Introduction to Telecoms

Introduction to Telecoms Course Code: TY2600

Duration: 3 days

Technical Level: 2

Other areas of expertise include: n

IP Networks and Protocols

n

Service Enablers

n

WiMAX

n

LTE

n

UMTS

n

GSM and GPRS

n

TETRA

n

CDMA

n

Transport and Signalling

n

Telecoms Industry Dynamics

www.wraycastle.com

INTRODUCTION TO TELECOMS

First published 2008 Last updated February 2011 WRAY CASTLE LIMITED BRIDGE MILLS STRAMONGATE KENDAL LA9 4UB UK

Yours to have and to hold but not to copy The manual you are reading is protected by copyright law. This means that Wray Castle Limited could take you and your employer to court and claim heavy legal damages. Apart from fair dealing for the purposes of research or private study, as permitted under the Copyright, Designs and Patents Act 1988, this manual may only be reproduced or transmitted in any form or by any means with the prior permission in writing of Wray Castle Limited.

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Introduction to Telecoms

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INTRODUCTION TO TELECOMS

CONTENTS Section 1

Telecommunication Services in the Modern World

Section 2

The PSTN and ISDN

Section 3

Transmission Networks

Section 4

Mobile Cellular Networks

Section 5

IP Packet Networks

Section 6

Fixed and Wireless Broadband Access Technologies

Section 7

VoIP, NGNs and the IMS

Appendix

Telecoms Evolution to Packet-Switched Data Networks

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SECTION 1

TELECOMMUNICATION SERVICES IN THE MODERN WORLD

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CONTENTS Telecommunication Network Evolution The Range of Telecoms Services

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.1

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.2

Requirements of a Telecommunication Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.3 Transmission Networks

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.4

Transmission Media . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.5 Traditional Legacy Networks and Transmission . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.6 The Importance of Data Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.7 Circuit Switching

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.8

Packet Switching

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.9

Traditional Telecommunication Signalling Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.10 Factors Affecting System Choice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.11 Bearer Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.13 Types of Bearer Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.14 Teleservices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.15 Supplementary Services (SSs) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.16 Value Added Services (VASs) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.17 Telecommunication Services Summary Telecommunications Regulation

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.18

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.19

Telecommunciations Standards Bodies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.20 Standards Bodies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .1.21

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Telecommunication Services in the Modern World

OBJECTIVES At the end of this section you will be able to:



list the basic services that are provided by today’s telecommunication networks



identify the main components of a telecommunication network



outline the operating principles of circuit-switched and packet-switched networks



describe the functions of signalling



identify the factors that affect the choice of transmission system



differentiate between bearer services, teleservices and supplementary services



discuss the similarities and differences between services offered by fixed-line operators and those offered by mobile networks



describe the features of supplementary services



explain the benefits to subscribers and operators of Value Added Services (VASs) and give examples of VASs



explain the functions of the various telecommunications standards bodies

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Telecommunication Services in the Modern World

Telecommunication Network Evolution The evolution of telecommunication technologies is a continuous process. The original telephone network was designed to provide voice communication and, in the very early days, a manually operated switchboard was used to interconnect the calling and the called parties. An electro-mechanical system was developed to allow the call to be connected by the calling party using a dial. This system became known as the Strowger switch (named after its inventor) and was the first automatic telephone switching system. The Public Switched Telephone Network (PSTN) was based on analogue technologies and was voice centric. Customers connected to the network over analogue copper wires. In the 1980s, the network evolved to a digital infrastructure, which paved the way to the Integrated Services Digital Network (ISDN). This allowed customers to have digital access to the network, and in addition to voice, computer data could be transmitted over the network at faster data rates. Until the 1990s, a limited number of services were available to customers. These were mainly provided through fixed networks. Since the emergence of digital mobile networks, and the rapid increase in mobile-phone usage, many more services have become available and the dividing lines between fixed and mobile services have become blurred. Fixed and mobile networks have some services in common, such as telephony and data, while other services, such as the Short Message Service (SMS) and location-based services, are particular to mobile networks. With the introduction of fixed and mobile broadband technologies, many services are now being provided on both fixed and mobile networks. The services themselves are carried by the Internet Protocol (IP), which provides a common protocol and technology, thereby reducing costs. The data rates available in mobile broadband networks are approaching those provided by the fixed broadband architecture. This trend is enabling a convergence of services between fixed and mobile networks. Access to the World Wide Web (WWW), video streaming, file downloads, online shopping, music downloads, social networking and general information exchange can now be carried over fixed access networks utilizing copper wire infrastructure and mobile networks utilizing radio access networks. IP and other emerging technologies are changing telecommunications dramatically. These packet-based systems allow a wide range of services to be provided over both fixed and mobile broadband networks.

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Introduction to Telecoms

Mobile Phones

Broadcast (TV and Radio)

Blogs Telephony

WWW Internet

Facsimile

£

File Transfer

Telebanking

Fixed Networks Mobile Networks IP Based Networks

E-mail

e-commerce

Video Streaming

File sharing Music Downloads

Social Networking

Instant Messaging

Telemetry Short Messages LANs

MANs

WANs

Alarm Calls

The Range of Telecoms Services The world has become accustomed to a sophisticated telecommunication system that delivers a wide range of services to its users. Although the most widespread service is still telephony, there is an increasing need to move information in the form of data for a variety of applications. The applications that a particular system or network supports and offers to its users are termed ‘services’. Each service has different information related to it, and each requires the use of different techniques. The provision of equipment within a telecoms system must therefore take this into account. The ideal system has the flexibility to use the same equipment for all services rather than providing separate, specialist equipment. By employing technologies such as IP, new, advanced services can be combined with traditional services into a converged network. With this network, much of the equipment can be simultaneously utilized for real-time and non-real-time traffic, for data rates which are variable or constant or for high-speed or low-speed data transfer. This makes for a far more efficient use of network resources. The way in which traditional services have been provided is changing. For example, in addition to receiving broadcast television over a radio link from the main transmitter, television programmes can be carried over the Internet using IPTV and Video on Demand (VoD) technologies. Social networking web sites and web-logs (blogs) are popular services. Blogs are mainly text based, but some focus on particular interests such as art, photography, videos, music and audio (podcasting). Services such as micro-blogging (e.g. Twitter) are expanding rapidly. The traditional PSTN voice network is evolving towards Voice over IP (VoIP), which can be combined with other services to create new multimedia services and content.

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Telecommunication Services in the Modern World

User (Application)

Transmission System

The general requirements of the network are to convey information reliably and faithfully between any two points, over any distance, and at a reasonable cost.

Switching System

Signalling System

Requirements of a Telecommunication Network As the services offered to customers become more sophisticated, so the equipment and network infrastructure needed to support them becomes more complex. Although some services may be supported within the standard telephony network, others require a dedicated network in order to satisfy these requirements. A number of network types have evolved to support a variety of applications. However, certain general requirements are common to most networks. These are to convey information reliably and faithfully between any two points, over any distance, and at a reasonable cost. To meet these requirements, there are four major components of a communications network: transmission, switching, signalling (or control), and user equipment with applications. The type of information conveyed depends upon the user application and its associated equipment. Reliable and faithful transport of information over a distance requires the use of transmission techniques, while switching and signalling ensures that any two points can be interconnected automatically, wherever they are in the network.

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Equipment A

Input Data

Customer Premises Equipment (CPE) Switches etc.

Output Data

Information Pipe

Equipment B Customer Premises Equipment (CPE) Switches etc.

Transmission Network high availability low Error Rate resilient easy circuit provisioning

Digital Transmission Networks Tributary Interfaces

Multiplexing Equipment

Copper Optical Radio Systems

Add/Drop Facility

Cross Connect

PDH SDH WDM OTN

Transmission Networks Transmission networks are designed to transfer information over any distance from one point to another. To the user, the ideal transmission system would be error free and available at all times. From an operator’s perspective a transmission network should enable ease of circuit provisioning, be resilient to failures and reduce the cost per bit transmitted. Transmission systems can be said to provide a simple information pipe between two devices. The size or capacity of the pipe will vary depending on the amount of data to be transferred between the input and the output. In general terms, transmission systems consist of tributary interfaces, multiplexers, add/drop facilities, cross-connects and line systems. Four main transmission technologies are in use in modern systems. They are Plesiochronous Digital Hierarchy (PDH) and Synchronous Digital Hierarchy (SDH), which are digital systems; Wavelength Division Multiplexing (WDM), which provides optical multiplexing, and Optical Transport Networks (OTNs), which may be considered as hybrid networks providing capabilities similar to SDH and WDM systems. PDH systems were originally introduced into trunk systems in the USA in the early 1960s. SDH systems reflect the modern approach to digital transmission systems, having been in use since 1985; again originating in North America under the title of Synchronous Optical Network (SONET). The physical components of a transmission system consist of termination equipment and the transmission link.

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Telecommunication Services in the Modern World twisted pair in the access network twisted pair on main distribution frames  coaxial cable between equipment in a transmission building  computer Local Area Networks (LANs)  USB cables  etc.  

Metallic Wire

between towns and cities international links  submarine cables  between equipment in secure data centres  etc. 

Fibre Optic Cable



radio access for mobile phones microwave links between towns and cities  outside broadcasts  satellite communication  etc. 

Radio



Transmission Media There are three main types of transmission media: copper wire, optical fibre and radio. Copper wire is now mainly to be found in the fixed access network between the user and the network. By employing Digital Subscriber Line (DSL) techniques, copper twisted pairs are capable of bit rates up to 100 Mbit/s, although this figure will depend on the length of the wire and its thickness (au). Optical fibre has a much greater capacity than copper wire and is used on major routes between towns, cities and countries. By employing WDM techniques using up to 160 optical channels, with each channel carrying 10 Gbit/s, the aggregate bit rate of an optical fibre can be very large. Total data rates can be several terabits per second, sufficient for transmitting many millions of telephone channels simultaneously. The OTN enables the multiplexing and switching of optical wavelengths between fibres. In addition, an optical fibre cable can contain multiple fibres. Radio has less capacity than fibre but it has the advantage of being cheaper and more flexible to install when compared with cables. Another advantage of radio is that it can enable mobility, as in a cellular network. Point-to-point radio links can carry data rates in excess of 600 Mbit/s.

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Traditional Legacy Networks and Transmission Traditional transmission networks concentrated on supporting circuit-switched services and were required to support access connections and, in the communication provider’s core network, inter-switch site connections. The term ‘access network’ refers to the part of the network used to connect the customer to the network. Traditionally this has been implemented with a pair of twisted metallic wires (copper or aluminium) between the customer’s premises and the local switch (exchange). The average length of this wire in the UK is about 4 km. When the customer picks up the telephone handset, the two copper wires are connected together forming a loop. Consequently, the access network is often known as the ‘local loop’. Other names have been given to the copper fixed access network, such as the ‘subscriber line’ and the ‘last mile’. The copper wires terminate in the local exchange on a Main Distribution Frame (MDF), which allows flexible connections inside the exchange building. A concentrator is often provided to maximize the use of revenue-generating switch ports; this also provides a range of customer interface types, a basic telephony line or a line termination from a PBX, for example, while allowing the switch to have a single interface type. The former function is achieved by dynamically connecting customer lines to switch ports when the customer picks up the phone to use the line. By doing this, the concentrator can support more customer-facing lines than switch-facing ‘lines’; ratios of between 4 and 10:1 are typical. Using a 10:1 concentration ratio, for example, there may be 100,000 customer-facing terminations and only 10,000 switch terminations. In this case it is more likely that 10,000 out of 100,000 customers will make simultaneous calls than would be the case if only 10,000 customers were directly connected to the switch, hence making best use of the revenue-generating capabilities of the switch. Calls set up between customers via the switch network are referred to as ‘switched services’. As well as connection to the local exchange some corporate customers require secure dedicated inter-site connections for their data. Typically this is provided as a ‘leased line’ service via the transmission network. Leased lines can be used for voice, data or Internet services. Typically, leased lines are available at data rates from 64 kbit/s increasing in increments to 2 Mbit/s (E1). Higher-bit-rate services in the access loop to the customer are often provided using optical fibre connections or microwave radio links as an alternative to metallic pairs. The transmission links between switches are normally implemented as multiple 2 Mbit/s (E1) connections. PDH and SDH transmissions systems provide an effective mechanism for combining multiple E1 connections onto a single higher-capacity signal. This signal may then be transmitted between towns and cities over optical fibre or microwave radio systems.

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Telecommunication Services in the Modern World GPRS 40 k e-mail

Web Page

(no attachments)

Broadband (DSL) 8 M

UMTS 384 k

50 kbytes

200 kbytes HSDPA 3.6 M

GPRS 40 k Photograph Broadband (DSL) 8 M

Video Clip

UMTS 384 k

700 kbytes

GPRS – General Packet Radio Service UMTS – Universal Mobile Telecommunications System DSL – Digital Subscriber Line

HSDPA 3.6 M

10 Mbytes

The Importance of Data Rates The time taken to transmit and receive data is dependent upon the capacity of a network and, if a radio system, on the spectrum bandwidth allocated. The higher the data rate of a system, the shorter the time taken to transmit and receive the data. Where services are offered using radio systems, sufficient spectrum bandwidth must be allocated to provide a data rate which satisfies the speed requirements of the service. For example, waiting 33 minutes to download a short video clip would be unacceptable. The diagram shows the data rates of various telecommunication systems and the length of time it takes to receive different types of information, assuming no system delays and no inclusion of additional data. It is important to recognize that the higher the data rate, the greater the range of multimedia services that can be provided, such as voice, video or IPTV.

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A

Sorry, all the lines are busy

Circuit Switching Initial set-up delay Short delay

B

Constant delay Single connection

D

Circuit Switching In any given network, switching points are usually interconnected to form a mesh through which calls are routed from one terminal to another, based on the terminals’ unique addressing (numbering) scheme. The network is provided for every terminal to use, but the way in which the resources are allocated may vary depending on the switching mode implemented. Different modes of switching may be present in the same transmission network. There are two main switching methods: circuit switching and packet switching. Circuit switching involves a physical circuit being established between two terminals for a period of time. The circuit is allocated exclusively to the users for the duration of the connection. These resources only become available to other terminals upon release of the connection. Circuit switching has several advantages. Firstly, the end terminals/devices/users are allocated the full data carrying capability of the connection for the duration of the call irrespective of whether they have data to send. In addition, although the initial routing process (or set-up) takes time, further data exchange via the switches is relatively short because it does not involve the analysis of addresses; the data simply flows through the physical connection. Finally, as all data for the connection takes the same physical route, the time taken for data to be transferred between terminals is kept constant throughout the call. There are also disadvantages with circuit switching. Although the parties may cease to talk, network resources are still allocated to them. This is a disadvantage to both the user and the network operator. From the user’s point of view, a connection is being paid for even though no data is being sent; from the network operator’s view, the network may have no resources available for waiting users. Also, should the network fail, the connection is lost completely and a new connection will need to be created.

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A

C

B

D

Packet Switching Long delay Variable delay Shared resources

Packet Switching Packet switching involves the segmentation of users’ information into smaller blocks of data known as ‘packets’. Routing is carried out on a packet-by-packet basis; each packet is fed into the network and passed from one switching point to the next until it reaches its destination. Each packet must therefore contain addressing information to allow the routing process to take place at every switch on the route through the network. Within this basic type of packet network the packets are referred to as datagrams. An advantage of packet switching is that terminals share the network, each terminal being allocated network resources only when they have packets (or datagrams) to send. From the users’ perspective this is advantageous as they pay only for the data sent as opposed to the length time they are connected, i.e. users can be charged per packet as opposed to per second. From the network operator’s point of view, packet switching allows all users to be given access to the network. It also allows for more efficient network dimensioning, which can lead to financial savings that can ultimately be extended to the user. In addition, as each packet is handled separately, any failure within the network need not affect the transfer of packets. The network can simply route the packets via an alternative path so bypassing any failed elements. One disadvantage of packet switching is that the delays can be relatively long within the switches themselves. Although packet switching does not incur the set-up and release times associated with circuit switching, routing must be performed on every packet arriving at every switch and this will incur a delay. In addition, the delay between a packet arriving at a switch and it being routed onwards may vary. This is because a packet switch may need to queue packets for sending onwards, the length of the queue varying with the number of packets involved with the onward leg.

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Introduction to Telecoms Intelligent Network (IN) Value Added Services (VAS)

Access signalling system

Network signalling system

Access signalling system

Network signalling e.g. SS7 Access signalling e.g. loop disconnect, DTMF

Access signalling DSS1

Private network signalling e.g. DPNSS, Q.SIG PBX

PBX

DPNSS – Digital Private Network Signalling System

Traditional Telecommunication Signalling Systems Signalling is used within a network to set up, maintain and terminate a call. Signalling comprises the access signalling system, for customer access, and the network signalling system, for use in the network itself. The access signalling system is extended out to the customer and works in conjunction with recognizable tones to keep the customer informed about the progress of the call. In general, this does not have to be as sophisticated as the network signalling system, but again, the greater the level of sophistication, the more flexible and complex services can be. Access signalling systems include Loop Disconnect with pulse or, more efficiently, Dual Tone Multi-Frequency (DTMF) dialling and Digital Subscriber Signalling System No 1 (DSS1). The main network signalling is based on Signalling System No. 7 (SS7), which is used between switches in the network. The introduction of SS7 allowed the implementation of new services such as Calling Line Identification (CLI). SS7 is the basis of Intelligent Networking (IN). An IN is a service-independent telecommunications network where intelligence is taken out of the switch and placed in computer nodes that are known as Service Control Points (SCPs), which are distributed throughout the network. This provides the network operator with the means to develop and control services more efficiently. New capabilities can be introduced rapidly into the network and then the services are easily customized to meet individual customers’ needs. With the IN infrastructure, service providers are equipped with mechanisms allowing them to create, test, deploy, and support value-added voice services. Examples are credit card calling, malicious call identification and selective call forward on busy or no answer. In addition to access and network signalling, it is possible to provide the means for end users to communicate with each other using a specified signalling system. This is usually achieved via the access and network signalling systems. An example of a private network signalling system is the Digital Private Network Signalling System (DPNSS). This is a signalling protocol used to connect two PBXs (Private Branch Exchanges).

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Telecommunication Services in the Modern World User (Application)

Factors information type bandwidth/data rate application analogue or digital real time/non-real time cost

Transmission System information type distance and terrain cost bandwidth/data rate security and reliability analogue or digital choice of transmission medium, copper/coaxial cable, optical fibre, radio

Switching System

network size network capacity information type analogue or digital dedicated circuit or only when data to send switching speed position – private/public switch

Factors Affecting System Choice Any potential network operator, whether fixed or mobile, needs to take several considerations into account in order to maximize their network’s potential and satisfy the needs of the customers they are targeting. These considerations include the application (also known as the ‘user’, such as voice); the transmission system; the switching system; the signalling system, and the cost. Although telecommunications usually deals with electromagnetic signals, the form, rate and use of the information varies greatly from application to application. The type of information to be transmitted has a great impact on how the system is engineered. For example, basic voice information needs to be sent from one telephone to another at the instant the user speaks, but a high degradation of the signal may be allowed because of the tolerance of the human ear. On the other hand, computers need to exchange accurate information, but often the time at which it is sent is not critical. The transmission techniques employed must take into account the type of information, the distances which need to be covered, the terrain and the cost of provision. If the information requires a high bandwidth (for example, video pictures) a medium capable of supporting these requirements must be chosen: fibre optics or satellites, for example. Long distances may be best served by providing satellite links or radio links, although cable systems could ultimately provide better quality. It is usual to have many users connected to a telecommunication system. Users need to be able to contact other users. Each terminal connected to the system must therefore have a unique identifier, for example a telephone number or an Internet address. This number or address is used to route the information to the correct end-point. Routing employs dynamic switches, which use this identifier to provide a path from the originating terminal to the destination. A hierarchical approach is usually used to provide switching (and transmission) throughout a network at reasonable cost. The switching technique varies depending on whether a dedicated circuit is needed, or whether the application allows the switched path to be provided only when there is information to send.

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Signalling System A

Factors B

information to be transferred complexity of the services offered position within the network (access or core network) cost security

Cost acceptable service at acceptable cost a balance between: – quality – reliability – features – services

Factors Affecting System Choice (continued) Signalling informs the switches of the paths that must be provided through the network. In addition, signalling is generally used to set up, supervise and release calls and to provide control of supplementary services and advanced features. The bottom line is to provide an acceptable service at a cost which the customer is willing to pay. A balance has to be struck between quality, reliability, features and services.

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Bearer

Bearer

The network provides the means to transfer information

Audio or data e.g. 3.1 kHz audio 64 kbit/s Datagram (real time/non-real time)

Circuit Switch or Packet Switch

Bearer Services Every telecommunication network offers services to its users. These can be identified as bearer services, teleservices and supplementary services. Networks may also choose to offer a range of value-added services to their customers. The bearer service is the most basic service offered by the network. It is simply a means of getting the user data (analogue or digital) from source to destination. The bearer can vary depending on the user’s requirement. For example, the network may offer a bearer service of a 3.1 kHz audio circuit; this could be via a pair of wires across the network. Or it could offer a bearer service of 64 kbit/s, down which the user could put different types of information (speech, data, etc.). Another type of bearer service is a packet-switched service, where the information is broken down into packets at one end of the circuit and reassembled at the other end. The information contained in the packet may either be real time (sensitive to delay), or non-real time (not sensitive to delay).

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Introduction to Telecoms

Data Communication Services

Telephony

Telephone Network (PSTN/ISDN)

PSPDN

VoIP

IP Network

Bearer Networks

Analogue GP3 Fax (ITU-T T.30)

Facsimile

Fax over IP (ITU-T T.38)

PSPDN – Packet-Switched Public Data Network

Types of Bearer Service Examples of bearer services are telephone networks, i.e. the Public Switch Telephone Network (PSTN) or the Integrated Services Digital Network (ISDN) and Packet-Switched Public Data Networks (PSPDNs). Customers may wish to subscribe to the bearer service (information transfer mechanism) and provide the means to complete the communications package themselves. In this case, the customer will have to ensure that any hardware, such as the terminal, is compatible with the bearer service. A bearer can be used for several teleservices at the same time. Thus, the telephone network may be used as a bearer for teleservices including telepnony and fax and other forms of data. A number of bearer services may exist within the same transmission network, perhaps occupying different lines within the same cable. These lines could belong to such bearers as the telephone network or a data network.

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Telecommunication Services in the Modern World

Telephony

Telephony

Bearer

Bearer SMS

SMS

Fax

Fax

End-to-end communication using the network’s bearer services

Switching

Teleservices Teleservices use a network’s bearer services for end-to-end communication. An example of a teleservice is speech, the speech being carried over the bearer service from source to destination. Networks offer a range of teleservices to their subscribers. Teleservices may include Telephony, Fax, Short Message Service (SMS) and Emergency Calls. Teleservices define the compatibility of terminals as well as a suitable bearer service type.

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Introduction to Telecoms

SOFTWARE Ring Back Call Waiting Advice of Charge Calling Line Identification Call Forwarding Ring Hold Call Barring Multiparty

Supplementary Services (SSs) Although basic teleservices include speech, fax and data, a network operator may want to make its overall service appear more attractive to subscribers. Supplementary Services (SSs) which modify or provide a basic telecommunications service, include:     

calling line ID (presentation or restriction) call divert advice of charge barring of outgoing calls barring of incoming calls

It is common for networks to offer a wide range of such services. Some of these are shown in the diagram.

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Telecommunication Services in the Modern World Problem: Call charges are going down. How can I generate new revenue streams to increase profitability?

Call Charges

Time

Solution: Value Added Services (VAS) INBOX Fax, SMS, Voicemail, e-mail

Multimedia Messaging

Unified Messaging

Intelligent Network (IN)

Location-based Services

Service Enabler

… and many others.

Value Added Services (VASs) Call costs are constantly being driven down due to competition. This is ideal from the customer’s perspective, but from the service provider or network operator’s view, it poses a question: ‘How can I generate new revenue streams to compensate for the drop in call charges?’ One solution is the provision of VASs in addition to teleservices and SSs. Examples of VASs are found in both fixed and mobile networks, and include such services as location-based services, multimedia messaging and Unified Messaging (UM). Typically, VASs are implemented using INs. INs do not define the services to be delivered; they provide an overlay infrastructure on existing circuit-switched networks. An IN acts as a service enabler allowing communication providers to develop, test and deploy attractive operator-specific services which may act as a differentiator between themselves and other providers. This helps the provider to retain their existing customers and generate new revenue streams from them, and also to attract new customers and further increase revenue.

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1.17

Introduction to Telecoms Telecommunication Services

PSTN PSPDN etc.

Teleservices

Bearer Service

Basic Service

Supplementary Service

telephony

call forwarding

data

multiparty

fax

call waiting

64 kbit/s data

call hold

SMS

calling line ID

9.6 kbit/s data

Basic Service

Supplementary Service

4.8 kbit/s data 3.1 kHz audio

Plus VASs to increase revenue

Telecommunication Services Summary In telecommunications, any service that is offered to a customer is known as a telecommunication service. This collective term can be divided into two main categories, bearer services and teleservices. The basic services for both teleservices and bearer services can be modified by a supplementary service. For example, the basic bearer service of 64 kbit/s (for voice) can be modified by the network building a bridging circuit to link more than two subscribers together, a service known as ‘multiparty’. A network operator may attempt to increase revenue by offering SSs and VASs to their customers.

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Telecommunication Services in the Modern World

Government

Appoints/ runs

Office of Telecommunications (Oftel) Independent Broadcasting Authority (IBA) Independent Television Commission (ITC) Broadcasting Standards Commission (BSC)

Telecoms

Radiocommunications Agency (RA) Radio Authority

Broadcast Radio Convergence of technologies and market leads to

Single (‘super’) Regulator

Telecommunications Regulation The telecommunications industry is highly regulated. Operators have critical obligations over and above providing a range of services. These include the provision of emergency call services (999, 112, 911, etc.) and, in some cases, ‘universality’ – the right for customers to be connected to networks at a fair and level cost for all (even the connection cost may vary dramatically). A telecommunications regulator is usually an agency or a department of central government. A regulator’s role encompasses regulation and dealing with competition issues; it often acts as a mediator in disputes between network operators. With the increasing convergence between services, networks and technologies the regulators in many countries are being restructured. Traditionally, different bodies or regulators were responsible for telecommunications, radio (spectrum) and broadcast, but these are sometimes now combined in a single agency. An example of this regulation convergence is the creation of the Office of Communications (Ofcom) within the United Kingdom. Ofcom assumed its authority at the end of 2003. It was created by bringing together the following roles:      

Radiocommunications Agency (RA) Independent Broadcasting Authority (IBA) Independent Television Commission (ITC) Broadcasting Standards Commission (BSC) Office of Telecommunications (Oftel) Radio Authority

The combined role of Ofcom covers all aspects of telecommunications and broadcasting including the regulation of content and the handling of complaints from customers.

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1.19

Introduction to Telecoms

International Telecommunication Union (ITU)

International Organization for Standardization (ISO)

Third Generation Partnership Projects (3GPP/3GPP2)

Telecoms Industry Association (TIA)

Regional Telecoms Institutes World Wide Web Consortium (W3C)

European Telecommunications Standards Institute (ETSI)

Internet Engineering Task Force (IETF) Regional Standards Institutes (ANSI/BSI)

Electronic Industries Alliance (EIA) Broadband Forum (BBF)

Telecommunciations Standards Bodies Standards are critical in an industry such as telecoms because they are key to providing a competitive environment. Standards create so-called ‘open systems’, which by means of standard interfaces allow a network operator to build a multi-vendor network and allow customers to take their terminal from one network to another network. However, the legacy of telecommunication development in different countries and regions means that there are variations in technical standards around the globe. For example, a telephone terminal purchased in the UK will not be capable of connection in other European countries or North America as the line jack is different; and even if this was replaced, line signal differences would still lead to incompatibility. The same is true for mobile telephony, where different standards are used in different global regions. A diverse range of national, regional and international bodies are responsible for standards that apply to telecommunications networks, terminals and services. There is often a complex relationship between these bodies and large manufacturers and network operators will ensure that they have representatives in the important groups within the standards committees. The subject of standards can be very ‘political’, particularly when a company wants its technology to become, or be included in, a standard. This way the company may gain a lead in the market by virtue of already having the technology working; secondly, they may earn revenue from licensing their technology to others (so-called Intellectual Property Rights – IPR). IPR has caused major legal issues in telecoms in recent years and most standards bodies insist that for a technology to be adopted within a standard it must be available for license on a fair and reasonable basis.

