Telecommunication NTC l7

 

TELECOMMUNICATION The Public Switched Telephone Network (PSTN)

The term Public Switched Telephone Network (PSTN) describes the various equipment and interconnecting facilities that provide phone service to the public. The network continues to evolve with the introduction of new technologies. The PSTN began in the United States in 1878 with a manual mechanical switchboard that connected different parties and allowed them to carry on a conversation. Today, the PSTN is a network of computers and other electronic equipment that converts speech into digital data and provides a multitude of sophisticated phone features, data services, and mobile wireless access. PSTN voice facilities transport speech or voice-band data (such as fax/modems and digital data), which is data that has been modulated to voice frequencies. At the core of the PSTN are digital switches. The term "switch" describes the ability to cross-connect a  phone line with many many other phone lines and switching switching from one connection connection to another. The PSTN PSTN is well known for providing reliable communications to its subscribers. The phrase "five nines reliability," representing network availability of 99.999 percent for PSTN equipment, has become ubiquitous within the telecommunications industry. Network Topology

The topology of a network describes the various network nodes and how they interconnect. Depending on geographical region, PSTN nodes are sometimes referred to by different names. The three node types include:    

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End Office (EO): also called a Local Exchange. The End Office provides network access for the subscriber. It is located at the bottom of the network hierarchy. Tandem: connects EOs together, providing an aggregation point for traffic between them. In some cases, the Tandem node provides the EO access to the next hierarchical level of the network. Transit: provides an interface to another hierarchical network level. Transit switches are generally used to aggregate traffic that is carried across long geographical distances.

There are two primary methods of connecting switching nodes.   The first approach is a mesh topology, in which all nodes are interconnected. This approach does not scale well when a large number of nodes needs to be connected. One must connect each new node to every existing node. This approach does have its merits, however; it simplifies routing traffic between nodes and avoids bottlenecks by involving only those switches that are in direct communication with each other.   The second approach is a hierarchical tree in which nodes are aggregated as the hierarchy traverses from the subscriber access points to the top of the tree. PSTN networks use a combination of these two methods, which are largely driven by cost and the traffic  patterns between between exchanges exchanges..

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Figure below shows a generic PSTN hierarchy, in which End Offices are connected locally and through tandem switches. Transit switches provide further aggregation points for connecting multiple tandems  between different networks. While actual network topologies vary, most follow some variation of this  basic pattern. pattern.

Figure: Generic PSTN Hierarchies PSTN Hierarchy

The PSTN hierarchy is implemented differently in the United States and the United Kingdom. The following sections provide an overview of the PSTN hierarchy and its related terminology in each of these countries.  PSTN Hierarchy Hierarchy in the United States In the United States, the PSTN is generally divided into three categories:      

Local Exchange Networks InterExchange Networks Internationall Networks Internationa

Local Exchange Carriers (LECs) operate Local Exchange networks, while InterExchange Carriers (IXCs) operate Inter-Exchange and International networks. Local Exchange Network

The Local Exchange network consists of the digital switching nodes (EOs) that provide network access to the subscriber. The Local Exchange terminates both lines and trunks, providing the subscriber access to the PSTN. A Tandem Office often connects End Offices within a local area, but they can also be connected directly. In the United States, Tandem Offices are usually designated as either Local Tandem (LT) or Access Tandem (AT). The primary purpose of a Local Tandem is to provide interconnection between End Offices in a localized geographic region. An Access Tandem provides interconnection between local End Offices and serves as a primary point of access for IXCs. Trunks are the facilities that connect all of the offices, thereby transporting inter-nodal traffic. 2

 

Inter-Exchange Network

The Inter-Exchange network is comprised of digital switching nodes that provide the connection between Local Exchange networks. Because they are points of high traffic aggregation and they cover larger geographical distances, high-speed transports are typically used between transit switches International Network

The International network consists of digital switching nodes, which are located in each country and act as international gateways to destinations outside of their respective countries. These gateways adhere to the ITU international standards to ensure interoperability between national networks. The international switch also performs the protocol conversions between national and international signalling. The gateway also performs PCM conversions between A-law and -law to produce compatible speech encoding  between networks, networks, when necessary. necessary.

Figure: Generic U.S. Hierarchies Telephone switching offices are often referred to by class class.. For example, an EO is commonly called a class 5 office, and an AT is called a class 4 office. Class 1 being the highest office category and class 5  being the lowest (nearest (nearest to subscriber access). access). Aggregation of transit phone tra traffic ffic moved from the class 5 office up through the class 1 office. Each class of traffic aggregation points contained a smaller number of offices. Class  Office Type 

1

Regional Center

2

Sectional Center

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Primary Center

4

Toll Center

5

End Office

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Local calls remained within class 5 offices, while a cross-country call traversed the hierarchy up to a regional switching center. This system no longer exists, but we included it to give relevance to the class terminology,, which the industry still uses often. terminology

PSTN Hierarchy in the United Kingdom

Figure  below shows the PSTN topology used in the United Kingdom. End Offices are referred to as Digital Local Exchanges (DLE). A fully meshed tandem network of Digital Main Switching Units (DMSU) connects the DLEs. Digital International Switching Centers (DISC) connect the DMSU tandem switches for international call connections.

Figure: U.K. PSTN Hierarchy

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Access and Transmission Facilities

Connections to PSTN switches can be divided into two basic categories: lines and trunks. Individual telephone lines connect subscribers to the Central Office (CO) by wire pairs, while trunks are used to interconnect PSTN switches. Trunks also provide access to corporate phone environments, which often use a Private Branch exchange (PBX) or in the case of some very large businesses, their own digital switch. Figure switch.  Figure below  below illustrates illustrates a number of common interfaces to the Cen Central tral Office. Office.

Figure: End Office Facility Interfaces  Lines Lines are used to connect the subscriber to the CO, providing the subscriber access into the PSTN. The following sections describe the facilities used for lines, and the access signalling between the subscriber and the CO.

 

 The Local Loop    Dialing    Ringing and Answer Answer   Voice Encoding   Trunks The Local Loop The local loop consists of a pair of copper wires extending from the CO to a residence or business that connects to the phone, fax, modem, or other telephony device. The local loop allows a subscriber to access the PSTN through its connection to the CO. The local loop terminates on the Main Distribution Frame (MDF) at the CO, or on a remote line concentrator.

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Remote line concentrators, also referred to as Subscriber Line Multiplexers or Subscriber Line Concentrators, extend the line interface from the CO toward the subscribers, thereby reducing the amount of wire pairs back to the CO and converting the signal from analog to digital closer to the subscriber access point. In some cases, remote switching centers are used instead of remote concentrators. Remote switching centers provide local switching between subtending lines without using the resources of the CO. Remotes, CO.  Remotes, as  as they are often generically referred to, are typically used for subscribers who are located far away While terminating theform. physical loop, remotes transport the digitized voice stream back to thefrom CO the overCO. a trunk circuit, in digital  Dialing When a subscriber dials a number, the number is signaled to the CO as either a series of pulses based on the number dialed, or by Dual Tone Multi-Frequency (DTMF) signals. The DTMF signal is a combination of two tones that are generated at different frequencies. A total of seven frequencies are combined to provide unique DTMF signals for the 12 keys (three columns by four rows) on the standard  phone keypad. keypad. Usually, Usually, the dialing dialing pla plan n of the CO CO determ determines ines when all all digits have have been collected. collected.  Ringing and Answer Answer To notify the called party of an incoming call, the CO sends AC ringing voltage over the local loop to the terminating line. The incoming voltage activates the ringing circuit within the phone to generate an audible ring signal. The CO also sends an audible ring-back tone over the originating local loop to indicate that the call is proceeding and the destination phone is ringing. When the destination phone is taken off-hook, the CO detects the change in loop current and stops generating the ringing voltage. This  procedure is commonly commonly referred to as ring trip. trip. The off-hook signals the CO that the call has been answered; the conversation path is then completed between the two parties and other actions, such as  billing, can can be initiated, initiated, if necessary. necessary. Voice Encoding An analog voice signal must be encoded into digital information for transmission over the digital switching network. The conversion is completed using a codec (coder/decoder), which converts between analog and digital data. The ITU G.711 standard specifies the Pulse Coded Modulation (PCM) method used throughout most of the PSTN. An analog-to-digital converter samples the analog voice 8000 times  per second and then assigns a quantization quantization value based on 256 decision levels. The quantization value is then encoded into a binary number to represent the individual data point of the sample. Two variations of encoding schemes are used for the actual quantization values: A-law and -Law encoding. North America uses -Law encoding, and European countries use A-law encoding. When voice is transmitted from the digital switch over the analog loop, the digital voice data is decoded and converted  back into an analog analog signal before transmitting transmitting over over the loop. Trunks Trunks carry traffic between telephony switching nodes. Digital trunks may be either four-wire (twisted  pairs) or fiber optic medium for higher capacity. capacity. T1 and E1 are the most common trunk types for connecting to End Offices. North American networks use T1, and European networks use E1.

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On the T1/E1 facility, voice channels are multiplexed into digital bit streams using Time Division Multiplexing (TDM). TDM allocates one timeslot from each digital data stream's frame to transmit a voice sample from a conversation. Each frame carries a total of 24 multiplexed voice channels for T1 and 30 channels for E1. The Central Office

The Central Office (CO) houses the digital switching equipment that terminates subscribers' lines and trunks and switch calls. and switch  calls. The functional areas of the CO are:        

The Main Distribution Frame The Digital Switch The Switching Matrix Call Processing

Main Distribution Frame Incoming lines and trunks are terminated on the Main Distribution Frame (MDF). The MDF provides a  junction point where the external facilities connect to the equipment equipment within the CO. Jumpers J umpers make the connections between the external facilities and the CO equipment, thereby allowing connections to be changed easily. Line connections from the MDF to the digital switching equipment terminate on line cards that are designed to interface with the particular type of line being connected such as POTS, ISDN BRI. Trunk connections from the MDF are terminated on trunk interface cards, providing the necessary functions for message framing, transmission, and reception. The Digital Switch The digital switch provides a software-controlled matrix of interconnections between phone subscribers. All digital switches are designed with some degree of distributed processing. A typical architecture includes a central processing unit that controls peripheral processors interfacing with the voice channels. Redundancy is always employed in the design to provide the high reliability that is expected in the telephony network. For example, the failure of one central processing unit results in the activation of an alternate processing unit. The line and trunk interface cards, mentioned previously, represent the point of entry into the digital switch. These cards typically reside in peripheral equipment that is ultimately controlled by the central  processor. Within the digital switch, all voice streams are digitized data. Some voice streams, such as those from ISDN facilities and digital trunks, enter the switch as digital data. Other voice streams, such as the analog phone, enter as analog data but undergo digital conversion at their point of entry. Analog lines interface with line cards that contain codecs, which perform the PCM processing to provide digital data to the switch and analog data to the line. Using the distributed processing architecture, many functions related to the individual voice channels are delegated to the peripheral interface equipment. This relieves the central processor of CPU intensive, low-level processing functions, such as scanning for on/off hooks on each individual line to determine when a subscriber wants to place a call. The central processing unit monitors information from peripheral processors on call events such as origination, digit collection, answer, and termination — and and orchestrates the actual call setup and release. Information from these events is also used to perform call accounting, billing, and statistical information such as Operational Measurements (OM). 7

 

Although the main purpose of the digital switch is to perform call processing, much of its functionality is dedicated to maintenance, diagnostics, and fault recovery to ensure reliability. An OM is a counter that records an event of particular interest, such as the number of call attempts or the number of a particular type of message received, to service providers. OMs can also be used to record usage in terms of how long a resource is used. Modern digital switches usually record hundreds, or even thousands of different types of OMs for f or various events taking place in the switch. Switching Matrix A modern digital switch can process many voice channels. The actual number of channels it processes varies with the switch vendor and particular model of switch, but they often process tens of thousands of voice channels in a single switch. A number of switches have capacities of over 100,000 connections. The switch is responsible for many tasks, but one of its primary functions is connecting voice channels to create a bi-directional conversation path between two phone subscribers. All digital switches incorporate some form of switching matrix to allow the connection of voice channels to other voice channels. Once a circuit is set up between the two subscribers, the connection remains for the duration of the call. This method of setting up call connections is commonly known as circuit switching. Call Processing Call processing is associated with the setup, maintenance, and release of calls within the digital switch. The process is driven by software, in response to stimulus from the facilities coming into the switch. Signaling indications, indications, such as on/off-hook, dialing digits, and answer, are all part of the stimuli that drive the processing of calls. Each call process can be represented as an originating call half and a terminating call half. When combined, the two halves are completely representative of the call. The originating half is created when the switch determines that the originator is attempting a call. The terminating call half is created when the destination has been identified, typically at the translations or routing phase.

Functions of Switching System The switching office performs the following basic functions irrespective of the system whether it is a manual or electromechanical or electronic switching system. 1. Identity. The local switching center must react to a calling signal from calling subscriber and must be able to receive information to identify the required destination terminal to seize. 2. Addressing. The switching system must be able to identify the called subscriber from the input information (train of pulses or multiple frequency depends on the dialing facility). The address may be in same local center or some other exchange. If the terminal or trunk group is  busy, a suitable signal must be returned to the calling subscriber.

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3. Finding and Path setup. Once the calling subscriber destination is identified and the called subscriber is available, an accept signal is passed to the switching system and calling subscriber. Based on the availability, suitable path will be selected. 4. Busy testing. If number dialed by the calling subscriber is wrong or the called subscriber is  busy (not attending the phone) or the terminal may be free (lifting the phone) but no response (not willing to talk or children handling), a switching system has to pass a corresponding voice message or busy tone after waiting for some time (status). 5. Supervision. Once the path is setup between calling and called subscriber, it should be supervised in order to detect answer and clear down conditions and recording billing information. 6. Clear down. When the established call is completed, the path setup should be disconnected. If the calling subscriber keeps the phone down first, the signal called clear forward is passed to the switching system. If the called subscriber keeps the phone down first, a signal called clear  backward signal is passed to the switching system. By clear signal, the switching system must disconnect the path setup between calling and called subscriber. 7. Billing. A switching system should have a mechanism to meter to count the number of units made during the conversation. The cumulative number of units made for a particular duration by the calling subscriber is calculated.

Classification of Switching System Switching systems are broadly classified as manual and automatic switching system. Early switching systems were manual and operator oriented. The need for speed of switching and  privacy brought automatic switching into existence. Automatic switching systems can be classified as electromechanical and electronic switching system. Electromechanical switching systems include step by step (Strowger) switching and crossbar switching system. The control functions in Strowger system are performed by circuits associated with the switching elements in the system. Crossbar systems have hard wired control subsystems which use relays and latches. These subsystems have limited capability and it is virtually impossible to modify them to provide additional functionalities. In electronic switching systems, the control functions are performed by a computer or a  processor. Hence, these systems are called stored program control (SPC) systems. New features and services can be added to the SPC by changing the control program. The electronic switching system is divided into space division and time division switching. In space division a dedicated path is established between the calling and the called subscriber for the entire call duration. In time division switching, switching, sampled values of speech are transferred at fixed intervals. The signal in time division switching can be analog or digital. In digital 9

 

switching, if the coded values are transferred during the same time interval from input to the output, the technique is called space switching. If the values are stored and transferred to the output at a later time interval, the technique is called time switching. A combination of the two is also possible.

Manual Switching During the early days of telephones one pair of wires were required to route signals from each individual telephone to a central office where multitudes of operators routed calls from the calling party to the destination. Users cranked a handle on the telephone which rang a bell on the switchboard in the central office. An operator then used to plug a headset into the incoming line and ask for a called party number. At which point the operator would make a connection  between the lines with jacks ja cks and the user would crank their telephone again to generate a ringing signal at the called party's telephone set.  

STEP-BY-STEP SWITCHING OR STROWGER SYSTEM Step by Step Switching or Strowger switching was the first automatic telephone system introduced by Almon B. Strowger. This system uses selectors for switching. The selectors used in Strowger exchange are mainly of two types 1.  Uniselector 2.  Two motion selector. Both the selectors belong to the same types of switches called rotary switches.

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UNISELECTOR This is called uniselector because the rotary motion of this switch is in one direction, i.e., the wiper assembly moves only in one direction. The uniselector consists of moving contacts called wipers. These are used to make electrical connections with any one of several contacts, called  bank contacts, in an arc around it. The arc in most cases consists of ten steps. These wipers are operated by an electromagnet, called driving magnet. When current flows through the windings of the driving magnet, it gets energized. The uniselector rotates as many steps as the electromagnet is energized and de-energized.

TWO MOTION SELECTOR The two-motion selector is a type of rotary switch, in which the motion of the wiper assembly is in two directions, vertical as well as horizontal. In the vertical direction the wipers move upward to the desired level and make no connections with the bank contacts. While in the horizontal direction the wipers make connection with the bank contacts. The two-motion selector has 10 levels each having 10 contacts thus a total of 100 contacts are accessible. Each contact represents the terminals of one switch of the higher stage or of one telephone line in the case of final selector. The dialing pulses cause the wiper assembly to step up or down to the desired level. If we take the example of a final selector, where up to 100 lines can be connected, the vertical and horizontal stepping of the selector are controlled by the digits dialed by the subscriber.   When the first digit is dialed, the dialing pulses energize and de-energize the vertical magnet. The vertical magnet step up the wiper assembly, corresponding to the digit dialed. This is called vertical stepping.   When the second digit is dialed the dialing pulses are now diverted to another magnet, called horizontal magnet, with the help of a relay. These pulses energize and de-energize the horizontal magnet, which causes the wiper assembly to rotate to the proper contact, corresponding to the second digit dialed. This is called horizontal stepping. 11

 

  Thus the wiper assembly makes connection with the required number dialed by the subscriber. After completion of the call the wiper assembly comes back to the home  position.

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THE CROSSBAR SWITCH The crossbar is actually a matrix switch used to establish a speech path. A crossbar switch with  N input lines and N output lines contains an N x N array of cross points that connect each input line to one output line. An electrical contact is made by actuating a horizontal and a vertical relay. A typical crossbar matrix concept is shown in Figure. To make contact at point B4 on the matrix, horizontal relay B and vertical relay 4 must close to establish connection. Such a closing is momentary but sufficient to cause ―latching‖. Two forms of latching were used in conventional crossbar practice: mechanical and electrical. The latch keeps the speech-path connection until an ―on―on-hook‖ condition results, freeing the horizontal and vertical relays to establish other connections, whereas connection B4 in the Figure ―busied out.‖  out.‖ 

In contrast to a step-by-step switch, a crossbar switch is one that used centralized, common control for switch path selection. As digits were dialed, the control element of the switch received the entire address before processing it. When an appropriate path through the switch was determined (which may have involved number translation or alternate routing), the control element transferred the necessary information in the form of control signals to the switching matrix to establish the connection. 12

 

Limitations of crossbar switches are as follows:   The number of cross points grows with the square of the number of attached stations.   Costly for a large switch.   The failure of a cross point prevents connection between the two devices whose lines intersect at that cross point.   The cross points are inefficiently utilized.   Only a small fraction of cross points is engaged even if all of the attached devices are active.

STORED-PROGRAM CONTROL Stored-program control (SPC) is a broad term designating switches where common control is carried out to a greater extent or entirely by computerware. Common control is a means of control of the interconnecting switch network, first identifying the input and output of the terminals of the network that are free and then establishing a path between them. Computerware can be a full-scale computer, mini-microcomputer, microprocessors, or other electronic logic circuits. Control functions may be entirely carried out by a central computer in one extreme for centralized processing or partially or wholly by distributed processing utilizing microprocessors. There are four basic functional elements of an SPC switching system: 1.  switching matrix 2.  call-store (memory) 3.   program store (memory) and 4.  central processor The switching matrix can be made up of electromechanical cross-points, such as in the crossbar switch. The call store is often referred to as the scratch pad memory. This is a temporary storage of incoming call information, availability and status of trunks and circuits. The program store  provides the basic instructions to the central processor.

