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Traditional telephone networks were designed to implement a single type of communication service, i.e., the telephone service. Today’s telecommunication networks implement a wide range of communication services. In this section we introduce Markov models of communication services that compete for the bandwidth of a finite-capacity communication link.

Long-term planning of backbone telephone networks has been an important area of application of combinatorial optimization over the last few years. In this chapter, we review polyhedral results for models related to these problems. In particular, we study classical survivability requirements in terms of k-connectivity of the network, then we extend the survivability model to include the notion of bounded rings that limit the length of the rerouting path in case of link failure.

People communicate. One way or another, they exchange some information among themselves all the times. In the past several decades, many electronic technologies have been invented to aid this process of exchanging information in an efficient and creative way. Among these are the creation of fixed telephone networks, the broadcasting of television and radio, the advent of computers, the rise of the Internet, and the emergence of wireless sensation. Originally, these technologies existed and operated independently, serving their very own purposes. Not until recently that these technological wonders have started to converge, and it is a well-known fact that a computer communication network is a result of this convergence.

Modern portable computers and wireless connections over a cellular telephone network have created a new platform for distributed information processing. We present a communication architecture framework which makes it possible to exploit the existing TCP/IP communication architecture but which also takes into account the specific features of wireless links. Our communication architecture is based on the principle of indirect interaction. The mediating interceptor, Mobile-Connection Host, is the bridge between the worlds of wireless and wireline communication. The interceptor also provides enhanced functionality that improves fault-tolerance and performance for applications aware of mobility. Prototypes of the architecture are implemented both for the Unix (Linux) and for the Windows (3.11) platform.

In the last few years great strides have been made in the development of algorithms that design telephone networks. As computational capabilities improve it is possible to produce better results, both from an engineering and an economic standpoint. This paper considers the design issues that cost model developers have addressed successfully. Many of these issues are illustrated by a detailed description of a model developed by FCC staff, known as the Hybrid Cost Proxy Model (HCPM). HCPM is capable of utilizing very precise customer location data. From these data, the model uses clustering algorithms to identify serving areas that satisfy appropriate engineering constraints. Within each serving area, the model uses a modified minimum-cost spanning tree algorithm to connect actual customer locations to a serving area interface. The same tree algorithm connects each interface point to a switch. Within each path, the model performs intensive integer searches to find the cost minimizing, yet engineering-feasible, choice of technology and electronics for that path. The result is a-low cost, feasible network plan that gives an appropriate estimate of the forward looking cost of providing wireline telephone service to a particular area. This estimate should prove particularly useful in the ongoing debate about the size and make-up of the Universal Service Fund, and for other regulatory purposes such as the pricing of interconnection and unbundled network elements.

Today’s industrial market is global. Manufacturing and measuring products are geographically distributed all over the world. Furthermore, the products are sophisticated and include know-how, often in the form of software. Thus the manufacturer is often the only party that can provide the maintenance services. In this paper, we focus on single intelligent and resource limited devices, which need to be remotely monitored or controlled. With the global Internet, satellite and telephone networks the access itself becomes possible. However, security is a challenge, because the remote devices’ limited resources are pitted against threats from the Internet. We discuss the assets of and threats against a globally distributed industrial information system. To protect the assets, three possible security architectures, Centralized connections, Layered architecture and Integration of centralized and layered architecture, are proposed. Of these three, the integrated architecture meets best the requirements of a globally distributed industrial information system.

The public telephone network has been evolving from manually switched wires carrying analog encoded voice of the I9th century to an automatically switched grid of copper-wired, fiber optical and radio linked portions carrying digitally encoded voice and other data. Simultaneously, as our security consciousness increases, so does our desire to keep our conversations private. Applied to the traffic traversing the globe on the public telephone network, privacy requires that our telephone companies provide us with a service whereby unintended third parties are unable to access other’s information. However, existing public telephone network infrastructures do not provide such a service. This paper proposes a security architecture that provides end-to-end voice privacy and authentication services within the boundaries of the existing public telephone network infrastructures. Proposed architecture uses public key cryptography for authentication and key distribution, and symmetric key cryptography for voice privacy. This work is a part of an on going project on securing telecommunication system architectures and protocols.

The convergence of public telephone networks (PTNs), wireless networks and the Internet makes it possible to disrupt vital information and communications systems from afar with software. PTN vulnerabilities are also growing due to system complexity, deregulation, increasing numbers of highly-skilled attackers, and the automation of attacks. This paper analyzes PTNs vulnerabilities, and presents an attack taxonomy. The taxonomy considers security threats posed by in-band attacks, out-of-band attacks, wireless attacks and Internet attacks.

The Universal Service Obligation (USO) is a regulation often imposed on the providers of network services. In the case of the telephone network, the Universal Service Obligation regulation can be traced back to the Communication Act of 1934. Where this act charged the FCC to provide all people of the United State a rapid and efficient telecommunication network facility. As stated, the main principle of this regulation is the requirement that the provider of the network services will make available the basic telecommunication services to all the customers residing within its exclusive territory. The practical meaning (profit wise) of this policy is that the provider of network services is not allowed to select his customers just according to their contribution to his profits but must serve all customers even if some of them are generating negative profits. That is, some customers must be served at a price “bellow” cost.1

The synthesis of switched telecommunication networks is the problem of finding the optimal transmission capacity on the links between switches and the optimal routing of communications on these links. For circuit-switched telephone networks, the first methods can be traced back almost 80 years ago in the pioneering work of Erlang. For data networks, the first analysis and synthesis techniques go back to the 1960’s with the work of Kleinrock (1964), Fratta, Gerla and Kleinrock (1973). For historical reasons, telephone and data networks methods were developed in isolation from each other.