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Telecommunication Services in the Modern World

International Telecommunication Union (ITU)

International Organization for Standardization (ISO)

Third Generation Partnership Projects (3GPP/3GPP2)

Telecoms Industry Association (TIA)

Regional Telecoms Institutes World Wide Web Consortium (W3C)

European Telecommunications Standards Institute (ETSI)

Internet Engineering Task Force (IETF) Regional Standards Institutes (ANSI/BSI)

Electronic Industries Alliance (EIA) Broadband Forum (BBF)

Standards Bodies The diagram shows some the major standardization bodies responsible for producing standards and recommendations used by the telecommunications industry. The principal organizations are described below. International Telecommunication Union (ITU) The ITU has its headquarters in Geneva and is part of the UN. It is responsible for the international coordination of telecommunications and networks. To achieve this aim it is divided into three entities: the Telecommunication Standardization Sector (ITU-T), the Radiocommunication Sector (ITU-R) and the Development Sector (ITU-D). European Telecommunications Standards Institute (ETSI) ETSI, based in France, is an independent, non-profit organization that is officially responsible for standardization of Information and Communication Technologies (ICT) within Europe. ETSI comprises 766 members from 63 countries including manufacturers, network operators, research bodies and users. Third Generation Partnership Project (3GPP) 3GPP, formed in December 1998, unites the telecommunications standards bodies ARIB, CCSA, ETSI, ATIS, TTA, and TTC, which are known as ‘Organizational Partners’. 3GPP was formed to produce globally applicable specifications and reports for a third-generation (3G) network, which evolved into the Universal Mobile Telecommunications System (UMTS). 3GPP also has responsibility for the specifications for the Global System for Mobile Communications (GSM), the General Packet Radio Service (GPRS) and Enhanced Data rates for Global Evolution (EDGE) technologies. Internet Engineering Task Force (IETF) The IETF is an international body of network designers, operators, vendors and researchers concerned with the evolution of the Internet architecture and effective Internet operation. The IETF’s work is done in Working Groups (WGs), which are organized by topic (e.g. routing, transport, security). International Organization for Standardization (ISO) The ISO identifies requirements for international standards and goes on to develop those standards in collaboration with the industry sectors that will put them into use. It is a non-governmental organization comprising the standards organizations of 162 counties (one per country) from every world region. In 2010, the ISO’s portfolio contained over 18,500 standards.

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1.21

Introduction to Telecoms

International Telecommunication Union (ITU)

International Organization for Standardization (ISO)

Third Generation Partnership Projects (3GPP/3GPP2)

Telecoms Industry Association (TIA)

Regional Telecoms Institutes World Wide Web Consortium (W3C)

European Telecommunications Standards Institute (ETSI)

Internet Engineering Task Force (IETF) Regional Standards Institutes (ANSI/BSI)

Electronic Industries Alliance (EIA) Broadband Forum (BBF)

Standards Bodies (continued) Telecommunications Industry Association (TIA) The TIA represents those who provide information and communications technology products and services to the global marketplace. It does this through standards development, domestic and international policy advocacy, and business opportunities for members. The Association enables convergence between communications networks as well as promoting a competitive and innovative market environment. Electronics Industries Alliance (EIA) The EIA is a trade organization comprising an alliance of trade associations for electronics manufacturers in the USA. Those associations in turn govern sectors of EIA standards activity. The associations include the TIA and the Electronic Components Association. The EIA is accredited by ANSI to help develop standards on electronic components, consumer electronics, electronic information, telecommunications and Cyber security. For more information see http://www.ecaus.org/eia/site/index.html The Broadband Forum The Broadband Forum is the central organization driving broadband wireline solutions and empowering converged packet networks worldwide to better meet the needs of vendors, service providers and their customers. Their aim is to develop multi-service broadband packet networking specifications addressing interoperability, architecture and management. For more information see http://www.broadband-forum.org/about/mission.php The World Wide Web Consortium (W3C) Established in 1994 and with over 400 member organizations from around the world, the World Wide Web Consortium (W3C) promotes the WWW application of the Internet by developing common protocols and ensuring interoperability. The W3C’s overall goal is to ensure that the Web is accessible to all and has a standardized design. The W3C plays an active role in setting standards for the Internet and fostering co-operation to make the standards freely available. W3C standards include: the language of the Web, Hypertext Markup Language (HTML), the ‘next generation’ language, eXtensible Markup Language (XML) and MathML for marking up mathematics for the Web.

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Introduction to Telecoms

SECTION 2

THE PSTN AND ISDN

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I

Introduction to Telecoms

CONTENTS Basic Structure of the Public Switched Telephone Network (PSTN) . . . . . . . . . . . . . . . . . . . .2.1 Analogue and Digital Telecommunication Networks

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.2

The Analogue World . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.3 Signal Concepts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.4 Speech Signals

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.5

Commercial Speech Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.6 Two-Wire Transmission in the Local Loop

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.7

Switching in Telephony Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.8 Public Switches Hierarchy and Functions

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.9

Private Switches . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.10 Numbering Plans . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.11 Call Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.12 Analogue and Digital Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.13 Digital Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.14 Analogue Signal Quality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.15 Digital Signal Quality

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.16

Bit Rate and Baud Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.17 Analogue to Digital Conversion

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.18

Interworking between A-Law and µ-Law

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.19

Other Types of Voice Coders . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.20 The Need for Multiplexing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.21 Time Division Multiplexing (TDM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.22 Primary Level Multiplexers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.23 Timeslot Interchange . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.24

II

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The PSTN and ISDN

CONTENTS PSTN and ISDN Digital Switches . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.25 Evolution from PSTN to ISDN

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.26

The ISDN Digital Subscriber Line – Basic Rate Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.27 BRA Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.28 The ISDN Subscriber Line – Primary Rate Access (PRA) . . . . . . . . . . . . . . . . . . . . . . . . . . .2.29 Signalling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.30 Channel Associated Signalling (CAS)

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.31

Common Channel Signalling (CCS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.32 Access/Network Signalling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.33 Basic Call Process and the ISDN User Part (ISUP) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.34 End-to-End Call Setup – Basic Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.35 Services for Revenue Generation The IN Concept

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.37

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.38

IN Implementation Simplified . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.39 An Example of an IN Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.40 IN-based Services

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2.41

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III

Introduction to Telecoms

IV

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The PSTN and ISDN

OBJECTIVES At the end of this section you will be able to:



state the general requirements of a telephony network



explain the terms ‘local loop’ and ‘final mile’



outline the architecture of the Public Switched Telephone Network (PSTN)



identify the respective roles of access and core networks



describe the international numbering plan for telephony services



describe how numbers are used by switches to route calls



list the advantages of digital signals over analogue signals



describe the principles of Pulse Code Modulation (PCM)



define the Integrated Services Digital Network (ISDN) concept



describe the concept of an Intelligent Network (IN)

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V

Introduction to Telecoms

VI

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The PSTN and ISDN Access Signalling e.g. DTMF

Telephone Local Exchange

PBX

Computer Copper Wire Local Loop Access Network

Core Transmission Networks (Fibre or Radio)

Core Exchanges

Core Signalling System No. 7 (SS7)

(National and International)

PBX Telephone Local Exchange DTMF – Dual Tone Multi Frequency

Computer Copper Wire Local Loop Access Network

Basic Structure of the Public Switched Telephone Network (PSTN) The Public Switched Telephone Network (PSTN), also known as the Plain Old Telephone Service (POTS), is the fixed telephone network over which landline telephone calls are made. The PSTN relies on the principle of circuit switching, where the connection is set up through the network by signalling protocols before speech can commence. To connect one phone to another, the call is routed through numerous switches or exchanges operating on a local, regional, national or international level. The connection established between the two phones is called a circuit. In the early days, phone calls were routed as analogue signals all the way to the receiving end. During the 1970s, analogue voice calls began to be converted into digital format and the switching was carried out by microprocessor-controlled digital switches. This provided the basis for the Integrated Services Digital Network (ISDN). Digital techniques allowed a more efficient way of combining different types of traffic such as speech and data, and enabled a better quality of service for the customer. A variety of terminal equipment can be used by the customer including telephones, fax machines, computers and Private Branch Exchanges (PBXs). Within a company or larger organization, each employee or department can have their own extension telephone. Extensions from the main phone number are routed through a PBX that is located at the company’s premises. The local loop has for the most part remained as a twisted copper pair and is dedicated to one customer. However, in some situations radio is used in the access network, which is known as Fixed Wireless Access (FWA). The development of optical fibre transmission techniques now allows thousands of calls to be transferred between switches. However, these high-capacity fibre and radio transmission links have not changed the basic nature of the circuit-switched network, which still requires a connection or circuit to remain open for the duration of the phone call. Various signalling systems have been developed to control the switches in the network. The access signalling protocols are used between the customer and the local exchange – some for basic telephone connections such as DTMF and others for private signalling from a PBX. In the core network between switches, Signalling System No. 7 (SS7) is used.

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2.1

Introduction to Telecoms (Integrated modem)

(Integrated modem)

Fax

Fax

Original Analogue PSTN

PBX

PBX

(voice centric) Telephone

Telephone Modem Computer

Analogue Access Network

Modem Computer

Digital Access Network Digital fax

Digital fax

Integrated Digital

Digital PBX

Digital PBX Internet Service Provider (ISP)

Network Computer A/D Digital Telephone

Analogue Telephone

D/A

Analogue Access Network

Telephone

Analogue and Digital Telecommunication Networks The original PSTN was analogue in operation and was voice centric. An electrical circuit was connected from one end to the other and analogue telephones and PBXs were connected directly to the access network. If a digital device such as a computer or fax machine required connection across this network, then the digital signal had to be converted to analogue using a modem (modulator/demodulator) to enable it to be carried over the network. The network itself was unaware it was carrying computer data and treated it just like any other analogue telephone signal. Another modem was required at the far end to recover the digital signal. With the development of the digital switching infrastructure, the customer could connect digital devices directly to the network without a modem. However, if an analogue telephone required connection then an analogue/digital converter was necessary. This is normally placed in the local telephone exchange. This way of connecting telephones to the network is still in common use. A wider variety of services could now be provided over this integrated digital network such as fax, computer data, images, compressed video and voice.

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The PSTN and ISDN

Sound

Velocity Light

Amplitude Pressure Optical Fibres

Time

The Analogue World The world is analogue in nature since all aspects of the real world, such as temperature, humidity, pressure, velocity, light and sound, vary continuously with time. It is possible to sense these variables electrically and represent them as a continuously varying signal.

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2.3

Introduction to Telecoms a) Sinusoidal Wave in the Time Domain

Time

Amplitude

1 millisecond

f =

1 t

b) Sinusoidal Wave in the Frequency Domain

Amplitude Frequency 0

1 kHz

5 kHz

f =

1 t

Signal Concepts In a modern telecommunication network, electrical signals, light and radio waves (sinusiodal waves) are used to transfer information between network nodes or points. Another important fact is the awareness of the relationship between the time-domain and frequencydomain representation of signals. The figure illustrates a 1 kHz sine wave in the time domain. Such is the display expected on an oscilloscope. The diagram illustrates the same sine wave in the frequency domain. Such is the display expected on a spectrum analyzer.

2.4

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The PSTN and ISDN (a)

Amplitude

Time

(b)

Amplitude

Frequency

0 Bandwidth of Speech Signal 100 Hz

5 kHz

Speech Signals The previous example was of a single audio tone of 1 kHz. Human speech is a much more complex signal. It consists of a range of frequencies from approximately 100 Hz to anywhere up to 10 kHz. When represented in the time domain it appears as a constantly varying complex signal. As speech consists of a number of frequencies constantly varying in amplitude and frequency, then the representation in the frequency domain at any instant may resemble that shown in the diagram. The time domain representation shows the instantaneous amalgam of all the frequency components present. The concept is that every waveform that is not sinusoidal in the time domain is a complex wave and consists of a number of sinusoidal frequency components to be seen in the frequency domain. The examples of a single tone and speech have introduced the concept of bandwidth. It follows that to hear (receive) the single tone all that needs to be received is that same single tone. To hear (receive) the speech signal exactly as transmitted all frequency components need to be received. All of these frequencies are constituent parts of the information. The speech signal occupies a wider bandwidth than the single tone.

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2.5

Introduction to Telecoms

Speech Analogue Signal

Speech energy Emotion

15 Hz 100 Hz 300 Hz 3.4 kHz

5 kHz

10 kHz

15 kHz

3.1 kHz Commercial Speech Human Speech Human Hearing

Commercial Speech Channels In basic telephony, speech is converted into an electrical signal and carried over a transmission medium as an electrical representation. The electrical information is analogue in nature, as the voltage level varies with the analogue speech itself. Certain constraints need to be applied to the electrical signal in order to control its transmission. The range between the lowest and highest frequencies used for a particular purpose may be defined as bandwidth. The human ear can detect frequencies in a range between 15 and 15,000 Hz. However, it would be too costly to design a telephone system for this bandwidth and therefore more moderate standards are aimed for. Measurements show that if certain frequency components are removed from the speech signal, it is still possible to retain its intelligibility. The diagram shows that the majority of voice energy is transferred in the low frequencies of 600–700 Hz. Although basic speech continues at a much lower level beyond 10,000 Hz, this adds little to the intelligibility of the signal to the human ear, as shown by the solid line. The dashed line shows the portion of the frequency band that carries emotion. From this it can be seen that for economical transfer of intelligible speech, it is possible to use a much narrower band than 15–15,000 Hz. Speech is therefore limited to the range of 300–3,400 Hz, which gives a bandwidth of 3.1 kHz, as specified in the ITU-T Recommendations G.132 and G.151 and often referred to as the nominal 4 kHz voice channel. It is considered a good compromise between quality and cost.

2.6

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The PSTN and ISDN Single two-wire copper pair connected together for continuous circuit

Local Telephone Exchange

Analogue representation of human speech or modem tones

Cross Connection Point

Average UK distance 3–4 km Distribution Point

Known as:

:Local Loop Final Mile Access Loop

Two-Wire Transmission in the Local Loop A telephone conversation requires transmission in both directions. This can be achieved on a two-wire cable and is known as two-wire transmission. However, as with any cable, attenuation occurs, giving the cable a finite effective length before the signal being carried requires amplification in the form of a repeater. For a two-wire cable used to connect a typical home telephone to the local exchange, the average UK distance is 3–4 km and is known as the local loop, final mile or access loop. The diagram shows a simplified connection to the local exchange.

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2.7

Introduction to Telecoms a)

UK

USA

The PSTN

Australia

France

Exchanges

b)

Local

TXMN Section

National International National

TXMN Section

TXMN TXMN Section Section

Local

TXMN Section

TXMN Section

Routing info

Incoming call indication

Switched connection Routing info

Routing info

Routing info

Routing info

End-to-end switched circuit

Switching in Telephony Networks Telephony networks provide global telephony services to customers connected to the network. The network can be viewed as a single switch which provides circuit-switched connections on demand between any two end users located in any country. In reality, however, the network is made up of many systems constituting thousands of interconnected switches throughout the world. The establishment of a complete end-to-end switched circuit between a calling and a called party may involve many switches on many levels, i.e. local exchanges, national exchanges and international exchanges. Each switch assigns and provides a connection between the different transmission channels connected to the switch. Thus, for the duration of any call, an end-to-end circuit-switched connection consists of a number of Transmission (TXMN) sections joined together by one or more switches. This connection is cleared down at the end of a call and the resources become available for other users.

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The PSTN and ISDN

Typical Functions ISC: Switching Trunk Termination Signalling Operations and Maintenance Charging TE: Switching Trunk Termination Signalling Operations and Maintenance

International Switching Centre (ISC)

Regulatory Interconnect

Transit Exchange (TE) CP

Local Exchange (LE)

LE: Switching Trunk Termination Signalling Operations and Maintenance Subscriber Termination End User Subscriber Services Charging Circuit-related signalling Traffic signalling and traffic

Public Switches Hierarchy and Functions Public switches (or telephone exchanges) can be categorized as one of three general types of exchange: Local Exchange (LE), Transit Exchange (TE) (trunk switching centre) and International Exchange (IE), also known as an International Switching Centre (ISC). All of these switch types need to be able to carry out a range of functions. Specific functions will vary from switch to switch depending on their place in the switching hierarchy. Local Exchange (LE) An LE must perform all the necessary actions to allow subscribers to access the public network. This includes signalling, routing, analogue-to-digital conversion, services such as call back, and auxiliary functions such as power supplies, charging, and operations and maintenance. Routing may be within the LE, or externally via a transit switch. Transit Exchange (TE) A TE transfers traffic between exchanges which are subordinate in the hierarchy, or between a subordinate exchange and one further up in the network. Some exchanges may be designated for both local and trunk traffic, combining the functionality of both the LE and TE. International Exchange (IE) An IE provides the interface between national networks and the international network. Charging and accounting are major features of an IE. It will also have an international operator service.

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2.9

Introduction to Telecoms PBX Functions: to connect telephones within same organization to provide connection to rented lines towards the network concentration (lower number of lines for larger number of extensions) additional services, such as – Call Logging – Call Diversion – Call Waiting – Abbreviated Dialling – Call Barring – Multi-Party

To Secondary Trunk Switching Centre

Primary Trunk Switching Centre Other LEs

Effectively an extra ‘branch’ in the switching hierarchy

Local Exchange

PBX

Other subscribers or PBXs Note: PBXs may be interconnected to form a network, with single or multiple access points to the PSTN.

Private Switches In large organizations there is a need for internal communications between staff in different locations. A public network could be used to provide this service, with telephones interconnected via transmission lines to the LE, and public switches providing the interconnection. However, this would be wasteful, especially when the communicating parties are located within the same building or even the same room. Therefore, private switches are normally used to connect telephones within the same organization. Not only does this eliminate the cost of internal calls, but the number of rented lines from the private switch to the public network is normally far fewer than the number of internal extensions. Usually, a private switch takes the form of a Private Branch Exchange (PBX), although it is also possible to use a manual version, the Private Manual Branch Exchange (PMBX). Many private switches offer supplementary services like those available in public networks such as call logging, call diversion, call waiting, abbreviated dialling, call barring and multi-party calls. If a suitable signalling system is used, these services may also be extended across a network of PBXs (perhaps supporting a larger business with more than one site in a local area).

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The PSTN and ISDN a)

Country Code

National Destination Code

Subscriber Number

National Significant Number International ISDN number Notes: 1 The national and international prefixes are not shown – they are not considered to be part of the international ISDN number. 2 Maximum length – 15 digits.

b) Example of E.164 Number Structure to Identify a Subscriber (Prefix International Call)

(Prefix) National Call

0044 (0)1539 Identifies a country

742742

Kendal Transit Switch

Wray Castle ‘Customer’

Numbering Plans Within a telephony network, each customer is given a personal number. This is an identifying reference for that customer on their local switch. It also provides routing information so that incoming calls can locate the correct local switch within a network. Each network is allocated its own group of numbers which it can distribute to its customers on a switchby-switch basis. The function of a line does not affect the number a customer is given, i.e. it does not matter if it is used for speech or data calls. The ITU has issued Recommendations on the use of numbering plans. It allocates codes to identify subscribers uniquely in different countries. It also recommends that numbering plans for new networks/services adhere to the ITU-T Recommendation E.164. The national significant number allocation is a matter for national authorities; the country code is only used when dialling between countries. The maximum length of the dialled number is 15 digits.

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Introduction to Telecoms ‘A’ Dials Number: 00 44 1234 567890 Note: The ‘00’ International prefix is not part of the international ISDN number and may vary from country to country. Route towards subscriber based on ‘7890’ (Number of subscriber on the LE)

Route towards ‘E’ based on 56 (Network’s own allocation for LE ‘E’)

Party ‘B’ Local Exchange

E Network Gateway

Route towards ‘D’ based on ‘1234’ (National destination code)

D

International Gateway

International Gateway

C

Route towards ‘B’ based on international prefix ‘00’

A B Route towards ‘C’ based on ‘44’ (UK code)

Usually via transit network

Local Exchange

Party ‘A’

Call Routing Routing analysis based on the E.164 numbering plan simply involves selecting routes towards the correct country as specified by the country code, and then towards the correct network within that country based on the national destination code. It then uses the subscriber number to identify the correct line at the local exchange.

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The PSTN and ISDN

Continuous time signal

(Analogue)

Discrete in both time and amplitude

(Digital)

Analogue and Digital Signals Most of the signals considered so far have been analogue signals. However, most telecommunications systems are now based on digital technology. An analogue signal has an infinite range of amplitude values in respect of time, which is continuous, whereas a digital signal has a discrete limited range of amplitude values in respect of time, which is discontinuous. The term ‘data’ is often used when people talk about digital technology; but what is data? The literal meaning of the word is simply ‘information’. However, in this context it means information which is conveyed in a digital form. It should be noted that information itself may be analogue. Modern telephony, music and television are examples of analogue information (speech, music and moving pictures) that is converted to a digital form.

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2.13

Introduction to Telecoms

Analogue Signal

Digital Signal Analogue to Digital Converter

0 11 0 111 00 1 0

Analogue Signal Digital to Analogue Converter

Digital Systems All modern telecommunication networks are digital. The benefits of digital networks include quality improvements and the integration of different services onto one network. Information, such as speech, does not originate as a digital signal – it is analogue. The process of digitization – conversion from analogue to digital – will be performed as close to the point of origin as possible. In a fixed telephone network, this would be at the local exchange. For a digital mobile network, it takes place in the handset.

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The PSTN and ISDN

Pure Signal

Signal and Noise

Signal and Noise Amplifier

Amplifier

Transmission Media

Transmission Media

Attenuation

Attenuation

Analogue Signal Quality In any electronic environment, whether it is a piece of equipment or a link between equipment (cable or radio), there will be random signals present referred to collectively as ‘noise’. This noise may originate from several sources but is always present to a greater or lesser degree. Once noise becomes mixed with an analogue signal, the two components (noise and signal) can never be fully separated. This means that any attempt to amplify an attenuated (diminished) analogue signal will amplify not only the wanted signal, but noise as well. Hence over any transmission path the quality of an analogue signal will continuously degrade.

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2.15

Introduction to Telecoms

Bit Stream

10110

Regenerator

Regenerator

10110

10110 Original Bit Stream

Transmission Media

Transmission Media

Attenuation

Attenuation

Digital Signal Quality Digital signals also become corrupted by noise. However, it is a relatively simple process to remove the noise by regenerating the digital waveform. Although a clean digital bit stream is transmitted, it becomes corrupted by noise along the transmission link to its destination. The receiver examines each bit as it arrives and interprets its value in relation to a threshold value. For example, if the value is above the threshold, the value will be interpreted as a logical ‘1’, and if it is below the threshold, it will be interpreted as a logical ‘0’. Assuming the correct decision is made, the digital bit stream may then be regenerated to its original noise-free form. If the signal is pushed too far along the transmission medium without regeneration, the received waveform will be so corrupt that the system will be unable to recognize it. During transmission, therefore, the signal needs to pass through a regenerator before it becomes too corrupt.

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The PSTN and ISDN a) Bit Rate 1

1

0

0

1

1

0

0

1

1 ms

Time Bit Rate =

1 Bit Duration

=

1 1 ms

= 1000 bps = 1000 baud

b) Baud Rate (Symbol Rate) 3V 2V 1V

1 1 0 0 1 0 0 1 1 0 1 1 0 0 0 1 0V

1 ms

Time 1 state Baud Rate = = 1000 baud 1 ms In this example 1 Baud represents 2 bits ∴ 1000 Baud = 2000 bps

State Data Bits

0 1 2 3

00 01 10 11

Bit Rate and Baud Rate The speed at which digital information is transferred can be quantified in terms of bit rate and baud rate. The bit rate in a digital system is a measure of the number of data bits transmitted each second. Bit rate may be calculated from: bit rate = 1/bit duration The baud rate is a measure of the number of times per second a signal makes a transition between states, the states being frequency, voltage/current levels or phase angles. By using more than two states, it is possible to represent several bits per state. This is illustrated in the figure where a four-state system conveys two bits of data per state. Therefore, the baud rate is half the bit rate. It should be noted that the bit rate will equal the baud rate if only one change in state is used to represent a single bit. In this course the bit rate is primarily considered, but the importance of baud rate should not be underestimated. The baud rate is often referred to as the ‘symbol rate’.

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2.17

Introduction to Telecoms Sampling

Quantization +127

Coding 1 1 1 1 1 1 1 1

+127

etc.

Filtered Voice Signal (300–3.4 kHz)

1 0 0 0 0 0 0 1

0

+1

1 0 0 0 0 0 0 0

+0

0 0 0 0 0 0 0 0

–0

0 0 0 0 0 0 0 1

–1

etc. 0 1 1 1 1 1 1 1

–127

Time

A-Law

–127

1011001

8 bits/125 µs = 64 kbit/s

74 Clock 8,000 samples/second

73 72 Sample

Rounded off to nearest quantum level

Analogue to Digital Conversion Most modern telecommunication systems rely on a technique referred to as Pulse Code Modulation (PCM) to convert a band-limited analogue voice signal to a digital format suitable for transmission/processing. With reference to the diagram, PCM can be regarded as three processes: sampling, quantization and coding. Sampling is the measurement of the analogue signal’s amplitude at discrete instants in time. In order to recover the original signal the sampling frequency, according to the mathematician Nyquist, must be at least twice the maximum signal frequency, 8 kHz (8,000 samples per second) being the accepted standard. The amplitude range to be transmitted by the system may be divided into a number of levels referred to as quantum levels. Samples falling into a given quantum level are rounded off to the nearest quantum level. If 8-bit PCM is used then there are 256 (28) different codes that may be sent using one of two coding laws, A-Law or µ-Law. In A-Law the codes sent range from +0 to +127 and from –0 to –127. The final process in PCM is coding. Coding of the signal is performed by converting each quantization level to a binary codeword. Each sample taken could be represented by a single 8-bit (or byte) binary sequence, e.g. 01001101. This gives us a data rate of 8000 samples/second x 8 bit/sample = 64 kbit/s. This rate is often referred to as a basic rate channel and is used throughout the ISDN and PSTN. Here we have outlined the general principle of PCM encoding techniques. In practice there are two types of encoder in use, known as A-law and µ-Law. These are incompatible so where interworking is necessary, A-Law to µ-Law converters are required. A-Law is used in the European-based systems whereas µ-Law is used in the North American-based systems.

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The PSTN and ISDN

Typically located at an ISC Network

Network

µ-Law/A-Law Interworking

USA

UK

ISC – International Switching Centre

Interworking between A-Law and µ-Law The distribution of A-Law and µ-Law is limited to global regions and countries. Within regions, countries and networks only one system will be found. Thus there is no requirement for terminal equipment to support interworking functions capable of transmitting with more than one system. Interworking is only required between networks located in regions with different PCM systems in place. For example, the diagram shows a call between terminals in the UK and the US. In this case the US network and terminal will be using µ-Law, whilst the UK network and terminal will be using A-Law. In order for the terminals to communicate, the connecting circuit must pass through an interworking function at some point on its route. Since both PCM systems will never be found within a network, this interworking function must be between rather than in the networks. In this example the physical location of the interworking function is most likely to be at the entry point to the US network.