Space-division switching  A method of switching circuits, in which, each connection through the switch takes a physically separate path.

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In space division switching, single transmission path is accomplished using a switch to  physically separate a set of matrix contacts or cross points. Space division is therefore closely linked to the concept of crossbar switch. A cross pint connects an incoming and an outgoing circuit to create link. A dedicated physical path is created for for each traffic stream, and several signals are switched in parallel. Space division circuits can carry either analog or digital traffic.  Number (N) input paths form a crossbar matrix with number (M) output paths forming a N-by-M matrix. The number of cross points in a matrix grows as a square of the number of stations increases. Space division switching is fast because each path is assigned to one call and exclusively to that call throughout the duration. Once a continuous path has been established signals are transferred between the input and output outpu t at the full transmission speed of the circuit. The process of identifying the cross points that must be closed to set up a continuous path  between two given channels without disturbing other existing connections is known as pathfinding. This requires that the system identify the links to and from the cross points. The system must therefore be able to determine at any given time which links are free and which links are  busy. There is a requirement for central control that can check the status of the links at the network. Space division switching has a number of disadvantages. If the matrix becomes too large setup time can be a problem. In addition, the loss of single cross point can prevent a connection.

Time Division Switching Time division switching is defined as the switching of time division multiplexed (TDM) channels by shifting bits between time slots in a TDM frame. In time division switching, slow data streams are partitioned into pieced, or data segments, that share time on a higher speed stream. The segments are switched one at a time. This switching method is based on synchronous TDM, and the source and destination of the data in each time slot are known. When data is being transmitted on one channel, the data for the next channel is prepared for transmission and may be buffered at the gate. The allocated to a slot must equal the transmission time for the slot data plus any propagation delay d elay across the bus. For time division switching to be successful, the transmitting station must connect in the same time slot as the appropriate receiving station. The process of coordinating time slots is called the time-slot interchange (TSI). Time division switching can be implemented only on digital links.

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Time Slot Interchanger Time slot interchanging involves moving the data contained in each time slot from the incoming  bit stream to an outgoing bit stream but with a different time-slot arrangement in accordance with the destination of each time slot. What is done, of course, is to generate a new frame for transmission at the appropriate switch outlet. Obviously, to accomplish this, at least one time slot must be stored in memory (write) and then called out of memory in a changed position (read). The operations must be controlled in some manner, and some of these control actions must be kept in memory together with the software managing such actions. Typical control functions are time-slot timeslot ―idle‖ or ―busy.‖ Three of the basic functional blocks of a time switch: switch:   1.  Memory for speech 2.  Memory for control 3.  Time-slot counter or processor There are two choices in handling h andling the time switch: 1.  Sequential write, random read and 2.  Random write, sequential read In the first case, sequential write, the time slots are written into the speech memory also called data memory (DM) as they appear in the incoming incoming bit  bit stream. For the second case, random write, the incoming time slots are written into memory in the order of appearance in the outgoing  bit stream. This means that the incoming time slots are written into memory in the desired output order. The writing of incoming time slots into the speech memory can be controlled by a simple time-slot counter and can be sequential (e.g., in the order in which they appear in the incoming 15

 

 bit stream). The readout of the speech memory memor y is controlled by the control memory (CM). In this case the readout is random where the time slots are read out in the desired output order. The memory has as many cells as there are time slots.

COMBINATION OF TIME AND SPACE SWITCHES To increase the switching capacity and flexibility a combination of space and time switches are used. As shown in the figure below say user 4 on incoming line A wants to connect to user 10 on outgoing line Y then the switching may be performed as A4 to B4 using a Space switch and then 16

 

from B4 to Y10 using a Time switch. Alternately we may perform the TS Interchanging first i.e. from A4 to A 10 and then perform a space switching from A10 to Y10.

Digital switches are composed of time and space switches in any order or time switches only. Letter T is used to designate a time-switching stage and S to designate a space-switching stage. For instance, a switch that consists of a sequence of a time-switching stage, a space-switching stage, and a time switching stage is called a TST switch. A switch consisting of a spaceswitching stage, a time-switching stage, and a space-switching stage is designated an STS switch. 2-stage combination switches

Time multiplexed time division space switches do not provide full availability as they are not capable of performing time slot interchange. Time slot interchange switches are not capable of switching sample values across the trunks without the help of some space switching matrices. Therefore, a combination of the time and space switches leads to configurations that achieve  both time slot interchange and sample switching across trunks. These structures also permit a large number of simultaneous connections to be supported su pported for a given technolog technology. y. A combination switch can be built by using a number of stages of time and space switches. A two-stage combination switch may be organized with time switch as the first stage and the space switch as the second stage or vice versa. Accordingly, the two switch configurations are known  by the nomenclature time-space (TS) or space-time (ST) switches respectively. Each time multiplexed inlet/outlet stream carries M channels. A subscriber on the input side is assigned to one of the inlets and a time slot in that inlet. An input subscriber assigned to line i at time slot j is identified by the label Iij . Similarly, a subscriber connected to the outlet m and time slot n is identified by 0mn.The corresponding time slots are identified as lS ij and OSmn. 3-stage combination switches

Three-stage time and space combination switches are more flexible than their two-stage counter  parts. The most common three-stage configurations are:   Those which place time stages on either side of a space stage giving rise to TST configuration   Those which place space stages on either side of a time stage giving rise to STS configuration. 17

 

A TST network is shown in the figure below. The two time stage exchange information between external channels and the internal space array channels. The first flexibility that becomes obvious in this arrangement is that there is no need to have a fixed space stage time slot for a given input or output time slot. There are many alternative paths between a prescribed input and output unlike a two stage network which has only one fixed path. This factor reduces the value of the  blocking probability of a three-stage.

A space-time-space (STS) architecture consists of an N x k space matrix at the input, an array of k TSI switches in middle and a k x N space matrix at the output as shown in the figure below. In this architecture, the choice of input and output time slots is fixed for a given connection. But the flexibility is provided by the ability to utilize any free TSl switch by space switching on the input and the output side. There are as many alternative paths for a given connection as there are TSI switches.

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The expansion and the concentration take place at the space switch level and not at the time slot level. The time slots are symmetrical throughout the switch. Based on a reasoning similar to the one used for TST switch, a STS switch is non-blocking if k=2N-1. Switches are designed to be concentrating when the utilization of the input links is low. As the input traffic intensity increases, less and less concentration is acceptable. When the input loading  becomes sufficiently high, space expansion in the STS switch and time expansion in the TST switch are required to maintain low blocking probabilities. Time expansion is cheaper than space expansion. Hence, TST architecture is more cost effective than STS architecture for higher loads. There are, of course, other factors like modularity, testability and maintainability which must also be taken into account before deciding on a particular architecture.

Multistage Switching Limitations of crossbar switches can be overcome with the help of multistage space division switches. By splitting the crossbar switch into smaller units and interconnecting them, it is  possible to build multistage switches with with fewer crosspoints.

Fig: Schematic of Crossbar Switch

Fig: 3-Stage Space Division Switch

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Figure above shows a three-stage space division switch. In this case the number of crosspoints needed goes down from 64 to 40. There is more than one path through the network to connect two endpoints, thereby increasing reliability. Multistage switches may lead to blocking . The  problem may be tackled by increasing the number or size of the intermediate switches, which also increases the cost. The blocking feature is illustrated in Fig below. As shown in, after setting up connections for 1-to-3 and 2-to-4, the switch cannot establish connections for 3-to-6 and 4-to5.

Comparison between STS and TST Switch

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Transmission Media Transmission medium is the physical path between transmitter and receiver in a data transmission system. Transmission media can be classified as guided or unguided. In both cases, communication is in the form of electromagnetic waves. With guided media, the waves are guided along a solid medium, such as copper twisted pair, copper coaxial cable, and optical fiber. The atmosphere and outer space are examples of unguided media that provide a means of transmitting electromagnetic signals but do not guide them; this form of transmission is usually referred to as wireless transmission.  transmission.  The characteristics and quality of a data transmission are determined both by the characteristics of the medium and the characteristics of the signal. In considering the design of data transmission systems, a key concern, generally, is data rate and distance: the greater the data rate and distance, the better. A number of design factors relating to the transmission medium and to the signal determine the data rate and distance: con stant, the greater the bandwidth of a signal, the Bandwidth.  All other factors remaining constant, higher the data rate that can be achieved. Transmission impairments.  Impairments, such as attenuation, limit the distance. For guided media, twisted pair generally suffer more impairment than coaxial cable, which in turn suffers more than optical fiber. Interference. Interference from competing signals in overlapping frequency bands can distort or wipe out a signal. Interference is of particular concern for unguided media, but it is also a  problem with guided media. For guided media, interference can be caused by emanations from nearby cables. For example, twisted pair are often bundled together, and conduits often carry multiple cables. Interference can also be experienced from unguided transmissions. Proper shielding of a guided medium can minimize this problem. Number of receivers.  A guided medium can be used to construct a point-to-point link or a shared link with multiple attachments. In the latter case, each attachment introduces some attenuation and distortion on the line, limiting distance and/or data rate.

Transmission Media

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KEY POINTS  

The transmission media that are used to convey conve y information can be classified as guided or unguided.   Guided media provide a physical path along which the signals are propagated; these include twisted pair, coaxial cable, and optical fiber.    

Unguided media employ an antenna for transmitting through air, vacuum, or water. Traditionally, twisted pair has been the workhorse for communications of all sorts. Higher data rates over longer distances can be achieved with coaxial cable, and so coaxial cable has often been used for high-speed local area network and for high-capacity longdistance trunk applications.   However, the tremendous capacity of optical fiber has made fiber medium more attractive than coaxial cable for long-distance applications. app lications.   Unguided transmission techniques commonly used for information communications that include broadcast radio, terrestrial microwave, and satellite.   Infrared transmission is used in some LAN applications.

GUIDED TRANSMISSION MEDIA Guided media are those that provide a conduit from one device to another, include twisted-pair cable, coaxial cable, and fiber-optic cable.

Twisted Pair Physical Description

A twisted pair consists of two insulated copper wires arranged in a regular spiral pattern. A wire  pair acts as a single communication link. Typically, T ypically, a number of these pairs are bundled together into a cable by wrapping them in a tough protective sheath. Over longer distances, cables may contain hundreds of pairs. The twisting tends to decrease the crosstalk interference between adjacent pairs in a cable. On long-distance links, the twist length typically varies from 5 to 15 cm. The wires in a pair have thicknesses of 0.4 to 0.8 mm. In the telephone system, individual residential telephone sets are connected to the local telephone exchange, or ―end office,‖ by twisted-pair wire. These are referred to as subscriber loops. 22

 

Twisted-pair installations were designed to support voice traffic using analog signaling. However, by means of a modem, these facilities can handle digital data traffic at modest data rates.

Key Points:                  

Separately insulated Twisted together Often "bundled" into cables Twisting reduces interference (two parallel wires constitute a simple antenna; a twisted  pair does not.) Cheap medium Commonly used for communications within buildings and in telephone networks Produced in unshielded (UTP) and shielded (STP) forms, and in different performance categories. Cables may hold hundreds of pairs. Neighbour pairs typically have different twist lengths to reduce crosstalk. Twisted pair may be used to transmit both analog and digital transmission. For analog signals, amplifiers are required about every 5 to 6 km. For digital transmission (using either analog or digital signals), repeaters are required every 2 or 3 km.

Unshielded and Shielded Twisted Pair Twisted pair comes in two varieties: unshielded and shielded. Unshielded twisted pair (UTP) is ordinary telephone wire. Unshielded twisted pair is subject to external electromagnetic interference, including interference from nearby twisted pair and from noise generated in the environment. A way to improve the characteristics of this medium is to shield the twisted pair with a metallic braid or sheathing that reduces interference. This shielded twisted pair (STP)  provides better performance at higher data rates.

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Categories of unshielded twisted-pair cables

Coaxial Cable Physical Description

Coaxial cable, like twisted pair, consists of two conductors, but is constructed differently to  permit it to operate over a wider range of frequencies. It consists of a hollow outer cylindrical conductor that surrounds a single inner wire conductor. The inner conductor is held in place by either regularly spaced insulating rings or a solid dielectric material. The outer conductor is covered with a jacket or shield. A single coaxial cable has a diameter of from 1 to 2.5 cm.

Coaxial cable is used transmittoboth analog and digital signals. Coaxialbecable frequency characteristics that aretosuperior those of twisted pair, and can hence used has effectively at 24

 

higher frequencies and data rates. Because of its shielded, concentric construction, coaxial cable is much less susceptible to interference and crosstalk than twisted pair. The principal constraints on performance are attenuation, thermal noise, and an d intermodulation noise. For long-distance transmission of analog signals, amplifiers are needed every few kilometers, with closer spacing required if higher frequencies are used. The usable spectrum for analog signaling extends to about 500 MHz.

Applications  

Television distribution   Long-distance telephone transmission   Short-run computer system links   Local area networks

Categories of coaxial cables

Optical Fiber Physical Description

An optical fiber is a thin, flexible medium capable of guiding an optical ray. Various glasses and  plastics can be used to make optical fibers. An optical fiber cable has a cylindrical shape and consists of three concentric sections: the core, the cladding, and the jacket. The core is the innermost section and consists of one or more very thin strands, or fibers, made of glass or  plastic. The core has h as a diameter in the range of 8 to t o 100 µm. Each fiber is surrounded by its own o wn cladding , a glass or plastic coating that has optical properties different from those of the core. The interface between the core and cladding acts as a reflector to confine light that would otherwise escape the core. The outermost layer, surrounding one or a bundle of cladded fibers, is the  jacket. The jacket is composed of plastic and other material layered to protect against moisture, abrasion, crushing, and other environmental dangers.

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Optical fiber transmits a signal-encoded beam of light by means of total internal reflection. Total internal reflection can occur in any transparent medium that has   a higher index of refraction than the surrounding medium. In effect, the optical fiber   acts as a waveguide for 14 15 frequencies in the range of 10 to 10 Hertz.

Characteristics that distinguish optical fiber from twisted pair or coaxial cable are: 1.  Greater capacity: The data rate of hundreds of Gbps over tens of kilometers is possible with optical fiber communication. Compare this to the practical maximum of hundreds of Mbps over about 1 km for coaxial cable and just a few Mbps over 1 km or up to 100 Mbps to 1 Gbps over a few fe w tens of meters for twisted pair. 2.  Smaller size and lighter weight: Optical fibers are considerably thinner than coaxial cable or bundled twisted-pair cable 3.  Lower attenuation: Attenuation is significantly lower for optical fiber than for coaxial cable or twisted pair and is constant over a wide range. 4.  Electromagnetic isolation: Optical fiber systems are not affected by external electromagnetic fields. Thus the system is not vulnerable to interference, impulse noise, or crosstalk. 5.  Greater repeater spacing: Repeater spacing in the tens of kilometers for optical fiber is common Coaxial and twisted-pair systems generally have repeaters every ever y few kilometers.

Fiber types

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Comparison of performance of Guided media   Point-to-point Transmission Media type Data Rate

Bandwidth

Twisted pair

4 Mbps

3 MHz

Repeater distance 2-10 km

Coaxial cable optical fiber

500 Mbps 2 Gbps

350 MHz 2 GHz

1-10 km 10-100 km

UNGUIDED TRANSMISSION MEDIA Unguided media transport electromagnetic waves without using a physical conductor. This type of communication is often referred to as wireless communication. communication . 

Electromagnetic spectrum for wireless communication communication 

Three general ranges of frequencies are of interest in our discussion of wireless transmission. 9 Frequencies in the range of about 1 GHz (Gigahertz = 10 Hz) to 40 GHz are referred to as microwave frequencies. At these frequencies, highly directional beams are possible, and microwave is quite suitable for point-to-point transmission. Microwave is also used for satellite communications. Frequencies in the range of 30 MHz to 1 GHz are suitable for omnidirectional applications. We refer to this range as the radio range. Another important frequency range, for 11 local applications, is the infrared portion of the spectrum.This covers, roughly, from 3x10 to 14 2x10 Infrared is useful to local point-to-point and multipoint applications within confined areas, such as a single room. For unguided media, transmission and reception are achieved by means of an antenna.

Terrestrial Microwave Physical Description

The most common type of microwave antenna is the parabolic ―dish.‖ A typical size is about 3m in diameter. The antenna is fixed rigidly and focuses a narrow beam to achieve line-of-sight transmission to the receiving antenna. Microwave antennas are usually located at substantial heights above ground level to extend the range between antennas and to be able to transmit over intervening obstacles.

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Microwave transmission covers a substantial portion of the electromagnetic spectrum. Common frequencies used for transmission are in the range 1 to 40 GHz. The higher the frequency used, the higher the potential bandwidth and therefore the higher the potential data rate. The most common bands for long-haul telecommunications are:

Broadcast Radio Physical Description

The principal difference between broadcast radio and microwave is that the former is omnidirectional and the latter is directional. Thus broadcast radio does not require dish-shaped antennas, and the antennas need not be rigidly mounted to a precise alignment.

Applications Radio is a general term used to encompass frequencies in the range of 3 kHz to 300 GHz. This

range covers FM radio and UHF and VHF television. This range is also used for a number of data networking applications.

Transmission Characteristic Characteristicss The range 30 MHz to 1 GHz is an effective one for broadcast communications. Unlike the case for lower-frequency electromagnetic waves, the ionosphere is transparent to radio waves above 30 MHz. Thus transmission is limited to the line of sight, and distant transmitters will not interfere with each other due to reflection from the atmosphere. Unlike the higher frequencies of the microwave region, broadcast radio waves are less sensitive to attenuation from rainfall.

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Bands

Transmission Impairments Each transmission medium has limitations brought about by impairments. In one way or another each limitation is a function of the length of a link employing the medium and the transmission rate (i.e. bit rate). Distortion

When the received waveform is not identical in shape to a source waveform the phenomenon is called distortion. A received waveform generally contains certain distortions not attributable to external disturbances such as noise and interference but that can be attributed to internal characteristics of the channel itself. In contrast to noise and interference, distortion is deterministic. It is repeated every time the same signal is sent through the same path in the network. Thus distortions can be controlled or compensated for once the nature of the distortion is understood. On metallic transmission links, such as coaxial cable and wire-pair cable, line characteristics distort and attenuate the digital signal as it traverses the medium. There are three cable characteristics that create this distortion: (1) Loss (2) Amplitude distortion (amplitude-frequency response) (3) Phase (delay) distortion 29

 

Amplitude distortion refers to attenuating some frequencies in the voices spectrum more than others. Amplitude distortion could also be introduced by spectrum-limiting filters in FDM equipment. Ideally these filters should uniformly pass all voice band frequencies up to 4 kHz and reject all others. Practical designs however, imply the need for gradual attenuation "roll-offs"  beginning at about 3 kHz Phase distortion is related to the delay characteristics of the transmission medium. Ideally a transmission system should delay all frequency components in a signal uniformly so the proper  phase relationships exist at the receiving terminal. If individual frequency components experience differing delays, the time-domain representation at the output becomes distorted  because superposition of the frequency terms is altered at the output. Noise

 Noise, in its broadest definition, consists of any undesired signal in a communication circuit. The subject of noise and noise reduction is probably the most important single consideration in transmission engineering. It is the major limiting factor in overall system performance.  Noise is broken down into following categories: 1. Thermal noise 2. Intermodulation noise 3. Impulse noise Thermal Noise:  Thermal noise occurs in all transmission media and all communication equipment, including passive devices such as waveguide. It arises from random electron motion and is characterized by a uniform distribution of energy over the frequency spectrum with a Gaussian distribution of levels. Every equipment element and the transmission medium itself contributes thermal noise to a communication system if the temperature of that element or medium is above absolute zero on the Kelvin temperature scale. Thermal noise is the factor that

sets the lower limit of sensitivity of a receiving system and is often expressed as a temperature, usually given in units referred to absolute zero.

