed at the receiver. This will reduce any noise picked up by a further 6dB for the transmitter / receiver pair. It is also possible to ensure that the common mode rejection is as good as it can possibly get, by making R10 variable. I suggest that you use an 8.2k fixed resistor, with a 5k multi-turn trimpot in series. To balance the circuit, you may use an oscillator and millivoltmeter, or just a small battery and a multimeter. Join the two inputs together, and connect the battery or audio oscillator between the two joined inputs and earth. Adjust the trimpot until there is 0V at the output - the common mode signal is


now gone completely. Typically, this circuit will give a common mode rejection of about 40dB if not trimmed as described, but trimming will let you improve on this considerably. Although this transmitter and receiver pair will probably allow the use of unshielded interconnects, I don't recommend this. Use a good quality shielded twin microphone cable. The earthing of the shield should normally be done at the receiver end, but in some cases you might find that the noise rejection is better if the transmitter end is earthed. Experimentation will be needed. Phantom Power (For the Experimenter) It is possible to run this unit with the signal leads also carrying the power for the receiver. We could use conventional phantom feed (using a 48V supply), but it is easier to use a differential feed, with the +ve and -ve supply voltages on the signal leads. The basic scheme is shown in Figure 3. This may be found to reduce common mode rejection, and it is essential that the power is completely noise free, or it will become part of the signal! If this method is to be tried, use the trimming option, so the supply feed resistors can be catered for. Alignment with a battery will no longer be possible, and a signal generator will have to be used - with coupling capacitors to each signal line.The resistor RO must be removed in this configuration. I would strongly recommend that an output coupling capacitor is used from the Out terminal of the receiver, since it is likely that there will be some DC offset due to capacitor leakage currents.

Figure 3 - Differential "Phantom" Powering

The voltage to the receiver opamp is reduced by this technique, and the maximum signal level will be reduced too. Only by experimenting will you be able to determine the exact power losses and maximum signal level attainable. The tests I did indicate that you should not expect more than about 1V RMS, but you might get more depending on the opamp used for the receiver. The power feed resistors also load the transmitter, and reduce its output capability somewhat. You might want to experiment with a low-power opamp (such as an LF351) as the receiver, as this will allow a higher supply voltage and more signal before distortion.


I would expect that the most likely use for this arrangement would be for a remote sub-woofer, where it may be very inconvenient to have to create an additional power supply. I can't say that I am completely happy with this arrangement, but it does work. A 48V phantom supply would be better, but it is not likely that too many constructors will want to go to this trouble.

Figure 4 - Overall Frequency Response of Differential Feed and Both Circuits

The shield will now have to be connected at each end, but one end can be earthed using a 10 Ohm resistor, which should be bypassed with a 100nF capacitor. Again, experimentation is needed to determine which end should have the "hard" earth. Make sure that the connectors are polarised so that power cannot be connected the wrong way around. Diodes may be added if desired to provide proper protection. These should be in parallel with the receiver filter caps (C+ve and C-ve), because a series connection will reduce the voltage further (there is not a lot to start with, so a further reduction would be a disaster). Use of a multi-cored cable and suitable connectors will allow you to run the power supply on separate wires in the cable, and the additional cost of the cable and connectors is likely to be offset by the simpler circuit and better performance. This may not always be possible, hence the differential phantom feed.


Lesson IV: Permanent Digital NetworksBT Kilo stream About BT Kilo Stream Private Services are specially designed for businesses which rely heavily on communications. They provide permanently connected analogue and digital, voice and data circuits, between different sites, for the exclusive use of the business. Speech Line and Keyline analogue circuits are used for straightforward voice or low-speed data applications. However, once you are regularly in touch with the same locations, making increased use of e-mail or exchanging larger and larger data files, then switching to Kilo Stream or the Kilo Stream N (the fastest Kilo Stream service for speech or data) digital services should result in substantial cost savings. In fact, because Kilo Stream circuits are leased for a fixed tariff, the more you use them, the more cost effective they become. Kilo Stream comes in a range of different speeds, from 2.4kbit/s to 1,024kbit/s, to suit the needs and the budget of any business customer.


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Key benefits of KiloStream include; Physical point-to-point connectivity - assuring high levels of security A state of the art network - providing very high levels of reliability and circuit availability Geographical coverage - extending over 99% of the UK 2-week provision Absence of modems - saving cost and adding reliability Connectivity applications, including multiplexors, a mixture of all three. Key features of KiloStream N include; Cost effectiveness where ordinary KiloStream is insufficient A smooth evolution path for network growth Easy accommodation of specialist applications such as CAD/CAM and video-conferencing High quality transmission, performance and reliability Resilience - both separation/diversity & disaster recovery service available TotalCare support Nation-wide geographical coverage 6 week provision The Private Service you choose will depend on the volume and kind of information you wish to communicate Analogue or digital circuits up to 64kbit/s are mainly used for low-speed voice or data applications, such as PC terminal users at branch offices who need on-line access to a host computer for electronic data interchange (EDI), file transfer or remote printing facilities. At 64kbit/s, you can transmit voice and data, linking together local area networks (LANs) for order processing and stock control, or make Internet access more widely available. And at speeds of 128kbit/s and above, KiloStream N can be used for voice or data applications, to connect complete systems, for high speed faxing, or video conferences. data, voice and image; and, with suitable


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