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2.19

Introduction to Telecoms

Waveform Coders

Source Coders

Type

Bit Rate

Type

Bit Rate

Pulse Code Modulation (PCM)

64 kbit/s

GSM Full Rate

13 kbit/s

Adaptive Differential Pulse Code Modulation (ADPCM)

32 kbit/s

GSM Enhanced Full Rate

12.2 kbit/s

GSM Half Full Rate

5.6 kbit/s

Adaptive Multi Rate (AMR)

4.8 to 12.2 kbit/s

G.729

8 kbits

PCM encoded voice

Transcoder AMR voice

Other Types of Voice Coders In general there are two types of voice codec (coder/decoder): waveform coders and source coders. Waveform coders such as PCM are source agnostic, i.e. they do not care what the type source of the information is. The PCM codec is a waveform codec that can code any waveform type where information content is within the codec’s bandwidth. This enables the PCM codec to digitize voice or data sources using voice band data signals generated such as group 3 fax machines and voice band modems. Voice source coders use algorithms specifically designed for coding voice. They are suitable for use with music sources or voice band data sources. PCM was the codec originally specified for supporting voice over digital telephony networks. However, over time a number of other codec types have been developed (examples are given above). Note that these other codec types represent voice at lower bit rates than PCM. While this is an advantage from a capacity point of view, there are also disadvantages. Generally speaking voice quality drops with bit rate, and some codecs introduce significant processing delays, which can affect conversational quality. Nevertheless source coders and ADPCM codecs are in widespread use. The full range of GSM coders, including AMR, are embedded in most newer mobile handsets, while G.729 is a popular choice in Voice over IP (VoIP) applications. ADPCM has been used for many years over satellite circuits to double the number of voice channels available in a given capacity compared to using standard PCM coding. It is sometimes necessary to connect a voice call between devices that do not support the same codec types. In this case a transcoder function is used to convert between coding formats.

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The PSTN and ISDN

Aggregate digital signal 2 Mbit/s (E1) 30 digital voice signals each at 64 kbit/s

63 digital signals each at 2 Mbit/s

Tributaries

Aggregate digital Signal of 155 Mbit/s (could contain 1890 voice signals) Tributaries

The Need for Multiplexing The digital transmission systems in the PSTN can be divided into two main technologies – the Plesiochronous Digital Hierarchy (PDH) and the Synchronous Digital Hierarchy (SDH). These two systems carry out a function known as ‘multiplexing’. The main function of multiplexing is to enable many different type of signals (voice, data, video, etc.) to be combined so that they can be simultaneously passed down a single wire, fibre or radio link. By using a multiplexer, the number of transmission links such as cables or radio systems can be dramatically reduced. As can be seen in the figure, the more signals multiplexed, the faster the aggregate digital signal in bits/second. Multiplexing provides efficiency in the way the signals are transferred from one location to another and the aim is to share the expensive resources in the network such as cables and radio links. A device that performs multiplexing is called a multiplexer (MUX), and a device that performs the reverse process is called a demultiplexer (DEMUX). The inputs to the multiplexer are known as ‘tributaries’.

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Introduction to Telecoms Extra information added by the multiplexer to identify the start of the sequence and to carry management information such as alarms.

+

Output from the multiplexer contains the bits (or bytes) interleaved in time from the four input tributaries.

Tributaries

Time Division Multiplexing (TDM) The process used for multiplexing in a digital network is known as Time Division Muliplexing (TDM). The individual bits (or sometimes bytes) from each tributary are interleaved in time on the output aggregate signal. Special synchronization signals and indicators are added to the aggregate signal in the multiplexer, which allows the demultiplexer at the far end to identify the individual tributaries within the higher speed signal and monitor the signal quality. This accounts for the fact that the output bit rate of the multiplexer is more than the sum of the tributaries. Note, however, that this is not always the case with synchronous networks.

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The PSTN and ISDN Input Channels (tributaries)

Primary Multiplexer (PDM)

Aggregate (E1)

Ch. 1 2.048 Mbit/s

PSTN Network

Ch. 30

Usually flexible allocation of inputs to timeslots Frame structure TS TS 0 1

TS TS TS 15 16 17

Channels 1 – 15 (15 x 64 kbit/s) Frame Sync. and Alarms

TS 31

Channels 16 – 30 (15 x 64 kbit/s) Signalling

The primary multiplexer includes a codec to code voice bandwidth into a 64 kbit/s PCM format which is carried within one of the 30 traffic channels of the 2.048 Mbit/s link.

Primary Level Multiplexers The ETSI primary multiplexer combines 30 input channels (or tributaries) onto a common 2.048 Mbit/s aggregate signal. Every timeslot on the aggregate may be considered as a 64 kbit/s basic rate channel. The 32 timeslots are used as follows:    

TS0: frame synchronization and alarms TS1–TS15: 64 kbit/s traffic channels (voice or data) TS16: signalling TS17–TS31: 64 kbit/s traffic channels (voice or data)

The input tributaries may be analogue or digital channels and are adapted via a suitable process to be transported in a nominated timeslot. Thus, the 2.048 Mbit/s bearer may be used to carry a mix of voice (analogue) and data services. Therefore, the Primary Digital Multiplexer (PDM) is a flexible means of providing access to the PSTN. A 2.048 Mbit/s bearer is often referred to as a 2 Meg, E1, CEPT 1, Megastream or Open Port.

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Introduction to Telecoms

Transmission link

Transmission link

Digital Cross Connect

E1 signal out

E1 signal in TS 3 RX

8 bits from TS3 to TS25

TS 25 TX

E1 signal in

E1 signal out TS 3 TX

8 bits from TS25 to TS3

TS 25 RX

Timeslot Interchange Each timeslot in an E1 (2.048 Mbit/s) frame contains eight bits of information. The individual data or voice channel is represented by these eight bits. Most multiplexers operate in four-wire mode with a transmit direction (TX) and a receive direction (RX). The timeslot that is allocated to a particular channel is the same on both the TX and RX over the same link. In order to allow flexibility of traffic in a network, a digital cross connect is used to provide timeslot interchange from one link to another. As the channel passes through the network it can be seen that the timeslot that is used on each link may be different. The connections across the digital cross connect device are normally set up manually. Therefore, this device can be regarded as a manually controlled switch (i.e. no signalling) and is used for setting up a leased line. The cross connect will remain in place until it receives another command to release the connection. This could be several years. Some timeslot interchange devices can be controlled by a timer, so a connection could be set up automatically for part of the day or night – for example between 10pm and 7am. This will depend on the contract with the customer.

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The PSTN and ISDN

PSTN/ISDN Local Exchange

PSTN/ISDN Switch Transit Exchange

Multiple E1 Bearers

PCM

Multiple E1 Bearers

Modem/Fax

PCM

Customer Access Lines

PCM CODECs convert analogue voice or voice band data to 64 kbit/s

Switch analyzes routing information and dynamically switches customer ‘line’ to any available timeslot

64 kbit/s cross connect any timeslot to any timeslot

Switch analyzes routing Information and dynamically assigns cross connects

PSTN and ISDN Digital Switches PSTN and ISDN switches are essentially 64 kbit/s cross-connect devices that dynamically allocate connections based on the analysis of routing information, i.e. dialled digits. In general, two types of exchange are used in the PSTN/ISDN, Transit Exchanges (TE) and Local Exchanges (LE). TEs terminate a high density of E1s but typically do not provide access to subscriber lines. A TE is able to establish a 64 kbit/s cross-connect between any timeslot on any E1 bearer and any other available timeslot on any other or the same E1 bearer. An LE provides access to customer lines through an appropriate subscriber interface card. In the case of analogue access lines, PCM codecs convert voice signals or voice band signals from modems or fax devices to 64 kbit/s before switching them to allocated timeslots on a route to another switch, or, if the call is local, to the LE to another customer line. Connections established through the PSTN/ISDN are established on demand and only exist for the duration of the call.

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2.25

Introduction to Telecoms a) Separate Networks Prior to ISDN Packet Switch Exch

Public Networks Telex

Packet Switch Exch

Packet Data Local Exch

PSTN

Local Exch Fax

Fax

b) Integrating Services via ISDN Public Network IDN ISDN Exch

Fax Others

Digital Transmission Digital Switching CCS – Typically SS7

lower cost greater flexibility dimensioning is more efficient single customer connection improved services

ISDN Exch

Fax Others

ISDN

Evolution from PSTN to ISDN The PSTN is a fixed telecommunication network (either national or privately owned) that enables the transmission of voice and data traffic between users. The initial requirement of the PSTN was to carry voice traffic. However, there is an increasing requirement from customers to use fixed lines to connect to data networks such as the Internet and intranets. This increasing trend has stimulated the evolution of networks from analogue to digital, facilitating higher data speeds. Basic telephony (speech) has traditionally been the most widespread service offered by telephone companies and, until relatively recently, most telecommunication traffic comprised this service. Initially, therefore, operators designed their networks to provide acceptable speech quality at minimum cost. This usually precluded using the basic network for anything else, so when services such as text (telex) were offered, they were provided primarily via dedicated networks. Even when implemented through the PSTN, each service required a separate and dedicated customer access point. Several types of network have evolved in order to extend a variety of services to users. Different services traditionally have different characteristics, such as data rates, protocols and interfaces. Before the advent of the ISDN, in order to deal with this each network had to be provided with its own transmission paths and hardware. The disadvantages arising from this approach included high cost and lack of hardware flexibility. In addition, each network had to be dimensioned for its own traffic requirements, often leading to a lot of hardware being required to guarantee a minimum grade of service for a low average usage. The idea of having separate networks and multiple customer connection points for each service became less acceptable as customers began to use more applications alongside basic telephony. A number of bearer networks coexisting within a single transmission cable, with each service requiring different transmission rates and interfaces, was recognized as extremely wasteful and inefficient. The ISDN was developed to provide one network that can handle different telecommunication services simultaneously.

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The PSTN and ISDN

ISDN Digital Subscriber Line – Symmetrical Up to eight devices e.g. telepnone, fax, computer etc. 1

TE

8

TE

D – 16 kbit/s B – 64 kbit/s

2B+D (BRA)

B – 64 kbit/s

Voice PCM coded at 64 Kbits by the TE

BRA Interface

Local Digital Switch

NxE1 to other switches

2 x simultaneous digital voice calls or 2 x 64 kbit/s data channels or 1 x voice and 1 x data channel or 1 x 128 kbit/s data channel

The ISDN Digital Subscriber Line – Basic Rate Access With the ISDN, the copper line to the customer is used to carry information in digital format. An ISDN digital subscriber line is often a traditional copper pair with special equipment at each end to enable digital signals to be carried over it. In this case there is no requirement to install a new cable to the customer. In some circumstances, higher data rates are required and in this case it is sometimes necessary to install a separate cable especially for business customers. There are two types of ISDN digital subscriber lines. Basic Rate Access ISDN (BRA) is designed for residential customers and small businesses, generally referred to as Small Office Home Office (SOHO) users. Primary Rate Access ISDN (PRA) is designed for larger businesses which require a higher speed connection. Basic Rate Access ISDN provides two full duplex 64 kbit/s channels to the customer over a standard copper line. These can be used to transmit voice, images, video or data calls and are referred to as the ‘B’ channels (Bearer channels). A 16 kbit/s signalling channel, called the ‘D’ channel (Data channel) is also sent which controls the set-up and operation of the two faster ‘B’ channels, enabling two simultaneous independent calls. With additional equipment, the two ‘B’ channels can be combined for higher transmission speeds of 128 kbit/s. BRA is also known as 2B+D. At the customer’s premises, the digital subscriber line terminates in a device known as a Network Termination (NT). It is possible to connect up to eight devices (known as ‘Terminal Equipment’ (TE)) which could include faxes, printers, telephones or computers to a single BRA connection. One telephone number comes as standard with BRA, and a total of eight numbers may be assigned, one for each TE.

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Introduction to Telecoms

File Transfer

LAN Interconnection

Interactive Editing

Leased Line Backup

Dial Back for Remote IT Users

Security Monitoring

BRA Applications BRA Applications Basic Rate Access is suited to regional offices, small businesses and individual users. Two independent calls can be made at any one time and calls can be received from the public telephone network as well as from ISDN subscribers. Examples of the applications of BRA are: LAN interconnect (WAN) Connects local area networks on a dial-up basis. A single remote user can access services such as printers and files on the main office’s LAN. They can then function just as if they were in the main office. There is scope for the connection of teleworkers and remote offices. File transfer BRA can enable sending and receiving of computer files to and from remote sites on a dial-up basis. Files containing colour graphics and architects’ plans are examples of file transfer. Interactive editing One or more users can dial up a remote personal computer to view a screen image and make changes. It is useful in educational interactive multimedia. Visual communication can be further enhanced by desktop video conference units. Security monitoring Cameras and alarms can be remotely controlled via ISDN connections. The operator can view and control the image on a monitor. Leased line dial back-up and overflow Dedicated data circuits can be backed up with dial-up ISDN circuits. If a 64 kbit/s leased line fails, special equipment can divert vital traffic through the ISDN. Remote Dial Back for IT Users Remote IT users dial into their company’s data network access device. The access device authenticates the user, drops the original call and then dials back a pre-stored number for that user. This provides a level of security and mechanism for controlling call costs as the company may get wholesale prices for calls made from a central location.

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The PSTN and ISDN ISDN Digital Subscriber Line

PRA (30B+D)

PBX

2 Mbit/s (E1)

PRA Interface

Local Digital Switch

NxE1 to other switches

1 Up to 15 x 2B+D BRA Lines to/from SoHo users

ISDN Mux

PRA (30B+D) 2 Mbit/s (E1)

PRA Interface

15

The ISDN Subscriber Line – Primary Rate Access (PRA) Primary Rate Access ISDN (PRA) operates at the E1 (2 Mbit/s) rate for European and international systems. PRA service is generally provided for corporate users who require eight or more B channels. In European-based networks, E1 PRA lines provide thirty ‘B’ channels for use. Each ‘B’ channel operates at 64 kbit/s and a single ‘D’ channel also operates at 64 kbit/s and is used for call signalling. The ‘D’ channel is usually transported in timeslot sixteen of the E1 signal. The PRA service is often referred to as 30B+D. The D channel carries control signalling for establishing services across the PRA or BRA interfaces. The signalling system in the D channel is called Digital Subscriber Signalling System No. 1 (DSS1). Typically PRA services are delivered using fibre optic cable or microwave links. Primary Rate Access ISDN is more suitable for larger sites which require concentration of large volumes of voice calls (e.g. a call centre) or data traffic from remote users, or a combination of the two. PRA is designed for large businesses’ networking requirements. It can facilitate the large scale connection of voice and data circuits to a Private Branch Exchange (PBX), router, bridge or other ISDNcompatible equipment. Each 64 kbit/s channel can simultaneously and independently carry voice, data and video signals between sites. Where a number of Small Office and Home Office (SOHO) users in a particular localized area require a BRA service it is possible to deliver a 30B+D service to a ISDN multiplexer (IMUX) within that area. The IMUX may be used to multiplex up to 15 x 2B+D services for connection to individual customers. Note that the D channel signalling on the 2B+D interfaces are all multiplexed onto timeslot 16 of the PRA interface, due to the redundancy in the signalling channels, i.e. they are not in continuous use. There is plenty of capacity in the 64 kbit/s D channel to cope with the signalling load on the 15 16-kbit/s BRA D channels.

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2.29

Introduction to Telecoms Phone

Off-Hook

Switch

Ready? Dial Tone

Dial Number

Pulses or Tones Call Subscriber (Ringing)

Ringing Signal

Connected, or Engaged or Busy Tone

not Connected

Conversation

Signalling Signalling is used within a network to set up, maintain and terminate a call. Specific signals (pulses or audio tones) are used to indicate to the switches the required connection or service, although some of these signals are for interpretation by the subscriber. An example follows and is illustrated in the diagram. To initiate a call, a telephone subscriber lifts the handset off its rest (off-hook). This tells the local exchange to be ready to receive the number of the called subscriber. As soon as the exchange signals the dial tone back to the calling subscriber, they dial the wanted number. On older exchanges, this information is passed via a rotary dial by a series of makes-and-breaks of the subscriber’s connection, interrupting current flow, i.e. pulses. On more modern exchanges, voice-frequency (audio) tones are usually sent to the exchange as push buttons are pressed. These tones are usually called DTMF or Dual Tone Multiple Frequency. In due course the subscriber receives advice from the exchange about the status of the call, either by a ringing signal, an engaged or busy tone signal, an equipment-busy signal, or some other specialized tone.

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The PSTN and ISDN

Channel Associated Signalling (CAS) e.g. loop disconnect

e.g. loop disconnect

PBX

e.g. R2MFC

Exchange

A

A1

B

B1

C

C1 Each traffic channel has its own signalling channel

Traffic Signalling

Channel Associated Signalling (CAS) Signalling systems may be categorized as being either Channel Associated Signalling (CAS) or Common Channel Signalling (CCS). With CAS, every traffic channel has a dedicated signalling channel. Because of this, CAS tends to be inflexible and relatively slow. A CAS system is limited by the number of signalling states, depending on the number of Direct Current (DC) states, Alternating Current (AC) tones, or bits allocated. This slows the system down and therefore limits the variety of information that can be transferred. CAS can be analogue or digital. With the analogue system it is possible to choose between DC and AC. DC systems rely on the voltage and current around the signalling circuit, whereas AC systems rely on tones. In general, CAS systems are inefficient in that they require live signalling to take place, either continuously or, in the case of digital CAS, at regular intervals. This means that signalling occurs even when there is no new information. Types of CAS are:  

DC – Loop Disconnect; E&M AC – Switching System AC15 (SSAC15); R2 Multi-Frequency Compelled (R2MFC)

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2.31

Introduction to Telecoms

Exchange with SPC

E1

Traffic Channels

Exchange with SPC

A

A1

B

B1

C

C1 ST

ST

Processor

Processor

Common signalling channel for all traffic channels typically implemented as a 64 kbit/s circuit on an E1 bearer

SPC – Stored Program Control ST – Signalling Terminal

Traffic Signalling

Common Channel Signalling (CCS) CCS is an improvement on the CAS system due to its greater flexibility in signalling and its ability to evolve. CCS systems are based on a secure data circuit that links together the call control processors in connected switches. The common channel carries data messages that convey signalling for the circuits between these switches, and may additionally carry network supervisory and management data. CCS only requires one signalling channel for many traffic channels, since it only signals when required (unlike CAS, which signals even if nothing is happening). The ratio of 1000:1 (plus 1 backup) is common, but within GSM networks the ratio is more like 200:1 (plus 1 backup). CCS is faster and more flexible and it allows more services. The CCS channel is usually implemented as a 64 kbit/s circuit on an E1 or T1 bearer. In CCS systems, the signals are generated only to convey new information and are, therefore, more efficient than CAS systems. Examples of CCS include SS7 (National and International), Digital Access Signalling System No. 2 (DASS2) (UK specific) and Digital Subscriber Signalling System No. 1 (DSS1) (ITU-T).

2.32

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TY2600/v4.1

The PSTN and ISDN

DTMF

SS7

DTMF

SS7

PBX Local Exchange

Access Signalling System

Network Signalling System

Local Exchange

Access Signalling System

Network Signalling SS7 Access Signalling (CCS) *

Access Signalling (CAS) * e.g. Loop disconnect DTMF

DSS1 Q.931

PBX Private network signalling e.g. DPNSS, Q.sig * Supervisory tones are provided to keep the customer informed about the progress of the call

Access/Network Signalling This figure illustrates how signalling within a network can be subdivided into systems providing customer access and systems for use within the networks themselves. Access signalling is a signalling system that is extended out to the customer and works in conjunction with a variety of recognizable tones to keep the customer informed about the progress of the call. In general, it does not need to be as sophisticated as the network signalling system, but the greater the level of sophistication, the more flexible and complex the services offered can be. The network signalling system is used solely within the network. It is not extended to the customer because it is too sophisticated for this purpose, and the network operators are understandably reluctant to extend the means to access network functions out to the customer. In addition to access and network signalling, it is possible to provide the means for end users to communicate with each other using a specified signalling system. Typically this signalling is used between the PBXs and is referred to as private network signalling.

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2.33

Introduction to Telecoms

Originating Exchange

Traffic Circuits (PCM and TS)

Intermediate Exchange

N x E1/T1 Bearers

Circuit 10

Destination Exchange

Circuit 20

Access Signalling

Set-up

ISUP

IAM CIC=10 B No.

IAM CIC=20 B No.

ISUP

SS7

ISUP

SS7

MTP

MTP

ISUP supports: Bearer Services – 64 kbit/s: 3.1 kHz Audio Teleservices – Telephony, Fax Supplementary Services – CLI, CF, CB

MTP

ISUP – ISDN User Part PCM – Primary Channel Multiplex TS – Timeslot IAM – Initial Address Message CIC – Circuit Identity Code

Basic Call Process and the ISDN User Part (ISUP) ISDN networks support bearer circuits suitable for transporting both voice and non-voice traffic. Basic call set-up involves call control signalling, where adjacent exchanges in a traffic route exchange information to set up, supervise or clear a traffic circuit. Call control signalling is provided by a call control protocol such as the ISDN User Part (ISUP). SS7 uses signalling messages between exchanges in the network. As SS7 is a CCS system, the messages must identify exactly the PCM bearer and timeslot of the traffic circuit with which the message is associated. The Message Transfer Part (MTP) is responsible for the reliable, unduplicated and insequence transport of signalling messages between nodes in the network. The example in the figure illustrates a call between two fixed-line phones. After picking up the phone and receiving the dial tone the customer enters the telephone number of the called phone. The originating exchange and intermediate exchange send Initial Address Messages (lAMs) during the first stages of a call to establish the traffic route through the switched network. The originating exchange allocates circuit 10 to the call and the intermediate node allocates circuit 20. When the call arrives the destination exchange triggers the appropriate access signalling to the called party. The lAM must also carry enough information to facilitate onward routing, including the called party number and call type. Additionally the lAM may include data such as the calling address digits. In addition to the establishment of 64 kbit/s bearers within a switched network, ISUP supports a range of teleservices including telephony and fax and a wide range of supplementary services defined for the ISDN, such as Calling Line Id (CLI), Call Forwarding (CF) and Call Barring (CB). ISUP can be used in mixed digital/analogue network environments.

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TY2600/v4.1

The PSTN and ISDN DSS1

ISDN User A

ISUP

Local Switch A

Set-up ('B' No: BCI)

ISUP

Transit Switch P

IAM (B.No)

DSS1

Local Switch B

ISDN User B

IAM (B.No)

Call Proceeding Set-up ACM

ACM

(BCI) Accepted Alerting

Alerting

CPG (Alerting)

CPG (Alerting) Answered Connect

Connect

Answer (Charge)

Answer

End-to-End Call Setup – Basic Procedure The diagram illustrates a call being set up between two switches that terminate the calling and called parties. The two parties are assumed to be ISDN-connected, but could be connected to the network by analogue technology. Overall, the call set-up procedure for both speech and non-speech connections is the same. En-bloc addressing (all address digits contained in a single message) and overlap addressing (address digits are transmitted over a number of messages) are supported. In the call set-up procedure, routing analysis may take place at the originating exchange or may be requested from a remote database. The route chosen will depend on the connection type, and signalling method required, and the called party number. If the subscriber has digital access, the Set-up message may contain Bearer Capability Information (BCI). The local exchange uses the BCI to determine the correct connection type and signalling requirements. BCI can be transferred in the ‘User Service Information’ parameter of the Initial Address Message (IAM) for data calls.Connection types are: Connection types are:    

speech 3.1 kHz audio alternative speech/64 kbit/s unrestricted alternative 64 kbit/s unrestricted/speech

Signalling Capabilities are:   

ISDN-UP Preferred ISDN-UP Required ISDN-UP Not Required (any signalling system)

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2.35

Introduction to Telecoms DSS1

ISDN User A

ISUP

Local Switch A

ISUP

Transit Switch P

DSS1

Local Switch B

ISDN User B

Call Phase

Disconnect*

Release

Release

Release

Release

Complete

Complete

Disconnect*

Release

Release

Release

Release

Complete

Complete

Call Clear Down Diagram notes: * Either party clears Disconnect – traffic channel is cleared but not yet available for reuse Release – traffic channel is cleared and available for reuse Release Complete – on receiving release complete the call reference is released

2.36

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TY2600/v4.1

The PSTN and ISDN

Supplementary Services

Value-Added Services

Additional Features Terminal Based

Terminal Based

e.g. last number redial

Software Function

Node Based

Telecoms Network

e.g. calling line ID

IN Based e.g. prepay or credit card calling

Services for Revenue Generation In order to attract more custom, fixed-network and mobile-network operators are adding new services to basic phone provision. These may take the form of Supplementary Services (SSs), which are services that have been defined by a standards body, such as call hold and call forwarding; or Value Added Services (VASs), operator– or network-specific features that are open to implementation, such as voicemail. These service types are implemented by means of a software function in the network. Traditionally, SSs were implemented into network nodes. However, with the requirement for more sophisticated service types other mechanisms are required. The modern approach is to implement them at the optimum point using any one or a combination of three methods: node based, terminal based or IN based. Examples of each approach are shown in the diagram.

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2.37

Introduction to Telecoms

Network

Service function

Switching function

The IN Concept INs provide a framework that allows for the cost-effective introduction, control and management of services. To achieve this, the IN concept aims to reduce implementation time and improve efficiency by separating service control from other network functions. INs offer flexible network architectures, standard network interfaces and rapid service introduction. An IN separates the switching and service functions within a network. This enables the rapid introduction of new services with minimum disruption to the existing network. An IN is overlaid on the existing network. It is intended to be applicable to all telecommunication networks, both circuit-switched and packet-switched. INs support a wide variety of services, including supplementary services, and utilize existing and future bearer services (e.g. as defined in the ISDN context).

2.38

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TY2600/v4.1

The PSTN and ISDN

LE

TE (SSP) TE (SSP)

LE

SCP Service software

LE

TE (SSP)

IN Implementation Simplified Any call requiring IN-based services is forwarded on to a Service Switching Point (SSP), where the required service can be controlled by separate service logic located in a centralized Service Control Point (SCP). In effect, the intelligence has been separated from the switching. This centralized approach allows new services to be introduced without having to update every network node or reconfigure hardware. This, in turn, allows for more efficient implementation and reduced lead times. The main elements of the IN architecture are described below. The Service Control Point (SCP) performs the service control function. It contains the service logic to control all IN-based services. The Service Switching Point (SSP) detects when an IN service is required and forwards the necessary information to the SCP for processing. In practice, SSPs are the exchanges within an IN that are connected to the SCP. The SSP and SCP functions may be combined within one network node. This combined physical entity is known as a Service Switching and Control Point (SSCP). Databases hold service parameters and subscriber data. They are accessed by the SCPs.

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2.39

Introduction to Telecoms Service Control Point SCP Translation of 0800 Number Translate 0800 556777

4

New Number 0121 66788xx

3 SSP

5

TE (SSP) LE

2

LE

(with additional control information) 6

1 Calling Party ‘0800 556777’ (non-geographic number)

Called Party (0121 66788xx) (geographic number)

An Example of an IN Service An IN may use two types of signalling, circuit-related signalling and non-circuit-related signalling. Circuit-related signalling is used to set up, maintain and clear down a telephony circuit, whereas noncircuit-related signalling is used between nodes to exchange call-related information. This is shown in the call flow in the diagram above. One of the first IN services to be implemented was the 0800 or Freephone service. This simply involves the translation of an 0800 number (which has no geographical significance) to another number representing a destination (with geographical significance). 1–2

Circuit-related signalling takes place between the calling party and the LE and then between the LE and the TE.

3

On detecting that 0800 has been dialled, the TE sends non-circuit-related signalling to the SCP, requesting a translation of the 0800 number.

4

The SCP performs the translation and then passes the new number back to the TE. Again, this is performed using non-circuit-related signalling.

5–6

The TE then routes to the LE, and the LE to the called party, in the normal manner using circuitrelated signalling.

2.40

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TY2600/v4.1

The PSTN and ISDN

VOTE

Yes

No

55%

45% Premium Rate

Freephone Televoting

VPN PSTN Credit Card Dialling

Virtual Private Network

Single No.

London

Single No.