Intermodulation Noise:  Intermodulation (IM) noise is the result of the presence of intermodulation products. If two signals with frequencies F1 and F2 are passed through a nonlinear device or medium, the result will contain IM products that are spurious frequency energy components. These components may be present either inside and/or outside the frequency  band of interest for a particular device or system. The products result when two (or more) signals  beat together or ―mix.‖ These products can be sums and/or differences.  differences. 

  Second-order products F1±F2

 

 Third-order products 2F1±F2; 2F2±F1 and   Fourth-order products 2F1±2F2; 3F1±F2 . . . . 30

 

Impulse Noise: Impulse noise is non-continuous, consisting of irregular pulses or noise spikes of short duration and of relatively high amplitude. These spikes are often called hits hits,, and each spike has a broad spectral content (i.e., impulse noise  smears  smears a  a broad frequency bandwidth). Impulse noise degrades voice telephony usually only marginally, if at all. However, it may seriously degrade error performance on data or other digital circuits. The causes of impulse noise are lightning, car ignitions, mechanical switches (even light switches), fluorescent lights, and so on. Crosstalk

Crosstalk is the unwanted coupling between signal paths. Crosstalk is interference generated when magnetic fields or current nearby wires interrupt electrical current in a wire. As electrical current travels through a wire, the current generates a magnetic field. Magnetic field from wires that are close together can interfere with each other. There are essentially three causes of crosstalk: 1.  Electrical coupling between transmission media, such as between wire pair on a voicefrequency (VF) cable system and on digital (PCM) cable systems. Shielding the wire and twisting wire pair around each other help decrease de crease crosstalk. 2.  Poor control of frequency response (i.e., defective filters or poor filter design) 3.   Nonlinear performance in analog frequency division multiplex (FDM) system. Excessive level may intensify crosstalk. Excessive level means that the level or signal intensity has been adjusted to a point higher than it should be. In telephony and data systems, levels are commonly measured in dBm. There are two types of crosstalk: 1. I nt nte ellig lligii ble: where at least four words are intelligible to the listener from extraneous conversation(s) in a seven-second period; and 2. Unintelligible:  crosstalk resulting from any other form of disturbing effects of one channel on another. Two basic forms of crosstalk of concern to telecommunications engineers are near end crosstalk (NEXT) and far-end crosstalk (FEXT). Near-end crosstalk refers to coupling from a transmitter into a receiver at a common location. This form of crosstalk is most troublesome because of a large difference in power levels between the transmitted and received signals. Far-end crosstalk refers to unwanted coupling into a received signal from a transmitter at a distant location.

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Hybrid Transformer and Circuits In normal telephone service, the local loops are two wire circuits, on which a single telephone call can be transmitted in both directions. If the distance between the subscribers is substantial, the amplifiers (repeaters) are necessary to compensate the attenuation.  As the amplifiers are unidirectional, for two-way communication, four-wire transmission is necessary. The switching equipment in the local exchange and the line from subscriber to local office (local loops) are two wire operation. The local exchange will switch the subscriber loop to a toll connecting trunk. This is also a two-wire transmission. The toll offices are interconnected with inter tool trunks (which connects towns and cities). These trunks are of four-wire transmission.

A four-wire circuit has amplifiers in its repeaters for each direction of transmission. The four wire circuits may be physical four wire or equivalent four wire. For short distances, actual four wires used for transmission is referred as physical four wire circuits. But for long distance trunks  physical four wire is undesirable and usually equivalent eq uivalent four wire transmission is used, needing one pair of wires only. The two directions of transmission use different frequency bands so that they do not interfere with each other. The two directions are separated in frequency rather than space. At the toll office, the two wires are converted into four wire for long transmission. A hybrid coil accomplishes this conversion.

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Hybrid transformer: While connecting the two wire circuit to the four wire circuit, a loop may  be created and the signal could circulate round the loop, loop , results in continuous oscillation known as singing. The hybrid transformer (two cross connected transformer) and balancing network together acts as a four wire/two wire terminating set and eliminates the singing problem. Crossconnected transformer windings results in zero current in the line balance impedance. The power thus divides equally between the input of the send amplifier and the output of the receive amplifier, where it has no effect. The price of avoiding singing is thus 3 dB losse in each direction of transmission together with other loss in transformers.

Noise  Noise in electrical terms may be defined as any unwanted introduction of energy tending to interfere with the proper reception and reproduction of transmitted signals.  Noise is mainly of concern in receiving system, where it sets a lower limit on the size of signal that can be usefully received. Even when precautions are taken to eliminate noise from faulty connections or that arising from external sources, it is found that certain fundamental sources of noise are present within electronic equipment that limit the receivers sensitivity.

Classification of noise EXTERNAL NOISE   

 Noise created outside the receiver   External noise can be further classified as:   Atmospheric   Extraterrestrial   Industrial

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ATMOSPHERIC NOISE   

Atmospheric noise or static is generally caused by lightning discharges in thunderstorms and other natural electrical disturbances occurring in the atmosphere.   Since these processes are random in nature, it is spread over most of the RF spectrum normally used for broadcasting.

EXTRATERRESTRIAL NOISE

  COSMIC NOISE   SOLAR NOISE  Solar Noise   

Under normal conditions there is a constant noise radiation from sun, simply because it is a large body at a very high temperature (Over 6000°C on the surface, it therefore radiates over a very broad frequency spectrum which includes frequencies we use for communication.

 

Due to constant changing nature of the sun, it undergoes cycles of peak activity from which electrical disturbances erupt, such as corona flares and sunspots. This additional noise produced from a limited portion of the sun, may be of higher magnitude than noise received during periods of quite sun.

Cosmic Noise   

Sources of cosmic noise are distant stars ( as they have high temperatures), they radiate RF noise in a similar manner as our Sun, and their lack in nearness is nearly compensated  by their significant number.   The noise received is called Black Body noise and is distributed fairly uniformly over the entire sky. INDUSTRIAL NOISE   

This noise ranges between 1 to 600 MHz (in urban, suburban and other industrial areas) and is most prominent.   Sources of such Noise: Automobiles and aircraft ignition, electric motors, switching equipment, leakage from high voltage lines and a multitude of other heavy electrical machines.   The Noise is produced by the arc discharge present in all these operations. ( this noise is most intense industrial and densely populated areas)

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INTERNAL NOISE   

 Noise created by any of the active or passive devices found in receivers.   Such noise is generally random, impossible to treat on individual voltage basis, but easy to observe and describe statistically. Because the noise is randomly distributed over the entire radio spectrum therefore it is proportional to bandwidth over which it is measured.

 

1.  2.  3.  4. 

Internal noise can be further classified as: Thermal Noise Shot Noise Low frequency or flicker Noise Burst Noise

Thermal Noise   

The noise generated in a resistance or a resistive component is random and is referred to as thermal, agitation, white or Johnson noise.   CAUSE :

  The free electrons within an electrical conductor possess kinetic energy as a result of heat

•

exchange between the conductor and its surroundings.   Due to this kinetic energy the electrons are in motion, this motion is randomized through collisions with imperfections in the structure of the conductor. This process occurs in all real conductors and gives rise to conductors resistance.   As a result, the electron density throughout the conductor con ductor varies

•

•

Shot Noise  

Shot noise is random fluctuation that accompanies any direct current crossing potential  barrier. The effect occurs because the carriers (electrons and holes in semiconductors) do not cross the barrier simultaneously but rather with random distribution in the timing of each carrier, which gives rise to random component of current superimpose on the steady current.   In case of bipolar junction transistors, the bias current crossing the forward biased emitter  base junction carries the shot noise.   When amplified, this noise sounds as though a shower of lead shots were falling on a metal sheet. Hence the name shot noise.

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Flicker Noise (1/f noise)   

This noise is observed below frequencies of few kilohertz and its spectral density increases with decrease in frequency. For this reason it is sometimes referred to as 1/f noise.   The cause of flicker noise is not well understood and is recognizable by its frequency dependence. Flicker noise becomes significant at frequency lower than about 100 Hz. Flicker noise can be reduced significantly by using wire-wound or metallic film resistors rather than the more common carbon composition type. In semiconductors, flicker noise arises from fluctuations in the carrier densities (holes and electrons), which in turn give rise to fluctuations in the conductivity of the material. i.e the noise voltage will be developed whenever direct current flows through the semiconductor, and the mean square voltage will be proportional to the square of the direct current.   In Electronic devices, it shows up as a low frequency phenomenon, as the higher frequencies overshadowed by white noise from other sources.

Burst Noise    It consists of sudden step-like transitions between two or more discrete voltage or current levels, as high as several hundred microvolts, at random and unpredictable times. Each shift in offset voltage or

Singing Singing is the result of sustained oscillation due to positive feedback in telephone amplifiers or amplifying circuits. The feedback is the result of excessive receive signal feeding back through the hybrid to the transmit side, which is then amplified setting up oscillations. Circuits that sing are unusable and promptly overload multichannel carrier (FDM) equipment. Singing may be regarded as echo that is completely out of control. This can occur at the frequency at which the circuit is resonant. Under such conditions the circuit losses at the singing frequency are so low that oscillation will continue, even after cessation of its original impulse.

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Jitter In the context of digital transmission, jitter  is defined as short-term variation of the sampling instant from its intended position in time or phase. Longer-term variation of the sampling instant is called wander . Jitter can cause transmission impairments such as:   Displacement of the ideal sampling instant. This leads to degradation in system error performance   Slips in timing recovery circuits, manifesting in degraded error performance   Distortion of the resulting analog signal after decoding at the receive end of the circuit. In IP network, simply stated, jitter  stated,  jitter   is the variation of packet inter-arrival time. Jitter is one issue that exists only in packet-based networks. While in a packet voice environment, the sender is expected to reliably transmit voice packets at a regular interval (for example, send one frame every 20 ms). These voice packets can be delayed throughout the packet network and not arrive at that same regular interval at the receiving station (for example, they might not be received every 20 ms). The difference between when the packet is expected and when it is actually received is jitter  is jitter .

In the Figure, the amount of time it takes for packets A and B to send and receive is equal (D1=D2). Packet C encounters delay in the network, however, and is received after   it is expected. This is why a jitter buffer , which conceals inter-arrival packet delay variation, is necessary. Voice packets in IP networks have highly variable packet inter-arrival intervals.

  Transmission Lines At high frequencies, the wavelength is much smaller than the circuit size, resulting in different phases at

different locations in the circuit. Hence Quasi-static circuit theory cannot be applied. We need to use transmission line theory. A transmission line is a two-port network connecting a generator circuit at the sending end to a load at the receiving end. Unlike in circuit theory, the length of a transmission line is of utmost importance in transmission line analysis. Common types of transmission tr ansmission lines are:   Coaxial line   Two wire line   Parallel plate line   Microstrip line   Waveguide 37

 

a short short section Δz of a transmission line can be drawn as :

Using KVL and KCL circuit theorems,

Transmission Line Parameters

Z0  is characteristic impedance and γis the complex propagation propagatio n constant whose real part attenuation constant (Np/m) and whose imaginary part quantities are functions of ω.

is the is the phase constant (rad/m). Generally, these

Z0 and γ are the two most important parameters of a transmission line. They depend on the distributed  parameters  paramete rs (RLGC) of the line itself and ω but not the length of the line.

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39

 

 

Input impedance seen ―looking‖ towards load at z = -l .

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Signal Multiplexing When more than one application or connections share the capacity of one link it is called multiplexing.   This results in better utilization of resources.   A typical example is, many conversations over telephone line, trunk line, wireless channel, etc.

 Mult  M ultipl iple exing xi ng  is a method of combining multiple streams of information for transmission over a shared medium.

Demultiplexing  performs the reverse function: split a combined stream arriving from a shared medium into the original information streams.

The concept of multiplexing in which independent pairs of senders and receivers share a transmission medium

Types of Multiplex Multiplexing ing   Frequency Division Multiplexing   Time Division Multiplexing   Wavelength Division Multiplexing

Frequency Division Multiplexing (FDM)   Several stations can transmit simultaneously without interfering with each other provided they use separate carrier frequencies (separate channels).   In data communications FDM is implemented by sending multiple carrier waves over the same copper wire.   At the receiver‘s end, demultiplexing is performed by filtering out the frequencies other than the one carrying the expected transmission.

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  Rather than a single frequency, each channel is assigned a contiguous range of frequencies.   Channels are separated from each other by guard bands to make sure there is no interference among the channels.

Time Division Multiplexing (TDM) It means dividing the available transmission time into time slots, and allocating a different slot to each transmitter. Time-division multiplexing (TDM) is the time interleaving of samples from several sources so that the information from these sources can be transmitted serially over a single communication channel. One method for transmitters to take turns is to transmit in roundrobin order.

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Wavelength Division Multiplexing   In optical transmissions, FDM is known as Wavelength Division Multiplexing (WDM).   With light different frequencies correspond to different colors.   Several transmissions can be sent over the same fiber by using different light colors ( ), and combining into a single light stream.   Prisms are used as multiplexors and demultiplexer.

North American TDM System: T1 8

The North American T1 (DS1) system is a 24-channel PCM system using 8-level coding (2 = 256 quantizing steps or distinct PCM code words). Supervisory signaling is ―in - band‖  band‖ where bit 8 of every sixth frame is ―robbed‖ for supervisory signaling. The T1 format shown in Figure  below has one bit added ad ded as a framing bit. (This is that indication to tell the distant end receiver where the frame starts.) It is called the ―S‖ bit. The T1 frame then consists of: (8 x24) + x24)  + 1 = 193  bits making up a full sequence or o r frame. By definition, 8000 frames are transmitted per second (i.e., 4000 x 2, the Nyquist sampling rate), so the bit rate of T1 is: 193 x 8000 = 1544000 bps or 1.544 Mbps

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European TDM System: E1 The E1 European PCM system is a 32-channel system. Of the 32 channels, 30 transmit speech (or data) derived from incoming telephone trunks and the remaining 2 channels transmit synchronization-alignment and signaling information. Each channel is allotted an 8-bit time slot (TS), and as tabulate Time Slot 0 through 31 as follows:

E1 in its primary rate format transmits 32 channels of 8-bit time slots. An E1 frame therefore has 8 X 32 = 256 bits. There is no framing bit. Framing alignment is carried out in TS0. The E1 bit rate to the line is: 256 X 8000 = 2048000 bps or 2.048 Mbps

Higher-order PCM A primary rate of 1.5 or 2 Mbps is usually too slow for transmission in trunk or even in local networks. Higher-order PCM multiplex is developed out of several primary multiplex sources. Primary multiplex is typically DS1 in North America and E1 in Europe. The first standardized digital higher-order transmission hierarchy is known as plesioch as plesiochronous (PDH).   ronous digital digital hierarchy hierarchy (PDH).

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European PDH for Higher-Order Multiplexing:   The basic principle of the European standard for higher-order multiplexers is that each multiplexer stage takes four signals of a lower data rate and packs them together into a signal at a data rate that is a little bit over four times as high, as shown in the Figure below.   The reason for high data rate is in addition to tributaries, aggregate frames contain frame alignment information and justification information. The tributary frequencies may differ slightly and their frequencies must be justified to the higher-order frame. This process, called justificati called justification on

or  stuffing, adds a number of justification bits to each tributary in order to make the average tributary data rates exactly the same.   In the demultiplexer these justification bits are extracted and the original data rate for each tributary is generated.   At each hierarchy level the tributary signals are bit interleaved to the aggregate data stream, which means that the aggregate data stream contains one bit from tributary 1, one bit from tributaries 2, 3, and 4, and then again from tributary 1, and so on.   Additional bits are needed in the frame for frame synchronization (frame alignment) and  justification,  justificat ion, and therefore the next level has a slightly higher rate than four times the nominal tributary rate

North American PDH for are Higher-Order The North American PDH isMbps), shown in  below. Higher-order rates DS1 (1.544 Multiplexing Mbps), DS2 :(6.132 Mbps), DS3 (44.736 andFigure DS4 (274.176 Mbps).

The higher-order bit rate for each multiplexer is a little bit higher than the sum of the tributary data rates. The aggregate data stream at each level contains, in addition to tributary signals, framing information and 45

 

the stuffing bits that are used to justify tributary data rates, which may have slightly different data rates, into the higher-order frame. In the demultiplexer these stuffing bits are stripped off and the original tributary rate is produced.

SDH and SONET

Problems with the PDH standards:   Access to a tributary rate requires step-by-step demultiplexing because of stuffing (justification).   Optical interfaces are not standardized but vendor specific.   To use optical cables, a separate multiplexer for each level (e.g., multiplexing from 2 to 140 Mbps in European PDH requires 21 pieces of multiplexing equipment) and separate line terminals are needed.   American and European standards are not compatible.    Network manageme management nt features and and interfaces interfaces are vendor vendor dependent. dependent.   High data rates (above 140 or 274 Mbps) are not standardized. ANSI started to study a new transmission method in the middle of the 1970s to utilize optical networks and modern digital technology more efficiently. This system is called the  synchronous optical network (SONET) and it is used in the United States. ITU-T made its own worldwide standard, called SDH, by the end of the 1980s. SDH is actually an international extension of SONET and it was based on SONET but adapted to European networks.

Multiplexing Scheme SDH of SDH are called  synchrono The transmission data in streams  synchronous us transport modules (STMs) and they are exact multiples of STM-1 at the 155.52-Mbps data  data  rate. STM-1 data are simply byte interleaved with  with   other STM-1 data streams to make up a higher transmission data rate; no   additional framing information is added. Byte interleaving means that, for   example, an STM-4 signal contains a byte (8 bits) from the first STM-1 tributary,  tributary, then from the second, third, and fourth tributaries, and then again from  from the first one. The demultiplexer receives all STM-1 frames independently.  independently.   The STM-1 frame is repeated 8,000 times a second, a rate equal to  to  the PCM sampling rate. This makes each 8-bit speech sample visible in a  a   155.52 Mbps data stream. When PCM coding is synchronized to the same  same  source as SDH systems, we can demultiplex one speech channel just by picking  picking   up 1 byte from each STM-1 frame. The frame contains frame alignment  alignment  information and other information such as management data channels and  and   pointers pointers that tell the location of tributaries in the frame.  frame.   If tributaries are not synchronous with the STM-1 frame, a  pointer (a  (a  binary binary number) in a fixed location in the STM-1 frame tells the location of   each tributary. By looking at the value of this pointer, we can easily find the   desired tributary signal. This is a great

advantage over PDH systems, which  which   require step-by-step demultiplexing (to separate information and

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stuffing  bits) stuffing  bits) to the level of the tributary tributary that we want want to take out from the high high data-rate stream. stream.  A single STM-1 carry 63 E-1 signals. signals. Multiplexing Scheme in SONET   The synchrono The  synchronous us transport signal signal level 1 (STS-1) is the basic SONET module that corresponds to STM-1 of SDH. These modules have a bit rate of 51.840 Mbps and they are multiplexed synchronously into higher-order signals STS-N. Each STS-N signal has a corresponding optical signal called an optical   carrier (OC-N) for optical transmission transmission

The basic unit of transmission The basic unit of framing in SDH is a STM-1 (Synchronous Transport Module, level 1), which operates at 155.52 megabits per second (Mbit/s). SONET offers basic unit of transmissio transmission, n, the STS-1 (Synchronous Transport Signal 1) operating at 51.84 Mbit/s — exactly exactly one third of an STM-1. SONET Frame Format STS-1 STS-3 STS-12 STS-48 STS-192

SDH Frame Format STM-0 STM-1 STM-4 STM-16 STM-64

Line Rate (Mbps) 51.840 155.520 622.080 2488.320 9953.280

Framing In STS-1, the frame is 810 octets in size, while the STM-1 frame is 2,430 octets in size. For STS-1, the frame is transmitted as three octets of overhead, followed by 87 octets of payload. This is repeated nine times, until 810 octets have been transmitted, taking 125 µs. In the case of an STM-1, which operates three times faster than an STS-1, nine octets of overhead are transmitted, followed by 261 octets of payload. This is also repeated nine times until 2,430 octets have  been transmitted, transmitted, also also taking taking 125 µs. For both SONET and SDH, this is often represented by displaying the frame graphically: as a block of 90 columns and nine rows for STS-1, and 270 columns and nine rows for STM1

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Fig: Frame STM-1

Fig: Frame STS-1 Transport overhead

The transport overhead is used for signaling and measuring transmission error rates, and is composed as follows: Section overhead

Called RSOH (regenerator section overhead) in SDH terminology: 27 octets containing information about the frame structure required by the terminal equipment.