BANK

Cardiff

Universal Access Number

BANK

Edinburgh

BANK

Universal Personal Telecommunications

IN-based Services The implementation of INs has enabled the rapid introduction of new, sophisticated services. Some IN services are described below. Freephone The served user can be reached with a freephone number from all or part of the country (or internationally) and be charged for the call. Virtual Private Network (VPN) This permits a private network to be built using the resources of the public network. The subscriber’s lines may be connected onto different switches, but still allow call transfer, call hold, etc. (As an option, access may be allowed from any point on the network). Credit Card Calling This allows subscribers to place calls from any telephone or access point to any destination and have the costs automatically charged to the account specified. The caller dials the card number and a PIN followed by the called number. Premium Rate Part of a call cost is paid back to the called party (considered an added value service provider). The served user can be reached from all or part of the country or internationally via translation of a premium rate number. Televoting This enables subscribers to assist in surveys of public opinion by ringing a specific number according to their choice. Alternatively they may be asked to ring a unique number and, after prompting, to indicate their choice by use of voice or keypad. Universal Personal Telecommunications (UPT) Personal mobility is achieved by use of a unique Personal Telecommunications Number (PTN), which allows any type of service to be used across multiple networks at any network access. The PTN is translated to an appropriate destination number for routing. Universal Access Number (UAN) This service allows a subscriber with several terminating lines in any number of locations or zones to be reached via a unique directory number. The subscriber can specify which incoming calls are routed to which terminating lines based upon geographical area. TY2600/v4.1

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2.41

Introduction to Telecoms

2.42

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TY2600/v4.1

Introduction to Telecoms

SECTION 3

TRANSMISSION NETWORKS

© Wray Castle Limited

I

Introduction to Telecoms

CONTENTS Plesiochronous Digital Hierarchy (PDH) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.1 PDH Multiplexing Stages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.2 Capacities, Bit Rates and Tolerances within PDH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.3 Primary-Level Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.4 64 kbit/s Cross Connection in the PDH

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.5

8 Mbit/s and 34 Mbit/s Leased Services over the PDH

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.6

The Synchronous Digital Hierarchy (SDH) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.7 Advantages of SDH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.8 SDH Payloads

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.9

Concept of Virtual Containers (VC) in SDH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.10 End-to-End Connection and Virtual Containers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.11 Multiplexing Virtual Containers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.12 The Synchronous Transport Module (STM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.13 Synchronous Multiplexers

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.14

SDH Rings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.15 SDH Local Access Ring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.16 Concatenation

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.17

Contiguous Concatenation Example

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.18

Next Generation SDH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.19 Virtual Concatenation (VCAT)

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.20

Comparison of Communication Media . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.21 Radio Fundamentals – The EM (Electromagnetic) Wave

. . . . . . . . . . . . . . . . . . . . . . . . . . .3.22

Properties of EM Waves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.23 Radio Fundamentals – The EM Spectrum . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.24

II

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Transmission Networks

CONTENTS General Propagation Concepts

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.25

Fading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.26 Modulation Techniques

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.27

Radio System Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.28 A Typical Digital Radio System

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.29

PTP Radio Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.30 SDH Radio Access Alternate Routing

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.31

Optical Fibre . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.32 Types of Optical Fibre

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.33

Options for Increasing Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.34 Wavelength Division Multiplexing (WDM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.35 Coarse WDM (CWDM) and Dense WDM (DWDM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.37 Optical Cross-Connects and Optical Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.38 Optical Transport Network (OTN) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.39 Client Signals in the OTN – Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .3.40

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III

Introduction to Telecoms

IV

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Transmission Networks

OBJECTIVES At the end of this section you will be able to:



describe the operating principles of the Plesiochronous Digital Hierarchy (PDH)



identify the main functions of the Synchronous Digital Hierarchy (SDH)



outline the advantages of SDH compared to PDH



identify the structure of typical SDH networks



explain the reasons for developing next-generation SDH



provide examples of how radio is used in PDH and SDH networks



identify the main types of optical fibre and where they are used



explain how optical fibre can provide high-capacity transmission and how this may be extended with Dense Wavelength Division Multiplexing (DWDM)



describe the main functions of an Optical Transport Network (OTN)

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V

Introduction to Telecoms

VI

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Transmission Networks

30 analogue speech band

E1 signal (2.048 Mbit/s +/- 50ppm)

PMUX

signals

4 x E1 signals (+/- 50ppm)

2-8 HOM

E2 signal (8.448 Mbit/s +/- 30ppm) Note: the output bit rate is more than the sum of the inputs as justification is carried out in the HOM.

Plesiochronous Digital Hierarchy (PDH) PDH networks employ a hierarchy of multiplexing stages to create high-capacity systems. ITU-T Recommendation G.703 specifies this hierarchy of frame-structured digital signals. Primary level Multiplexers (PMUX) use Time Division Multiplexing (TDM) to generate a byte-interleaved frame structure – either E1 or T1. Before the development of a digitally switched network, E1 signals operated between PMUXs and the receiver recovered its timing (clock) from the transmitting PMUX. Each PMUX was independently operated from a quartz timing source, which meant that all the E1s in a network were operating at a slightly different speed. The term used to describe this is ‘plesiochronous’. The variation of the E1 speed was limited to +/– 50ppm (parts per million). When Higher Order Multiplexers (HOMs) were developed, it could not be guaranteed that all the E1 inputs were running at exactly the same speed. When multiplexing takes place it is very important to ensure that no bits are ‘lost’. Therefore, if four plesiochronous E1 signals (2.048 Mbit/s) are to be multiplexed in time, then some extra bits are added to the higher speed signal to ensure that no ‘real’ data bits are lost. This means that the output bit rate is more than four times the input. This process of adding the extra bits is known as ‘justification’. The HOMs also employ TDM, but generate bit-interleaved frame structures. They are unaware of the frame structures of the lower speed tributaries and just take each bit and multiplex it into the higher speed signal. If there are justification bits or traffic bits in the tributary signal, the HOM is unaware of these – it just multiplexes every bit in turn. A hierarchy of multiplexers was developed over the years, each one building on the principles of justification. This is known as the Plesiochronous Digital Hierarchy (PDH). The PDH may be considered as a mechanism for transporting 2.048 Mbit/s, or indeed any other PDH signal rate. With the development of the digital switch, all the E1 inputs and outputs to the switch had to be time synchronized. This means that now, the vast majority of E1’s and primary level equipment operates synchronously, i.e. their clocks are referenced to an accurate common source (an atomic clock). However, the PDH still carries out justification at each multiplexing stage. It is important to recognize that the E1 signal is the only signal in the hierarchy where there are no justification bits present.

TY2600/v4.1

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3.1

Introduction to Telecoms Higher Order Multiplexer (Plesiochronous) Line 34-140 DM System 140-34 DM

Multiplexing

140

34 140 3 DM

34

Virtual 34 Mbit/s ± 20 ppm

8

3 DM 34

34 2 DM

Same average rate 34.368 Mbit/s ± 20 ppm

8 2 DM

Virtual 8.448 Mbit/s ± 30 ppm

2 8

1

Demultiplexing

8 2

Same average rate 8.448 Mbit/s ± 30 ppm

Prim

Prim

1

Virtual 2.048 Mbit/s path 30

TX

RX

Same average rate 2.048 Mbit/s ± 50 ppm Trib out

Trib in

For example:

30

TE

e.g. 140 Mbit/s

TE

Transmission System

PDH Multiplexing Stages A 2.048 Mbit/s path is implemented via connections through the HOM environment. A 2.048 Mbit/s signal is multiplexed via a 2–8 Digital Multiplexer (DM), an 8–34 DM and a 34–140 DM equipment to occupy a position in the 140 Mbit/s signal. The demultiplexing process via the 140–34 DM, 34–8 DM and 8–22 DM equipment results in the presentation of a 2.048 Mbit/s tributary signal whose average rate is the same as its corresponding input tributary. Even though the HOM environment is plesiochronous, it maintains the timing characteristics of the primary level input signals (E1s). A main aim of the PDH is to ensure that for a given 2.048 Mbit/s path through the network, the output tributary signal operates at the same average rate as its corresponding input signal. This is also true of higher-rate signals. In the diagram, it can be seen that the 140 Mbit/s signal is passed over a transmission line system. This transmission system could be either a cable or radio network. As each digital multiplexer operates independently of the one above or below, it is not possible to find the E1 signal inside the 140 Mbit/s bit stream. The signal must be completely demultiplexed back to the E1 (2.084 Mbit/s) level.

3.2

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TY2600/v4.1

Transmission Networks Hierarchy Level

Multiplexer

O/P Bit Rate (Mbit/s)

Equivalent # No. of 64 kbit/s Chs.

Tolerance (ppm)

1

Primary

2.048

30

± 50

2

Second Order

8.448

120

± 30

3

Third Order

34.368

480

± 20

4

Fourth Order

139.264

1920

± 15

5

Fifth Order *

564.992

7680

*

* Currently there are no standalone fifth-order multiplexers in use, this stage of multiplexing is performed inside Optical Line Terminating Equipment (OLTE). # This number represents the number of simultaneous telephone calls possible at each level of the PDH hierarchy.

Capacities, Bit Rates and Tolerances within PDH Although the digital rates in the CEPT hierarchy are defined by the ITU-T, these are nominal rates and a slight variation is permitted. The allowance for rate variations results in a series of higher multiplexers (those above the primary rate), which do not operate on four synchronous input streams. The inputs to these multiplexers all have the same nominal rate, and are therefore termed ‘plesiochronous’ (nearly synchronous). The HOMs are described as plesiochronous multiplexers. In this situation, all HOM clocks are free running, i.e. not synchronized, but operate within a predetermined range. The diagram also indicates the number of 64 kbit/s channels which may be carried at each level of the PDH.

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3.3

Introduction to Telecoms Through-connect the contents of E11 TS2 to E1 2 TS5 and vice versa (64 kbit/s – cross connect)

Terminal Multiplexer 1

Voice or Data Tributaries

TS2

Terminal Mux

Aggregate E1

TS5

E11 Aggregate

Aggregate E1 2

ADM

TS1

TS8

30 Tributary signals flexibly assigned to timeslots

Voice digitized and assigned Timeslot 1 on E1 1

Manually configured static connection map

Tributary Cards Voice

Data

Data assigned to Timeslot 8 on E1 2

Manually configured static cross-connect map

Digital Access Cross Connect System (DACCS)

Multiple E1 Bearers

Multiple E1 Bearers

64 kbit/s cross connect any timeslot to any timeslot

Manually configured static cross-connect map

Primary-Level Devices The timeslot interchange function enables the development of a number of primary level equipment types: terminal multiplexers, Add and Drop Multiplexers (ADM), and Digital Access Cross Connect Systems (DACCS). A terminal normally consists of a single E1 aggregate and a range of interface cards to enable the support of voice and data services. Traffic applied to an interface card will be mapped to an available timeslot on the aggregate. Speech signals applied to a voice card are converted to 64 kbit/s PCM. Add and drop multiplexers, also called drop and insert multiplexers, are versatile devices typically deployed in access networks. They can support multiple E1 aggregates and a range of tributary cards to support voice and data services. Traffic from any tributary card can be flexibly assigned to any available timeslot on any aggregate. This assignment is bidirectional, i.e. input to a tributary is added (inserted) to an assigned timeslot in the transmit direction, while traffic received in the assigned timeslot is dropped to the appropriate tributary card. In addition to add and drop connections it is also to through-connect the contents of a timeslot on one aggregate to an available timeslot on another aggregate. This is known as a 64 kbit/s cross connect. The connection map for the device is manually configured and statically stored in local memory. This can be reconfigured from time to time by the operator to provision new circuits or to re-route existing ones. It is possible to equip an ADM with an E1 tributary card. In this case it is possible to establish 64 kbit/s cross connect between a timeslot on the tributary card and any available timeslot on any aggregate. DACCS devices may terminate a high density of E1 bearers. However, they do not provide access to tributary cards. They are normally used in an operator’s core network to route and groom traffic; for example, they can be used to provide 64 kbit/s leased services. The DACCS device is able to establish a 64 kbit/s cross connect between any timeslot on any E1 bearer to any other available timeslot on any other or the same E1 bearer. The connection map for the device is manually configured and statically stored in local memory. This can be reconfigured from time to time by the operator to provision new circuits or to re-route existing ones. The DACCS and ADM are 64 kbit/s cross connect devices. While it is possible to build N x 64 kbit/s (N=1 to 30) circuits, they cannot cross connect an entire E1 bearer.

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Transmission Networks Timeslot 3 cross connected to Timeslot 6

64 kbit/s Tributary signal assigned to Timeslot 2

DACCS

Through connect TS2

ADM

E1

Customer Site A

TS3

E1

TS3

PDH HOM

TS6

E1

E1 64 kbit/s Cross Connect

Terminal Mux

PDH HOM

Timeslot 6 cross connected to Timeslot 24

64 kbit/s Tributary signal assigned to Timeslot 5

DACCS

Through connect

Server

TS5

E1

Customer Site B

TS24

ADM

E1

TS24

PDH HOM

TS6

E1

E1 64 kbit/s Cross Connect

Terminal Mux

64 kbit/s Cross Connection in the PDH Leased lines at various bit rates can be provided for customers. For example, a 64 kbit/s leased line can be implemented across the network using a range of primary layer transmission equipment including terminal multiplexers, ADMs and DACCS systems. Terminal multiplexers map tributary inputs to a nominated timeslot while the ADMs and DACCS devices cross connect the contents of a timeslot from one E1 bearer into another timeslot in another E1 bearer. In the diagram, a corporate customer is linking two offices at A and B using a 64 kbit/s leased line with timeslot mappings and cross connections between E1 bearers. E1 bearers between network sites are connected via a PDH HOM network. The 64 kit/s connection will remain in place for the duration of the lease, which could be many years. The cross connects can be controlled remotely from a network management system, which allows a leased line to be provisioned relatively quickly across the network.

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3.5

Introduction to Telecoms 8 Mbit/s

3 DM

CPE 8M Radio Access Link

34 Mbit/s

8M DDF

140 Mbit/s

4 DM

OLTE

34 M DDF

140 M DDF

140 Mbit/s Optical Fibre Link

CPE 34 Mbit/s

34 Mbit/s

140 Mbit/s

OLTE

CPE

4 DM 34 M DDF

140 M DDF 8 Mbit/s

3 DM

CPE 8 Mbit/s Optical Fibre Link

34 Mbit/s

8M DDF

8 Mbit/s and 34 Mbit/s Leased Services over the PDH To enable flexibility in the provision of digital signals through the PDH network, Digital Distribution Frames (DDFs) are employed between each multiplexing stage. These DDFs contain coaxial jumper cables used to interconnect the multiplexers and other equipment at each stage. In the figure, there are two customers, one with an 8 Mbit/s leased line and another with a 34 Mbit/s leased line. For the 8 Mbit/s leased line, the Customer Premises Equipment (CPE) produces the 8 Mbit/s signal and in this case an 8 M radio access link is used to access the network. The signal is then connected to an 8–34 DM (in this case using tributary four). The DDF is used to make this connection. The 34 Mbit/s signal is then passed to a 34 M DDF to be connected to tributary one on the 34–140 DM. The 140 Mbit/s signal is then passed to the 140 M DDF and in this case the signal is passed to an Optical Line Terminating Equipment (OLTE). The OLTE converts the electrical signal to optical and transmits it over an optical fibre to a distant location where the process is reversed down to the 8 Mbit/s level. This signal in this case is then transferred to the customer’s premises over an optical fibre access link. For the 34 Mbit/s leased line, the signal from the customer is taken to the 34 M DDF and then connected to tributary three on the 34–140 DM. Circuit provisioning for these leased lines can be time consuming as the DDFs must be manually configured using coaxial jumper cables. It is important to recognize that in the PDH, justification is carried out at each stage of multiplexing. This means that it is not possible to drop out an 8 Mbit/s signal directly from a 140 Mbit/s signal. It must be demultiplexed one stage at a time until the 8 Mbit/s signal can be recovered.

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TY2600/v4.1

Transmission Networks PDH Input

SDH Transmission Network

PDH Output

2 Mbit/s

2 Mbit/s

34 Mbit/s

34 Mbit/s

140 Mbit/s

140 Mbit/s

high capacity flexible built-in management and protection backward compatibility with PDH future-proofed

The Synchronous Digital Hierarchy (SDH) As defined by the ITU-T in Recommendation G.708, ‘The SDH is a set of hierarchical transport structures, designed to transport suitably adapted payloads in a physical (managed) transmission network.’ From the user’s perspective, the SDH is a transmission network that provides information pipes connected from input to output. As in PDH, these pipes may operate at a number of different bit rates. However, the bit rates used in SDH are far higher than PDH. In SDH the information pipes are referred to as payloads. SDH may also be used to carry several types of payload, including PDH signals. From the point of view of the operator, SDH is a high-capacity, flexible network with management and protection facilities built in, and backward compatibility with PDH. It is future-proofed to interwork with high-speed services. By examining the relationships between each of the three key elements of the formal ITU-T definition, it is possible to build a simple conceptual model of an SDH transmission network and the facilities that such a network could be designed to support.

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3.7

Introduction to Telecoms Managed Network Extensive overhead facilities permit in-service monitoring of: – lines – section – paths

Flexibility SDH offers flexibility of operation by: – remote configuration/reconfiguration (via control channels embedded in overheads) – add-and-drop and cross-connect functionality

Network Resilience Built-in protection mechanisms are implemented at: – media level (line system) – path level (individual circuit) – equipment redundancy

Advantages of SDH SDH contains many functions that overcome the shortcomings of PDH systems, including network management capability, flexibility and resilience. The SDH transmission structures contain sufficient overhead elements to allow the monitoring of lines, sections and paths, so that each of these entities can be managed separately. SDH is based on a synchronous multiplexing technique, which means that justification is only carried out once, when the tributary signal is inserted in the frame. This means that the lower-order payloads can be identified and directly dropped out of or added into the higher bit rate signal. Flexible multiplexing arrangements include add-and-drop facilities, and along with remote configuration capabilities, it means that SDH is more efficient than PDH systems, which had a very static configuration. With the SDH, reconfiguration is carried out under software control, but in the PDH, reconfiguration was carried out by manually moving jumpers on the DDFs. SDH incorporates sophisticated protection mechanisms which can be used to protect traffic at both the circuit and line level.

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Transmission Networks

Network Management Functions Possible payloads (to be adapted) European PDH 2, 34 and 140 Mbit/s North American PDH 1.5, 6, 45 Mbit/s Asynchronous Transfer Mode (ATM) cells LAN and MAN Payloads (e.g. FDDI and IEEE 802.6)

Network Elements multiplexers cross-connects regenerators optical fibre radio (rings are often used)

Transport Structures STM 0 STM 1 STM 4 STM 16 STM 64 STM 256

51.84 Mbit/s 155.520 Mbit/s 622.08 Mbit/s 2.48832 Gbit/s 9.95328 Gbit/s 39.81312 Gbit/s

Synchronization and Clocking

SDH Transmission Network

SDH Payloads SDH frame structures have been optimized to transport structured PDH payloads, LAN/MAN and Broadband ISDN (B-ISDN) service payloads. The SDH frame structures have been designed to transport payloads structured in accordance with both CEPT and ANSI PDH standards. The exception to this is the 8.448 Mbit/s signal (E2), which is not supported as an SDH payload. A number of levels have been defined in a hierarchical set of transport structures. Each level is defined in terms of a specific transmission, or transport, bit rate and by a unique frame-structured signal referred to as a Synchronous Transport Module (STM). The nominal transmission bit rates for each of the hierarchical levels 1 to 4 are specified in ITU-T Recommendation G.707. It is interesting to note the simple harmonic relationship between each hierarchical level of the SDH, and to compare this with the relationships that exist between the various levels of the conventional PDH. For example, an STM-4 is exactly four times the bit rate of an STM-1 and no justification is carried out between the higher STM levels. This can be achieved because all the STM signals in the network are derived from the same atomic clock and are therefore accurately synchronized. STM-0 (51.84 Mbit/s) of the SDH was defined in an annex of the 1993 issue of ITU-T Recommendation G.708 in response to a requirement for a transport structure to support light to medium aggregate traffic in an SDH network.

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3.9

Introduction to Telecoms Justification Bits 1.544 Mbit/s (T1)

Payload

VC-11 Justification Bits

2.048 Mbit/s (E1)

Payload

VC-12 (2.24 Mbit/s) Justification Bits

6.312 Mbit/s (T2)

Payload

VC-2 Justification Bits

34.368 Mbit/s (E3) or 44.736 Mbit/s (T3)

Payload

VC-3 Justification Bits

139.264 Mbit/s (E4)

Payload

VC-4 (150.336 Mbit/s)

Concept of Virtual Containers (VC) in SDH The frame structure of an SDH signal is based on the concept of Virtual Containers (VCs). All virtual containers in a network are derived from the same clock source and are therefore synchronous. Different sizes of VC have been designed to carry the various bit rates of the PDH. For example, a VC-12 is designed to carry the first level of the PDH, second option which is E1. A VC-11 is designed to carry the first level of the PDH, first option which is T1. Other containers are available to carry most of the PDH payloads from both the European and North American hierarchies. These are shown in the diagram. Note that a VC-2 has been designed to carry 6.312 Mbit/s from the North American PDH but there is no facility to carry an 8 Mbit/s signal from the European PDH. The bit rates of all the VCs are synchronized, i.e. all VC-12s have a bit rate of 2.24 Mbit/s. This means that the E1 tributary signal can have a variation of +/– 50 pmm but the container VC-12 will always be 2.24 Mbit/s. Justification bits are therefore added to ensure the tributary signal fills up the VC-12. If the E1 signal is slow, then more justification bits are added, and if the E1 signal is fast then fewer justification bits are added. All VCs also carry a Path Overhead (POH). This is a set of labels containing information about the VC such as its identifier, whether there are errors present, various alarms, etc. In SDH, it is the VC which is cross-connected through the network.

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Transmission Networks VC-N cross-connected into appropriate slots within the STM-N

STM-4 Input/ Output

SDH X-Con

STM-4

STM-4 ADM 2

ADM 1

ADM 3

Input/ Output

2M

VC-12

VC-12

VC-12

2M

34M

VC-3

VC-3

VC-3

34M

140M

VC-4

VC-4

VC-4

140M

X-Con

PDH Payloads mapped into a virtual container

X-Con

Transported end-to-end VC-N

PDH Payloads mapped into a virtual container

Tributary Card

End-to-End Connection and Virtual Containers A Virtual Container of order N (VC-N) may be considered as the unit of transport for SDH networks. All payload types are connected to a tributary card on an SDH device such as an ADM and are mapped into the appropriate VC-N, as shown above. The VC-N is then transported end to end across the SDH network. Intermediate SDH nodes such as other ADMs and Cross-Connects (X-con) cross connect the VC-N from one STM signal to another. When the VC-N reaches its destination the terminating SDH device recovers the payload from the VC-N and transmits it out of the appropriate tributary card.

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3.11

Introduction to Telecoms VC-12

E1 tributary (+/-50ppm)

X3 (m) 2 x 2M

6M tributary VC-12

VC-2

X7 (l)

21 x 2M

VC-12

VC-3

34M or 45M tributary

X3 (k) VC-4 VC-4

63 x 2M or 3 x 34M or 1x 140M etc.

VC-12

140M tributary

Multiplexing Virtual Containers As all virtual containers are derived from a common clock source, they are synchronous in nature. Although there are containers of different bit rates, these bit rates are directly related and remain constant. The smaller containers can therefore be placed inside larger containers without the addition of justification bits. A simplified diagram is shown: although the actual process is more complex and other stages are employed, the principles of multiplexing can be seen. An E1 tributary is placed in a VC-12 along with some justification bits. This VC-12 can be synchronously multiplexed with two other VC-12s to make a VC-2. Seven VC-2s can then be multiplexed into a VC-3 and three VC-3s can be multiplexed into a VC-4. Note that justification is only carried out when a tributary is placed in a container. The location of the VC-12 inside the VC-4 can be identified by a k,l,m number:   

k is the number of the VC-3 inside the VC-4 l is the number of the VC-2 inside the VC-3 m is the number of the VC-12 inside the VC-2

In practice, a unit known as a Tributary Unit (TU) is used which employs a device known as a pointer. The pointer ensures that if there is any phase variation between networks and countries, the virtual containers can be identified. A VC-4 does not have to contain 63 E1 signals. For example, a VC-3 could contain 21 E1s and another VC-3 could contain a 34 Mbit/s signal. A VC-2 does not have to contain three E1s; it could contain a 6 Mbit/s signal. There are many combinations that could be used, but the important thing is that it is always possible to identify each type of tributary inside the VC-4. This process is the key that makes SDH very flexible when combining signals of different bit rates.

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Transmission Networks

VC-4

STM-1

150.336 Mbit/s

5.184 Mbit/s

STM-1 155.52 Mbit/s

VC-4

20.736 Mbit/s

VC-4

VC-4

VC-4

STM-4

150.336 Mbit/s STM-4 622.08 Mbit/s

The Synchronous Transport Module (STM) The STM is used to carry the VC-4 over an optical fibre or radio network. Extra overhead information is added to the VC-4 to form an STM-1. This extra overhead contains information related to the fibre or radio link such as alarms, identities, automatic protection switching, etc. An STM is not passed through a multiplexer as the STM overhead is removed at the input to a multiplexer and a new overhead is added at the output. Only VCs are cross-connected in SDH networks. An STM-4 is exactly four times the size of an STM-1 but it is important to recognize that an STM-4 does not contain four STM-1s. An STM-4 overhead is four times the size of an STM-1 overhead but the STM-4 carries four VC-4s. Each VC-4 can be cross-connected into other STM-4s (or STM-1s) as it passes through the network.    

STM-1 capacity = 1 x VC-4 STM-4 capacity = 4 x VC-4 STM-16 capacity = 16 x VC-4 STM-64 capacity = 64 x VC-4, etc.

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3.13

Introduction to Telecoms a)

STM 1 (Aggregate)

STM 1 ADM

West

STM 1 (Aggregate) East

Tributary Cards 21 x 2 Mbit/s

21 x 2 Mbit/s

STM 1 (Trib) (Partially loaded with 2 Mbit/s signals)

b)

STM 4 (Aggregate)

STM 4 ADM

STM 4 (Aggregate) East

West

Tributary Cards 1 x140 63 x2 3 x 34 Mbit/s Mbit/s Mbit/s

STM 1 (Trib)

Synchronous Multiplexers Synchronous multiplexers in the SDH environment may be configured to operate either as a terminal multiplexer, an ADM or a Digital Cross Connect (DXC). A DXC does not have the facility of tributary access. Note that as the rate of information flow increases, i.e. as the STM rate rises, the choice of physical medium used in the line system drops to a single option, which is fibre. ADMs support multiple SDH aggregate signals and also provide access to tributary traffic (tributary inputs may be PDH or SDH signals). The 2 Mbit/s signals applied to the STM 1 ADM may be assigned to either the West or East aggregate in protected systems. Similarly, traffic signals in the STM 1 tributary may be connected to either or both of the main aggregates (West/East). Traffic signals arriving on the West aggregate may be cross-connected to the East aggregate (and vice versa), connected to tributary inputs/outputs, or connected to transport structures in the STM 1 tributary. By definition, a Network Element (NE) must support at least one SDH physical transport interface operating at one of the SDH hierarchical levels. Each medium (fibre, copper or radio) will support the transmission of suitably adapted payloads at one of the SDH hierarchical levels (STM 1/4/16/64). Standard optical interfaces have been specified for operation of NEs at each of the three hierarchical levels, to support intra- and inter-office working. Two electrical (copper) interfaces have been defined for intra-office working at the STM 0 (51 Mbit/s) and STM 1 (155 Mbit/s) levels only. No common air interface has been specified for SDH radio systems. An SDH DXC provides the functions of cross-connecting the VCs contained in several separate STM-N signals connected to the DXC. It does not provide the function of adding or dropping tributary signals from the fibre.

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Transmission Networks STM 4 Transport Ring

Network Access Point STM 4 ADM

West

East

STM 1 Tributary

W

E ADM 1

W ADM 2

2 Mbit/s E

E ADM 4

2 Mbit/s Customer site A W

ADM 3

W

STM 1 Access Ring

E

Leased Line between site A and B

2 Mbit/s Customer site B

SDH Rings ADMs may be connected in ring configurations and deployed as an access mechanism to the main backbone network. Ring structures have the advantage of having at least two paths between any point on the ring enabling protection schemes to be relatively easily and cost-effectively deployed. Rings are probably the most commonly deployed topology in SDH networks. Each ADM has at least two ports connected to the ring, which are typically referred to as East and West, but they just represent the direction around the ring. SDH rings may be interconnected using either PDH or SDH tributary ports, although in order to maintain SDH monitoring functions on an end-to-end basis the preference would be to use the SDH option. In the example shown here, an STM-1 tributary interface is used to interconnect an STM-1 access ring to an STM-4 ring. Provisioning circuits using rings is relatively easy. For example, a customer may require a leased line service between customer Site A and Site B. To satisfy this requirement a 2 Mbit/s tributary from ADM 3 is dropped off at a suitable tributary at ADM 4. As another example, it may be necessary to provide a 2 Mbit/s connection from customer Site B to the STM-4 ring for onward routing to Site C (not shown). In this instance a 2 Mbit/s tributary is connected to the West-bound aggregate and cross connected via ADM 2 to ADM 1 East-bound aggregate. From here, the 2 Mbit/s signal is connected to the STM 1 tributary and finally onto either the West-bound or Eastbound aggregates associated with the STM 4 ADM. Additionally, the 2 Mbit/s signal could be broadcast both ways around the ring. The West aggregate may be declared the main path; should this fail, the East aggregate provides a protection path, which may be selected automatically.

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3.15

Introduction to Telecoms 30 analogue telephones

PSTN Switch

PBX

PMUX

2 Mbit/s 2 Mbit/s Tribs

2 Mbit/s

ADM #1

VC-12

ADM #8

VC-12

ADM #2

ADM #3 VC-12

ADM #4 Optical Fibre

STM-16 Local Access Ring ADM #7

ADM #6

ADM #5 VC-3

34 Mbit/s Trib

OLTE

2 Mbit/s 34 Mbit/s Trib

PBX

34 Mbit/s Optical Fibre Link

34 Mbit/s Radio Link

OLTE 34 Mbit/s

CPE

34 Mbit/s Leased Line

CPE

SDH Local Access Ring The diagram shows an example of an SDH fibre ring located in an access network. A 34 Mbit/s leased line has been set up between two CPEs. The signal is passed to ADM#8 where it is placed in a VC-3. This VC-3 could be transported either way around the ring to ADM#5, where the 34 Mbit/s tributary is recovered and passed to the customer over a radio link. PBXs send 2 Mbit/s signals to ADM#7 and ADM#3. These are placed in VC-12s, sent around the ring and recovered at ADM#1. The 2 Mbit/s signals are then passed to a PSTN switch. Thirty analogue telephone lines are multiplexed into a 2 Mbit/s signal by a PMUX. This E1 signal is then put into a VC-12 at ADM#4 and recovered at ADM#1 for access to the local PSTN switch. This process has many advantages, for instance the length of the copper analogue telephone wire can be reduced by placing the ADM#4 in a street cabinet in the neighbourhood. This would mean that the customers do not have to be ‘local’ to the PSTN switch. A ring network can provide a certain amount of resilience. If there is a fibre failure between ADM#7 and ADM#6 then the SDH network can reroute the VC-3 around the other side of the ring and restore service.

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Transmission Networks

VC-4-4c

Payload 599.04 Mbit/s

20.736 Mbit/s

150.336 Mbit/s STM-4 622.08 Mbit/s

Concatenation Concatenation is the process of logically combining a number of VC-4s so that they remain locked together as they pass across the network. In this case a cross connect moves the concatenated VC-4s as one block. This would be used to provide a payload of 599.04 Mbit/s across the network when four VC-4s are concatenated. There is only one path overhead attached to the concatenated VC-4s. If four VC-4s are concatenated, the new VC is known as a VC-4–4c and is carried inside an STM-4 or higher STMs. If sixteen VC-4s are concatenated, the new VC is known as a VC-4–16c and is carried inside an STM-16 or higher STMs. Concatenation is used to provide very high speed Internet backbone data links, for example, using protocols such as ATM or MPLS.

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3.17

Introduction to Telecoms ATM Switch

ADM

VC-41

ADM

VC-42

VC-43

VC-44

ADM

VC-41

VC-42

VC-43

VC-44

STM-16 Section Overhead

VC-4-4c VC-41

VC-42

VC-43

VC-44

VC-41

VC-42

VC-43

VC-44

VC-45

VC-46

VC-47

VC-48

VC-49

VC-410

VC-411

VC-412

VC-413

VC-414

VC-416

VC-416

VC-41

VC-42

VC-43

VC-44

ADM

STM-16 Ring

600 Mbits/s

ADM ATM Switch

Note that the contiguously concatenated VC-4 must remain on the same path as an ordered block throughout the entire connection

Concatenation options for VC-4-Xc: VC-4-64c, VC-4-16c, VC-4-4c

Contiguous Concatenation Example In this example four contiguously concatenated VC-4s (written as VC-4-4c) are used to connect an ATM ‘circuit’ at around 600 Mbit/s between two ATM switches. The VC-4-4c group must be consecutively numbered VC-4s within the STM-16 and they must be switched as a block at intermediate ADMs. When planning this type of connection it should be ensured that all sections of the path can support the required ‘block capacity’.

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Transmission Networks 10 Mbit/s Ethernet 20% bandwidth utilization

VC-3

Payload 48.384 Mbit/s

100 Mbit/s Ethernet 66.7% bandwidth utilization

VC-4 (149.76 Mbit/s)

Payload 149.76 Mbit/s

1000 Mbit/s Ethernet 41% bandwidth utilization

VC-4-16c

Payload 2.396 Gbit/s

Next Generation SDH Many new network products are being developed under the general title of ‘Next Generation SDH’. Most Next Generation SDH products support recently developed protocols that can significantly enhance the bandwidth management and data handling capabilities of service provider networks. Next Generation SDH has been deployed by carriers as a way to support data protocols that traditionally were limited to relatively short distances such as Ethernet. Next Generation SDH enables the efficient transport of high-speed, high-bandwidth data within very tight budget constraints. The popularity of Ethernet in local area networking is largely due to its simplicity and cost-effectiveness. Standard Ethernet line rates operate at 10/100/1000 Mbit/s and Next Generation SDH is designed to carry these rates efficiently. The main aim of Next Generation SDH technologies is to improve the bandwidth granularity beyond the traditional concatenation boundaries, while maintaining all of the major characteristics of SDH such as fast protection and restoration and performance monitoring capabilities. SDH was designed with Virtual Containers whose size reflected the PDH bit rates. If a 10 Mbit/s Ethernet signal has to be transported over SDH then a VC-2 is too small so a VC-3 would have to be used. In this case, the actual transport capacity efficiency is about 20%. If a 100 Mbit/s signal has to be transported over SDH then a VC-3 is too small so a VC-4 would have to be used. In this case, the actual transport capacity efficiency would be about 66.7%. If a 1 Gbit/s signal has to be transported over SDH then an STM-16 link is normally used with concatenation. In this case, the actual transport capacity efficiency is about 41%.

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3.19

Introduction to Telecoms VCAT Transmitter

VCAT Receiver VC-4 cross connect equipment in the network is unaware of VCAT

Gigabit Ethernet (1Gbit/s)

Places VC-4s back into the correct sequence

VC-4

1

VC-4

3

VC-4

1

VC-4

2

VC-4

2

VC-4

2

VC-4

3

VC-4

1

VC-4

3

VC-4

4

VC-4

4

VC-4

4

VC-4

5

VC-4

6

VC-4

5

VC-4

6

VC-4

7

VC-4

6

VC-4

7

VC-4

5

VC-4

7

Gigabit Ethernet (1Gbit/s)

Virtual Concatenation (VCAT) The ITU has defined a process known as ‘Virtual Concatenation’ (VCAT), which addresses the problems of payload inefficiency. Virtual concatenation maps individual virtual containers into a ‘virtually concatenated’ link. Any number of containers can be logically grouped together, which provides more efficient bandwidth granularity than can be achieved using traditional techniques. It also enables network operators to adjust the transport capacity to the required customer service for greater efficiency. If seven VC-4s are concatenated together into a Virtual Concatenated Group (VCG) it is referred to as VC-4–7v. This is a VCG, where VC-4 defines the basic granularity and 7v defines the number of members in the group which would provide approximately 95% bandwidth utilization. In the figure, a Gigabit Ethernet signal (1000 Mbit/s) is split into seven parallel bit streams. Each of these is placed into a VC-4. The seven VC-4s are ‘virtually concatenated’. In other words they are bound together in a logical group with each one having a number to identify its place in the group. These VC-4s are transported over the SDH network. The network is unaware that these VC-4s are part of a VCG – it just carries and cross connects the VC-4s through the network. Each VC-4 can therefore follow different routes through the network. At the receiving end, the VC-4s may arrive out of sequence if they have been routed over different paths. The VCAT receiver places the VC-4s back into the correct sequence and recovers the Gigabit Ethernet signal.

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Transmission Networks a) Coaxial

Input Signal

Output Signal

Amplify Signal Repeater

LTE

LTE

Use: Intra-Office/Local Loop LTE = Line Terminal Equipment Distance: 200 m at 140 Mbit/s Typical loss: 18 dB/km

b) Optical Fibre System

Output Signal

Input Signal OLTE

Regenerator

OLTE

OLTE = Optical Line Terminating Equipment Use: Trunk Network/Access Network Distance (Single Hop): Typical 75–100 km Typical capacity: 2.5–40 Gbit/s (STM 1 to STM 256) Typical loss: 0.2 dB/km

c) Microwave Radio System

Parabolic Antenna

Parabolic Antenna

TX/RX

TX/RX

Digital Interface

Use: Trunk Access/Network Access Distance (Single Hop): 1–40 km Maximum capacity (currently): 565 Mbit/s Typical loss at 13 GHz is 115 dB at 1 km

Digital Interface

Comparison of Communication Media Radio has a relatively low capacity in view of the limited spectrum, and what spectrum is available is very expensive. Radio transmissions are subject to environmental effects such as man-made or atmospheric noise. Signals may also be reflected or refracted from buildings. However, radio is easy to install and is relatively cheap. Optical fibre has a very large capacity and is robust, i.e. not susceptible to interference. It is, however, expensive to produce, and in an urban environment it could be slow and expensive to install. Cable is cheap to produce but is susceptible to some interference. Like fibre it is slow and expensive to install. Cable has a fairly low capacity compared to fibre but is better than radio. Geography, terrain, installation costs, production costs and traffic volume are all areas that need to be considered when planning a network or a point-to-point link.

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3.21

Introduction to Telecoms

( +

Radio Systems Requirements: Radio transmitter 

generate EM wave



modulate waveform to carry information

Radio Receiver 

detect EM Wave



demodulate signal and recover information Direction of Propagation

Radio Fundamentals – The EM (Electromagnetic) Wave Radio communication requires three things. The first is that Electromagnetic (EM) waves need to be produced. These are a particular kind of disturbance set up when electrons are caused to change their position rapidly. EM waves do not communicate any intelligence simply by existing. It is necessary to alter the waves in order to convey information. For example, switching the waves on and off in accordance with the dots and dashes of the Morse code or producing variations corresponding to the sound waves of speech or music. This process is called ‘modulation’. It is also necessary to construct equipment at the distant point that will detect the presence of the waves and convert the small amount of energy in the waves into electrical currents of practical magnitude. This receiving equipment must not only detect the waves, but must also convert the modulation into suitable variations of current so as to reproduce the original information in the required form. This process is known as ‘detection’ or ‘demodulation’. If an alternating current is applied to a conductor, an alternating magnetic (H) field will be created around the wire. The alternating H field, due to the current in the wire, will create an alternating electrostatic (E) field in space further out from the conductor. The first transition from conduction fields to space fields has been made. The alternating E field, just born in space, will create an alternating H field further away from the conductor, which will then create another alternating E field, etc. This process, which continues on away from the conductor, is called ‘EM wave propagation’. The EM wave travels through free space at a speed of approximately 300,000,000 metres per second. An EM wave is a simultaneous disturbance in both the E field and the H field. This disturbance travels (propagates) as wave energy.

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Transmission Networks Frequency = Cycles (wavelengths) per second Polarization refers to the alignment of the Electric field component Frequency (f) = Velocity (c) / Wavelength ( )

Direction of travel at velocity ‘c’ Amplitude

Wavelength ( )

Properties of EM Waves Wavelength is defined as the distance in metres between any two similar points on the wave and is given the Greek symbol λ (lambda). Velocity is the measurement of the distance a crest has moved from a fixed point in a given time, for example per second. EM waves travel at the speed of light (c) in free space. The frequency (ƒ) of waves is the number of complete cycles passing a fixed point in one second. If one cycle passes a fixed point in one second this corresponds to a frequency of 1 Hertz (Hz). The relationship between λ, c and ƒ is: Frequency = velocity/wavelength or ƒ = c / λ The amplitude expresses the peak magnitude of the wave. Polarization refers to the alignment of the electric field in an EM wave. If an EM wave is transmitted with its E field oriented in the vertical plane the EM wave is vertically polarized. If the E field is transmitted in the horizontal plane it is horizontally polarized.

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3.23

Introduction to Telecoms

Radio

Millimetric Waves

Light

Gamma Rays

X-Rays

Infra Red 100 µm

VLF

LF

MF

HF

VHF

3 kHz

30 kHz

300 kHz

3 MHz

30 MHz

100 km

10 km

1 km

100 m

10 m

UHF

300 MHz

1m

Cosmic Rays

Visible

Ultra Violet

700 nm 400 nm

SHF

3 GHz

10 cm

10 nm

EHF

30 GHz 300 GHz

1 cm

1 mm

Used in telecom transmission networks for Point-to Point Radio links Satellite communication

Radio Fundamentals – The EM Spectrum Radio occupies just one small part of the complete EM spectrum. That part of the spectrum that is referred to as radio can be defined in terms of frequency as extending up to 300 GHz or in terms of wavelength as having a wavelength greater than 1mm. EM waves at the top end of the UHF band and in the SHF band are referred to, strangely, as microwaves despite having wavelengths in the range circa 10 cms to 1 cm. EM waves at frequencies higher than 300 GHz and wavelengths less than 1 mm are generally referred to as millimetric waves. At even higher frequencies, and once the wavelength reaches 100 µm, the waves are classified as light. Because the frequencies are so high, conventional descriptions of light waves are usually given in terms of wavelength. The lowest frequency light waves, infra red, have wavelengths from 100 µm down to 700 nm. Visible light has wavelengths from red light at 700 nm to violet light at approximately 400 nm. The remaining light spectrum is described as ultra violet and continues until the wavelength reaches 10 nm. Beyond this point in the EM spectrum waves are usually classified in terms of particle energy levels rather than frequency or wavelength. At energy levels higher than ultra violet light there are three loosely defined bands described as X-rays, gamma rays and cosmic rays. There are many ways of classifying different blocks of radio spectrum but a common one is to use decade divisions in wavelength and frequency. However, the names of these bands have more historical than practical significance in the context of modern wireless technologies. Microwave radio systems used as point-to-point radio links in telecoms transmission networks and satellite communications operate in the SHF and EHF bands. These links, due to their propagation characteristics, are often referred to as line-of-sight systems.

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Transmission Networks a) Ground Wave: 30 MHz Satellite

Ionosp here

Earth

Space Wave Line of Sight

General Propagation Concepts The lowest and highest radio frequencies propagate between transmitter and receiver in markedly different ways. In general, three basic loose divisions are made. Frequencies below about 3 MHz tend to propagate mainly as waves which follow the contours of the earth’s surface. The waves are commonly termed Ground Waves. At these frequencies, antennas and power levels need to be large and ranges of several hundred miles are possible. At these frequencies, between about 3 MHz and 30 MHz the ground wave is rapidly absorbed and the main mechanism of propagation is the sky wave. The sky wave is relatively high-angle radiation which undergoes refraction in the ionosphere, a layer of ionized gases high above the earth. The waves return to earth at some distant point and worldwide coverage is possible with relatively low transmitter powers and small antennas. At frequencies above about 30 MHz the wavelength becomes very short. Towards the top of the UHF section, the term ‘microwaves’ is used. At these frequencies, the ground wave is negligible, being absorbed very rapidly. Sky waves tend to escape into space, because of the different properties of the ionosphere at these frequencies. This allows satellite relays to be used, extending the range well over the horizon. Low-angle radiation from the antenna propagates as space waves. Although not strictly correct, space waves may be considered to propagate (travel) in straight lines and require a direct Line of Sight (LoS) between transmitting and receiving antennas. Radio systems using space waves are used extensively in transmission networks to implement point-to-point microwave radio links.

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3.25

Introduction to Telecoms 15 GHz and Above

Dominant Fading Cause

Precipitation

10 GHz and Below

Dominant Fading Cause

Multipath

10–15 GHz

Precipitation and Multipath

With good planning, link availabilities of 99.999% are typical

Fading In urban areas, most fixed customer access links are usually no more than a few kilometres long and operate in the higher microwave frequency bands, e.g. in the 18, 25, or 38 GHz bands. At frequencies above 15 GHz, radio waves are scattered by raindrops and other precipitates in the atmosphere. The effects of precipitation fading impose an upper limit on the availability of links operating above 15 GHz. At frequencies below 10 GHz, the dominant cause of fading is frequency selective multipath fading caused by the simultaneous reception of a direct wave and one or more reflected waves. Although the effects of precipitation fading can be seen and measured, precipitation fading is normally ignored when preparing path performance estimates for radio paths operating below 10 GHz. In temperate climates, for fixed links operating at frequencies between 10 GHz and 15 GHz the effects of multipath and precipitation fading are broadly similar. When preparing path performance estimates, allowances are made for the inclusion of both forms of fading on the basis that the conditions that favour precipitation fading do not tend to support simultaneous multipath fading. Nowadays the transmission characteristics of radio paths are modelled in software using functional algorithms that predict the probability, depth and duration of fading events, whilst other models are used to quantify the susceptibility of microwave radio receivers to such events. Path planning tools have largely reduced the complicated tasks of producing path performance estimates and optimizing other hardware parameters to an automated process. A fixed radio link is more than capable of meeting the performance and availability targets to support services in local access, i.e. customer access networks. In practice, link availabilities of up to 99.999% are routinely met, even when subjected to the restrictions placed on transmitter output powers by national regulatory bodies.

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Transmission Networks a) Amplitude Shift Keying (ASK) 0

1

0

1

1

0

1

0

b) Frequency Shift Keying (FSK) 0

Data

1

0

1

1

0

1

0

Data

Carrier

Carrier

Modulated Carrier

Modulated Carrier

c) Phase Shift Keying (PSK) 0

1

0

1

1

0

1

0

Data

Carrier

Modulated Carrier

Modulation Techniques To transmit information over a radio link, in either analogue or digital form, it must be added to, or placed upon, a carrier. A carrier in this respect is a single radio frequency. The three characteristics of a radio (sine) wave that may be changed in order to represent a modulated signal are its frequency, amplitude and phase. Thus the options for digital modulation of a carrier are Frequency Shift Keying (FSK), Amplitude Shift Keying (ASK) and Phase Shift Keying (PSK), or more sophisticated variants of these basic themes. ASK alters the amplitude in accordance with modulating signal; FSK alters the frequency in accordance with the modulating signal, and PSK alters the phase in accordance with the modulating signal. ASK carries the information in its amplitude so is corrupted by the effect of additive noise during radio transmission. The bandwidth is the same as PSK. FSK and PSK carry the information in their frequency or phase so are more robust against the effects of noise and are the preferred options. In general, PSK is the favoured option. The modulation schemes shown here are ‘low level’ modulation schemes. There are only two symbols available and each symbol represents a single bit, therefore the bit rate and symbol rate are the same. Higher level modulation schemes use more than two symbols and each symbol may be used to represent more than one bit, e.g. 16 QAM uses 16 symbols where each symbol represents four bits. For a given symbol rate, higher level modulation schemes are capable of higher bit rates. However, they are more susceptible to errors in noisy environments.

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3.27

Introduction to Telecoms 7.5 GHz 50 km

Link Length

3–4 km 58 GHz

Bit Rate

Bit rate α bandwidth Bandwidth determined by modulation scheme fc = 38 GHz

Bandwidth: 28 MHz May support 16 x 2 Mbit/s circuits (At lower frequencies a 28 MHz channel may support 156 Mbit/s)

ƒ

Radio System Parameters In temperate climates, such as NW Europe and much of the continental United States, a 7.5 GHz radio link between a customer’s premises and a radio distribution node might be up to 50 km long. At the other end of the scale, at 58 GHz, the maximum length of the access link is more likely to be in the order of three to four kilometres. The Radio Frequency (RF) spectrum is a finite resource. The use of the spectrum is strictly regulated by international agreements enforced by national regulatory bodies. Each radio link is licensed to operate on a designated RF channel with a specific bandwidth. For any given digital modulation scheme it is the bandwidth of the RF channel that determines the maximum bit rate of the traffic that can be delivered by the radio link. In the higher frequency bands, relatively simple modulation schemes are allowed and a typical 38 GHz PTP system might be allocated a 28 MHz wide channel to support a 16 x 2 Mbit/s PDH access link. In the lower frequency bands, a 28 MHz wide RF channel is now routinely expected to support traffic at up to 156 Mbit/s, i.e. at approximately four times the rate of 38 GHz systems. A characteristic of almost all fixed microwave radio links is that a clear LoS must be established between the antennas at each end of the link. At microwave frequencies, any obstruction in the LoS between a transmitter and a receiver has a marked effect on the level of the received signal. The clearance between any LoS path and a potential obstruction has to be calculated taking into account the probability of fading.

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Transmission Networks

Digital signal used to modulate radio carrier

Digital signal demodulated from the radio signal

Modulated radio carrier

Radio Carrier

Radio Carrier

Feeder NxE1

MUX

Modulator

TX

Feeder RX

MUX

Demodulator

NxE1

Point-to-Point Radio Link (Line of Sight)

Primary Mux

Primary Mux

1 30 Voice or Data Tributaries

30 1 Voice or Data Tributaries

A Typical Digital Radio System The growth in the number of digital LoS microwave links is expanding due to the digitalization of existing telephone networks and the phenomenal rise in mobile radio communications. Typical information carried on microwave links includes telephone speech, data, television/video and telemetry. Modern digital networks are based on a PCM waveform which is produced when speech is converted from an analogue into a digital 64 kbit/s signal typically multiplexed with other PCM signals to form an E1 bearer. A number of E1 bearers may be further modulated onto a higher rate signal, which may then be modulated onto a radio frequency and transmitted across the link. The figure shows a basic block diagram of a microwave link system.

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3.29

Introduction to Telecoms Corporate Customer’s Premises

25 GHz

Radio Distribution Node

12.0 km

N x 2 Mbit/s G.703

M Data Services

FMUX

Nx2 Mbit/s DRT

Nx2 Mb/s DRT

PDH/SDH MUX

PSTN Leased Line Network Services

–50 dBm

P-ISDN and Inter-PABX Trunks

+16 dBm

PABX

FMUX DRT

Flexible Access Multiplexer Digital Radio Terminal

PTP Radio Access The figure shows a PTP radio link operating at 25 GHz over a 12 km path between a radio distribution node and a corporate customer’s premises. The digital radio terminal supports ‘N’ x 2 Mbit/s traffic streams presented as standard G.703 interfaces to the customer’s terminal equipment and to higher-order PDH or SDH multiplexing equipment at the radio distribution node. Normally ‘N’, the number of 2 Mbit/s traffic streams, would be between 1 and 4 or 8 and 16. In the figure, the PTP link is shown supporting P-ISDN services and a range of other undefined services via a flexible PDH multiplexer or FMUX.

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Transmission Networks 38 GHz

Corporate Customer’s Premises

Radio Distribution Node

4.0 km

STM-1 DRT

STM-1 DRT

STM-1 155 Mbit/s Optical Line System

W ADM 1

STM-1 155 Mbit/s W

E

ADM 2

E

PABX

SDH Core Network

Router STM-16 Access Ring (622 Mbit/s)

Router

PMUX

PMUX DRT

STM-1 155 Mbit/s

Primary Multiplexer Digital Radio Terminal

SDH Radio Access Alternate Routing In this example a Point-To-Point (PTP) SDH microwave radio link is used to provide additional network security in cases where a second optical fibre route with sufficient diversity cannot be provided. An STM-1 Add-and-Drop Multiplexer (ADM) installed at a corporate customer’s premises is supported by an optical line connection and a diverse route provided by the PTP radio link. Customer traffic transported in appropriate VCs are mapped to both East and West-bound aggregates at ADM1. At the radio distribution node the VCs are re-mapped by the STM-1 DRT onto an STM-1 connection to ADM2. ADM2 routes them to the network operator’s SDH core transmission network via a high capacity STM-16 access ring along with other PDH or SDH traffic collected at the radio distribution node. SDH protection mechanisms at the far end of a Virtual Circuit connection allows customer traffic to be received from the best path, for example if the path associated with the fibre link fails traffic will still be available via the path associated with the PTP radio link.

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3.31

Introduction to Telecoms

Total internal reflection of light wave Cladding On/Off pulses of light

Light source (LED or Laser)

Light detector (photodiode or phototransistor) Data

Amplifier Core

Optical Fibre The overriding limiting factor in the use of wire pairs is the resistivity of the conducting material (usually either copper or aluminium). The resistance and thus the loss per unit length of a wire pair can be decreased by increasing the cross-sectional dimensions of the pair, but this is limited by cost and practical factors. In addition, as the frequency of the transmitted signal increases, it has a tendency to travel in a very thin skin at the surface of the conductors, which greatly reduces the cross-sectional area through which it travels, thus increasing the resistance. Furthermore, at frequencies of the order of GHz, there is a significant extra loss in the insulating material, which separates the conductors. Optical fibres use a fundamentally different form of transmission to that in wire pairs, i.e. the information is sent as a fluctuating beam of light along a glass tube. It is now possible to manufacture pure glass, which causes very little loss in the transmitted light. As light frequencies are very high it follows that fibre can be used for signals with much greater bandwidths (i.e. hundreds of megabits per second). Further advantages of optical fibres are immunity from EM interference, which makes it suitable for use even at relatively low bit rates in electrically noisy environments, and that it does not create a spark hazard and is therefore suitable for use in areas subject to fire risk. It also generates negligible interference and is therefore extremely difficult to tap. The fibre itself consists of glass core, a glass cladding of lower reflective index than the core and one or more protective coatings. The light is contained within the core of refraction; the precise way it travels down the fibre depends on the dimensions and the refractive profile of the core. The transmitter is a Light Emitting Diode (LED) or a high-performance laser (Light Amplification by Simulated Emission of Radiation); the receiver is a light-sensitive photo-diode or photo-transistor. Most optical fibre systems are used for digital transmission with the information sent as a sequence of pulses and spaces. Typical wavelengths used are between 800 and 1,600 nm, although in practice three wavelengths are commonly used, i.e. 850 nm, 1,300 nm (1.3 µm) and 1,500 nm (1.55 µm), each having a bandwidth of 100 GHz.

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Transmission Networks a) Multimode Stepped Index

Electrical Output Pulse

Electrical Input Pulse

Optical Receiver

Optical Transmitter

b) Multimode Graded Index

c) Monomode

Types of Optical Fibre Each type of optical fibre cable is classified by one of the methods of rating the index mode. In practice there are three commonly used types of optical fibre cable. These are multimode stepped index, multimode graded index and monomode. Multimode Stepped Index In a multimode stepped index fibre the cladding and the core have a different but uniform index (the index steps up between the cladding and the core). Some of the light emitted by the core strikes the core/cladding interface at an angle greater than the critical angle (critical angle is the angle at which total internal reflection first occurs) and is refracted into the cladding where it is absorbed; the rest of the light is propagated along the core by means of total internal reflections at this interface. Depending on the angle at which it strikes the interface, the propagated light takes a variable amount of time to reach the receiver. This stretching out, or dispersion as it is commonly referred to, tends to cause the pulses and spaces to merge. The effect restricts the bit rate and/or transmission distance. Graded Index Dispersion can be reduced by using a core material that has a graded refractive index, i.e. it declines in a particular way from a maximum at the core axis down to the value of the cladding’s index at the interface. The light is refracted by an increasing amount as it moves away from the axis. The average refractive index on the longer paths is less than that on the shorter paths because the former go closer to the core/clad interface. The average velocity on the longer paths is therefore greater than that on the shorter ones with the result that the propagation time along the different paths is almost constant. The received pulse widths in graded index fibre are therefore less than those in stepped index fibre, allowing an increase in the bit rate and/or transmission distance. Monomode A multi-mode fibre is so called because it allows light to travel down the fibre in a number of different modes, each of which is associated with a particular propagation angle, i.e. angle relative to the core axis. As the diameter of the core is decreased there is a reduction in the number of modes and when the diameter becomes less than approximately 10 µm only one mode is possible. There is very little dispersion with this type of fibre and used with a high laser diode, a very high performance can be obtained, i.e. bit rates in the order of hundreds of megabits per second combined with transmission of the order of tens of kilometres. Monomode is also known as ‘single mode fibre’.