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Line overhead

Called MSOH (multiplex section overhead) in SDH: 45 octets containing information about error correction and Automatic Protection Switching messages (e.g., alarms and maintenance messages) as may be required within the network. AU Pointer

Points to the location of the J1 byte in the payload (the first byte in the virtual container).

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SIGNALLING SYSTEM In order to route telephone traffic through the Public Switched Telephone Network (PSTN), it is necessary to communicate with the switches that make up the PSTN. Signaling is a means for transferring network-related information between switching nodes, and also between the end office switches and their subscribers. Signaling is used to do the following: •  •  •  •  • 

Request service from the central office switch (via going off-hook). Provide central office switch with the information necessary to route a telephone call (via DTMF addressing digits in a specific format). Alert destination address of incoming call (ringing). Provide status information and call supervision for billing. Manage network lines/trunks (set up and teardown calls).

The ITU-T defines signaling as, "The exchange of information (other than by speech) specifically concerned with the establishment, release and other control of calls, and network management, in automatic telecommunications operation."   All the control signals used within or between communication equipment, whose function are to setup communication   Signals except bearer information (voice, video, data, fax) which are used to control the action of the exchange.

SWITCHING CONCEPT The Exchange   

serves to connect various forms of access devices analog telephone handsets, digital subscriber lines, primary and basic rate ISDN, etc.

Signaling   

is the process of communication between user and network or between two networks

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In order to be able to perform the switching function, a communication will be required between the calling subscriber and his own switching unit. This is the User-to-Network Interface (UNI). A communication will also be required between each switching unit and the next one in the call sequence. This is the Network-to-Network Interface (NNI) FUNCTIONAL CLASSIFICATION

In Channel Signalling (CAS = Channel Associated Signalling)

  Call setup information (off-hook, dial tone, address digits, ring back, busy) is transmitted in the same band of frequencies as used by the voice signal.   Voice (talk) path is cut over only when the call setup is complete, using the same path that the call setup signals used.   SF (single-frequency) signalling uses tones to represent on-hook or payphone deposits.   MF (multi-frequency) signalling is used for switch-to-switch call setup

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Common Channel Signalling

CCS employs a separate, dedicated path for signaling. (See Figure below) Voice trunks are used only when a connection is established. ‖Separate‖ packet-switched packet -switched signalling signalling network which is independent of circuit switched connections is used to establish a call. If part of the switching network is out of operation, CCS can search for a different route between originating and terminating exchanges and restore the lost circuit connection.

Comparison: CCS7 Vs CAS

CCS #7

CAS

Special hardware and software is required for the message transport facility

 No Special Special hardware and software is required for the message transport facility

Only few signalling channels are required  between to Exchanges Exchanges

Line signalling requires all channel 16 of all PCM links between the 2 Exchanges

Very fast signalling (1 message is sent in 2 - 4 ms)

Very fast signalling (100ms per digit)

CCS can be used for other type of information, e.g. charging, maintenance etc.

CAS only used for call handling information

Failure will have major impact upon the system. They will influence many circuits. Therefore, protection mechanism has to be foreseen.

Failures will have a limited impact upon the system. They will only affect the related PCM link

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CCS7 PROTOCOL STACK

 

Each layer is designed to perform certain group of functions, and each designed to be able to offer its functionalities to other modules.

 

The fundamental principle of CCS #7 is the division of functions into a common "Message Transfer Part (MTP)" on one hand, and separate "User Parts" on the other hand. The same common MTP module is extracted and serves as a transport system, even for different User Parts

MTP Level 1   

Is equivalent to the OSI Physical Layer. MTP Level 1 defines the physical, electrical, and functional characteristics of the digital signaling link.

MTP Level 2   

Ensures accurate end-to-end transmission of a message across a signaling link. Level 2 implements flow control, message sequence validation, and error checking. When an error occurs on a signaling link, the message (or set of messages) is retransmitted. MTP Level 2 is equivalent to the OSI Data Link Layer.

MTP Level 3   

Provides message routing between signaling points in the SS7 network. MTP Level 3 re-routes traffic away from failed links and signaling points and controls traffic when congestion occurs. MTP Level 3 is equivalent to the OSI Network Layer.

SCCP provides connectionless and connection-oriented network services and global title translation (GTT) capabilities 3. A global title A is an address which is translated SCCP into a destination pointabove codeMTP and Level subsystem number. subsystem number uniquely by identifies an 53

 

application at the destination signaling point. SCCP is used as the transport layer for TCAP-based services. The ISDN User Part (ISUP) defines the protocol used to set-up, manage, and release trunk circuits that carry voice and data between a calling party and a called party. ISUP is used for both ISDN and non-ISDN calls. TCAP supports the exchange of non-circuit related data between applications across the SS7 network using the SCCP connectionless service. Queries and responses sent between SSPs and SCPs are carried in TCAP messages.

CCS7 ARCHITECTURE

Signalling End Point (SEP)

  This is a local node in a telecommunication network to which subscriber lines are attached and served by CCS7. It is a source or a sink of signalling traffic. Signalling Transfer Point (STP)

  All CCS7 messages travel from one SEP to another through the services of STP. The STP switches the message as received from the various SEP‘s through the network to their appropriate destination, that is, to the SEP, or perhaps another STP.   In STP, there is no switching oriented processing of the signalling message, that is, the contents of the signalling messages is not examined. Three levels of STP exist: National, International and Gateway. Signalling points are deployed in pairs for redundancy and diversity. To make sure the CCS7 network is always operational, alternate and multiple paths are provided. Thus, SEP‘s are connected to atleast 2 STP‘s. Further, multiple paths exist between different STP‘s.  STP‘s.  

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Signalling Unit

Signalling messages delivered by superior hierarchical levels (the user parts) are transferred over the signalling links in variable length ―Signalling Units‖. A Signal Unit is nothing more than a packet, to be transmitted over the CCS7 network, but because of many applications, different packet structures and capabilities are anticipated. In fact, CCS7 uses 3 different structures of Signal Units. 1.  Message Signal Unit (MSU) MSU's will transport information sent from a certain User Part of a Signalling Point to a User Part of another Signalling Point

2.  Link Status Signal Unit (LSSU) LSSU are sent between between two Signalling Signalling Points to indicate indicate the status of the signalling link on which it is carried. Therefore the LSSU is only of significance between two Signalling Points and will not be transmitted further over the network. More precisely, the LSSU will be used for the following reasons: In case of congestion, a busy status will be sent by means of LSSU until the condition has disappeared.   During the initial alignment phase of the link, LSSU are used in order to properly start up the link  

3.  Fill-in Signal Unit (FISU) The signalling link will continuously send Signal Units between the two Signalling Points even when there is no payload to be delivered and the CCS 7 network is idle. In this case empty empty Signal Units, called FISU will be sent. They do not contain any information informatio n part. By means means of these FISU, the integrity of signalling links is constantly monitored in CCS 7. Otherwise, a link could degrade in the absence of traffic and this would only be noticed at a transmission, which would fail.

Fields of Signalling Unit   An 8 bit opening flag "01111110" is used as message separator. There is only one opening flag, and no closing flag.   The "Forward Sequence Number (FSN)" is the sequence number of the Signal Unit in which it is

carried.

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 

The "Backward Sequence Number (BSN)" is the sequence number of a Signal Unit being acknowledged.   Both the FSN and the BSN are 7 bits in length and thus can span a cyclic sequence ranging from 0 to 127   The "Forward Indicator Bit (FIB)" and the "Backward Indicator Bit (BIB)" , together with the FSN and the BSN are used to perform the error control The indicator bits will be used to request a retransmission. During normal conditions, both indicator bits should be the same   When a retransmission is being requested, the Signal Unit being sent by the Signalling Point requesting the retransmission will have an inverted BIB.   The "Length Indicator (LI)" of 6 bits is included in the Signal Unit. It indicates the number of octets following the LI and preceding the check bits. LI = 0 FISU LI = 1 or 2 LSSU LI  3 MSU   Spare fields are always coded as 0.   Information Part of the message contains the actual information to be sent to the destination. Depending on the type of Signal Unit, three different contents can be found in the information    

 part The SIO ―Service Information Octet‖ is only on ly present in an MSU and is used by MTP-3 to identify the type of protocol, used at level 4 (i.e. TUP, ISUP, and TCAP)   The ―Signalling Information Field‖ SIF is only present in an MSU and is used to transfer the actual control information, as well as the routing label used by MTP-3   The ―Status Field‖ SF carries the link status information for the link on which it is carried. It is only present in an LSSU   Check Bits are used to detect errors in the information, information , CRC the message message is added to extended with a frame check sequence

 

PULSE DIALING

Pulse Dialing is an inband signaling technique. It is used in analog telephones that have a rotary dialing switch. The large numeric dial wheel on a rotary dial telephone spins to send digits to place a call. These digits must be produced at a specific rate and within a certain level of tolerance. Each pulse consists of a "break" and a "make", which are achieved when the local loop circuit is opened and closed. The break segment is the time during which the circuit is open. The make segment is the time during which the circuit is closed. Each time the dial is turned the bottom of the dial closes and opens the circuit leading to the CO switch or PBX switch. The number of consecutive opens and closes or breaks and makes represents the dialed digits Therefore, if the digit 3 is dialed, the switch is closed and opened three times. Figure next represents the sequence of  pulses that occur when a digit 3 is dialed with pulse dialing. dialing.

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This illustration displays the two terms, make and break. When the telephone is off−hook, a make occurs and the caller receives a dial tone from the CO switch. Then the caller dials digits, which generate sequences of makes and breaks that occur every 100 milliseconds (ms). The break and make cycle must correspond to a ratio of 60 percent break to 40 percent make. Then the phone stays in a make state until another digit is dialed or the phone is put back to an on-hook (equivalent to a break) state. Dial pulse addressing is a very slow process because the number of pulses generated equates to the digit dialed. So, when a digit 9 is dialed, it generates nine make and break pulses. A digit 0 generates ten make and break  pulses. DTMF DIALING

DTMF dialing is an inband signaling technique just like pulse dialing. This technique is used in analog telephone sets that have a touch-tone pad. This dialing technique uses only two frequency tones per digit, as shown in Figure below. Each button on the keypad of a touch-tone pad or a push-button telephone is associated with a set of high and low frequencies. On the keypad, each row of the key is identified by a low frequency tone, and each column is associated with a high frequency tone. The combination of both tones notifies the telephone company of the number called, hence the term dual tone multi-frequency. Therefore, when digit 0 is dialed, only frequency tones 941 and 1336 are generated instead of the ten make and break pulses generated by pulse dialing. The timing is still a 60ms break and 40ms make for each frequency generated. These frequencies were selected for DTMF dialing based on their insusceptibility to normal background noise.

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Telephone Traffic One of the most important steps in telecommunication engineering practice is to determine the number of trunks required on a route or connection between exchanges termed as dimensioning the route. To dimension a route correctly, we must have some idea of its usage, that is, how many people will wish to talk at once over the route. The usage of a transmission route or a switch brings us into the realm of traffic engineering, and the usage may be defined by two  parameters: (1)  Calling rate or rate or the number of times a route or traffic path is used per unit period, or, more  properly defined, ―the call intensity per traffic path during the busy hour‖.  hour‖.  (2)  Holding  Holding time  time  or ―the duration of occupancy of a traffic path by a call,‖ or sometimes, ―the average duration of occupancy of one or more paths by calls.‖ A traffic path is ―a channel, time slot, frequency band, line, trunk, switch, or circuit over which individual communications pass in sequence.‖ Carried traffic is the volume of traffic actually carried by a switch, and Offered traffic is volume of traffic offered to a switch. To dimension a traffic path or size a telephone exchange, we must know the traffic intensity representative of the normal busy hour.

Busy Hour Definitions 1. Busy Hour . The busy hour refers to the traffic volume or number of call attempts, and is that continuous 1-h period lying wholly in the time interval concerned for which this quantity (i.e., traffic volume or call attempts) is greatest. 2. P eak B usy H our . The busy hour each day; it usually is not the same over a number of days. 3. T i me C ons onsii ste stent nt B usy H our . The 1-h period starting at the same time each day for which the average traffic volume or call-attempt count of the exchange or resource group concerned is greatest over the days under consideration.

Blockage, Lost Calls, and Grade of Service Assume that an isolated telephone exchange serves 5000 subscribers and that no more than 10% of the subscribers wish service simultaneously. Therefore, the exchange is dimensioned with sufficient equipment to complete 500 simultaneous connections. Each connection would be, of course, between any two of the 5000 subscribers. Now let subscriber 501 attempts to originate a call. He/she cannot because all the connecting equipment is busy, even though the line he/she wishes to reach may be idle. This call from subscriber 501 is termed a lost call or blocked 58

 

call . He/she has met blockage. Grade of service expresses the probability of meeting blockage during the BH and is expressed  by the letter p letter p.. A typical grade of service is is p  p = 0. 0.01. This means that an average of one call in 100 will be blocked or ―lost‖ during the BH. the BH.

In loss system, the carried traffic by the network is generally lower than the actual traffic offered to the network by the subscribers. The overload traffic is rejected and hence not carried by the network. The amount of traffic rejected by the network is an index of quality of service offered  by the network. This is termed as GOS and is defined as the ratio of lost traffic to the offered traffic           Where,  AO = Offered Traffic  AC  = Carried Traffic  AO - AC = Lost Traffic Call Completion Rate (CCR) CCR is defined as the ratio of the number of successful calls to the number of calls attempts. Busy Hour Call Attempts (BHCA) BHCA is defines as the number of calls attempts in thse BUSY HOUR. It is a parameter that measures the processing capacity of a common control or a stored program control system of an exchange. Average Busy Hour Calls = BHCA X CCR Busy Hour Calling Rate = (Average Busy Hour Calls)/ (Total Number of Subscribers)

The telephone traffic is defined as the aggregate of telephone calls over a group of circuits or trunks with regard to the duration of calls as well as their number. The traffic measured over a  period of one hour is called traffic intensity (A) expressed as: as:          Where, A = Traffic intensity  = Call Arrival Rate H = Call Holding Time

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Units of Traffic Intensity Hour. r. Erlang: One erlang of traffic is generated when a circuit is occupied for one Hou CCS: Centum Call Second is traffic generated when a circuit is occupied for 100 seconds. Cm: Call Minute is traffic generated when a circuit is occupied for one minute

Smooth, Rough, and Random Traffic Traffic probability distributions can be divided into three distinct categories: (1) smooth, (2) rough, and (3) random. Each may be defined by α, the VMR (Variance to Mean Ratio). For smooth traffic, α is less than 1. For rough traffic, α is greater than 1. When α is equal to 1, the traffic distribution is called random random.. The Poisson distribution function is an example of random traffic where VMR = 1. Rough traffic tends to be peakier than random or smooth traffic. For a given grade of service, more circuits are required for rough traffic because of the greater spread of the distribution curve (greater dispersion).

Lost Call Cleared Model: Erlang B This is a type of trunked systems which offers no queuing for call requests. That is, for every user who requests service is given immediate access to a channel if one is available. If no channels are available, the requesting user is blocked without access and is free to try again later. This type of trunking is called lost calls cleared or blocked calls cleared and it assumes that calls arrive as determined by a Poisson distribution. 60

 

Furthermore, it is assumed that   There are infinite number of users   Memoryless arrivals of request   System has full availability for every request   The probability of a user occupying a channel is exponentially distributed   There are a finite number of channels and   Inter-arrival time of calls are independent of each other This is known as M/M/m queue used in Erlang B traffic model. The Erlang B formula determines the  probability that a call is blocked and is a measure of the GOS for f or a trunked system which provides no queuing for blocked calls.

    []        ∑  Where, C is the number of trunked channels offered by a trunked radio system and A is the total offered traffic. 

DELAY SYSTEMS: Erlang C The delay system places the call or message arrivals in a queue if it finds all  N servers (or lines) occupied. This system delays non-serviceable requests until the necessary facilities become available. These systems are variously referred to as delay system, waiting-call systems and queueing systems. The delay systems are analysed using queueing theory which is sometimes known as waiting line theory. Consider that there are k calls (in service and waiting) in the system and  N lines to serve the calls. If k ≤ N , k lines are occupied and no calls are waiting. If k > N , all N all N lines are occupied and k  –   N N calls waiting. Hence a delay operation allows for greater utilization of servers than does a loss system. Even though arrivals to the system are random, the servers see a somewhat regular arrival pattern. A queueing model for the Erlang delay system is shown in the figure below:

The basic purpose of the investigation of delay system is to determine the probability distribution of waiting times. From this, the average waiting time was random variable can be easily determined. The waiting times are dependent on the following factors: 1. Number of sources 2. Number of servers 3. Intensity and probabilistic nature of the offered traffic 61

 

4. Distribution of service times 5. Service discipline of the queue. The probability of waiting (the probability of finding all lines occupied) is

ROUTING PLAN Routing planning refers to the procedures that determine which path in a network are assigned to  particular connections. The T he switching centres may use fixed routes to each e ach destination. Adaptive routing may be employed in which each exchange may use different routes for the same destination, depending upon traffic conditions. For effective routing of a call, some form of interconnection of switching exchanges are required. Three basic topologies are adopted for interconnecting exchanges. Exchanges are interconnected  by group of trunk lines referred as trunck groups. Two trunk groups are required between any two exchanges. Mesh, star and mixed or hierarchical are the three basic topologies. Mesh-connected network. This is also called fully connected topology. The advantage of mesh network is that each station has a dedicated connection to other stations. Therefore, this topology offers the highest reliability and security. If one link in the mesh topology breaks, the network remains active. A major or disadvantage of this topology is that it uses too many connections and therefore requires great deal of wiring, espeically when the number of station increases. The mesh topology requires N(N –  N(N –  1)/2  1)/2 connections.

Star topology. In star network, the number of lines is equal to the number of stations. As shown, a star connection utilises an intermediate exchange called a tandom exchange. Through the tandom exchange (TE) all other exchanges communicate.

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Hierarchical networks. Many star networks may be inter connected by using an additional tandom exchange, leading to two level star network. An orderly construction of multilevel star networks leads to hierarchical networks. Hierarchical networks are capable of handling heavy traffic with minimal number of trunk groups. The hierarchical network requires more switching

nodes, but achieves significant savings in the number of trunks.

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NUMBERING PLAN The numbering plan is used to identify the subscribers connected in a telecommunication network. The main objective of numbering plan by any nation is to standardise the number length wherever practical according to CCITT recommendations. Other objectives includes (a) to meet the challanges of the changing telecom environment (b) to meet subscriber needs for a meaningful and user friendly scheme (c) to reserve numbering capacity to meet the undefined future needs.