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Introduction to Telecoms

Install additional fibre –

additional fibre or ducting not always feasible or cost-effective

Migrate to higher speed TDM –

dispersion at high bit rates

Deploy another technology –

Dense Wavelength Division Multiplexing (DWDM)

Options for Increasing Capacity One option for increasing capacity is to install additional fibre. This is not a cost-effective option, however, as the fibre infrastructure is one of the most expensive parts of the network. Also, adding more fibre or ducting is not always feasible. Another option is to migrate to higher-speed TDM. Unfortunately, the problem of dispersion at high bit rates means that regenerators may have to be closer together. For this to work effectively, more regenerators would be required. Given these difficulties, it may be necessary for operators to deploy different technologies. Advances in optical transmission technology have led to the technique of Wavelength Division Multiplexing (WDM) to tackle capacity problems.

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Transmission Networks

Ultraviolet

Visible

Infrared

x-rays gamma rays

Radio long waves 700

800

900 1000 1100 1200 1300 1400 1500 1600 1700

850

1310

nm

1550 1625

Wavelength Division Multiplexing (WDM) WDM uses different wavelengths of light to enable a number of bearers to be combined in a single fibre. The typical electromagnetic spectrum used for WDM is in the infrared region. Groups of wavelengths are used in this process as denoted by the first, second, third and fourth windows. WDM allows for the merging of optical traffic onto a single common fibre. It also allows high flexibility in expanding bandwidth whilst reducing costly mux/demux functions, of possibly independent bit rates and formats. The multiplexing process is entirely optical (no electronics are involved) and often uses a device called an Arrayed Waveguide Grating (AWG).

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3.35

Introduction to Telecoms

Transmitters

Receivers

λ1

λ1 λ2

Combining Signals

Transmission on Fibre

λn–1

Separating Signals

λ2

λn–1 Optical Amplifiers

λn

λn

Wavelength Division Multiplexing (WDM) (Continued) With WDM technology each digital signal modulates a separate laser light source, each laser generating a slightly different wavelength and the different optical wavelengths are fed into one fibre using a passive combiner. At the receiving end the different optical wavelengths are separated by filters and then passed to optical detectors so that the digital electrical signal can be recovered. A receiving filter is a component which acts as a wavelength-dependent reflector and will achieve precise wavelength separation. As this is a passive device, fabricated into glass fibre, it is very robust and durable. Another advantage of WDM is that different bit rates and framing structures can be used on each optical channel. Optical amplifiers have been developed which can be used on long routes to boost all the wavelengths of the optical signal. Note: The Greek letter lambda (λ) is used to represent wavelength.

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Transmission Networks 2.5 Gbit/s

2.5 Gbit/s

2.5 Gbit/s

2.5 Gbit/s 2.5 Gbit/s 2.5 Gbit/s

fibre pair (10 Gbit/s)

2.5 Gbit/s 2.5 Gbit/s

2.5 Gbit/s

2.5 Gbit/s

10 Gbit/s

10 Gbit/s

2.5 Gbit/s

Optical amplifiers

2.5 Gbit/s

Ethernet

Ethernet

2.5 Gbit/s

2.5 Gbit/s

Ethernet

Ethernet

2.5 Gbit/s

2.5 Gbit/s

10 Gbit/s

10 Gbit/s

Coarse WDM (CWDM) and Dense WDM (DWDM) The first networks deploying WDM technology multiplexed signals from lasers in the 1310nm region, which is now referred to as Coarse WDM (CWDM). Current CWDM, systems such as those with over 20nm channel spacing, are used for short-range transmissions where no amplification is required, such as metro networks. CWDM is a cost-effective solution to increase capacity on a fibre. The basic concept of CWDM technology is to offer acceptable optical bandwidth scalability at reduced cost. This is achieved by defining a much broader channel spacing between wavelengths, typically 2500 GHz (20nm), over the available single mode fibre spectral bands. Due to the wide optical spectrum employed it is not normally possible to use optical amplifiers. DWDM technology uses very close channel spacing and operates in the wavelength bands 1530–1625nm. These bands were chosen in part due to the operating range of optical amplifier devices which enabled the distance of the cable to be extended. DWDM channel frequencies are defined by a standardized ITU-T channel grid, and are typically spaced at either 200 GHz (1.6nm), 100 GHz (0.8nm), 50 GHz (0.4nm) and 25 GHz (0.2nm) separation. This close channel allocation provides large capacity transmission bandwidth, which allows for more than 100 optical channels that provide over one terabit/s of bandwidth on a single fibre. DWDM signals can be amplified by optical amplifiers as the close channel spacing occupies less optical bandwidth when compared with CWDM. However, DWDM techniques increase the cost of the laser transmitters and filter technologies. DWDM has evolved to dominate the long-haul core networks and is also moving into the metro and regional networks. It is predicted that there will be a steady reduction in the component prices (filters, lasers, amplifiers) of DWDM allowing it to gain wider adoption in the metro-access networks. Advances in DWDM continue and systems with the following capacities have been tested:   

320 x 2.5 Gbit/s (total: 800 Gbit/s) 160 x 10 Gbit/s (total: 1.6 Tbit/s) 128 x 40 Gbit/s (total: 5.12 Tbit/s)

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Introduction to Telecoms

Optical signals with different wavelengths

WDM Optical Multiplexer

Optical Fibre

Optical Add Drop Multiplexer

Optical Cross Connect

Optical Fibre

Fibre #3

Fibre #1

Fibre #4 Fibre #2

Optical Cross-Connects and Optical Switching Optical cross connects operate at the optical layer and are used to switch high-capacity optical signals. Various technologies have been developed to allow all-optical switching of input ports to output ports. Some of the competing technologies include Micro-Electro-Mechanical System (MEMS) technology, which use microscopic mirrors to reflect light, Frustrated Total Internal Reflection (FTIR), fibre gratings, holographic switches, waveguide switches and bubble-jet technology. Wavelength selectors split an incoming DWDM signal into its component wavelengths. One or multiple numbers of these wavelengths can be accessed at the selector’s ports. The signals are then recombined at the wavelength selector’s main output port. Service providers and cable operators are looking for a flexible and cost-effective solution to dynamically provide high bandwidth connectivity to any network site without interrupting the existing services or re-engineering the network. As the pace of optical technology advances, many equipment vendors have developed Reconfigurable Optical Add/Drop Multiplexers (ROADMs), which allow operators to terminate or insert a specific wavelength at a certain site directly from a fibre.

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Transmission Networks Fibre to the Neighbourhood (FTTN)

ROADM

ROADM

Fibre Ring

ROADM

Gigabit Ethernet (1GbE)

ROADM

Backbone Network SDH Optical Transport Network (OTN) The amount of data traffic relative to voice traffic on optical networks and the total traffic volume continues to increase. These factors are the drivers behind emerging, flexible technologies to supplement the mature, SDH transport infrastructure and help manage network complexity. At the edge of the network, where data and voice combine in a common infrastructure, new data-centric applications have emerged. An example is the combination of Virtual Concatenation (VCAT), which provides flexible bandwidth groupings for SDH virtual containers, Link Capacity Adjustment Scheme (LCAS), which provides dynamic bandwidth settings, and the Generic Framing Procedures (GFP), which provides a protocol agnostic frame container. In the transport core, bandwidth requirements have been expanded by the creation of the Optical Transport Network (OTN) described in general terms in ITU-T G.872. ITU-T G.709 provides the network interface definitions. An OTN is composed of a set of optical network elements connected by optical fibre links, which can provide the functionality of transport, multiplexing, routing, management, supervision and survivability of optical channels (wavelengths) carrying client signals. The figure shows a simplified OTN where ROADMs are used to add and drop particular wavelengths from a fibre. Each wavelength can be carrying different types of client signals such as Gigabit Ethernet and SDH, and all of these can be passed over a backbone network. On the other hand, a number of client signals can be multiplexed onto a single carrier in order to maximize the capacity of a single wavelength. G.709 was designed to improve transport network performance and facilitate the evolution to higher backbone bandwidths. The G.709 OTN frame includes a transport overhead that provides Operation, Administration, and Maintenance (OAM) capabilities, and Forward Error Correction (FEC). FEC helps reduce the number of transmission errors on noisy links, which enables the deployment of longer optical spans. Within Optical Transport Networks (OTNs), each wavelength is managed as a discrete optical channel (OCh) with its own Operations, Administration and Maintenance (OAM) functions. In the digital wrapper concept, a small, efficient OAM structure is wrapped around the data payload to uniformly define critical transport-related information, such as control functions, restoration signals, traffic types and traffic destinations. TY2600/v4.1

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3.39

Introduction to Telecoms TDM Electrical

OTN Terminal Multiplexer 1 (T1)

STM-16

STM-16 1

2.5 Gbit/s Gbit/s 2.5

2

GEthernet GEthernet

3

2.5 Gbit/s Gbit/s 2.5

4

2.5 Gbit/s Gbit/s 2.5

OTU21 1

2

3

4

λ1

WDM λ1

10 GE GE 10

λ2

λ2

OTU22

OTU22

GEthernet GEthernet

2

2.5 Gbit/s Gbit/s 2.5

3

2.5 Gbit/s Gbit/s 2.5

4

1 2

λ2

WDM λ2

λ1

Optical X-con

λ2

1

TDM Electrical

Optical X-Con OTN Terminal Multiplexer 2 (T2)

2.5 Gbit/s Gbit/s 2.5

3

OTU21

4 Optical X-con

10 GE GE 10

λ1

λ1 WDM

OTU2 is a digital transport frame, aka digital wrapper, with a capacity of around 10 Gbit/s. The OTN supports the following : OTUs: OTU1 – 2.5 Gbit/s; OTU2 – 10 Gbit/s; OTU3 – 40 Gbit/s under consideration is the OTU4 – 100 Gbit/s

OTN Terminal Multiplexer 3 (T3)

Client Signals in the OTN – Example The Optical Transport Network (OTN) is the latest generation of transmission network to be standardized by the IUT-T. It may be considered to be a network that incorporates the best elements of SDH and DWDM (maximizing the capacity of fibre optic cables) into one package. It is flexible, scalable and provides an efficient way of multiplexing high-bandwidth client payloads onto an optical carrier. In the example shown here, a number of client signals are to be transported through the OTN from OTN terminal 1 to OTN terminals 2 and 3 as shown. The T1 to T3 connection is required to support 3 SDH STM-16 clients and one Gigabit Ethernet (GE) client. The other connection form T1 to T2 is used to transport a 10 Gigabit Ethernet (10GE) client. One of the transport elements in the OTN is called an Optical Channel Transport Unit level 2 (OTU2). The capacity of this ‘box’ is sufficient to carry 10 Gbit/s of traffic. The four client signals at T1 to be connected to T3 are multiplexed into OTU21 and allocated optical channel λ1, the 10GE client is mapped into OTU22 and allocated optical channel λ2. λ1 and λ2 are then multiplexed onto the same fibre and connected to the an optical cross-connect switch. The optical crossconnect switches are used to route λ1 and λ2 to T3 and T2 respectively where the client signals are demultiplexed and connected to the appropriate output tributary.

3.40

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TY2600/v4.1

Introduction to Telecoms

SECTION 4

MOBILE CELLULAR NETWORKS

© Wray Castle Limited

I

Introduction to Telecoms

CONTENTS The EM Spectrum and Cellular System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.1 The Mobile Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.2 Mobile Network Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.3 Base Stations and Cells . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.4 Wide Area Coverage and Capacity

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.5

Increasing Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.6 Example Realistic Cell . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.7 The Mobile Phone The USIM

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.8

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.9

Network Registration

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.10

Making a Call from a Mobile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.11 Making a Call to a Mobile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.12 Making a Call to a Roaming Mobile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.13 Maintaining a Call on the Move

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.14

The Handover . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.15 GSM Network Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.16 The GSM Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.17 GPRS System Architecture

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.18

GPRS Network Element Basic Functions

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.19

SMS Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.20 GSM Radio Frequencies

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.21

Multiple Access Techniques . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.22 GSM Physical Channels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.23 GSM Physical Channel Offset and Data Rates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.24

II

© Wray Castle Limited

Mobile Cellular Networks

CONTENTS GSM Cell Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.25 GPRS Resource Allocation

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.26

EGPRS Resource Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.27 UMTS Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.28 Basic UMTS System Architecture

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.29

The UMTS Access Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.30 UMTS Core Network (CN) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.31 The Evolution of Core Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.32 Code Division Multiple Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.33 UMTS Frequencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.34 CDMA Code Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.35 HSPA and HSPA+ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.36 CDMA2000™ Development Phases

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .4.37

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III

Introduction to Telecoms

IV

© Wray Castle Limited

Mobile Cellular Networks

OBJECTIVES At the end of this section you will be able to: 

list potential issues that arise from using radio, or wireless, to access networks and services



describe the concept of cellular networks



identify the architecture of an example cellular network and state the function of each network element



describe how cellular networks manage user mobility



understand the sequence of events for incoming and outgoing calls from mobile phones, including roaming



briefly discuss the roles of network elements found in GSM and GPRS and UMTS architecture



list the UMTS service aims



discuss the use of CDMA2000™ as a 3G radio access technology

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V

Introduction to Telecoms

VI

© Wray Castle Limited

Mobile Cellular Networks

Radio

Millimetric Waves

Light

Gamma Rays

X-Rays

Infra Red 100 µm

VLF

LF

MF

HF

VHF

3 kHz

30 kHz

300 kHz

3 MHz

30 MHz

100 km

10 km

1 km

100 m

10 m

UHF

300 MHz

Visible

Ultra Violet

700 nm 400 nm

SHF

3 GHz

1m

Cosmic Rays

10 cm

10 nm

EHF

30 GHz 300 GHz

1 cm

1 mm

Best compromise for:  antenna size  power – efficient radiation from small antenna less power  small battery size with good battery life  capacity  relatively short propagation distance

The EM Spectrum and Cellular System The part of the radio spectrum that is most commonly used for mobile cellular networks occupies a relatively small part of the Ultra High Frequency (UHF) band. This part of the radio spectrum is very valuable and heavily used because it offers the best overall compromise between capacity, antenna size and propagation characteristics. Antennas radiate most efficiently and require less power for a given propagation distance where the antenna size is around the same size as the wavelength; quarter wavelength antennas are typical. At the frequencies used by mobile cellular systems, GSM and UMTS for example, antenna sizes are a few centimeters in length. Efficient radiation with minimum power means battery sizes can be relatively small with a reasonable lifetime. Capacity is of paramount importance in cellular systems. Operating in the higher frequency bands enables more channels of higher bit rates to be used. For example, GSM channels are spaced at 200 kHz, where each channel may support 271 kbit/s. This in turn may be used to support eight simultaneous users. Many channels of this size can be accommodated in the UHF band, 125 such channels occupy 25 MHz of spectrum, which is quite small compared to the overall bandwidth of the entire UHF band. The same number of channels allocated in the HF band would occupy almost the entire band. Additionally, at UHF frequencies terrestrial propagation distances are relatively short, allowing channels of the same frequency to be reused many times. This is critical to obtaining both capacity and coverage in cellular networks with a limited number of available channels allocated per operator.

TY2600/v4.1

© Wray Castle Limited

4.1

Introduction to Telecoms

Air interface

temporary ‘wires’ hostile environment

Home Network Roaming Advantages:

Disadvantages:

mobility

radio environment

roaming

locating the mobile spectrum costs infrastructure

Visited Network The Mobile Environment In a fixed-line network, the user is connected to the network by an installed set of wires. In the mobile environment these wires between the user and the system do not exist in a permanent form – they have to be ‘created’, using radio, every time the user wants to make or receive a call. This requirement for a radio link, or ‘air interface’, offers many advantages to users, principally mobility: the user is free to move while using the phone. However, any radio environment is hostile, and the cellular radio air interface is no exception. It is susceptible to problems that face any radio system, such as interference and variations in signal strength. These matters require technical solutions and careful network planning to minimize their effects. Additionally, while mobility makes cellular networks attractive to subscribers, it creates problems for the networks themselves. As a mobile moves, or roams into other networks, its whereabouts need to be monitored for call routing purposes, and its power needs to be monitored to ensure it is transmitting neither too much nor too little. Far more signalling needs to take place in a mobile network than in the fixed network. The radio spectrum is a finite resource and its availability is problematic. The shortage of available spectrum increases its cost when auctions are held, and these costs are inevitably met by the end user. Finally, an extensive infrastructure is required to provide good coverage. This includes the base station transmitters and receivers and all their associated equipment, as well as the transmission links to and from the base station sites.

4.2

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TY2600/v4.1

Mobile Cellular Networks

Databases Mobility

User Verification

Equipment Verification

Core Network

Radio Access Network

Mobile Network Overview A mobile cellular network comprises three main functional areas: the radio access network, the core network and the databases. The radio access network represents the collection of base stations, or masts, that provide the connection between user mobiles and the network. This part of the network is distributed across a city, a region or, as is common in Europe, a whole country to provide radio coverage. The core network performs many functions, but key among these is the network of interconnected switches, or telephone exchanges. Once a call has reached the core network via the radio access network it must be routed on to its intended destination. This is the main function of the core network. However, it is also responsible for a number of other functions such as billing and the monitoring of system performance. The databases are really part of the core network. There are three main database functions, but all relate to the maintenance of accurate and up-to-date information about users and their equipment. This information is used for billing, security functions, service provision and for keeping track of users’ locations so that incoming calls can be routed to them.

TY2600/v4.1

© Wray Castle Limited

4.3

Introduction to Telecoms

Maximum 35 km

Typically carrying about 60 simultaneous calls.

Mobile

for GSM

Base Station (Mast)

Base Stations and Cells A base station may also be referred to as a mast, a tower or a transmitter. Typically it consists of an equipment cabinet along with the mast and radio aerials. However, there are many ways in which the visual impact of a base station can be reduced. A ‘cell’ is the term used to describe the area of radio operation or coverage from a base station. Many factors influence the size of a cell; some are related to equipment performance, some to the geography of the area and some to the number of potential users in the area. In a mature system it is usually the last factor that is most important. In terms of equipment limitations, the maximum radius for a GSM cell is approximately 35 km. Typically, however, a radius of a few kilometres is likely; in an urban area it could be just a few hundred metres. The capacity of a cell varies a great deal because there are so many configuration options available to an operator. A typical configuration could support about 60 simultaneous calls. It is important to understand that since not all users will be using their phones at the same time, such a cell would be able to support far more than 60 users, and could be more than one thousand.

4.4

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TY2600/v4.1

Mobile Cellular Networks

12 cells x 60 calls = 720 simultaneous calls

Wide Area Coverage and Capacity The geography of an area can have a great impact on radio coverage characteristics and therefore on the size of a cell. However, the need to provide service to a sufficient number of users is usually the most significant driver of cell size and therefore the number of cells required by an operator. Consider an area in which acceptable radio coverage can be provided by 12 cells, as shown. Assuming each cell can carry 60 simultaneous calls, the total area capacity will be 720 simultaneous calls. This may be sufficient when the network is built, but congestion could occur as the number of users increases. When the second-generation Global System for Mobile communications (GSM) networks were first built there were relatively low subscriber numbers. Today, these same networks have to cope with many millions of subscribers. In addition, typical subscribers will use their phones much more today than they would have done in the past. Operators must therefore continue to seek ways to get more capacity out of their networks.

TY2600/v4.1

© Wray Castle Limited

4.5

Introduction to Telecoms

24 cells x 60 calls = 1440 simultaneous calls

Increasing Capacity Although technological improvements in GSM and the arrival of UMTS have helped operators improve capacity, this alone does not solve the problem. The most effective strategy is to build a larger number of smaller cells. In this example the size of the existing cells is reduced and then 12 new cells are added to fill in the coverage gaps. There are now 24 cells in the area, each still supporting 60 simultaneous calls. Thus the capacity in the area is doubled. As well as providing more capacity, other benefits arise from this strategy:      

reduced base station transmit power reduced mobile phone transmit power reduced visual impact more consistent and reliable coverage more competitive pricing more advanced services

When cells are smaller, the mobile phones and the base stations will tend to be closer together, therefore less transmit power is required. This significantly extends battery life and further reduces any possible health risks. Base stations can also be physically smaller, making it much easier to reduce their visual impact. The service level to the user is improved in terms of reliability of coverage and competitive pricing. The extra capacity not only enables more users to be supported, but also makes it easier to introduce advanced services.

4.6

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TY2600/v4.1

Mobile Cellular Networks

Cell A

Example Realistic Cell Diagrammatically, cells are often shown as being circular or hexagonal in shape, but real cells are much more complex. Sophisticated software is used to predict the coverage of cells so that operators can plan their cell locations accurately. The diagram shows a coverage plot of a cell located near the seashore in the town of St. Helier on Jersey. The area shown in the aerial photograph is about 2.5 km by 1.5 km. The coverage is shown in colours that are used to give a coarse indication of signal level. Notice how irregular the shape is and how greatly the radio coverage is affected by the terrain and by buildings. If this cell were to be moved, perhaps only 50 m, the pattern would be quite different. This is why operators need to choose preferred cell locations very carefully.

1

A colour version of this diagram is included on the colour CD accompanying the notes.

TY2600/v4.1

© Wray Castle Limited

4.7

Introduction to Telecoms

Mobile Equipment (ME)

Universal Subscriber Identity Module (USIM)

Mobile Station (MS) or User Equipment (UE)

The Mobile Phone The mobile phone, known as the Mobile Station (MS) for GSM and User Equipment (UE) for UMTS, consists of two elements, each with its own functionality. These are the Mobile Equipment (ME), which incorporates hardware and software functions to allow it to operate over the air interface, and the Universal Subscriber Identity Module (USIM) card. Neither on its own offers the user much in the way of a useful phone system, but bring them together and they operate as one to provide a a full range of telecommunication and bearer services, with supplementary services as appropriate. In the case of many phones, features such as a digital camera, music playing capability, portable computer capabilities, satellite navigation or game playing capabilities are included.

4.8

© Wray Castle Limited

TY2600/v4.1

Mobile Cellular Networks

Temporary Mobile Subscriber Identities (TMSI) International Mobile Subscriber Identity (IMSI)

Phone Book

Cipher Keys Forbidden networks

Last dialled number list Authentication Key (Ki)

Location Area Identity (LAI)

List of carriers for cell selection

The USIM The USIM performs vital tasks in providing the user with access to the network. Possibly the most important is authentication, the process of validating the subscriber prior to use of the network. Authentication is done by means of a cryptographic challenge response mechanism. For security reasons, this procedure is carried out entirely on the USIM. Other tasks performed by the USIM mainly involve assisting the ME in its operation. For example, it stores network parameters that the equipment refers to during the initial cell selection process when the mobile is turned on. The USIM card is removable and stores such details as:        

phone book International Mobile Subscriber Identity (IMSI) Temporary Mobile Subscriber Identities (TMSI) Cipher Keys Authentication Key (Ki) Location Area Identity (LAI) list of carriers for cell selection forbidden networks

TY2600/v4.1

© Wray Castle Limited

4.9

Introduction to Telecoms 1

Phone switched on and scan

2

Access Request

3

Access Grant

4

Location update request

5

Retrieve subscriber data

6

Central Database (HLR)

Mobile changes location area Location update request and subscriber data retrieval

11

Delete subscriber data

10

5 11

Authenticate

Local Database (VLR)

Location area 1

9 10

Local Database (VLR)

Location area 2 6

Location area 3 8

4

10

2

3

7

1

9 7

Mobile changes location area

8

Location update and authentication

Network Registration Before a mobile can be used to make a call on a network, it must first have registered on that network. Among other things this process will involve a security check of the user’s identity and, potentially, an independent check of the validity of the user’s mobile phone. When the mobile is switched on it will hunt for a usable network and then for the best cell on that network. Once the mobile has found the best cell it will access and request registration. In most cases the subscriber will be authenticated at this point and issued with a TMSI. The network is divided into geographical regions known as location areas. Typically each location area will contain a large number of cells and several location areas will be managed by a single core network node. Once a mobile has registered it drops back into idle mode. The network’s databases maintain information limited only to the location area in which the mobile is located. As the mobile moves and detects that it has changed location area it will perform a location update procedure to keep the network’s databases up to date about its current location.

4.10

© Wray Castle Limited

TY2600/v4.1

Mobile Cellular Networks 1

Access Request

2

Access Grant

3

Request for a call

4

Verify user

5

User verified

PSTN/ISDN

7

Databases

9 8 9

Mobile Network

5 4

6

8 2 9

1

6

Request for call passed on

7

Incoming call

8

Ringing

9

Answer

3

Making a Call from a Mobile The user dials the called number and presses the send button. The phone’s first task is to request a radio channel from the access network. Once this channel is established the phone requests service from the core network. This process may involve a further security check on the user’s identity and again a potential check on the validity of the user’s phone. Additionally, encryption may be used from this point on in the call to prevent eavesdropping. Once the network is satisfied with the user’s validity the phone will indicate the required call details. These include the called and calling party numbers as well as a description of the call type, i.e. voice or data. The mobile network will use the called party number to determine how to route the call onward to its destination. In this example the call is to a fixed phone, so it routes it out to the fixed network. This network will in turn use the called party number to route the call to its ultimate destination.

TY2600/v4.1

© Wray Castle Limited

4.11

Introduction to Telecoms 1

Call Request

2

Call Request

3

Request for routing information

4

Routing information

5

Incoming call

PSTN/ISDN

Databases 2

Mobile Network

1

3

4

5 7

6 8

7

7 7

6

Page Request

7

Paging

8

Access Request/Grant

Call established after security check, ringing and answer.

Making a Call to a Mobile The calling party dials the number of the called party’s mobile phone. This number identifies the network to which the called party belongs but does not identify the current geographical location of the user. Once the call arrives at the user’s network their location must be separately determined. All mobile networks are divided into geographical regions known as location areas. The size of these areas varies but they usually consist of quite a large number of cells. While a mobile phone is switched on and in standby mode it monitors its location area. It will then keep the network informed about its current location area. The network stores this information in its databases. Once the call has arrived at the user’s network the user’s current location area will be read from the database. This is then used to route the call to the nearest switch. However, the network still doesn’t know which cell the phone is in. The solution is to page the phone in all the cells in its current location area. The phone will be listening so that when it hears the page containing its identity it will respond. Once it has responded, a radio channel can be allocated, security checks can be carried out, encryption can be started and the call can be routed through the correct cell to the called party’s phone.

4.12

© Wray Castle Limited

TY2600/v4.1

Mobile Cellular Networks 1

Call Request

2

Call Request

3

Request for routing information

4

Routing information

5

Incoming call

PSTN/ISDN

Databases 2

Mobile Network

4

1

3

5

Visited Mobile Network

Making a Call to a Roaming Mobile In order to call a mobile user when they are travelling it is not necessary to know where they are. The user’s normal number is dialled and the call will initially be routed to the user’s home network. The network will refer to the database that contains information about the actual whereabouts of the user. This information is used to route the call onward, and usually internationally, to the visited mobile network. This network will in turn page the mobile and complete the call. All the relevant security checks will still be carried out in this scenario. Since the calling party is not aware that they are making an international call they can only be billed at the standard rate. Nevertheless, an international call has been made and must be paid for. The usual solution is to bill the called party for the international leg that is required to complete the call.

TY2600/v4.1

© Wray Castle Limited

4.13

Introduction to Telecoms

Maintaining a Call on the Move When users make or receive calls it is often the case that they are moving. Since cells can be small and calls may go on for several minutes, a process that can cope with users moving from one cell to another is required. This will maintain calls seamlessly as users cross cell boundaries by swapping radio channels from one cell to another. This is known as handover. Once the call is established, the phone begins a monitoring process. It sends regular measurement reports to the network about the quality of its current connection and the signal level observed from specified neighbouring cells. The network uses this information to assess which cell is best to handle the call at any given time.