International Numbering Plan This plan has to be implemented irrespective of a country‘s national numbering plan and   implemented in accordance to the recommendations of ITU. With some standard international framework, subscribers from different countries can call each other. This plan makes it possible to access all countries with the same country code anywhere in the world. For the international numbering plan, the world has been divided into nine geographical area as given below. The general rule is that within each global region each country code starts with the same digit.

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The numbering format for international telephone number is

An internation telephone number starts with one to three digit country code followed by 9 to 12 subscriber number. The dialling procedure is that the international prefix pre fix ‗00‘ should be dialed first followed by the telephone number.

National Numbering Plane Each country decides for itself what kind of numbering plan it can have. A numbering plan may  be open, semi open or closed. Each country decides what rules to follow when issueing telephone numbers. Such a numbering plan is called national numbering plan. National numbering format is

 National significent number (NSN) is the combination of trunk code, exchange code and the number. The exchange code and line number together called as subscriber number (SN). NSN length varies from country to country.

CHARGING PLAN The cost of providing a telecommunication network consists of the capital cost and the current operating expenses. The capital cost includes switching systems, buildings, lines and land. Operating cost includes staff salaries, maintenance costs, water and electricity charges and miscellaneus expenses. All of these costs must be met by the income obtained by the telecom operator from its subscribers. The telecom operator charges the subscribers for its services by the following three ways. 1. An initial charge for providing a network connection (as installation charges) 2. A rental or leasing charge 3. Call charges. The charging methods for individual calls fall under two broad categories. 1. Duration independent charging 2. Duration dependent charging

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Data Communication SWITCHING TECHNIQUES When there are many devices, it is necessary to develop suitable mechanism for communication  between any two devices. One alternative is to establish point-to-point communication between each pair of devices using mesh topology. However, mesh topology is impractical for large number of devices, because the number of links increases exponentially given by        Where, n is the number of devices and L is the number of links connecting with the devices. A  better alternative is to use switching techniques leading to switched communication commun ication network. In the switched network methodology, the network consists of a set of interconnected nodes, among which information is transmitted from source to destination via different routes, which is controlled by the switching mechanism. The end devices that wish to communicate with each other are called st  sta ati ons. The switching devices are called nodes. Some nodes connect to other nodes and some are connected to stations.

The switching performed by different nodes can be categorized into the following three ttypes: ypes: 1.  Circuit Switching 2.  Packet Switching 3.  Message Switching

Circuit switching Technique Communication via circuit switching implies that there is a dedicated communication path  between the two stations. The path is a connected through a sequence of links between network nodes. On each physical link, a logical channel is dedicated to the connection. Circuit switching is commonly used technique in telephony, where the caller sends a special message with the address of the callee (i.e. by dialing a number) to state its destination. It involved the following three distinct steps:

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C i r cuit E st sta ablishm li shme ent nt::   To establish an end-to-end connection before any transfer of data. Some segments of the circuit may be a dedicated link, while some other segments may be shared.

Data transfer: Transfer data from the source to the destination. The data may be analog or digital, depending on the nature of the network. The connection is generally full-duplex.

C i r cuit di di sco sconne nnect ct:: Terminate connection at the end of data transfer. Signals must be propagated to deallocate the dedicated resources.

Thus, the actual physical electrical path or circuit between the source and destination host must  be established before the message is transmitted. This connection, once established, remains exclusive and continuous for the complete duration of information exchange and the circuit  becomes disconnected only when the source wants to do so.

 Message Switching Switching

In this switching method, a different strategy is used, where instead of establishing a dedicated  physical line between the sender and the receiver, the message is sent to the nearest directly 67

 

connected switching node. This node stores the message, checks for errors, selects the best available route and forwards the message to the next intermediate node.

The line becomes free again for other messages, while the process is being continued in some other nodes. Due to the mode of action, this method is also known as store-and-forward technology where the message hops from node to node to its final destination. Each node stores the full message, checks for errors and forwards it. In this switching technique, more devices can share the network bandwidth, as compared with circuit switching technique. Temporary storage of message reduces traffic congestion to some extent. Higher priority can be given to urgent messages, so that the low priority messages are delayed while the urgent ones are forwarded faster. Through broadcast addresses one message can be sent to several users. Last of all, since the destination host need not be active when the message is sent, message switching techniques improve global communications. However, since the message blocks may be quite large in size, considerable amount of storage space is required at each node to buffer the messages. A message might occupy the buffers for minutes, thus blocking the inter-nodal traffic. Basic idea: Each network node receives and stores the message

Determines the next leg of the route, and Queues the message to go out on that link. Advantages: Line efficiency is greater (sharing of links). Data rate conversion is possible. Even under heavy traffic, packets are accepted, a ccepted, possibly with a greater delay in delivery. Message priorities can be used, to satisfy the requirements, if any. Disadvantages : Message of large size monopolizes the link and storage

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Packet Switching The basic approach of packet switching is not much different from message switching. It is also  based on the same store-and-forward approach. However, to overcome the limitations of message switching, messages are divided into subsets of equal length called packets. In packet switching approach, data are transmitted in short packets (few Kbytes). A long message is  broken up into a series of packets as shown in Figure below. Every packet contains some control information in its header, which is required for routing and other purposes.

Main difference between Packet switching and Circuit Switching is that the communication lines are not dedicated to passing messages from the source to the destination. In Packet Switching, different messages (and even different packets) can pass through different routes rou tes

There are two basic approaches commonly used to packet Switching: 1.  Virtual-circuit packet switching 2.  Datagram packet switching. In virtual-circuit packet switching a virtual circuit is made before actual data is transmitted, but it is different from circuit switching in a sense that in circuit switching the call accept signal comes only from the final destination to the source while in case of virtual-packet switching this call accept signal is transmitted between each adjacent intermediate.

Virtual Circuit Packet Switching Networks An initial setup phase is used to set up a route between the intermediate nodes for all the packets  passed during the session between the two end nodes. In each intermediate node, an entry is registered in a table to indicate the route for the connection that has been set up. Thus, packets  passed through this route, can have short headers, containing only a virtual circuit identifier (VCI), and not their destination. Each intermediate node passes the packets according to the information that was stored in it, in the setup phase. In this way, packets arrive at the destination 69

 

in the correct sequence, and it is guaranteed that essentially there will not be errors. This approach is slower than Circuit Switching, since different virtual circuits may compete over the same resources, and an initial setup phase pha se is needed to initiate the circuit. As in Circuit Switching, if an intermediate node fails, all virtual circuits that pass through it are lost. The most common forms of Virtual Circuit networks are X.25 and Frame Relay, which are commonly used for public data networks (PDN).

Datagram Packet Switching Networks  This approach uses a different, more dynamic scheme, to determine the route through the network links. Each packet is treated as an independent entity, and its header contains full information about the destination of the packet. The intermediate nodes examine the header of the packet, and decide to which node to send the packet so that it will reach its destination. Thus, in this method, the packets don't follow a pre-established route, and the intermediate nodes don't have pre-defined knowledge of the routes the packets should be passed through.

Packets can follow different routes to the destination, and delivery is not guaranteed (although  packets usually do follow the same route, and are reliably sent). Due to the nature of this method, the packets can reach the destination in a different order than they were sent, thus they must be sorted at the destination to form the original message. This approach is time consuming since every node has to decide where to send each packet. The main implementation of Datagram Switching network is the Internet, which uses the IP network protocol.

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Advantages:

  Call setup phase is avoided (for transmission of a few packets, pa ckets, datagram will be faster).   Because it is more primitive, it is more flexible.   Congestion/failed link can be avoided (more reliable). Problems:   Packets may be delivered out of o f order.   If a node crashes momentarily, all of its queued packets are lost.

Integrated Services Digital Network Integrated Services Digital Network (ISDN)  (ISDN)  is comprised of digital telephony and data-transport services. ISDN involves the digitization of the telephone network, which permits voice, data, text, graphics, music, video, and other source material to be transmitted over existing telephone wires. ISDN applications include high-speed image applications, additional telephone lines in homes to serve the telecommuting industry, high-speed file transfer, and videoconferencing. Voice service is also an application for ISDN.

ISDN Devices ISDN devices include terminals, terminal adapters (TAs), network-termination devices, line termination equipment, and exchange-termination equipment. ISDN terminals come in two types. Specialized ISDN terminals are referred to as terminal equipment type 1 (TE1). NonISDN terminals, such as DTE, that predate the ISDN standards are referred to as terminal equipment type 2 (TE2). TE1 connect to the ISDN network through a four-wire, twisted-pair digital link. TE2s connect to the ISDN network through a TA. The ISDN TA can be either a standalone device or a board inside the TE2. If the TE2 is implemented as a standalone device, it connects to the TA via a standard physical-layer interface. Examples include EIA/TIA-232-C (formerly RS-232-C), V.24, and V.35. Beyond the TE1 and TE2 devices, the next connection point in the ISDN network is the network termination type 1 (NT1) or network termination type 2 (NT2) devices. These are networktermination devices that connect the four-wire subscriber wiring to the conventional two-wire local loop. The NT2 is a more complicated device that typically is found in digital private branch exchanges (PBXs) and that performs Layer 2 and 3 protocol functions and concentration services. An  NT1/2 device also exists as a single device that combines the functions of an NT1 and an NT2. ISDN specifies a number of reference points ( specifications for connecting two network elements) that define logical interfaces between functional groups, such as TAs and NT1s. ISDN reference points include the following:

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  R (Rate) —  (Rate) —  reference  reference point between non-ISDN equipment and a TA.   S (System) —  (System) —   reference reference point between user terminals and the NT2.   T (Terminal) —  (Terminal) —  reference  reference point between NT1 and NT2 devices.   U (User) —  (User)  —  reference   reference point between NT1 devices and line-termination equipment in the carrier network. The U reference point is relevant only in North America, where the NT1 function is not provided by the carrier network. Not N ot defined by ITU Figure below illustrates a sample ISDN configuration and shows three devices attached to an ISDN switch at the central office. Two of these devices are ISDN-compatible, so they can be attached through an S reference point to NT2 devices. The third device (a standard, non-ISDN telephone) attaches through the R reference point to a TA. Any of these devices also could attach to an NT1/2 device, which would replace both the NT1 and the NT2. In addition, although they are not shown, similar user stations are attached to the far-right ISDN switch.

Services There are two types of services associated with ISDN:   BRI   PRI

ISDN BRI Service The ISDN Basic Rate Interface (BRI) service offers two B channels and one D channel (2B+D). BRI B-channel service operates at 64 kbps and is meant to carry user data; BRI D-channel service operates at 16 kbps and is meant to carry control and signaling information, although it 72

 

can support user data transmission under certain circumstances. The D channel signaling  protocol comprises Layers 1 through 3 of the OSI reference model. BRI also provides for framing control and other overhead, bringing its total bit rate to 192 kbps. The BRI physical layer specification is ITU-T I.430.

ISDN PRI Service ISDN Primary Rate Interface (PRI) service offers 23 B channels and 1 D channel in North America and Japan, yielding a total bit rate of 1.544 Mbps (the PRI D channel runs at 64 kbps). ISDN PRI in Europe, Australia, and other parts of the world provides 30 B channels plus one 64kbps D channel and a total interface rate of 2.048 Mbps. The PRI physical layer specification is ITU-T I.431.

Type of ISDN Channels ISDN structure includes a central ISDN office. All the users are linked to this office through a digital pipe. This digital pipe may be of different capacities and may have different data transfer rates. These digital pipes between the customers and central office are organized into multiple channels of different size. ISDN standard defines the following three channel types: t ypes: 1.  B Channel 2.  D Channel 3.  H Channel

1.  B Channel  A Bearer Channel (B Channel) is defined at a rate of 64 kbps. It is the basic user channel and can carry any type of digital information in full duplex dup lex mode as long as the required transmission rate does not exceed 64 kbps. It can be used to carry digital data, digital voice and any other low data rate information. Four kinds of connections can be set up over a B channel: (a)  Circuit switched  The user places a call and a current switched connection is established with another network user. (b) Packet switched  The user is connected to a packet switching mode and data are exchanged with other users via X.25.

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(c)  Frame mode  The user is connected to a frame relay mode and data are exchanged with other users via LAPF. (d) Semi-permanent  This is a connection to another user set up by prior arrangement and not requiring-a call

establishment protocol. It is equivalent to a leased line.

2.  D Channel   A data channel (D channel) chann el) carry control signal for bearer channels.   This channel does not carry data and can be either 16 or 64 kbps, depending upon the user's need.   If we are using in-band signaling, the same cable carries data as well as control signals.   If we are using out-of-band signaling (as in ISDN), then two different cables carry data and control information.

3.  H-Channel 

  A Hybrid channel (H-channel) is provided for user information at higher bit rates. Hybrid channels are used for high data rate applications such as video, teleconferencing.

  There are three types of H-Channels depending on the data rates: i.  H0 with data rate of384 kbps ii.  H11 with data rate of 1536 kbps iii.  H12 with data rate of 1920 kbps

Broadband ISDN  –    –    –    –    –  

Second generation (broadband) ISDN Supports very high data rate ( 600 Mbps)  It uses fiber optic cable at all levels  Has packet switching orientation  It uses ATM to move data from one end point to another. 

Broadband ISDN Services

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Interactive services:   –   Conversational  •  Services that support real time data exchange ( phone class, video conferencing, real time data transfer)  –   Messaging  •  These services are store-and-forward services (voice mail, data mail, video mail)  –   Retrieval:  •  Information are retrieved from a central source (libraries, shared resources) Distributive services: are unidirectional    –   Without user control  •  Commercial TV, programming contents and times are decided by provider.  –   With user control  •  Pay TV •  Educational broadcasting Broadband ISDN Access Methods

• 

• 

Symmetrical:   –   155.52 Mbps full-duplex  •  Suitable for residential and small businesses subscribers  –   622 .08 Mbps full-duplex  •  Suitable for businesses subscribers that provide and receive distributive services Asymmetrical:   –   155.52 Mbps output/622.08 Mbps input full-duplex 

DSL DSL stands for ‗digital subscriber line‘. The term is a general te rm applied to a variety of different technologies used to achieve ‗broadband‘ or high speed digital transmission over 2-wire 2 -wire or 4-wire 4-wire ‗standard copper‘ public telephone network access lines.. lines..   A Digital Subscriber Line makes use of the current copper infrastructure to supply broadband services. DSL requires two modems, one at the phone company end and one at the subscribers end. The use of the term modem is not entirely correct because technically a DSL modem does not do modulation/demodulation as in a modem that uses the normal telephone network.

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DSL also have the added benefit of transmitting telephone services on the same set of wire as data services. DSL's comes in many flavor, and are referred to as xDSL, the x standing for the specific type.   SDSL (Symmetric Digital Subscriber Line)   ADSL (Asymmetric Digital Subscriber Line)   HDSL (High bit rate Digital Subscriber Line)   VDSL (Very high bit rate Digital Subscriber Line)   RADSL (Rate Adaptive Digital Subscriber Line) DSL technologies can be subdivided into two broad classes:   Symmetric. Within this class, the data rate transmitted in both directions (downstream and upstream) is the same. This is a typical requirement of business customers.   Asymmetric. In this case, there is asymmetry between the data rates in the downstream and upstream directions, with the downstream data rate typically higher than the upstream (usually appropriate for applications such as Web browsing).

* New technologies have reported up to 24 Mbps For years, it has been believed that the upper limit for transmitting data on analog phone lines was 56 kbps. This limit is set using the maximum possible bandwidth and no compression. The reason for this limit is that POTS or Plain Old Telephone Service uses the lower 4 KHz only.

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The limit imposed by the POTS lines does not take advantage of all the bandwidth available on copper, which is on the order of 1 MHz. The xDSL technology takes advantage of this difference and uses the upper frequencies for data services. Previously this was not possible because of the interference that the data services would cause in the POTS band. Advances in digital signal processing have eliminated the near-end crosstalk that results from the use of the upper bandwidth for data. The new DSP technologies allow data and POTS to be transmitted on the same set of copper wires without interfering with each other.

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Asymmetric Digital Subscriber Line (ADSL) ADSL is an Internet access technology that makes use of the existing telephone network to  provide broadband Internet service to homes and businesses. Asymmetric DSL (digital subscriber line) is a technology that leverages a conventional telephone line to provide "always on", high-speed Internet connectivity. Because it doesn't require a dedicated phone line, ADSL is able to share a line with an existing telephone service without impacting that phone service in any way. The word "asymmetric" means that the capacity to send data upstream and downstream is not equivalent. "Digital" means that data or voice is converted into a binary format where the audio or video data is represented by a series of "1"s and "0"s. The term "subscriber line" simply refers to the copper pair telephone wires that are used for conventional phone service. Working Principle

  ADSL works on the principle that since voice does not use all of the bandwidth available

•

from a standard copper twisted pair it‘s possible to maintain a high-speed high-speed data connection

 

•

 

•

   

•

•

at the same time. To do this, ADSL splits the 1MHz maximum bandwidth of a copper wire connection into 4 KHz channels Only the bottom 4 KHz channel is used for plain old telephone system (POTS) conversations or for fax and analogue modem data. The other 255 available channels are used for parallel digital communication. Being asymmetric 223 channels of 4 KHz each are used for the downlink and only 32 channels of 4 KHz each for the uplink.

  Discrete multi-tone (DMT) modulation is used as standard line code   ADSL is a fixed quality (fixed BER of 10 -7), variable rate service. During training, the

•

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ADSL system (ATU-R and ATU-C) evaluates the quality of the line by measuring the SNR and attenuation/ gain per ton. It can then decide on the maximum data rate -7 sustainable on the copper loop and still maintain a BER of less than 10 .

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Architecture

Components   ATU-R (ADSL Terminal Unit-Remote)   ATU-C (ADSL Terminal Unit-Central)

 

  SPLITTER



Splitter has a set of two filters. There is a low pass filter to avoid interference from signals of ADSL high-frequency and high-pass filter to act as a barrier that eliminates the POTS line from ADSL signals. There is a splitter on the user side and one at the local exchange.

DSLAM is a chassis that contains a large number of cards, each of which has several modems called ATU-C, and also concentrates all ADSL ADS L traffic links towards a WAN.

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OSI MODEL An ISO standard that covers all aspects of network communications is the Open Systems Interconnection model. An open system is a set of protocols that allows any two different systems to communicate regardless of their underlying underlying architecture. The purpose of the OSI model to show to facilitate communication between different without requiring changes to the is logic of thehow underlying hardware and software. The OSI model systems is not a  protocol; it is a model for understanding understanding and designing a network arc architecture hitecture that that is flexible, robust, and interoperable. The OSI model is a layered framework for the design of network systems that allows communication  between all types of computer systems. It consists of seven separate but related layers, each of which defines a part of the process of moving information across a network

Organization of the Layers The seven layers can be thought of as belonging to three subgroups. Layers I, 2, and 3-physical, data link, and network-are the network support layers; they deal with the physical aspects of moving data from one

device to another (such as electrical specifications, physical connections, physical addressing, and transport timing and reliability). Layers 5, 6, and 7-session, presentation, and application-can be thought of as the user support layers; they allow interoperability among unrelated software systems. Layer 4, the transport layer, links the two subgroups and ensures that what the lower layers have transmitted is in a form that the upper layers can use. The upper OSI layers are almost always implemented in software; lower layers are a combination of hardware and software, except for the physical layer, which is mostly hardware. Physical Layer The physical layer coordinates the functions required to carry a bit stream over a physical medium. It deals with the mechanical and electrical specifications of the interface and transmission medium. It also defines the procedures and functions that physical devices and interfaces have to perform for transmission to Occur.