4.14

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Mobile Cellular Networks

The Handover As a mobile in call moves through the system it moves from one cell to another. The network detects that this is occurring when the measurement report shows that one of the measured neighbour cells is giving a better signal level than the current serving cell. When this happens a handover is triggered. The system allocates a new radio channel on the new cell. On a command from the network the phone retunes to the new channel. The call then continues on the new cell. This process happens very fast and is usually imperceptible to the user. As can be imagined, the handover process is potentially very complex; many measurements and calculations have to be carried out very quickly. At any one time there are thousands of cells being accessed by tens of thousands of users, some moving fast and some moving slowly. Each case is slightly different requiring slightly different treatment, so it is not surprising that occasionally handovers fail and calls are dropped. Operators work continuously analyzing the performance of their networks and fine-tuning the mechanisms that operate this process.

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4.15

Introduction to Telecoms

Other PLMN (GSM)

PSTN/ ISDN

GSM (CS) elements

Network Switching System

RAN

GPRS (PS) elements

VAS Provider

Other PLMN (GPRS)

Intranet

Internet

GSM Network Overview The GSM network can be considered as comprising a number of distinct areas: the Mobile Station (MS), the Radio Access Network (RAN) and the Network Switching System (NSS). The NSS consists of the Circuit-Switched (CS) and Packet-Switched (PS) domains. As shown, the CS domain connects to other Public Land Mobile Networks (PLMNs) for circuit-switched services such as speech, and to the Public Switched Telephone Network (PSTN) and Integrated Services Digital Network (ISDN). The PS domain connects to Internet Protocol (IP) networks such as the Internet and intranets, the packet-switched elements of other PLMNs, and VAS providers.

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Mobile Cellular Networks RAN

NSS Databases AuC

BSS N

HLR

EIR

VLR

BSS 2

E1 MSC E1

BSS 1

BTS

PSTN/ISDN E1

E1 E1

B S C

E1 GMSC NSS CS Domain

Other PLMN

RAN provides signalling and traffic channel connections between MS and MSC

The GSM Network An operator’s RAN provides signalling and traffic connections between the MS and the Mobile-services Switching Centre (MSC). The RAN consists of one or more Base Station Systems (BSS). A typical BSS consists of one or more Base Transceiver Stations (BTS) connected using E1 bearers to a Base Station Controller (BSC). The BTS provides the air interface radio resource into the RAN. The BSC connects to a single MSC via a number of E1 bearers. The BSC dynamically schedules air interface resources to MSs and connects them to an MSC. The BSC also manages handovers procedures. The MSC’s main function is switching: connecting mobile subscribers to other subscribers, either fixed or mobile. Like ISDN exchanges the MSC switches at 64 kbit/s and is capable of simultaneously connecting many thousands of circuits. The MSC may have a large number of BSCs connected to it. There is, therefore, the potential for a very large number of subscribers to be within the MSC’s service area. Operators nominate some or all of their MSC as Gateway MSCs (GMSC). GMSCs provide interworking functions with other circuit-switched networks, i.e. the ISDN/PSTN or other PLMNs. Connections to other networks are implemented using E1 bearers. Subscribers are managed by a number of databases. The Home Location Register (HLR) and Visitor Location Register (VLR) maintain a record of a subscriber’s location together with the types of service they are allowed to access within a particular service area. The AuC generates parameters allowing the HLR/VLR to authenticate a user trying to access the network and to cipher data transmitted between the MS and the BTS. The Equipment Identification Register (EIR) enables the validity of the Mobile Equipment to be checked. For example, the EIR may know that a mobile is stolen, in which case the operator may choose to bar access to the network.

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4.17

Introduction to Telecoms NSS Database AuC

HLR

EIR

SGSN Internet/ Intranet

IP Backbone

Radio Access Network P C U

GGSN

E1

Third Party Content Provider

SGSN

QoS Parameters:

IP Datagram

Throughput

IP Bearer

Server

Delay

DIP SIP Data

Reliability Priority

DIP – Destination IP Address SIP – Source IP Address

IP Bearer + QoS – PDP context

GPRS System Architecture The addition of GPRS has required significant modifications to the GSM network architecture to enable it to handle both packet- and circuit-switched connections. Three new entities have been added: the Gateway GPRS Support Node (GGSN), Serving GPRS Support Node (SGSN), and Packet Control Unit (PCU). These elements are connected with each other and with the GSM network elements. The SGSN and GGSN are connected via the operator’s backbone IP network. The SGSN is connected to the PCU using Frame Relay over an E1 physical interface. The GSM RAN provides ‘connections’ from the MS to the SGSN for both signalling and traffic. GPRS shares some network resources with GSM, including the HLR, EIR and AuC. The HLR has been modified to cater for GPRS subscription data and in cooperation with the SGSN keeps track of a mobile’s location for GPRS services. The PCU allocates air interface resources for GPRS operation like the BSC does for CS GSM resources. Since the air interface is a shared resource, the PCU is generally co-located with the BCS to coordinate air interface resource allocation more efficiently . GPRS provides a packet-switched bearer service between the MS and an external packet network accessed via a GGSN. For example, GPRS may be used to provide an IP bearer service to the public Internet or to a private IP network, intranet, or third-party content provider. To achieve this the MS initially attaches itself to an SGSN, essentially a similar process to location updates. The MS then requests an IP service via the SGSN indicating which network it wishes to reach and what Quality of Service (QoS) it requires. The SGSN forwards a request to the appropriate GGSN in order to establish the bearer. The GGSN allocates the appropriate QoS and an IP address for the MS, following which IP datagrams may be transferred between the mobile and external IP-based devices. The IP bearer together with its QoS definition is referred to as a Packet Data Protocol (PDP) context. The PDP context is managed by the SGSN and GGSN; the PCU measures the QoS delivered to the end user against the negotiated QoS for the PDP context.

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Mobile Cellular Networks

SGSN Basic Functions: serves attached users mobility management, e.g. authentication

SGSN

ciphering of data to/from the MS PDP context management data packet counting (for charging)

Internet/ Intranet GGSN

BSS 1

BTS

MS

SGSN E1

E1

B S C

IP Backbone

E1

Third Party Content Provider

P C U

E1

GGSN Basic Functions: can act as a firewall

PCU Functions:

can allocate MS IP address

radio channel allocation

route PDUs to SGSN

QoS measurements

data packet counting (for charging)

GPRS Network Element Basic Functions The PCU provides radio access control. It allocates radio channels for data transfer, ensures packets are the correct size for transmission over the radio interface, and makes QoS measurements in respect of the radio link with the user’s mobile. Like an MSC, an SGSN is responsible for a service area containing a number of mobiles. Within this service area the main functions of the SGSN include the authorization and authentication of mobiles; ciphering of packets across the air interface; the routing of data packets to and from mobiles; and location management, noting the location of mobiles new to the service area and tracking their subsequent position within the service area. The SGSN also has charging functionality. It gathers data relating to a subscriber’s use of the radio network. The GGSN performs the functions necessary to allow mobiles to communicate with external networks. For incoming calls it contains routing tables so that incoming packets of data can be routed to the SGSN that is supporting the destination mobile. In addition, the GGSN can allocate IP addresses to the served mobile and also act as a firewall to prevent unwanted access. In respect of charging, the GGSN is responsible for gathering data relating to packets transmitted to and received from external networks.

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4.19

Introduction to Telecoms ಫ6WRUHDQG )RUZDUGಬ

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SMS Architecture All Short Message Service (SMS) messages, whether Mobile-Originated (MO) or Mobile-Terminated (MT), must pass through a Short Message Service Centre (SMSC). This has the effect of splitting the delivery of the message into two point-to-point procedures. GSM does not specify the functionality of the SMSC or the transport protocols that connect it to the GSM network. It simply identifies the information elements that must be passed between the mobile station and the SMSC. An SMS gateway function is used to connect the SMSC to the network. For MT messages, this gateway is similar in function to a GMSC; for MO messages, the gateway provides the interworking between GSM and the SMSC, which is still essentially a gateway process. It is important to note that only one SMSC is involved in receiving and forwarding short messages to the final recipient. This SMSC will reside in the sender’s network. This is in contrast to Multimedia Messaging Service (MMS), in which more than one service centre is involved, one residing in the sender’s network and another in the receiver’s network.

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Mobile Cellular Networks Examples Uplink

Downlink

ARFCN

890.2 MHz

935.2 MHz

1

890.4 MHz

935.4 MHz

2

Primary Band – GSM900 MHz Uplink band 890–915 MHz Downlink band

935–960 MHz

200 KHz channel spacing

Bidirectional RF channel

125 channels

Extended Band – E-GSM900 MHz Uplink band

880–890 MHz

Bidirectional

Downlink band

925–935 MHz

RF channel

50 channels

GSM1800 MHz Band Uplink band

1710–1785 MHz

Bidirectional

Downlink band

1805–1880 MHz

RF channel

375 channels

GSM1900 MHz Band Uplink band

1850–1910 MHz

Bidirectional

Downlink band

1930–1990 MHz

RF channel

300 channels

GSM Radio Frequencies The frequency bands allocated for GSM operation are shown above. The original Primary GSM (P-GSM) spectrum allocation was agreed in 1979. This consists of two subbands 25 MHz wide. With duplex spacing of 45 MHz and frequency spacing of 200 kHz capable of supporting 125 carriers with the inclusion of guard bands, this reduces to a total of 124 carriers. In accordance with the GSM Memorandum of Understanding (MoU), this spectrum in the UK is split between at least two operators. In response to a perceived future demand for more capacity, the Primary GSM (P-GSM) spectrum was extended to form Extended GSM (E-GSM). This represents an extension of the lower end of the two sub-blocks by 10 MHz, giving a further 50 carriers. Two further GSM blocks of 4 MHz have been included at the lower end of the E-GSM allocation for GSM Railway (GSM-R) applications. The above bands are collectively known as GSM900. At a later stage in GSM development the technology was modified to meet the need for further networks. This involved changes to the radio interface, which moved spectrum allocation up to around 1.8 GHz. More spectrum is available in this frequency range, for two sub-blocks of 75 MHz with a duplex spacing of 95 MHz, giving a total of 375 carriers, 374 including guard bands. As with GSM900, this spectrum is divided between two or more operators. GSM1900, often known as Personal Communication Service (PCS) 1900, refers to the digital cellular phone technology used in the USA. Spectrum is available in this frequency range, for two sub-blocks of 60 MHz with a duplex spacing of 80 MHz, giving a total of 300 carriers, 299 including guard bands. The Federal Communications Commission’s (FCC) licensing arrangements for this spectrum are complex splitting the bands between 51 Major Trading Areas (MTA) and 493 Basic Trading Areas (BTA). An MTA is broadly equivalent in size to a state, while a BTA approximates to a large city.

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4.21

Introduction to Telecoms a) FDMA – Frequency Division Multiple Access f 1

MS1

2

MS2

3

MS3

4

MS4

t

b) TDMA – Time Division Multiple Access f

MS3

MS8

1

MS4 MS1

t

MS2

Multiple Access Techniques GSM is a telephone system. Therefore users need both to talk and to be heard. The spectrum allocation consists of two bands: the lower band is used by the mobiles to transmit to the network (uplink), and the upper band is used by the network to transmit to the mobiles (downlink). This use of separate frequencies in the uplink and downlink directions is a full duplex system. GSM and GPRS use a combination of two techniques to allow a number of users to access the system at a time. These techniques are Frequency Division Multiple Access (FDMA) and Time Division Multiple Access (TDMA). A mobile is allocated a channel number, which refers to a pair of frequencies, one from the lower band and the other from the upper band. Each band contains carriers, spaced by 200 kHz. It is the allocation of a frequency from within the whole spectrum that provides the FDMA part of the GSM radio interface. When the network indicates a channel for the mobile to use, it not only indicates what frequencies to use, but also which timeslot. This is possible as each of the frequencies is further divided by time. Hence the concept of TDMA as well as FDMA is introduced to the GSM radio interface. The time structure on which the GSM air interface is based is a frame containing eight timeslots over a period of 4.62 ms.

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Mobile Cellular Networks

Time

Doma

in

20

2 3 0 1 7 6 4 5 s 3 m 2 2 4.6 0 1 6 7 5 nel 3 4 Chan 1 2 l a o i c 0 i d s Ra Phy 1 2 Ch. 1 7 0 6 4 5 2 3 1 7 0 5 6 4 2 3 0 1 Radio Ch. 2 5 6 3 4 2 0 1 6 7 5 3 4 1 2 0 o i Rad Ch. 3 0k

qu

Hz

Fre

en

cy

20

Do

0k

in

Hz

ma

577 µ

s

GSM Physical Channels In GSM and GPRS, each carrier, or radio channel, is divided in the time domain to produce eight timeslots, each known as a ‘physical channel’. It is the physical channel that carries a logical channel. The P-GSM allocation supports 992 (8 x 124) physical channels. The 4.62 ms frame structure repeats, giving each mobile an opportunity to transmit and receive information for 577 µs every 4.62 ms.

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4.23

Introduction to Telecoms

0 1

Dow

nlink 2 3 4 5

6 7

Ch.n 0 1

Ch.n + duplex separation

2 3 4 5 6 7 Uplin k

Physical Channel MS

Bit Rate Per Timeslot pre-EDGE Voice

Up to 13 kbit/s

CS data

Up to 14.4 kbit/s

GPRS

Up to 21.4 kbit/s

GSM Physical Channel Offset and Data Rates The physical channel is the channel in which an MS transmits and receives information. It consists of one timeslot on a downlink carrier, and the corresponding timeslot on the uplink carrier. This full duplex method of operation is known as Frequency Division Duplex (FDD). The example shows timeslot 2 being used. The MS in this case will only transmit and receive in timeslot 2. Note the three-timeslot offset in uplink and downlink. This is so the MS is not transmitting and receiving at the same time, allowing for simpler and cheaper equipment. The diagram also indicates the maximum, pre-EDGE, bit rate that may be supported in each timeslot for GSM/GPRS services. Enhanced Data rates for Global Evolution (EDGE) is a technique used for increasing bit rates across the air interface.

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Mobile Cellular Networks

Control 8 x Dedicated Signalling Channels C0

0

C1 0

C2 0

Traffic Channels 1

2

3 4 5 6 BCC H Ca rrier 1 2 3 4 5 6 Traff ic Ch anne 1 2 ls 3 4 5 5 6 Traffic Chan nels

7

7

7

GSM Cell Structure Each BTS may transmit and receive a number of carriers, each of which may carry up to eight simultaneous phone calls on its eight timeslots. However, not all timeslots are available for traffic as at least one of them (often several) is reserved exclusively for control channels. These control channels are normally transmitted on one of the cell’s carriers, known as the Broadcast Control Channel (BCCH). These channels provide facilities to the mobile phone which are necessary for it to establish telephone calls, such as informing the MS of an incoming call (paging), allowing the MS to contact the BTS requesting service, providing information, such as which channels are available and allowing the BTS and the MS to exchange information necessary to set up a call (e.g. phone number). Every GSM cell has one of its carriers designated as the BCCH carrier. Whereas all other carriers have eight timeslots available for telephone calls (although some may be allocated for GPRS use), the BCCH carrier has a reduced number (typically six) due to the presence of the control channels. If more than three radio channels are used, it is possible to use extended control and signalling channels. The dedicated signalling channels are used for signalling between the BTS and the mobile after the initial communication on the BCCH.

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4.25

Introduction to Telecoms Note: this is without error protection and may be vary between 9.1 kbit/s (CS1) and 21.4 kbit/s (CS4) depending on interference conditions in the radio channel.

MS2 MS1

GMSK Modulation

MS3

Maximum available bit rate per timeslot 21.4 kbit/s*

Mobiles may share the available resources in a single timeslot or multiple slots.

TS0

TS1

TS2

TS3

TS4

TS5

TS6

TS7

Multislot allocation

Maximum theoretical bit rate: 8 timeslot allocation= 171.2 kbit/s

MS2 MS1

MS3

Reality check: Four timeslot allocation is typical with data rates of 40–50 kbit/s.

* Note: due to protocol stack overheads the actual user data rate will be less than 21.4 kbit/s.

GPRS Resource Allocation In GSM, a timeslot – effectively a channel – is allocated to a user for their sole use for the duration of their call. In GPRS, users may share resources within a timeslot or across multiple timeslots. This concept is illustrated in the diagram. Note that in ‘standard GPRS’ the modulation scheme used on the air interface is Gaussian Minimum Shift Keying (GMSK). GPRS provides the ability to vary the bit rate within a timeslot by using four coding schemes, CS1 to CS4. With CS1 the timeslot contains 9.1 kbit/s of data with a high level of forward error correction (suitable for poor radio channel performance) while CS4 has 21.4 kbit/s data with no error correction (suitable for excellent radio channel condition). CS4 is rarely used. This is because there is no error protection provided with this coding scheme and the system has no way of a dynamically and quickly adapting between coding schemes in order to combat changes in radio channel conditions. Operators therefore err on the side of caution and use the lower-level coding schemes. The maximum theoretical bit rate, 171.2 kbit/s, is achieved by allocating all eight timeslots. However, in typical operation four timeslots may be allocated with a user rate of between 40 and 50 kbit/s. It should be noted that as in GSM, GPRS resources are finite. If many subscribers are using the network, fewer timeslots can be allocated to each subscriber, resulting in slower data rates. Highest data rates are likely to be achieved when the network is least busy.

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Mobile Cellular Networks Note: this is without error protection and may be vary between 8.8 kbit/s (MCS-1) and 59.2 kbit/s (MCS-9) depending on interference conditions in the radio channel.

MS2 MS1 8PSK A fast adaptive modulation and coding scheme

MS3

Mobiles may share the available resources in a single timeslot or multiple slots.

TS0

TS1

TS2

TS3

Maximum available bit rate per timeslot 59.2 kbit/s*

TS4

TS5

TS6

TS7

Multislot allocation

Adaptive abilities: switch between 8PSK and GMSK modulation with 8PSK rapidly switch through MSC-5 to MCS-9 with GMSK rapidly switch through MSC-1 to MCS-4

Maximum theoretical bit rate: eight timeslot allocation= 473.6 kbit/s

MS2 MS1

MS3

Reality check : four timeslot allocation is typical with data rates of 110–120 kbit/s

* Note: due to protocol stack overheads the actual user data rate will be less than 59.2 kbit/s.

EGPRS Resource Allocation EDGE is an enhancement to GSM’s air interface that may be applied to both CS and PS operation. However, in practice it has only been applied to GPRS and is commonly referred to as EGPRS. EGPRS introduces a new coding scheme, Octogonal Phase Shift Keying (8PSK), which provides a three-times increase in the air interface bit rate but is more susceptible to errors in noisy radio channels. Note that EDGE-enhanced devices may switch between GMSK and 8PSK modulation schemes. EDGE also introduces nine new coding schemes, MCS-1 (8.8 kbit/s, high error protection) to MCS-9 (59.2 kbit/s, no error protection). MCS-1 to MSC-4 are used with GMSK while MCS-5 to MCS-9 are used with 8PSK. EDGE is able to rapidly adapt between both modulation schemes and coding schemes in order to combat a noisy radio channel albeit at the expense of higher bit rates. For example, it may be that all eight timeslots are allocated to EGPRS using 8PSK and MCS-9 for the highest bit rate when the radio conditions are favourable. However, as the channel worsens, the coding scheme may be ‘throttled’ down to say MCS-5. If inference increases further then this may cause a change of modulation scheme to GMSK and, say, MCS-2. The maximum theoretical bit rate, 473.6 kbit/s, is achieved by allocating all eight timeslots. However, in typical operation four timeslots may be allocated with a user rate of between 100 and 120 kbit/s.

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4.27

Introduction to Telecoms

New air interface = new services and service enhancement Multimedia

Multisession

Simultaneously in a session:

Simultaneous sessions:

Audio Video Data Speech

Video message Voice call Data download

UMTS General Service Aims Integrated Telecommunication System Personal Communication Regardless of Location Differentiation of Operators’ Offerings Narrowband or Broadband Simple to Operate Continuity of Service while Roaming PBX and LAN Emulation

UMTS Services Voice has been the dominant traffic type in 2G technology. Speech will continue to be an important revenue generator for UMTS operators, but it is enhanced by still images and video telephony. Because the UMTS air interface can offer higher data rates than GSM/GPRS, new services can be developed. UMTS is designed to allow for future growth and the introduction of new services, including those that are not yet perceived or developed. It is difficult to predict which services will prove most popular; the phenomenal success of SMS as a communication medium, initially unpredicted by the industry, indicates that take-up of services may not always be as expected. Services in UMTS are not simply multimedia, they may also be multisession: different applications can run simultaneously on a terminal. For example, without losing any of the connections to the network, a subscriber may hold a voice call while downloading data, read a message while engaged in online chat, or move from a gaming environment to speech. The main aim for UMTS is that while introducing new applications to mobile communication, it should carry services and features that are generally associated with fixed networks into the mobile and roaming environments. Overall, UMTS should provide an integrated telecommunication system that can support a wide range of applications. The throughput should be variable to provide a range from narrowband to wideband capability; in developing seamless roaming, the user should experience true personal communication regardless of location. If UMTS is to succeed, then all aspects of service provision need to be considered. This means allowing for the types of services that will relate to probable applications, and considering their ease of use for the subscriber. Although 3GPP has acknowledged many of these considerations, they do not set them out as requirements, but rather as design aims for manufacturers and operators. 3GPP therefore standardize service capabilities, but the teleservices themselves are selected and offered by operators. Services should be accessible in a uniform and easy-to-understand way, which will impact on both service design and UE design. Also, with much attention being paid to the roaming environment, an important objective is that users will experience constant service quality irrespective of their location. 4.28

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Mobile Cellular Networks Core Network (CN) Domain

Access Network (AN) Domain Uu

User Equipment (UE) Domain

Basic UMTS System Architecture The UMTS system architecture¹ is divided into three key domains: the User Equipment (UE) domain, the Access Network (AN) domain and the Core Network (CN) domain. In UMTS, the mobile station is known as the UE. The UE is connected to the Universal Terrestrial Radio Access Network (UTRAN) via the Uu interface. It encompasses the radio equipment, application platform and Man–Machine Interface (MMI) that a subscriber will use to access services provided by the network. There are classes and physical forms of UE, which have different capabilities according to application. The UTRAN is involved with the radio aspects of UMTS, providing the means for the UE to access the CN. The main function of the CN is to provide switching, routing and transit for user traffic. It also contains the databases and network management functions. The CN is divided into the CS and PS domains.

¹

3GPP TS 23.101 General UMTS Architecture

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4.29

Introduction to Telecoms

Core Network

RNS

Node B

Node B

Node B

RNS

Node B

Node B

Node B

The UMTS Access Network The UTRAN facilitates high-bit-rate services over the air interface. It comprises one or more Radio Network Subsystems (RNS), which in turn include one Radio Network Controller (RNC) and a number of attached Node Bs. The RNC is responsible for all of the radio functionality in the UMTS system and controls one or more Node Bs. RNCs are connected via the Iur interface and connect to the core network over the Iu interface. A Node B provides the radio interface into the UTRAN. A single Node B may support one or more cells. It connects to the RNC via the Iub interface.

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Mobile Cellular Networks UTRAN VLR

Circuit-Switched Core Network

RNS

PSTN/ ISDN Node B

EIR

UE

HLR AuC

RNS

Internet/ Intranet Firewall

Other PLMN Packet-Switched Core Network

Firewall

UMTS Core Network (CN) The CS and PS domains share some entities, such as the EIR, HLR and AuC. The CS domain entities have evolved from GSM, while the PS domain entities come from GPRS networks. The interfaces are derived from GSM/GPRS networks, except the lu-PS and lu-CS interfaces, which link the CN to the Radio Access Network (RAN). There is also an Iur interface, which links RNCs together. The Gn interfaces interconnect the GSNs via the PLMN backbone network, which is based on IP. External connections to the Internet and intranets is via the Gi interface, while connections to other PLMNs is via a Border Gateway (BG) and the Gp interface. The Iu-CS interface may not be required if VoIP is introduced. The remaining interfaces, used for signalling only, are the same as those in use in GSM/GPRS networks.

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4.31

Introduction to Telecoms IMS

CSCF

Other IMS or IP Networks

Mw Mm HSS

CSCF

Cx Mg

Gr

MGCF

Gi

PS Domain

Gi

Gn

Mc

Iu

UTRAN

Gi

PSTN/ Legacy

Iu

CS Domain MGW

CS Domain

Nb

MGW

Mc

3G MSC

MSC Server

TDM

Mc Nc

SGW

GMSC Server

HSS

R99 MSC

Ru MSC (Split Architecture)

The Evolution of Core Architecture In many existing UMTS networks, the core can be considered as a circuit-switched network evolved from GSM and a packet-switched network evolved from GPRS. However, this initial architecture will evolve. The UMTS circuit-switched core network still provides circuit-switched connectivity for voice and circuitswitched data, but with alterations to the nodes. The traditional MSC, with its 64 kbit/s group switch, may be replaced with a soft switch. This new node will consist of an MSC server, in essence a Media Gateway Controller (MGC), and a Media Gateway (MG). This approach means minimal impact on the radio access network, as signalling towards the MSC server is unchanged. The traffic will be converted from circuit switched to packet switched by the media gateway. This means that the transport of traffic through the core can make use of the efficiencies of packet switching. Interworking with legacy networks such as the PSTN can be facilitated by using media gateways and signalling gateways. The packet-switched core will have fewer architectural changes, with connectivity still being provided by GPRS. The main additional architecture is defined at Release 5 of the standards. The IP Multimedia Subsystem (IMS) can be accessed via the packet-switched core, providing a variety of services and applications based on a packet-switched bearer. These will include peer-to-peer applications such as VoIP, video and audio streaming and Push-to-talk over Cellular (PoC). The diagram shows the 3GPP UMTS architecture at Release 5. The PS core is unchanged and connects into the IMS. Note, the HLR evolves into the Home Subscriber Server (HSS) at Release 4 onwards.

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Mobile Cellular Networks Sender

Receiver FDMA 1

2

Frequency

In GSM, channels are uniquely identified by frequency and time

3

Time TDMA

1

2

3

1

2

3

1

2

CDMA 1, 2 and 3

3

In CDMA, channels are uniquely identified by codes

Code Division Multiple Access The frequency spectrum can be divided in two ways. There are two finite resources, frequency and time. Division by frequency, so that each pair of communicators is allocated part of the spectrum for all of the time, results in Frequency Division Multiple Access (FDMA). Division by time, so that each pair of communicators is allocated all (or at least a large part) of the spectrum for part of the time results in Time Division Multiple Access (TDMA). Whilst these systems can be very complex, the principles on which they are based are simple to visualize. In Code Division Multiple Access (CDMA), however, every communicator is allocated all of the spectrum all of the time. It is much harder to visualize how this could result in anything but unacceptable interference. Imagine a room full of people who are all simultaneously in conversation, and all speaking at about the same level. Each listener’s partner is at least matched in volume by the ambient background noise level in the room. If they are all speaking a different language, then it is reasonable to assume that despite the noise, communication will be possible. This is because of the uniqueness of each individual’s information compared to the noise. In CDMA the uniqueness of a channel is identified by a code or combination of codes.

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4.33

Introduction to Telecoms Frequency MHz 1885

1920

1980

2010 2025

2170

2110

2200

1900 DECT TDD

FDD

SAT

Uplink

TDD

FDD

Duplex Spacing 190 MHz

SAT

Downlink

Ch. 1

Ch. 2 Nominal 5 MHz channel spacing 200 kHz Raster

FDD

TDD

WCDMA 3.84 Mcps

WB-TDMA/CDMA 3.84 Mcps

UMTS Frequencies The physical layer within UMTS maps transport channels from layer 2 into a series of Uplink (UL) and Downlink (DL) physical channels. Both FDD and Time Division Duplex (TDD) modes are defined, FDD operation using a physical layer based upon Wideband CDMA (WCDMA), and TDD using a hybrid Time Division CDMA (TD-CDMA) technology. A physical channel in FDD mode has a chip rate of 3.84 Mcps and is characterized by the code-set used, the frequency and, in the UL direction, the relative phase in terms of I and Q (modulation parameters). In TDD mode, the physical channel is also characterized by the timeslot used. The CDMA part of UMTS is used in the paired radio spectrum, i.e. FDD. As such, it may be referred to as FDD mode 1. Mode 1 distinguishes it from the other technologies that are part of the 3GPP family. It may also be referred to as DS mode. The TD-CDMA part of UMTS is used in the unpaired radio spectrum, i.e. TDD. It is simply referred to as TDD mode. TDD mode has found limited market applications, typically for fixed radio access and mobile TV trials. Carrier frequencies are designated by a UMTS Absolute Radio Frequency Channel Number (UARFCN). The general formula relating frequency to UARFCN is: UARFCN = 5 x (frequency in MHz) This is applicable in any radio band and for both FDD and TDD modes. Note that for FDD mode a duplex channel will be denoted by a pair of UARFCNs.