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The physical layer is also concerned with the following:   Physical characteristics of interfaces and medium   Represent Representation ation of bits   Data rate   Synchroniz Synchronization ation of bits   Line configuration configuration

 

  Transmission Physical topology  mode Data Link Layer The data link layer transforms the physical layer, a raw transmission facility, to a reliable link. It makes the physical layer appear error-free to the upper layer (network layer).

Other responsibilities of the data link layer include the following:   Framing   Physical addressing   Flow control   Error control   Access control Network Layer The network layer is responsible for the source-to-destination delivery of a packet, possibly across multiple networks (links). Whereas the data link layer oversees the delivery of the packet between two systems on the same network (links), the network layer ensures that each packet gets from its point of origin to its final destination. If two systems are connected to the same link, there is usually no need for a network layer. However, if the two systems are attached to different networks (links) with connecting devices between between the networks (links), there is often a need for the network layer to accomplish source-todestination delivery.

Other responsibilities of the network layer include the following:   Logical addressing   Routing Transport Layer The transport layer is responsible for process-to-process delivery of the entire message. A process is an application program running on a host. Whereas the network layer oversees source-to-destination delivery of individual packets, it does not recognize any relationship between those packets. It treats each one independently, as though each piece belonged to a separate message, whether or not it does. The transport layer, on the other hand, ensures that the whole message arrives intact and in order, overseeing both error control and flow control at the source to-destination to-destination level.

Other responsibilities of the transport layer include the following:   Service-point addressing   Segmentation and reassembly   Connection control   Flow control   Error control

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Session Layer The services provided by the first three layers (physical, data link, and network) are not sufficient for some processes. The session layer is the network dialog controller.  controller.  It establishes, maintains, and synchronizes the interaction among communicating systems.

Specific responsibilities responsibilities of the session layer include the following:   Dialog control   Synchronization Presentation Layer The presentation layer is concerned with the syntax and semantics of the information exchanged between two systems.

Specific responsibilities of the presentation layer include the following:   Translation   Encryption   Compression Application Layer

The application layer enables the user,such whether human or software, to access the network. It provides user interfaces and support for services as electronic mail, remote file access and transfer, shared database management, and other types of distributed information services. Specific services provided by the application layer include the following:    Network virtual virtual terminal terminal   File transfer, access, and management   Mail services   Directory services

X.25 Packet Switched Network The concept of a packet-switched network is based on the idea that the network switching nodes will have multiple choices for routing of data packets. If a particular route becomes congested or has degraded operation, a node can send a packet on another route, and if that route becomes congested, possibly a third route will be available to forward the packet to its destination. At a data source, a file is segmented into comparatively short data packets, each of the same length and each with its own header and trailer. As these packets may take diverse routes through various nodes to their destination the destination node is responsible for data message reassembly in its proper order. Introduction to X.25

X.25 is ITU-T protocol standard for WAN communications. X.25 network device fall into three general categories 1.  DTE (Data Terminal Equipment) Equipment) 2.  DCE (Data Circuit-terminating Equipment) 3.  PSE (Packet Switching Exchange)

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Data terminal equipment (DTE) devices are end systems that communicate across the X.25 network. They are usually terminals, personal computers, or network hosts, and are located on the premises of individual subscribers. Data communication Equipment (DCEs) are communications devices, such as modems and packet switches that provide the interface between DTE devices and a PSE, and are generally located in the carrier's facilities. f acilities. PSEs are switches that compose the bulk of the carrier's network. They transfer data from one DTE device to another through the X.25 PSN.

X.25 standard specifically calls for three layers of functionality 1.  Physical layer 2.  Link layer 3.  Packet layer Layer

Description

 Network

X.25 Packet Packet Level

Link

X.25 Frame Level (LAPB)

Physical

RS232 etc

Data terminals defined by X.25 operate in a synchronous full-duplex mode with data rates of 2400, 4800, 9600, 14,400, 28,800, and 33,600 bps. X.25 spans the lowest three layers of the OSI reference model, as illustrated in Figure. It can be seen in the figure that X.25 is compatible with OSI up to the network layer.

Physical Layer  The physical layer is layer 1, where the requirements are defined for the functional, mechanical, procedural, and electrical interfaces between the DTE and DCE. Physical layer deals with  physical interface interface between between computer and and the link that attaches attaches tha thatt station to the packet packet switching switching node. Link Layer  The link layer of X.25 uses the LAPB protocol. The information field in the LAPB frame carries the user data, in this case the layer3 packet. X.25 packet is carried within the LAPB frame as the information field (i). The link layer (also called level 2, or frame level) ensures reliable transfer of data  between the DTE DTE and the DCE, by transmitting transmitting the data as a sequence of of frames (a frame is an individual individual data unit which contains address, control, information field etc.). The functions performed by the link level include: • Transfer of data in an efficient and timely fashion.

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• Synchronization of the link to ensure that the receiver is in step with the transmitter. • Detection of Detection of transmission errors and recovery from such errors • Identification and reporting of procedural errors to higher levels, for recovery. LAPB Frame Format

Flag: (8 bits) Indicates start and end of frame (01111110) Address: (8 bits) DTE address is maintained in higher layer so this field is used to identify command and responses between DTE and DCE. A value of 0x01 indicates a command from DTE and responses from DCE while a value of 0x03 indicates commands from DCE and responses from DTE. Control: (8 bits) Contains sequence numbers, commands and responses for controlling data flow. Data: (varies is size) Contains upper layer data. FCS: (16 bits) Frame Check Sequence used to determine if an error has occurred in transmission (variation of CRC). LAPB Frame Types: Three types of frames   I-Frames (Information Frames): Carry data as well as Next Send (NS) and Next Receive (NR) counts S-Frames (Supervisory Frames): Controls flow of data with Receiver Ready (RR), Receiver Not Ready (RNR), and Reject (REJ) frames U-Frames (Unnumbered Frames):  Establish and maintain communications with Set Asynchronous Balanced Mode (SABM), Unnumbered Acknowledgment (UA) and Disconnect Mode (DISC) Packet layer this level governs the end-to-end communications between the different DTE devices. Layer 3 is concerned with connection set-up and teardown and flow control between the DTE devices, as well as network routing functions and the multiplexing of simultaneous logical connections over a single physical connection. PLP is the network layer protocol of X.25. X.25 session establishment and virtual circuits

There are three approaches used with X.25 operation to manage the transfer and routing of packet streams: datagrams, virtual connections (VCs), and permanent virtual connections (PVCs) . Datagram service uses optimal routing on a packet-by-packet basis, usually over   diverse routes. In the virtual circuit approach, there are two operational ―modes‖: virtual  connection and permanent virtual connection. These two are analogous to a dial-up telephone   connection and a leased line connection, respectively. With the virtual connection  a logical connection is established before any packets are sent. The packet originator   sends a call request to its serving node, which sets up a route in advance to the desired   destination. All packets of a particular message traverse this route, and each packet of   the message contains a virtual circuit identifier (logical channel number) and the packet   data. At any one time each station can have more than one virtual circuit to any other   station and can have virtual circuits to more than one station. With virtual circuits routing decisions are made in advance. With the datagram approach ad hoc decisions are   made for each packet at each node. There is no callsetup phase with datagrams as there  is with virtual connections. Virtual connections are advantageous for high community-of-interest connectivity, datagram service for low community-of-interest relations.   Datagram service is more reliable because traffic can be alternately routed around network congestion

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 points. Virtual circuits circuits are fixed-routed for a particular call. call. Call setup time at each node is eliminated on a packet basis as with the virtual connection technique. X.25 also allows the possibility of setting up permanent virtual connections and is network assigned. This latter alternative is economically viable only for very high-traffic relations; otherwise these permanently assigned logical channels will have long dormant periods. 

Figure: X.25 Connection Establishment

Frame Relay Frame Relay is a virtual-circuit wide-area network that was designed in response to demands for a new type of WAN. Prior to Frame Relay a virtual-circuit switching network called X.25 was in use that  performed switching switching at at the network network layer. X.25 has several drawbacks: a.  X.25 has a low 64-kbps data rate. By the 1990s, there was a need for higher data-rate WANs.  b.  X.25 has extensive flow and error control at both the data link layer and the network layer. c.  Originally X.25 was designed for private line, not for the Internet. X.25 has its own network layer. This means that the user's data are encapsulated in the network layer packets of X.25. The Internet, however, has its own network layer, which means if the Internet wants to use X.25, the Internet must deliver its network layer packet, called a datagram, to X.25 for encapsulation in the X.25 packet. This doubles the overhead. d.   Not suitable for bursty data as bandwidth cannot be provisioned as per demand (bandwidth on demand) In response to the above drawbacks, Frame Relay was designed. Frame Relay is a wide area network with the following features f eatures:: 1.  Frame Relay operates at a higher speed (1.544 Mbps and recently 44.376 Mbps). This means that it can easily be used instead of a mesh of T-I or T-3 lines.

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2.  Frame Relay operates in just the physical and data link layers. This means it can easily be used as a backbone network to provide services to protocols that already have a network layer protocol, such as the Internet. 3.  Frame Relay allows bursty data. 4.  Frame Relay allows a frame size of 9000 bytes, which can accommodate all local area network frame sizes. 5.  Frame Relay has error detection at the data link layer only. There is no flow control or error control. There is not even a retransmission policy if a frame is damaged; it is silently dropped. Frame Relay was designed in this way to provide fast transmission capability for more reliable media and for those protocols that have flow and error control at the higher layers. Frame Relay Devices   Data Terminal Equipment (DTE) Terminals, Personal Computers, routers and bridges typically at the customer location   Data Circuit-terminating Equipment (DCE) Typically packet switches owned by the carrier that transmit data through the WAN

The connection between a DTE device and a DCE device consists of both a physical layer component and a link layer component. The physical component defines the mechanical, electrical, functional, and  procedural specifications specifications for the connection connection between the devices. One of the most commonly used  physical layer layer interface interface specifications specifications is the the recommended recommended standard standard (RS)-232 specificatio specification. n. The link llayer ayer component defines the protocol that establishes the connection between the DTE device, such as a router, and the DCE device, such as a switch. Virtual Circuits

Frame Relay is a virtual circuit network, so it doesn‘t use physical addresses to define the DTEs connected to the network. Frame Relay provides connection-oriented data link layer communication. This means that a defined communication exists between each pair of devices and that these connections are associated with a connection identifier. However, virtual circuit identifiers in Frame relay operate at the data link layer, in contrast with X.25, where they operate at the network layer. This service is

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implemented by using a Frame Relay virtual circuit, which is a logical connection created between two data terminal equipment (DTE) devices across a Frame Relay packet-switched network (PSN). Virtual circuits provide a bidirectional communication path from one DTE device to another and are uniquely identified by a data-link connection identifier (DLCI). A virtual circuit can pass through any number of intermediate DCE devices (switches) located within the Frame Relay PSN. Switched Virtual Circuits Switched virtual circuits (SVCs) are temporary connections used in situations requiring only sporadic data transfer between DTE devices across the Frame Relay network. A communication session across an SVC consists of the following four operational operational states:

1.  Setup — The The virtual circuit between two Frame Relay DTE devices is established. 2.  Data transfer   — Data Data is transmitted between the DTE devices over the virtual circuit. 3.  Idle — T The he connection between DTE devices is still active, but no data is transferred. If an SVC remains in an idle state for a defined period of time, the call can be terminated. 4.  Call termination — T The he virtual circuit between DTE devices is terminated. After the virtual circuit is terminated, the DTE devices must establish a new SVC if there is additional data to be exchanged.

Permanent Virtual Circuits

Permanent virtual circuits (PVCs) are permanently established connections that are used for frequent and consistent data transfers between DTE devices across the Frame Relay network. Communication across PVC does not require the call setup and termination states that are used with SVCs. PVCs always operate in one of the following two operational states: 1.  Data transfer: Data is transmitted between the DTE devices over the virtual circuit. 2.  Idle: The connection between DTE devices is active, but no data is transferred. Unlike SVCs, PVCs will not be terminated under any circumstances when in an idle state. DTE devices can begin transferring data whenever they are ready because the circuit is permanently established.

Frame Relay Layers Frame Relay has only 2 layers, namely Physical layer and Data Link layer. And as compared to other layer of packet switching network such as X.25 it eliminates eliminates all network layer functions Physical Layer  No specific protocol is defined for physical layer in frame relay. Frame relay supports any one of the  protocols recognized recognized by ANSI, and thus the choice of of physical layer protocol protocol is up to the implementer. implementer. Data Link Layer At Data-link Layer Frame employs a simpler version of HDLC. Simpler version is used because HDLC  provides extensive extensive error error and flow control fields fields that aare re not needed needed in frame frame relay. relay.

To understand much of the functionality of Frame Relay, it is helpful to understand the structure of the Frame Relay belowThree depicts the basic format ofmake the Frame the  beginning andframe. end ofFigure the frame. primary components up the Relay Frameframe. Relay Flags frame:indicate the header and address area, the user-data portion, and the frame check sequence (FCS). The address area, which is 2

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 bytes in length, is is comprised of 10 bits representing representing the actual actual circuit identifier identifier and 6 bits of fields fields related to congestion management. This identifier commonly is referred to as the data-link connection identifier (DLCI).

Flags - Delimits the beginning and end of the frame. The value of this field is always the same and is represented either as the hexadecimal number 7E or as the binary number 01111110. Address - Contains the following information:   DLCI - The 10-bit DLCI is the essence of the Frame Relay header. This value represents the virtual connection between the DTE device and the switch. Each virtual connection that is multiplexed onto the physical channel will be represented by a unique DLCI.   Command/response (CIR) - The command/response (C/R) bit is provided to allow upper layers to identify a frame as either a command or a response.   Extended address (EA) - The extended address (EA) bit indicates whether the current byte is the final byte of the address. An EA of 0 means that another address byte is to follow   Forward explicit congestion notification (FECN) - The forward explicit congestion notification (FECN) bit can be set by any switch to indicate that traffic is congested. This bit informs the destination thatofcongestion delay or a loss packets. has occurred. In this way, the destination knows that it should expect   Backward explicit congestion notification (BECN) - The backward explicit congestion notification (BECN) bit is set (in frames that travel in the other direction) to indicate a congestion  problem in the network. network. This bit informs informs the sender that congestion congestion has occurred. occurred. In In this way, the source knows it needs to slow down to prevent the loss of packets.   Discard eligibility (DE). The discard eligibility (DE) bit indicates the priority level of the frame. In emergency situations, switches may have to discard frames to relieve bottlenecks and keep the network from collapsing due to overload. When set (DE=1), this bit tells the network to discard this frame if there is congestion. This bit can be set either by the sender of the frames (user) or by any switch in the network.

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FRAD To handle frames arriving from other protocols, Frame Relay uses a device called a Frame Relay assembler/disassembler (FRAD). A FRAD assembles and disassembles frames coming from other  protocols to allow them them to be carried by Frame Frame Relay frames. A FRAD FRAD can be implem implemented ented as a separate separate device or as part of a switch. Figure shows two FRADs connected to a Frame Relay network.

Asynchronous Transfer Mode

 Asynchronous Transfer Mode (ATM)  Asynchronous (ATM) is an ITU-T standard for cell relay wherein informa information tion for multiple service types, such as voice, video, or data, is conveyed in small, fixed-size cells. ATM networks are connection-oriented. Asynchronous transfer mode (ATM) can be viewed as an evolution of packet switching. Like packet switching protocols for data (e.g., X.25, frame relay, Transmission Control Protocol and Internet protocol (TCP IP]), ATM integrates the multiplexing and switching functions, is well suited for bursty traffic (in contrast to circuit switching), and allows communications between devices that operate at different speeds. Unlike packet switching, ATM is designed for high-performance multimedia networking. The most basic service building block is the ATM virtual circuit, which is an end-to-end connection that has defined end  points and routes but does not have bandwidth dedicated to it. Bandwidth Bandwidth is allocated allocated on demand by the network as users have traffic to transmit. ATM also defines various classes of service to meet a broad range of application needs. ATM Service

ATM services are categorized into the following classes related to the statistical nature of the data rate requirements of the respective sources and the quality of service (QoS) that the network can provide for those services:        

Constant-bit-rate (CBR) Services Variable-bit-rate (VBR) Services Available bit rate (ABR) Services Unspecified bit rate ( UBR) Services

Constant-Bit-Bate Services This class is used for emulating circuit switching. The cell rate is constant with time. CBR applications are quite sensitive to cell-delay variation. Examples of applications that can use CBR are telephone traffic (i.e., nx64 kbps), videoconferencing, and television.

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Variable-bit-rate Variable-bitrate (VBR) Services This class allows users to send traffic at a rate that varies with time depending on the availability of user information. Statistical multiplexing is provided to make optimum use of network resources. Multimedia e-mail is an example of VBR

Available bit rate (ABR) services This class of ATM services provides rate-based flow control and is aimed at data traffic such as file transfer and e-mail. Although the standard does not require the cell transfer delay and cell-loss ratio to be guaranteed or minimized, it is desirable for switches to minimize delay and loss as much as possible. Depending upon the state of congestion in the network, the source is required to control its rate. The users are allowed to declare a minimum cell rate, which is guaranteed to the connection by the network. Unspecified bit rate (UBR) Services UBR services are inherently intended for non-real-time applications wherein no specific quality of service is desired or implied. Neither delay parameters nor cell loss ratios are specified.

ATM Applications

ATM is used in both LANs and WANs. Few of the possible applications are ATM WANs: ATM is basically a WAN technology that delivers cell over long distances. Here ATM is mainly used to connect LANs or other WANs together. A router between ATM network and the other network serves as an end point. This router has two stacks of protocols: one belonging to ATM and other  belonging to other protocol. protocol. ATM LANs: At the surface level, to implement an ATM LAN ATM switch will replace the traditional Ethernet switch, in a switched LAN. But few things have to be kept in mind and software modules would  be needed to map the following differences differences between between the two technologies: technologies:

  Connectionless versus connection-oriented: ATM is a virtual connection oriented technology, while traditional Ethernet uses connectionless protocols.   Physical address versus virtual circuit identifier : In the Traditional LAN packets are routed  based on the source source and destination destination addresses, addresses, while in ATM cells cells are routed based based on the virtual circuit identifiers (VPI-VCI pair).

LAN Emulation:  LAN Emulation (LANE) is a standard defined by the ATM Forum that gives to stations attached via ATM the same capabilities that they normally obtain from legacy LANs, such as Ethernet and Token Ring. As the name suggests, the function of the LANE protocol is to emulate a LAN on top of an ATM network. Specifically, the LANE protocol defines mechanisms for emulating either an IEEE 802.3 Ethernet or an 802.5 Token Ring LAN.

Frame-relay service areinfrastructure deploying ATM to meet the rapid growth of theirbackbones: frame-relayFrame-relay services to use as aproviders networking for abackbones range of data services and to enable frame relay to ATM service internetworking services.

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Internet backbones: Internet service providers are likewise deploying ATM backbones to meet the rapid growth of their frame-relay services, services, to use as a networking infrastructure for a range r ange of data services, and to enable Internet class-of-service offerings and virtual private intranet services. ATM Signalling

 

n

ATM is a connection-oriented protocol, which means that virtual channels must be set up before any data cells can be sent on the channel

n  Connection setup is done using a signalling protocol

Protocol consists of two parts

n  User-Network Interface (UNI) n  defines how hosts talk to switches n  Network-Network Interface (NNI) n  defines how switches talk to other switches Cell formats are slightly different

 ATM  AT M UNI Cell Cell F orma rmat  

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ATM Cell Header Fields

  Generic Flow Control (GFC) — Provides Provides local functions, such as identifying multiple stations that share a single ATM interface. This field is typically not used and is set to its default value of 0 (binary 0000).   Virtual Path Identifier (VPI) — IIn n conjunction with the VCI, identifies the next destination of a cell as it passes through a series of ATM switches on the way to its destination.   Virtual Channel Identifier (VCI) — In In conjunction with the VPI, identifies the next destination of a cell as it passes through a series of ATM switches on the way to its destination.   Payload Type (PT) —   — Indicates Indicates in the first bit whether the cell contains user data or control data. If the cell contains user data, the bit is set to 0. If it contains control data, it is set to 1. The second  bit indicates congestion (0 = no congestion, 1 = congestion), and the third bit indicates whether the cell is the last in a series of cells that represent a single AAL5 frame (1 = last cell for the frame).   Cell Loss Priority (CLP) —   — Indicates Indicates whether the cell should be discarded if it encounters extreme congestion as it moves through the network. If the CLP bit equals 1, the cell should be discarded in preference to cells with the CLP bit equal to 0.   Header Error Control (HEC) —   — Calculates Calculates checksum only on the first 4 bytes of the header. HEC can correct a single bit error in these bytes, thereby preserving the cell rather than discarding it.

B asic si c Signalling Si gnalling Ope Oper ation n  Connection requests proceed hop-by-hop through the switches of the network en route to destination n  Switches perform Call Admission Control (CAC) based on traffic descriptor, QOS requirements, and available resources at that switch n  If connection is acceptable, then request is forwarded on, otherwise ―reject‖ is returned   n  If destination accepts connection, then ―accept‖ is returned  returned  

 

n

VPI and VCI assigned

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Multi-protocol Label Switching (MPLS)

The fundamental concept behind MPLS is that of labeling packets. In a traditional routed IP network, each router makes an independent forwarding decision for each packet based solely on the packet‘s network-layer header. header. Thus, every time a packet arrives at a router, the router has to ―think through‖ where to send the packet next. Before explaining basic MPLS functionality, three drawbacks of traditional IP forwarding should be highlighted:   Routing protocols are used on all devices to distribute the routing information.   Regardless of the routing protocol, routers always forward packets based on the destination address only.   Routing lookups are performed on every router. Each router in the network makes an independent decision when forwarding packets. MPLS helps reduce the number of routing lookups, possibly changes the forwarding criteria, and eliminates the need to run a particular routing protocol on all the devices.

This figure illustrates how routers in a service provider network forward packet based on their destination addresses. The figure also shows that all the routers need to run a routing protocol (BGP) to get all the Internet routing information. Every router in the path performs a destination-based routing lookup in a large forwarding table. Forwarding complexity is usually related to the size of the forwarding table and the switching mechanism. MPLS is a new switching mechanism that uses labels (numbers) to forward packets. Labels usually correspond to L3 destination addresses (equal to destination-based routing). Labels can also correspond to other parameters (Quality of Service [QoS], source address, etc.). MPLS was designed to support other protocol stacks than IP as well. Label switching is performed regardless of the L3 protocol.

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This figure illustrates a situation where the intermediary router does not have to perform a timeconsuming routing lookup. Instead this router simply swaps a label with another label (5 is replaced r eplaced by 3) and forwards the packet based on the received label (3). In larger networks the result of MPLS labeling is that only the edge routers perform a routing lookup. All the core routers forward packets based on the labels. With MPLS, the first time the t he packet enters a network, it‘s assigned to a specific forwarding equivalence class (FEC), indicated by appending a short bit sequence (the label) to the packet. Each router in the network has a table indicating how to handle packets of a specific FEC type, so once the packet has entered the network, routers don‘t need to perform header analysis. Instead, subsequent routers use the label as an index into a table that provides them with a new FEC for that packet. This gives the MPLS network the ability to handle packets with particular characteristics (such as coming from particular ports or carrying traffic of particular application types) in a consistent fashion. Packets carrying real-time traffic, such as voice or video, can easily be mapped to low-latency routes across the network —  network  —  something that‘s challenging with conventional routing. I. MPLS Terminologies I. 

FEC

MPLS groups packets, to be forwarded in the same manner, into a class called the forwarding equivalence class (FEC). That is, packets of the same FEC are handled in the same way. The classification of FECs is very flexible. It can be based on any combination of source address, destination address, source port, destination port, protocol type and VPN. II. 

Label

A label is a short fixed length identifier for identifying a FEC. A FEC may correspond to multiple labels in scenarios where, for example, load sharing is required, while a label can only represent a single FEC. A label is carried in the header of a packet. It does not contain any topology information and is of local significance.

A 32-bit label contains the following fields:

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  20-bit label: The actual label   3-bit experimental field: It is used to define a class of service (i.e. IP precedence)   Bottom-of-stack bit: MPLS allows multiple labels to be inserted; this bit is used to determine if this is the last label in the packet

III. 

  8-bit time-to-live (TTL) field: It has the same purpose as the TTL field in the IP header LSR Label switching router (LSR) is a fundamental component on an MPLS network. All LSRs support MPLS.

IV. 

LSP

Label switched path (LSP) means the path along which a FEC travels through an MPLS network. Along an LSP, two neighboring LSRs are called upstream LSR and downstream LSR respectively. In Figure, R2 is the downstream LSR of R1, while R1 is the upstream LSR of R2.

An LSP is a unidirectional path from the ingress of the MPLS network to the egress. It functions like a virtual circuit in ATM or frame relay. Each node of an LSP is an LSR. V. 

LDP

Label Distribution Protocol (LDP) means the protocol used by MPLS for control. An LDP has the same functions as a signaling protocol on a traditional network. It classifies FECs, distributes labels, and establishes and maintains LSPs. Structure of the MPLS network

As shown in Figure, the element of an MPLS network is LSR. LSRs in the same routing or administrative domain form an MPLS domain. In an MPLS domain, LSRs residing at the domain border to connect with other networks are label edge routers (LERs), while those within the MPLS domain are core LSRs. All core LSRs, which can be routers running MPLS or ATM-LSRs upgraded from ATM switches, use MPLS to communicate, while LERs interact with devices outside the domain that use traditional IP technologies.

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Each packet entering an MPLS network is labeled on the ingress LER and then forwarded along an LSP to the egress LER. All the intermediate LSRs are called transit LSRs.

Next Generation Network (NGN) 

The term NGN is generally used to depict the shift to higher network speeds using broadband, the migration from the PSTN to an IP-network, and a greater integration of services on a single network.  NGN is defined defined by ITU as a Packet based network able to provide services including telecommunication services and able to make use of multiple broadband, QoS-enabled transport technologies and in which service related functions are independent from underlying transport-related technologies.  NGN offers access by users to different service providers, and supports ― generalize  generalized d mobility which will allow  consistent and ubiquitous provision of services to users.‖ allow users. ‖   NGN is also defined as as Broadband managed IP networks‖, includes next generation ―core ― core‖‖ networks, which evolve towards a converged IP infrastructure capable of carrying a multitude of services, such as voice, video and data services, and next generation ―access ― access‖‖ networks, i.e. the development of high-speed local loop networks that will guarantee the delivery of innovative services. Fundamental characteristics of NGN

             

 packet-based transfer  packet-based transfer decoupling of service provision from transport support for a wide range of services real time, streaming, non-real time and multimedia  broadband capabilitie capabilitiess with end-to-end end-to-end QoS generalized mobility interworking with legacy networks via open interfaces

  unrestricted access by users to different service providers   converged services between fixed/mobile

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  independence of service-related functions from underlying transport technologies   support of multiple last-mile technologies Decoupling of service provision from transport Existing Network –  Network –  Vertical  Vertical Network

NGN Network –  Network –  Horizontally  Horizontally Integrated Network

NGN Concept

  A unified packet transport layer for all types of services o  A session based control architecture   For user to user voice , video and data services over the packet infrastructure o  A common Service delivery platform

o

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Evolution of NGN

NGN architecture

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Transport Layer

• 

Transport Layer of NGN is based on IP.

• 

Transport Layer forms the core of the Network.

• 

It basically consists of Routers, which are responsible for carrying traffic originated by access layer.

• 

it should be able to make use of bandwidth policies and QoS policies. Operator has to think of managed Network for its subscribers.

• 

It is basically an assembly of routers connected with optical network.

• 

Traffic coming from gateways is properly routed by those routers

Access Layer

• 

Access Layers is responsible for direct subscriber attachment function. Equipment that provides various interfaces interfaces to the users or other network lies in this layer.

• 

Access Gateway(AG)

• 

Trunk Gateway (TG)

• 

Signalling Gateway (SG)

• 

SIP Terminals / Softphone, Hardphone etc.

•   NGN can support all kind of existing access as well as upcoming access and is capable of  processing traffic originated originated from PSTN, GSM, CDMA, xDSL, WiMAX or any other access system. • 

Depending upon the type of access, protocol conversion and/or media conversion may be required at the NGN Gateways

Control Layer

• 

The Control Equipments lies in thi thiss layer

• 

Softswitch

• 

Softswitch (MGC) is for voice service and controls MGWs

Application Layer

• 

Various Application Servers lies in this layer.

• 

To provide various value added services such as prepaid services, CRBT, gaming, news, VoD etc.

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VOICE OVER IP  VoIP is packetisation and transport of classic public switched telephone system audio over an IP network.   It allows 2-way voice transmission over broadband connection.   It is also called IP telephony, internet telephony, voice over broadband, broadband telephony.  PSTN vs. INTERNET PSTN

 

Voice network use circuit switching.

 

Dedicated path between calling and called party.

 

Bandwidth reserved in advance.

 

Cost is based on distance and time.

INTERNET

 

Data network use packet switching.

 

No dedicated path between sender and receiver. rece iver.

 

It acquires and releases bandwidth, as it needed.

 

Cost is not based on distance and time.

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The 1-2-3s of VoIP   

1. Compression –   –  voice   voice is compressed typically with one of the following codecs, G.711 64k, G.729 AB 8k, G.723.1 6.3k

 

2. Encapsulation –  the  the digitized voice is wrapped in an IP packet

 

3. Routing –   –  the  the voice packet is routed through the network to its ffinal inal destination destination

VoIP functions 1.  Signaling comprises all functions to set up, control and teardown a VoIP call/session. Examples of VoIP signaling protocols: H.323, SIP, MGCP, H.248, NCS, Skype. UDP and TCP are used for signaling transport. 2.  The data path is responsible for encoding, packetizing and compressing the voice. UDP is always used for the data path since: a. TCP would introduce too much delay and  b. Retransmissions Retransmissions are not necessary necessary and only distort distort the voice in case case of packet packet loss. c. The RTP protocol is the one that really transports encoded audio. d. RTP is transported over UDP.

VOIP Protocol Suite

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IP Protocol   

One of the most well-known protocols

 

Its name comes from Internet Protocol.

 

This protocol offers a service with ‖no guarantees‖ and is also referred to as ‖best effort‖.

 

The packets can arrive out of order, and be reordered at the destination.

 

Some packets can even be lost on the way.

 

This disorder and loss of packets can affect the quality qu ality of the voice.

 

Despite all of this, intelligent ways have been found fou nd to solve these problems in the best way possible.

 

It's a unique number that identifies a host connected to an IP network.

 

It's made up of 32 bits, or 4 octets. In practice a notation is used in which each octet is translated into decimal, and separated with a period. An example of an IP address a ddress is 130.5.5.26.

 

An IP address is composed in two parts: One identifies the host, and the other identifies the network to which that host belongs.

 

To find these parts another parameter is used, called a network mask.

 

The network mask is a 32 bit b it binary number that is also represented in a no notation tation that is similar to the IP address.

 

It begins with ones and continues this way until it reaches a number of ones eq equal ual to the  portion of the IP address that corresponds to the network.

 

Therefore, in the previous example we obtain an IP address of 130.5.5.26 with a mask of 255.255.255.0 belonging to the 130.5.5.0 network.

 

The previous mask was a 24 bit mask, since there were 24 ‖ones‖.

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 

Once the network where the host ho st is located is known it is easier to send the IP packets to their destination.

 

Routers store tables of routes or rules on how to find other networks

 

An IP packet contains all the necessary information to arrive to its destination.

 

It can be divided into two parts: parts: the header and the payload.

 

The header contains information that allows the packet to be delivered to its destination, and the payload retrieved.

 

This header decreases slightly the quantity of information that can be transported since it occupies space.

 

The payload can be used to encapsulate other protocols like UDP or TCP.

Addressing in IP

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Reserved Internet Addresses

TCP Protocol  

Transport Control Protocol

 

It is an IP IP protocol that allows packet retransmission, retransmission, packet order management, and receipt acknowledgement.

 

To achieve this goal, TCP carries additional information that adds weight to the packet. That is why it is not recommended for real time applications a pplications like voice.

 

However, it can work for voice signaling.

 

TCP introduces the concept of port.

 

A port is an abstraction that allows us to relate flows of data with services on the network.

 

For example, port 80 corresponds to the Web service, or HTTP protocol.

UDP Protocol   

UDP (User Datagram Protocol) is another transport protocol.

 

It divides information into packets called datagrams.

 

This protocol doesn't care if the data arrives with errors or if it doesn´t do esn´t arrive at all. That is the main difference with TCP.

 

This is why it introduces little extra weight to the IP packet which makes it ideal for realtime applications like voice.

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TCP/IP Vs OSI Model

Voice coding  

Voice is adequately encoded for transmission.

 

After encoding, it is transmitted over RTP.

 

Encoding can serve to reduce the chance of error, as well as to minimize the amount of  bandwidth used.

 

To encode this data, an algorithm called a codec is used.

 

There are several different codecs, each has pros and cons.

G.711  

One of the most widely used codecs of all time.

 

Comes from the ITU-T standard that was released in 1972. 1972 .

 

Comes in two types called μ-Law μ -Law (or u-law, primarily used in Europe) and a-Law (used in the USA).

 

Advantage 1: Good voice quality since it uses 64kbit/s, that is a sampling of 8 bits aatt 8kHz.

 

Advantage 2: It is already enabled in Elastix, it's not necessary to pay for it.

 

Disadvantage: It uses a lot of bandwidth. It is not recommended for connections with low  bandwidth.

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VoIP standards:  

 

 

 

IEEE developed H.323 IETF developed SIP MGCP IEEE/IETF Megaco/H.248

H.323 Describes terminals and other entities that provide multimedia communications services over Packet Based Networks which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications. H,323 defines, 

•  • 

Call establishment and teardown. Audio visual or multimedia conferencing.

H.323 Components

H.323 Terminals H.323 terminals are client endpoints that must support: •  H.225 call control signaling. •  H.245 control channel signaling. •  RTP/RTCP protocols for media packets. •  Audio codecs. •  Video codecs support is optional. H.323 Gateway A gateway provides translation: •  For example, a gateway can provide translation between entities in a packet switched network (example, IP network) and circuit switched network (example, PSTN network).

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• 

Gateways can also provide transmission formats translation, communication procedures translation, H.323 and non-H.323 endpoints translations t ranslations or codec translation.

H.323 Gatekeepers  All devices (clients, gateways, MCU) register with gatekeeper. For each new call the clients contact the gatekeeper for address resolution. The gatekeeper mainly has following functions: •  Address translation (phone number to IP) •  Admission control (who is allowed to place calls and to whom) •  Registration •  Bandwidth control •  Zone management •  Call control signaling (optional) •  Call authorization (optional) •  Bandwidth management (optional) •  Call management (optional) H.323 also allows direct signaling between clients without a gatekeeper in between (gatekeeper- less signaling). Thus, Gatekeepers are optional but if present in a H.323 system, all H.323 endpoints must register with the gatekeeper and receive permission before making a call.

H.323 Multipoint Control Unit MCU provide support for conferences of three or more endpoints.   An MCU consist of:  (MC) –  provides  provides control functions. •  Multipoint Controller (MC) –  (MP) –  receives  receives and processes audio, video and/or data •  Multipoint Processor (MP) –  streams.

Process for Establishing Communicat Communication ion Establishing communication using H.323 may occurs in five steps:   •  Call setup •  Initial communication and capabilities exchange •  Audio/video communication establishment •  Call services •  Call termination

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H.323 call setup

Both endpoints have previously registered with the gatekeeper.  •  Terminal A initiate the call to the gatekeeper. (RAS messages are exchanged).   •  The gatekeeper provides information for Terminal A to contact Terminal B.   •  Terminal A sends a SETUP message to Terminal B.  •  Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission.  •  Terminal B sends a Alerting and Connect message.   •  Terminal B and A exchange H.245 messages to determine master slave, terminal capabilities, and open logical channels.  •  The two terminals establish RTP media paths.  

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Session Initiation Protocol (SIP) SIP is an application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users  

SIP Distributed Architecture

User Agents An application that initiates, receives and terminates calls.

• 

User Agent Clients (UAC) –  (UAC) –  An  An entity that initiates a call.

• 

User Agent Server (UAS) –  (UAS) –  An  An entity that receives a call. 

Both UAC and UAS can terminate a call.  

Proxy Server An intermediary program that acts as both a server and a client to make requests on behalf of other clients. •  A SIP proxy server receives a SIP request (thus acting as SIP UA server), performs some application-specific action on the SIP message (e.g. changing the URLs) and forwards the SIP request to another SIP server (thus acting as SIP UA client).

• 

Requests are serviced internally or by passing them on, possibly after translation, to other servers.

• 

Interprets, rewrites or translates a request message before forwarding it.

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Location Server • 

A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s).  location(s). 

Redirect Server • 

A server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client.

• 

Unlike a proxy server, the redirect server does not initiate its own SIP request.

• 

Unlike a user agent server, the redirect server does not n ot accept or terminate calls.

Registrar Server • 

A server that accepts REGISTER requests.

• 

The register server may support authentication.

• 

A registrar server is typically co-located with a proxy or redirect server and may offer location services.

SIP Messages –  Messages –  Methods  Methods and Responses SIP components communicate by exchanging SIP messages: me ssages: SIP Methods:

• 

INVITE –  INVITE  –  Initiates  Initiates a call by inviting user to participate in session.

• 

ACK - Confirms that the client has received a final response to an INVITE request. request .

 

•

BYE - Indicates termination of the call.

• 

CANCEL - Cancels a pending request.

• 

REGISTER –  REGISTER  –  Registers  Registers the user agent.

• 

OPTIONS –  OPTIONS  –  Used  Used to query the capabilities of a server.

• 

INFO –  INFO  –   Used Used to carry out-of-bound information, such as DTMF digits.

SIP Responses:

• 

1xx - Informational Messages.

• 

2xx - Successful Responses.

• 

3xx - Redirection Responses.

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• 

4xx - Request Failure Responses.

• 

5xx - Server Failure Responses.

• 

6xx - Global Failures Responses.

SIP Addressing

• 

The SIP address is identified by a SIP URL, in the format: user@host.

• 

Examples of SIP URLs:

 –   sip:[email protected]  –   sip:[email protected]  –   sip:[email protected]

Process for Establishing Communicat Communication ion Establishing communication communication using SIP usually occurs in six steps: • 

Registering, initiating and locating the user

• 

Determine the media to use  –   involves delivering a description of the session that the user is invited to

• 

Determine the willingness of the called party to communicate  –   the called party must send a response message to indicate willingness to communicate (accept or reject)

• 

Call setup

• 

Call modification or handling –  handling  –  example,  example, call transfer (optional)

• 

Call termination

SIP Registration • 

Each time a user turns on the SIP user client (SIP IP Phone, PC, or other other SIP device), the client registers with the proxy/registration server.

• 

Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location.  

• 

The registration information is periodically refreshed and each user

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client must re-register with the proxy/registration server. 

• 

Typically the proxy/registration server will forward this information to be saved in the location/redirect server.

SIP Call Setup User 1 uses his softphone to reach the SIP phone of user2. Server1 and server2 help to setup the session on behalf of the users. This common arrangement of the proxies and the end-users is called "SIP Trapezoid" as depicted by the dotted line.