4.34

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TY2600/v4.1

Mobile Cellular Networks Frequency 1 Scrambling Code Cell 2

UE3

Cell 2

Scrambling codes uniquely identify one cell from another.

Channel code 10

UE4 Channel code 3

Frequency 1 Scrambling Code Cell 1

UE1

Cell 1

Channel code 1

UE2 Channel code 3

Channelization codes: all cells use the same channelization codes identify individual channels allocated to a UE code length determines channel rate: shorter code – higher bit rate, less users per cell longer code – lower bit rate, more users per cell

UE2 will not listen to this channelization code as it is covered by Cell 2’s scrambling code.

UMTS Modulation

QPSK

Bit rates up to

2 Mbit/s UL/DL

Realistic

384 kbit/s

CDMA Code Allocation In a CDMA system codes are used to differentiate between channels. Two types of code are used, channelization codes and scrambling codes. Physical channels are formed through the application of CDMA codes to the channel data stream. These codes are referred to as channelization codes. The codes permit a receiver to isolate a particular incoming physical channel data stream. In the downlink direction the same channelization codes are used in all cells. Since all cells may use the same frequency and all cells use the same channelization codes it is possible for two neighbouring cells to allocate the same channelization code on the same frequency at the same time. If this is the case then mutual interference would occur on the channel. To prevent this, each cell is allocated its own unique cell scrambling code. Thus in the diagram UE 1 will read the channelization code 3 covered by the scrambling code for cell 1 and not the other the channelization code 3 covered by the scrambling code for cell 2.

TY2600/v4.1

© Wray Castle Limited

4.35

Introduction to Telecoms

HSPA techniques: higher level modulation fast scheduling fast retransmission

HSPA+ techniques: higher level modulation MIMO (Multiple Input Multiple Output) advanced antenna techniques dual cell

HSPA

HSPA+

HSDPA (R5)

HSUPA (R6)

HSDPA+ (R7/R8)

HSUPA+ (R7/8)

Modulation

16QAM

QPSK

64QAM

16QAM

Max bit rate

10 Mbit/s

6 Mbit/s

42 Mbit/s

12 Mbit/s

Typical bit rate

7.2 Mbit/s

1 Mbit/s

*

*

* No statistics available to date

Modulation schemes shown are those needed to achieve the maximum bit rate the system can support.

HSPA and HSPA+ A number of enhancements have been made to increase the data rates available across the air interface in UMTS systems. High Speed Packet Access (HSPA) has two sub-categories: High Speed Downlink Packet Access (HSDPA), which was introduced in 3GPP R5, and High Speed Uplink Packet Access (HSUPA), which was introduced at R6. These include a number of techniques to increase available bit rates, including higher-level modulation schemes, fast scheduling of available cell resources, and fast retransmission. Fast scheduling enables the Node B to allocate unused cell resources rapidly (once every 2 ms). Fast retransmission is used to optimize the error protection overhead. HSPA+ was introduced in 3GPP R7/8 with techniques including higher-level modulation schemes, Multiple Input Multiple Output (MIMO) advanced antenna techniques, which allow data to be delivered over multiple paths, at the same frequency, simultaneously; and dual-cell operation, which allows data to be delivered using two frequencies simultaneously (in CDMA a frequency is a cell). Currently, either MIMO or dual-cell may be used, but not both at the same time, although this may be the case in future recommendations.

4.36

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Mobile Cellular Networks

3.5G CDMA2000 1xEV-DO (Revision B) >9 Mbit/s 3.5G CDMA2000 1xEV-DV (Revision C/D)

3.5G CDMA2000 1xEV-DO (Revision 0/A) >3 Mbit/s 3G 2G cdmaOne (IS-95-B)

CDMA2000 1x (Revision 0)

>5 Mbit/s 3G CDMA2000 3x (Revision 0) >2 Mbit/s

>300 kbit/s

>100 kbit/s

CDMA2000™ Development Phases There are a number of variations within the main technology description for CDMA2000 that offer different service level capabilities. To a degree, these can be seen as an evolutionary path for an operator looking to introduce new capabilities with a phased approach. The first step is to upgrade cdmaOne™ radio access equipment to CDMA2000 1x. This is achieved with a relatively simple upgrade to an existing IS-95-B network. The chip rate remains unchanged at 1.2288 Mcps, but a number of changes, including more sophisticated power control and modified code usage, contribute to greater capacity from the same spectrum. The higher layers are modified to allow more flexible service provision, including efficient packet data operation although forward and reverse data rates are typically limited to 153.6 kbit/s (307.2 kbit/s is theoretically possible). In the early stages of standards development for CDMA2000, an upgrade was defined that involved the concatenation of three radio carriers, hence the term 3x. This modification was intended to provide true 3G data capability, but development of this option has now ceased. 1xEV now represents the mainstay of CDMA2000 capability development. There are two variants of 1xEV, the first being 1xEV-Data Optimized (1xEV-DO), and this variant is now well established in a high proportion of CDMA2000 networks. Development work on the second variant, 1xEV Data and Voice (1xEV-DV), has now ceased.

TY2600/v4.1

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4.37

Introduction to Telecoms

4.38

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TY2600/v4.1

Introduction to Telecoms

SECTION 5

IP PACKET NETWORKS

© Wray Castle Limited

I

Introduction to Telecoms

CONTENTS Why Move to Packet-Based Technologies? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.1 Typical Structure of a Packet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.2 ISO Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.3 Functions of the Layers

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.4

TCP/IP Protocol Stack . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.5 Connectionless and Connection-Oriented Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.6 Layer 2 and Layer 3 Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.7 Connecting Hosts, Clients and Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.8 LANs, MANs and WANs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.9 Ethernet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.10 MAC Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.11 Ethernet Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.12 Ethernet Physical Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.13 The Internet Protocol (IP) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.14 IP Addressing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.15 IP Datagram Forwarding

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.16

Allocation of IP Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.17 Transporting IP over Different Networks

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.18

IP over an Ethernet LAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.19 Routing IP between two Ethernet Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.20 Internet Service Providers (ISPs) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.21 The Internet Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.22 Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.23 Web Browsing

II

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.24

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IP Packet Networks

CONTENTS E-mail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.25 Real-Time Internet Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.26 IP Network Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.27 Enterprise Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.28 Multi-Protocol Label Switching (MPLS) The MPLS Architecture

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.29

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.30

Label Processing in an LSR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.31 MPLS Label Stacks

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.32

The MPLS Control Planes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.33 MPLS VPNs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.34 MPLS Traffic Engineering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .5.35

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III

Introduction to Telecoms

IV

© Wray Castle Limited

IP Packet Networks

OBJECTIVES At the end of this section you will be able to:



identify why packet switching techniques make efficient use of network resources



describe the principles of connectionless and connection-oriented packet networks



identify the differences between LANs, MANs and WANs



describe the functions of a client and server



explain the basic format of an Ethernet MAC address



describe the facilities provided by an Ethernet switch



identify some common Ethernet physical interfaces



explain the purpose of IP



outline the operating principles of a router



describe how IP and MAC addresses are associated in a LAN



identify the main elements in an MPLS network



explain the functions of a Label Switched Path (LSP)



outline the functions of traffic engineering

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V

Introduction to Telecoms

VI

© Wray Castle Limited

IP Packet Networks Circuit-Switched Network – inefficient use of network resources (Each user has a dedicated connection through the network)

User information such as voice, video and data

Predictable delay across the network

Packet-Switched Network – efficient use of network resources (Each user has a virtual connection through the network)

Variable delay across the network. Not a significant problem for data but causes difficulty for voice and other real-time applications User information such as voice, video and data placed in packets

Why Move to Packet-Based Technologies? The world of telecommunications is moving towards packet-based networks. Packet switching is the approach used by most computer data networks to deliver data across a local or long-distance connection. Examples of packet based technologies are X.25, Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and Internet Protocol (IP). Packet switching uses the principle of placing data into specially formatted units called packets. These packets are typically routed from the source to the destination using network switches and routers. Each packet contains some form of identity, such as a label or address, that identifies the packet to the network. This identity may, for example, identify the sending and receiving computers and indicate how the packet should be handled during its journey through the network. Packet switching is an alternative to circuit switching, which has been used in telephone networks such as the PSTN. When compared to circuit switching, packet switching provides a more efficient use of overall network resources and bandwidth due to the flexibility in routing the packets over shared links. The packet switch, or router, multiplexes packets from different sources over a single link. Packet switching networks are normally cheaper to build due to the fact that less equipment is required as various types of traffic can share the available bandwidth over a link in a statistical way. Packet networks have traditionally been used for data transfer where the transmission delay was considered a problem. The packet switches and routers have to make decisions on where to route packets and provide queuing functions. For example, if an e-mail is delayed by 10 seconds then it is regarded as an inconvenience, but if voice was transferred over a conventional packet network, delays may cause difficulties with the conversation.

not the not the

For many applications, delays are not long enough to be significant, but for high-performance applications like real-time voice or video, additional data compression and QoS technologies are often used to achieve the required performance levels. Packet networks may have the potential for network security risks due to the use of shared physical links. Protocols and other related elements in packet switching networks must be designed with the appropriate security precautions.

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5.1

Introduction to Telecoms

User Data

Header

Data

Trailer

Address or Identifier and Control Information

User Data

Error Checking Information

Typical Structure of a Packet Everything carried over the Internet is transferred in a series of packets: every web page that is downloaded, every email and every video clip. The computer places the data into packets of a certain size in octets (bytes). Each packet carries information in its header that is used to get it to its destination, such as a label or address, along with information about how the packet has been assembled. Some types of packet contain a trailer that carries out functions such as checking to see if the packet contains errors. If there are errors in a packet, the networks and protocols can treat this situation in a variety of ways. Some protocols are designed to ignore the errors because they expect another protocol to correct them. Some protocols can attempt to correct the errors in real time while others may request the packet to be retransmitted. Each packet is sent to its destination by the best available route. With some protocols, packets for a particular virtual connection may all take the same route through the network, while other protocols may allow the packets for a virtual connection to take different routes across the network. If different routes can be taken, this makes the network more flexible as the network can balance the load across various pieces of equipment on a millisecond-by-millisecond basis. If there is a problem with one piece of equipment in the network while a packet is being transferred, packets can be routed around the problem, ensuring the delivery of the entire message. Depending on the type of network, packets may be referred to by other names such as frame, block, cell, segment and datagram. Packet switching uses virtual circuits. The circuits are known as virtual because they are not electrical circuits where there is a direct electrical connection from end to end. Rather, there is a ‘logical’ connection, or virtual connection, where the user data moves from end to end, but without a direct electrical circuit.

5.2

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TY2600/v4.1

IP Packet Networks Applications communicate using services of seven layers

A

B

Application

Application Application-Centric Layers

Application Layer

Application Layer

Presentation Layer

Presentation Layer

Session Layer

Network-Centric Layers

Transport Layer

Session Layer Transport Layer

Network Layer

Network Layer

Network Layer

Link Layer

Link Layer

Link Layer

Physical Layer

Physical Layer

Physical Layer

each layer provides a service to the layer above each layer is self-contained – modularity high layer data is encapsulated in lower layer PDUs

ISO Model The traditional means of modelling communications protocols is by use of the International Standards Organization (ISO) Seven-Layer Model. Protocol stacks such as Transmission Control Protocol/Internet Protocol (TCP/IP) and Signalling System 7 (SS7) do not align exactly with this model, so it is typically not used in a formal sense to specify protocols and implementations. However, as a reference model against which to understand and explain the role of real-world protocols, it remains very useful. The model can be viewed in various ways. From a functional perspective, each layer provides a set of services to the layer above it in the stack, culminating in the service provided by the application layer protocols to the applications. Likewise, each layer expects to receive a defined service from the layer beneath it in the stack. From an implementation perspective, each module is a self-contained piece of software (or hardware) which makes available a set of primitives (function calls) to the layer above. These together form an Application Programming Interface (API), which allows any third party to write a protocol making use of the services available from the layer. This means that separate vendors can provide different layers in the stack and there should be good interoperability. From a data perspective, each layer encapsulates the user data received from the layer above with control information relevant to its own layer as it passes down through the stack onto the network. The corresponding layer at the receiving side processes that information and removes it before passing the payload up to the next layer in the stack. The ISO model describes the packages of data and control information as Protocol Data Units (PDUs).

TY2600/v4.1

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5.3

Introduction to Telecoms

Encapsulation (Sending node)

De–Encapsulation (Receiving Node)

Application

Application

Application

AH

Application

Interface to application, file transfer

(7)

PH

AH

Application

Syntax Conversion

(6)

SH

PH

AH

Application

Dialogue Establishment and synchronization

(5)

TH

SH

PH

AH

Application

End-to-end data transfer with known characteristics

(4)

NH

TH

SH

PH

AH

Application

Addressing and Routing

(3)

LLH

NH

TH

SH

PH

AH

Application

Framing, error detection and (potentially) retransmissions

(2)

LLH

NH

TH

SH

PH

AH

Application

Physical connections, bit rates, etc.

(1)

Application Layer PDU

Application Layer (7)

Presentation Layer PDU

Presentation Layer (6)

Session Layer PDU

Session Layer (5)

Transport Layer PDU

Transport Layer (4)

Network Layer PDU

Network Layer (3)

Link Layer (L2)

Link Layer PDU

Physical Layer (L1)

PH

Functions of the Layers The application layer in the model provides an interface to user applications and provides basic operations such as file transfer and messaging transfer. Note that this layer is not providing the actual application, but application layer services; it may be viewed as making data exchanges with remote systems behave in the same way as an operating system call to resources on the local machine. The presentation layer is responsible for any syntax conversion necessary between the local and remote system. The analogy of a translation service is often used to explain this. Suppose the local application speaks English and the remote application speaks Spanish; they cannot communicate directly, but if the local machine has an interpreter who can translate between English and French, and the remote system had an interpreter to translate between Spanish and French, then the communication can proceed. The interpreters in this example are acting as a presentation layer. Note the presentation layer need not understand the meaning (or semantics) of the conversation, just the syntax (or grammar and structure of each language). The session layer is responsible for establishing and synchronizing the dialogue between two end points, including whether the dialogue is full duplex, half duplex, etc. It can also provide synchronization points, so that if the dialogue develops a fault, it can be resumed from the most recent ‘known good’ point. The transport layer provides a message exchange between the two end points with known characteristics. A range of transport layers have been defined in the model, ranging from a Minimal Transport Layer (TPO), through to a transport layer with full error and flow control (TP4). These are roughly equivalent to the transport layer functions of User Datagram Protocol (UDP) and TCP respectively in the TCP/IP protocol suite. The network layer is responsible for delivery of session PDUs between the two end points and for the addressing, etc. associated with this. Both connectionless and connection-oriented network level protocols have been defined. The network layer is also responsible for interworking between different network protocols, where this is required. The data link layer provides delivery of network-layer PDUs across a single physical connection and provides framing, error detection and (potentially) retransmission of errored frames. The physical layer protocol defines the electrical/optical and physical characteristics of connections, such as bit rates, contention mechanisms, signal rise times and encoding schemes.

5.4

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TY2600/v4.1

IP Packet Networks

OSI

DoD

TCP/IP

Application Layer Process/ Application

HTTP, FTP, SMTP, TFTP, Telnet

Transport Layer

Host-to-Host

TCP or UDP

Network Layer

Internet Layer

Internet protocol

Presentation Layer Session Layer

Layer 1 and 2 alternatives Link Layer Physical Layer

Network Access

Ethernet

Frame Relay, ATM

Ethernet

NG SDH, OTN

TCP/IP Protocol Stack The Department of Defence (DoD) model is a shortened version of the OSI model containing 4 layers as shown. A wide range of protocols may be used at the Process and Application layer, which corresponds to layers 5-7 in the OSI model. These protocols include the Hyper Text Transfer Protocol (HTTP), File Transfer Protocol (FTP), Trivial File Transfer Protocol (TFTP), Simple Mail Transfer Protocol (SMTP) and Telnet. The Host-to-Host layer maps to the OSI Transport layer and may be used to provide reliable or unreliable transport services between hosts in the TCP/IP model. UDP provides basic connectionless data transfer capability for the application that sits above UDP. It is intentionally very simple and is appropriate for such applications as domain TFTP and network management data. UDP provides basic error control for the application by detecting and rejecting corrupt data. TCP provides a highly reliable connection-oriented transport mechanism. TCP establishes a ‘connection’ between the hosts before data transfer commences and then by sequencing data and providing error protection it creates a reliable connection suited to applications such as web browsing. The Internet layer corresponds to the OSI Network Layer, layer 3, and deals with the logical transmission of packets over the entire network. In the TCP/IP model this layer is implemented using the Internet Protocol (IP). IP addresses are assigned to host and the IP layer handles routing of packets, potentially, across multiple networks. The Network Access layer corresponds to layers 1 and 2 of the OSI model. It oversees hardware addressing and defines protocols for the physical transmission of data including media access rules. In the TCP/IP model Ethernet may be used to implement layer 1 and 2 protocols. However, alternative protocols may be used at layer 1 and layer 2 including ATM or Frame at layer 2 and Next Generation SDH or OTN at layer 1.

TY2600/v4.1

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5.5

Introduction to Telecoms Packet 2 Packet 1

Connection-Oriented Service Packet 1

Connectionless Service (packets may arrive out of sequence)

Connectionless and Connection-Oriented Services When a packet arrives at a switch, the switch reads the packet’s header information and then forwards it on to an appropriate destination based on routing tables within the switch. Once the packet has been sent, the switch has no further dealings with it. With a connectionless service, the switch makes no association between packets that it receives and sends. Every packet is treated individually and packets from a single source may take different routes through the network. Because of this, packets may arrive at their destination out of sequence. With a connectionless service, it is difficult to maintain quality of service, especially in terms of delay, but it is more flexible in the way it uses network resources. Examples of connectionless protocols are IP and Ethernet. With a connection-oriented service, set-up information is provided to the network prior to data transfer to establish a path for the user’s data packets. The network makes an association between all the packets sent to and from a specific user and these will follow the same route across the network. Because all the packets follow the same path, they arrive at the destination in sequence. At the end of the data exchange, information is sent across the network to release the connection. Although with a connection-oriented service users have their own connection through the network, it is a logical connection only, it does not consume physical resources unless it is actively transmitting data. This enables many logical connections to share the same physical connection between packet switching devices. Connection-oriented networks require set-up procedures which establish an end-to-end connection, confirms end user availability and guarantees sequenced packet delivery. Examples of connection-oriented protocols are X.25, Frame Relay, ATM and MPLS.

5.6

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TY2600/v4.1

IP Packet Networks

Data from higher layers

Data to higher layers

L3

L3

L3

L3

L2

L2

L2

L2

L2

L2

L2

L2

L1

L1

L1

L1

L1

L1

L1

L1

Terminal Device

Layer 2 Switch

Layer 3 Switch

Layer 2 Switch

Terminal Device

Physical transmission link (radio, fibre, etc.)

Layer 2 and Layer 3 Switching The layered model could be compared to the postal network. A layer three packet is an envelope, a layer two packet is a sack which contains the envelope, and the layer one packet is a truck that carries the sack along a road. In the postal network, sometimes sacks are switched from one truck to another – this is a layer two switch. Sometimes, the sack is unwrapped and the envelope is switched from one sack to another sack. This is a layer three switch. Data packets are switched across a packet network in a similar way. A layer two switch terminates the layer one packet and only looks at the header of the layer two packet (or frame). It uses this information to decide where to send the packet next. It is not interested in the layer three packet header. A layer three switch on the other hand, terminates the layer one and two packets and looks at the header of the layer three packet. It uses this to decide where to send the layer three packet next. An example of a layer one packet could be a Virtual Container in SDH. A layer two packet could be an Ethernet frame, frame relay frame or an ATM cell. A layer three packet could be an X.25 packet or IP datagram.

TY2600/v4.1

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5.7

Introduction to Telecoms Host

Host

Host

Network

Host

Client asks server for information

Server

Server returns information to client

Client

Host

Connecting Hosts, Clients and Servers In general, all machines connected to the Internet can be categorized as either clients or servers. Client devices are typically computers with network software applications installed that request and receive information over the network from a server. Mobile devices and desktop computers can both function as clients. A server device typically stores files and databases including more complex applications like web sites. Server devices often feature higher-powered central processors, more memory and larger disk drives than clients. The client–server model can be used on the Internet as well as LANs. Examples of client–server systems on the Internet include web browsers and web servers, File Transfer Protocol (FTP) clients and servers, and Domain Name Servers (DNS). Client/server networking grew in popularity many years ago as personal computers became common. A host is a device that is connected to a network; both clients and servers are hosts of a particular network. Every host on the Internet has a unique IP address containing a network part and a host part. The routers on the Internet first send the IP packets to the appropriate network, and then routers in the network find the appropriate end device (host).

5.8

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TY2600/v4.1

IP Packet Networks Typical range 200 metres Local Area Network (LAN) Typical range 15 miles

LAN LAN

Metropolitan Area Network (MAN)

LAN

Maximum range 12,000 miles

LAN LAN

LAN

Wide Area Network (WAN)

LANs, MANs and WANs The various types of computer network designs have been categorized by their range or scale. The computer networking industry traditionally refers to nearly every type of network as some kind of area network. The most common are Local Area Network (LAN), Metropolitan Area Network (MAN) and Wide Area Network (WAN). A LAN connects networking devices over a relatively short distance, normally within a building. An office, school, or home is usually within the range of a single LAN. However, some buildings may contain a few small LANs. In addition to operating over a limited distance, LANs are also normally owned and managed by a single person or organization. Common technologies that are used for LANs are Ethernet and Token Ring. A version of the LAN that uses radio technologies is the Wireless LAN (WLAN) commonly referred to as WiFi. A WAN spans a large geographical distance, so the Internet could be regarded as the largest WAN, spanning the earth. A WAN is usually a geographically-dispersed collection of LANs connected by longdistance transmission networks. A device called a Router (R) connects LANs to a WAN. In IP networking, the router maintains both a LAN address and a WAN address. A WAN differs from a LAN in several important ways. Most WANs are not owned by any one organization but rather exist under collective or distributed ownership and management. WANs tend to use technology like ATM, Frame Relay and X.25 to make connections over the longer distances. Residential customers typically employ one LAN and connect to the Internet (WAN) via an Internet Service Provider (ISP) using a broadband modem and router. The ISP provides a WAN identity to the modem such as an ATM label. All computers on the home LAN can communicate directly with each other but must go through a central gateway, normally a device like a broadband router, to reach the ISP. A MAN is a network spanning a physical area larger than a LAN but smaller than a WAN, such as a city or university campus. A MAN is typically owned and operated by a single entity such as a government body, university or other large organization. LAN and WAN were the original categories of area networks. Others that have gradually emerged over many years of technical network evolution are Storage Area Network (SAN), Campus Area Network (CAN) and Personal Area Network (PAN) e.g. Bluetooth. The dominant technology in LANs is Ethernet. Ethernet is also becoming a competing technology in the MAN and WAN environments. TY2600/v4.1

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5.9

Introduction to Telecoms

Ethernet II (DIX) Sync (2 bits – 11) Preamble – 62 bits (10101....)

Type (2 bytes)

Destination Source Address Address

T

Data Field

FCS

46–1500 bytes

Minimum 64 bytes maximum 1518 bytes

IEEE 802.3 Preamble -7 bytes (10101....)

Destination Source Address Address

SFD (1 byte – 10101011)

L

Length (2 bytes)

Data Field

FCS

46–1500 bytes

Ethernet Ethernet has become the most widely used LAN technology because of its speed, low cost and relative ease of installation. Ethernet is a layer 2 protocol and can be regarded as connectionless in its operation. In 1980 DEC, Intel and Xerox (DIX) issued a DIX Ethernet standard for 10 Mbit/s Ethernet systems. That same year, the Institute of Electrical and Electronics Engineers (IEEE) commissioned a committee to develop open network standards. In 1985, this committee published the portion of the standard pertaining to Ethernet (based on the DIX standard) – IEEE 802.3 Carrier Sense Multiple Access with Collision Detection (CSMA/CD). Even though the IEEE title does not mention the Ethernet, the original term has caught on, and IEEE 802.3 is referred to as the Ethernet standard. (The IEEE standard was called 802 because work on it started in February 1980). The main difference between the Ethernet II (DIX) frame format and IEEE 802.3 format is the field that follows the addresses. The Ethernet II (DIX) format designates this field as ‘Type’, which identifies the protocol type of the data field to enable forwarding to the appropriate higher layer, but the IEEE 802.3 frame assigns this field to indicate the ‘Length’ of the data minus any padding.

5.10

© Wray Castle Limited

TY2600/v4.1

IP Packet Networks

Binary Hex

Sometimes written

0000 1000 0000 0000 0000 1100 1001 1110 0001 0111 1000 1010

0

8

0

08 08

0

0

00 :

00

c

9

0c :

0c

e

1

9e :

Assigned by the IEEE

9e

7

8

17 :

17

a

8a :

8a

Assigned by the manufacturer

MAC Addresses LAN addressing is a layer 2 function and is known as Medium Access Control (MAC) address. A MAC address consists of a 6 byte (48 bit) field which contains a globally unique number. The MAC address is associated with a host device but, more specifically, with an Ethernet interface on that device. MAC addresses are sometimes called physical addresses because they are burned into Read-Only Memory (ROM) in the Ethernet device. This means that for the most part MAC addresses are unique.

TY2600/v4.1

© Wray Castle Limited

5.11

Introduction to Telecoms Hub

Frame from A to E

Frame from A to E Full Duplex

Full Duplex

Port 1

Port 2

Half Duplex

Full Duplex

Port 3

Half Duplex

Port 4

Half Duplex

Frame forwarded to port 4 Filter Database Port 1

A

Port 2

B

Port 3

C

Port 4

D, E, F

Switch

Ethernet Switching A modern Ethernet LAN employs layer 2 switches to connect terminal devices together. Normally, only one device is connected to each interface on the switch, but if a hub is employed then more than one device can be connected. A hub is a layer 1 device that just takes the data and broadcasts it out to every port on the hub. A hub does not look at layer 2 information such as a MAC address – it is just a physical layer repeater. Devices connected via a hub operate in half duplex mode. Logically devices connected via a hub appear to be connected to a common transmission path. Collisions occur when multiple devices attempt to transmit information onto the line at the same time; collisions reduce the performance of the network. A switch isolates devices at the physical layer. If a single device is connected to a switch port then collisions will not occur, network performance throughput is improved and full duplex operation may be used. A switch needs to know the MAC addresses of devices that are accessible via the segments attached to its ports, so it learns these MAC addresses by looking in the Source Address field in the Ethernet frame. It then builds a database of addresses associated with each segment and port. This is called a Filter Database, as it is used to filter out data. When a frame arrives the switch tries to match the destination address MAC with an entry in the filter database; if an entry exists the frame is forwarded on the matching post only. If a switch does not have an entry in the filter database for a specific MAC address, i.e. it is an unknown address, then it floods the frame to all ports except the one the frame was received on. The switch can check the Frame Check Sequence (FCS) for errors in the Ethernet frame. A frame that contains errors is filtered and discarded by the switch and any collisions in a half-duplex port do not reflect across a switch. Frames that are travelling between nodes connected to the same halfduplex segment are also filtered and they are not allowed to cross the switch.

5.12

© Wray Castle Limited

TY2600/v4.1

IP Packet Networks

Ethernet Type

Transmission Rate

Cable Type

Typical Span Distance

Duplex

10BASE-2

10 Mbit/s

Coax (thick or thin)