The transaction user1 making an INVITE request user2. But user1 doesn't know the exact location of user2starts in thewith IP network. So it passes the request to for server1. Server1 on behalf of user1 forwards an INVITE request for user2 to server2. It sends a TRYING response to user1 informing that it is trying to reach user2. Receiving INVITE M2 from server1, server2 works in a similar fashion as server1. It forwards an INVITE request to user2 (note: Here server2 knows the location of user2. If it didn't know the location, it would have forwarded it to another proxy server. So an INVITE request may travel through several  proxies before reaching the recipient). After forwarding INVITE M3 server2 issues a TRYING response to server1. The SIP phone, on receiving the INVITE request, starts ringing informing user2 that a call request has come. It sends a RINGING response back to server2 which reaches user1 through server1. So user1 gets a feedback that user2 has received the INVITE request.

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User2 at this point has a choice to accept or decline the call. Let's assume that he decides to accept it. As soon as he accepts the call, a 200 OK response is sent by the phone to server2. Retracing the route of INVITE, it reaches user1. The softphone of user1 sends an ACK message to confirm the setup of the call. This 3-way-handshaking (INVITE+OK+ACK) is used for reliable call setup. Note that the ACK message is not using the proxies to reach user2 as by now user1 knows the exact location of user2. Once the connection has been setup, media flows between the two endpoints. Media flow is controlled using protocols different from SIP e.g. RTP When one party in the session decides to disconnect, it (user2 in this case) sends a BYE message to the other party. The other party sends a 200 OK message to confirm the termination of the session.

Media Gateway Control Protocol (MGCP)  MGCP is a protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents. MGCP is a master/slave protocol assumes limited intelligence at the edge (endpoints) and more intelligence at the core (call agent). It is used between call agents and media gateways. Interoperates with SIP and H.323.

Components

Call agent or media gateway controller 

• 

 

•

Provides call signaling, control and processing intelligence to the gateway.  Sends and receives commands to/from t o/from the gateway. 

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Gateway 

• 

Provides translations between circuit switched networks and packet switched networks. 

• 

Sends notification to the call agent about endpoint events.  

• 

Execute commands from the call agents. 

Megaco/H.248 Megaco is a (master/slave) protocol for control of gateway functions at the edge of the packet network. Examples of this are IP-PSTN trunking gateways and analog line gateways. The main function of Megaco is to allow gateway decomposition into a call agent (call control) part (known as Media Gateway Controller, MGC) - master, and an gateway interface part (known as Media Gateway, MG) - slave. The MG has no call control knowledge and only handles making the connections and simple configurations.

Voice over IP (VoIP) technology is currently finding its place in the telecommunication market. VoIP technology is gaining popularity in both commercial and residential markets because the voice quality resulting from packets transmitted over the IP network is comparable to the voice quality resulting from analog signals sent over the Public Switched Telephone Network (PSTN). The MEGACO/H.248 signaling protocol was introduced by the Internet Engineering Task Force (IETF) and International Telecommunication Union (ITU) to help control and manage the increasing volume of VoIP traffic. The MEGACO/H.248 signaling protocol employs a call control concept.

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The call control ―intelligence‖ or the master server resides in the Media Gateway Controller (MGC), while the Media Gateway (MG) serves as the slave device (dumb terminal). This concept reduces the complexity of the gateway, making it easier and more suitable for mass deployment. Gateway Architecture

The MEGACO/H.248 protocol employs the master/slave architecture, where the MGC acts as a master server, while the MG behaves like a slave device. In a deployed telecommunication network, one MGC may control multiple MGs. Media Gateway Controller: The MGC is the central point of intelligence for call signaling. It maintains the states of each MG and responds appropriately to any event notification. For instance, upon receiving an off-hook event from an MG, the MGC instructs the MG to play the dial tone and listen for the dual tone multi-frequency (DTMF) tones. Media Gateway: The master/slave architecture was designed to eliminate processor-intensive functionalities from the MG. Due to the reduced complexity, the cost of MG is much lower than the cost of MGC, making it more affordable to the commercial and residential markets. Essentially, the MG is a dumb terminal awaiting commands from the MGC for its next actions. Upon the successful creation of a

connection, the MG is also responsible for streaming the voice packets over the IP backbone using various encoding/compression algorithms.

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Realtime Transport Protocol (RTP) Multimedia services, such as video conferencing, Internet telephony and streaming audio has revealed that some modifications and extensions to the current internet protocols are needed to be able to support real-time applications better. Minimization of the end-to-end delay, accurate synchronization of the voice and video streams and a feedback mechanism for the quality of service monitoring are some of the main requirements of these various multimedia applications. The Transmission Control Protocol (TCP) is the most widely used transport-level protocol in the Internet. However, there are several facts that make TCP quite unsuitable for the real-time traffic. Firstly, TCP includes an in-built retransmission mechanism, which may be useless with strict real-time constraints. Secondly, TCP is a point-to-point protocol without direct support for multicast transmission. Thirdly, there is not any timing information carried, which is needed by most real-time applications. The other widely-used transmission protocol, User Datagram Protocol (UDP), does not either include any timing information. So, a new transport level protocol, called Real Time Transport Protocol (RTP) came into existence to cope with the aforementioned problems with the real-time traffic. RTP is a real-time end-to-end transport protocol. However, considering RTP as a transport protocol may  be misleading because it is mostly used upon UDP, which is also considered as a transport protocol. On the other hand, RTP is very closely coupled to the application it carries. So, RTP is best viewed as a framework that applications can use to implement a new single protocol. RTP doesn't guarentee timely delivery of packets, nor does it keep the packets in sequence. RTP gives the responsibility for recovering lost segments and resequencing of the packets for the application layer. There are a couple of benefits in doing so. The application may accept less than perfect delivery and with video or speech there usually is no time for retransmissions. Also the sender may provide, instead of retransmission, new or updated data that tries to fix the consequences of the original loss. What RTP then provides is •  •  •  • 

Payload type identification Source identification Sequence numbering Time-stamping

which are required by most multimedia applications. The accompanying RTP Control Protocol (RTCP)  provides feedback of the quality of the data delivery and information about session participants. A RTP session usually is composed of a RTP port number (UDP port), a RTCP port number (consecutive UDP  port) and the participant's participant's IP address. address.

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Packet Structure of RTP

The real-time media that is being transferred forms the 'RTP Payload'. RTP header contains information related to the payload e.g. the source, size, encoding type etc. However the RTP packet can't be transferred as it is over the network. For transferring a transfer protocol called User Datagram Protocol (UDP). To transfer the UDP packet over the IP network, it is encapsulate to IP packet.

The first 32 bits of the header consists of several control bits. The version number (V) is currently 2 bits. The padding bit (P) indicates if there is padding octets inserted at the end of this packet. Padding may be required by some applications with fixed length packet sizes. The extension (X) bit indicates if there is an experimental extension after the fixed header. The count field (CC) tells the number of contributing source identifiers (CSRC) following the fixed header. The marker bit (M) may be used as general marker, for instance indicating the beginning of a speech burst. The payload type (PT) field identifies the payload format. The sequence number is an incrementing counter which is started by a source from a random number. The timestamp corresponds to the generation instant of the first octet in the payload. The synchronization source identifier (SSRC) is a randomly generated value that uniquely identifies the source within a session. Even if it is very unlikely that two sources generate the same SSRC number, every RTP implementation should have a mechanism to cope with this chance. Following the fixed header there are one or more contributing source identifiers which are supplied by the mixer and the  payload. Payload Types

Before RTP may be used for a particular application the payload codes and the actual payload formats should be defined in a profile specification, which may also describe some application specific extensions or modifications to RTP. These payload types include for example G.721, GSM Full Rate, G.722 and G.728 speech codecs and JPEG and H.261 video codecs.

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Mixer in RTP As RTP is designed to support multicast transmission the RTP packet includes a source identifier (SSRC) which identifies the particular sender from the group. There are, however two special kinds of sources: a mixer and a translator.

It may so happen that all participants in a conference do not have the connection of same bandwidth. So how do they take part simultaneously? One solution is that all of them use a lower bandwidth. But this leads to reduced-quality audio encoding. A smarter solution exists in the use of a RTP-level relay called a mixer. A mixer may be placed near the low-bandwidth area. This mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packe packett stream across the low-speed link.

Translator in RTP

A problem occurs if one or more participants of a conference are behind a firewall which won't allow an IP packet containing the RTP message to pass. For this situation translators are used. Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site's internal network.

RTCP Overview   RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries

•

the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization synchroniza tion of multiple streams.

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  RTP  is originated and received on even port numbers and the associated RTCP 

•

communication uses the next higher odd port number. 

 

•

The RTP control protocol (RTCP) is based on the periodic transmission of control  packets to all participants in the session, using the same distribution mechanism as the

data packets.   It is recommended that the fraction of the session bandwidth allocated to the RTCP is 5%.

•

  The primary function of this protocol is to provide

•

feedback on the quality of the data distribution.

  RTCP specifies report PDUs exchanged between

•

sources and destinations of multimedia information

   

•

•

receiver reception report sender report

source description report   Reports contain statistics such as the number of RTPPDUs sent, number of RTP-PDUs lost, inter-arrival  jitter

 

•

•

 

•

Used by application to modify sender transmission rates and for diagnostics purposes

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Intelligent Network (IN) An intelligent network (IN) is a service independent telecommunication network. That is, intelligence is taken out of the switch and placed in computer nodes that are distributed throughout the network. Intelligent network is an architecture and implementation technology that allows services to be introduced, controlled and managed more effectively, economically and rapidly than permitted  by traditional network architecture. The greatest benefit of the IN architecture is that new services do not need to be implemented separately in each exchange. Instead, the services are introduced at the level of database. Standarised database interfaces permit the realization of services independent of the underlying technologies of the database and the exchange. In a telecommunications network, the main task for the switch is to process large volumes of calls. It is not designed for complicated tasks such as processing, accessing large databases, interaction through user friendly Web interfaces or creating new intelligent value added services. This is why we require Intelligent Networks (IN) for these tasks. IN performs the following functions:   Call routing: IN informs the switch on which number or carrier a call should be routed.   Call blocking: IN informs the switch if a call should be blocked or not.   Call monitoring: IN monitors calls and collects information, such as billing or invoicing data.

Basic IN Architecture

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Service Switching Point (SSP)  are stored-program control switches, either local exchanges or access tandem exchanges, that are able to interface with the SS7 signaling network. The Service Switching Point recognizes IN service calls and routes the corresponding queries to the Service Control Point via the SS7 network, which consists of Signaling Transfer Points (STPs). Service Control Point commands will be used by the Service Switching Point to further process the call. Service Control Point (SCP)   is typically an on-line, fault-tolerant, transaction-processing data base which provides call handling information as response to Service Switching Point queries. The service control point is where intelligence (Logic) of IN resides. The SCP receives requests from the SSP via the SS7 network. It then executes Service Logic Programs (SLP) in order to generate the instructions needed to carryout the requested service. The SCP send these instructions to the SSP, which uses them for call control functions that, in turn, perform the IN service.

In order to manage IN platforms a Service Management System (SMS) containing the reference service databases is required. Supervision, remote operations and maintenance of Service Control Points, and (coordinated) software downloading are part of the Service Management System features. The Service Management System is integrated in an Operations Support System which supports network operation, administration and maintenance functions, and normally resides in a commercial host computer. An additional Intelligent Peripheral (IP)  may be connected to a Service Switching Point, providing enhanced services/functions, such as announcements, speech synthesizing, speech recognition, etc. under the control of a Service Switching Point or Service Control Point. The motivation for the introduction of this network element is an economical aspect, because it might be better for several users to share an Intelligent Peripheral, when the capabilities of the Intelligent Peripheral are too expensive to be implemented at all Service Switching Points.

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IN Services 1.  IN services provide capabilities beyond basic telephony services (Plain Old Telephony Service POTS), e.g.: –  e.g.: –     flexible routing   flexible charging   advanced user interaction   enhanced customer control over specific service elements (customer profile management) 2.  For rapid service provision IN defines an extensible set of generic and reusable service components   Service Independent Building Blocks (SIBs) which can be combined to higher level service elements, i.e. Service Features   Service Features (SFs) which can be combined to build IN services

IN Service Examples The ―classic‖ centralised IN Services (without and with differentiation): –     Freephone, Premium Rate   Universal Access Number / One Number, Televoting   Calling Card Services, Virtual Private Networks IN infrastructure based services:    Number Portability, Portability, Mobility Mobility Services Services Combined and integrated IN services    Number Portability Portability + Personal Personal Communica Communication tion Services, Services, Unified Messaging, Messaging, Convergent Convergent Services, etc.

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IMT 2000 The next generation cellular, 3G, is envisioned to enable communication at any time, in any place, with any form, as such, it will:  

allow global roaming   provide for wider bandwidths to accommodate different types of applications   support packet switching concepts The ITU named this vision: IMT-2000 (International Mobile Telecommunications 2000) with the hope of having it operational by the year 2000 in the 2000MHz range.

IMT 2000 Vision  

Common spectrum worldwide (2.8 –  (2.8 –  2.2  2.2 GHz band)   Multiple environments, not only confined to cellular, encompasses: cellular, cordless, satellite, LANs, wireless local loop (WLL)   Wide range of telecommunications services (data, voice, multimedia, etc.)    

Flexible radio bearers for increased spectrum efficiency Data rates of:   9.6Kbps or higher for global (mega cell)   144Kbps or higher for vehicular (macro cell)   384Kbps or higher for pedestrian (micro cell) and   up to 2Mbps for indoor environments (pico cell)   Global seamless roaming   Enhanced security and performance   Full integration of wireless and wireline

Main Technical Implementati Implementation on 1.  2.  3.  4. 

UMTS (EUROPE) CDMA2000 (AMERICA) TD-SCDMA (CHINA) UWC-136

Universal Mobile Telecommunication System (UMTS)  

UMTS is developed by 3GPP (3 Generation Partnership Project) a joint venture of several organization

 

3G UMTS is a third-generation (3G): broadband, packet-based transmission of text, digitized voice, video, multimedia at data rates up to 2 Mbps

 

Also referred to as wideband code division multiple access(WCDMA) 124

 

 

Allows many more applications to be introduce to a worldwide

 

Also provide new services like alternative billing methods or calling plans.

 

The higher bandwidth also enables video conferencing or IPTV.

Once UMTS is fully available, computer and phone users can be constantly attached to the Internet wherever they travel and, as they roam, will have the same set of capabilities. Features  

It uses FDD/TDD duplexing method.

 

It uses the Bandwidth of 5 MHz.

 

The Chip rate is about 3.84 Mbps

 

It is very much flexible with 100/200 kHz carrier spacing.

 

The Frame length unit is 10 ms.

 

It uses BPSK for uplink and QPSK for downlink.

 

It has variable spreading factor.

 

The maximum data rat for indoor is 2 Mbps and for mobile it is 384 kbps.

 

The channel coding is convolution coding, turbo code for high data rate.

Types of Cells and Data Rates Macro Cell  

These cover a large area and will give slow access.

 

144 Kbps –  Kbps –  max  max speed of 500 Km/h. Low data rate.

Micro Cell  

These should cover a medium area.

 

384 Kbps max speed 120 Km/h. Medium data rate.

Pico Cell    

Less than 100 metres. 2 Mbps –  Mbps –  max  max speed of 10 Km/h. High data rate

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UMTS network services have different QoS different QoS classes for four types of traffic:

  Conversational class (voice, video telephony, video gaming)   Streaming class (multimedia, video on demand, webcast)   Interactive class (web browsing, network gaming, database access) 

 

Background class (email, SMS, downloading)

Architecture of UMTS

A UMTS network consist of three interacting domains; Core Network (CN), UMTS Terrestrial Radio Access Network (UTRAN) and User Equipment (UE). (UE). The main function of the core network is to provide switching, routing and transit for user traffic. Core network also contains the databases and network management functions. The basic Core Network architecture for UMTS is based on GSM network with GPRS. All equipment has to be modified for UMTS operation and services. The UTRAN provides the air interface access method for User Equipment. Base Station is referred as Node-B and control equipment for Node-B's for Node-B's is  is called Radio Network Controller Con troller (RNC).

Core Network

The Core Network is divided in circuit switched and packet switched domains. Few of the A. Circuit switched elements are   Mobile services Switching Centre (MSC),   Visitor location register (VLR) and

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  Gateway MSC. B. Packet switched elements are   Serving GPRS Support Node (SGSN) and   Gateway GPRS Support Node (GGSN). C. Some network elements, like EIR, HLR, VLR and AUC are shared by both domains.

Radio Access Network

Wideband CDMA technology was technology was selected for UTRAN air interface. UMTS WCDMA WCDMA is  is a Direct Sequence CDMA system where user data is multiplied with quasi-random bits derived from WCDMA Spreading codes. Spreading codes. In UMTS, in addition to channelisation, Codes Codes are  are used for synchronisation and scrambling.

UE  

It is not a simple mobile phone but rather, a mobile multimedia terminal provides simultaneously voice, video and data services.

 

UE is composed of two parts  

Mobile Equipment(ME)   Universal subscriber identity module (USIM). Information located in USIM are:   The personal identification Number(PIN).   The preferred languages   The codes to enable emergency call   One or several IMSI and MSISDN.   The user’s temporary identities allocated.  allocated.     Circuit and packet switched temporary location information. UTRAN  

The UMTS(UMTS Terrestrial Radio Access network) has two elements:   RNC   Node B.   UTRAN is subdivided into individual radio network (RNS), where each RNS is controlled by RNC.   The RNC is connected to a set of Node B elements, each of which can serve one or

several cells. 127

 

RNC  

The RNC enables autonomous radio resource management (RRM) by UTRAN.   The RNC handles protocol exchanges between Iu, Iur and Iub interfaces   The RNC uses the Iur interface for eliminating the burden from CN.  

Provide air interface between UE’s and Core Network). Core Network). Node B  

Node B is the physical unit for radio TX/RX with cells.   A single Node B can support both FDD and TDD modes.   The Main task of Node B is the conversion of data to and from th Uu radio interface, including forward error correction (FEC)   Node B also participates in power control.

Core Network The UMTS core network may be split into two different areas: Circuit switched elements: Carry data in a circuit switched manner, i.e. a permanent channel for the duration of the call. Mobile switching center (MSC): An exchange performing all the switching and signalling functions Functions  

call management   mobility management(handling attach and authentication)

 

subscriber administration maintenance of charging data(for radio network usage)   supplementary call services (call forwarding, etc.)

 

Gateway MSC (GMSC) Provides interconnection between the UMTS core network and external PSTN/ISDN networks. Packet switched elements: Carry packet data. This enables much higher network usage as the capacity can be shared and data is carried as packets which are routed according to their destination.

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Serving GPRS Support Node (SGSN) The SGSN provides a number of functions within the UMTS network architecture.  

Mobility management   Session management:   Interaction with other areas of the network:   Billing: Gateway GPRS Support Node (GGSN):  

Central element in UMTS.

 

It handles inter-working between the UMTS packet switched network and external packet switched networks.

Shared Elements The shared elements of the 3G UMTS core network architecture include the following network entities: Home location register (HLR):  

Contains all the administrative information about each subscriber along with their last known location

Equipment identity register (EIR):  

The EIR is the entity that decides whether a given UE equipment may be allowed onto the network or not on the basis of IMEI.

Authentication centre (AuC) :  

The AuC is a protected database that contains the secret key also contained in the user's USIM card.

Major Interface There are four major new interfaces defined in UMTS:

  Iu The interface between UTRAN and the CN   IU-CS between RNC and MGW/MSCS   IU-PS between RNC and SGSN   Iur The Interface between different RNCs   Iub The interface between the Node B and the RNC

•

•

•

•

•

•

  Uu The air interface between UE and Node B