The purpose of this chapter, like the previous chapter and all subsequent chapters, is to apply the thesis of the book to a subcategory of applications, in this case automation. Computers and robotics subtended by automation are obvious physical systems, but the principles and rules by which they operate are abstract. More importantly, these principles underwrite evolution and genetics, as well as man-made machines.

The capacity to adapt to the extensive and dynamic network conditions is a requirement of intelligent networks (INs). The network environment may change due to a large number of internal and/or external conditions. The network may become overloaded or faulty; it may experience switching delays or inadequate standby channel capacity or any other network condition. Further, the user or the source and the destination of the information may lead to an extraneous search before the right information is conveyed to the customer. In an overall sense, the network monitors its own performance without user, operator, or any other human intervention. In essence, the network adaptation responds to the environment in controlling and executing an entire range of communication functions.

Various coding techniques are being considered for higher rates ranging from DS-1 over the existing twisted wire-pairs spanning the ISDN Central Offices to the Subscriber, and up to 155 Mbps from data distribution centers to desk tops. Rates higher than the basic information rate at 144 kbps, i.e., the H0 (384 kbps), 768 kbps, are discussed in Chapter 11. Most telephone networks have at least one metallic connection to the customers from the Central Office or a remote distribution facility. Enhanced use of this existing copper facility to transmit high-speed data appears to be a viable choice towards the ISDN video and imaging services. Simulation results for increased rate with ideal equalizers are presented in this Chapter. The use of fiber in the distribution plant facilitates the availability of very high rate data ranging from 51.64 Mbps (or the standard OC-1 rate) through 2.4 Gbps (OC-48 rate) at the network distribution centers. More recently, the OC-3 rate is gaining most popularity for the fiber coaxial ATM systems. However, the use of fiber to every home is not economical at this stage, because of the existing copper wire-pairs to customer residences and almost all businesses. There is also a lack of justification for such high rates (even the OC-1 rate) to every customer.

Chapter 10 shows that to encode speech digitally requires a bit-rate of 64 kbps. Most public telephone networks have now changed from analogue transmission to digital exchanges using PCM switching and transmission. The interface to many telephones is still analogue, as shown in Figure 11.1(a). Over the coming years this will change so that the speech is converted to digital data within the consumer’s equipment, as illustrated in Figure 11.1(b). The consumer’s equipment will also provide signalling information.

Adaptive filtering has been successfully used in canceling line echos in the telephone network since the 1960’s [179, 443, 387]. In these cases, the adaptive filter is used to model the unknown echo path in a communications channel and then cancel the echo signals from the conversation. An adaptive filter is required because the echo path is not only unknown, but also because it is different for each network path. However, the acoustic echo cancellation problem is not as easily solved with conventional adaptive filtering systems due to the requirements for a much longer adaptive impulse response and for faster convergence since the acoustic echo path is time-varying during the communication transmission. In this chapter, we summarize key results from adaptive filter theory, and discuss the least mean square (LMS) and normalized least mean square (NLMS) algorithms used to adjust an adaptive filter. We then examine adaptive filtering in subbands as one method for acoustic echo cancellation. Simulations (using a measured room response with speech signals) demonstrate that the oversampled subband adaptive system provides improved performance for acoustic echo cancellation applications.

Communications networks of any kind — from early telegraph and circuit-switched telephone networks to future integrated broadband networks — are represented most naturally by a graph G(V,E), where vertices, or nodes, of V are essentially switches (telephones or computer terminals) and the edges or arcs of E are the transmission links. Classification of networks, for example into local area networks (LANs), metropolitan area networks (MANs) or wide area networks (WANs), will result in a change of the technical definitions of network nodes and their geographical coverage, but the graph representation preserves the concepts of ‘interconnectivity’ and ‘reachability’ in terms of existing paths leading from any one node to any other node. This is the precise reason why graph-theoretical methods are of great importance for design and routeing in telecommunications networks.

It may well have been Alexander Graham Bell’s successful demonstration on March 10, 1876, of what was later claimed to be the world’s first demonstration of the electronic transmission of intelligible speech—enshrined in Bell’s historic call to his assistant: “Mr. Watson, come here, I want to see you”—that set in motion what eventually became a major step forward in communication at distance. However, the history of the invention of the telephone is complex and Bell’s was by no means the only contribution.

AM and DSBSC-AM modulation do not use the frequency spectrum efficiently. Their spectral components equal distances above and below the carrier frequency contain identical information because they are complex conjugates of each other. The portion above the carrier frequency is called the upper sideband and the portion below the lower sideband. In this experiment you will see how a baseband message can be transmitted by using only one of the sidebands and, consequently, half the bandwidth of AM or DSBSC-AM. This type of modulation is called single-sideband (SSB) modulation. It has been extensively used in many radio transmission systems and in the telephone network.

To say that telephony is in a state of transition is to describe the history of telephony. It always has been in a state of transition and will continue so for the foreseeable future. As the world has been increasingly compressed in size through television and telecommunications, the trend of telephony standards has been toward enabling worldwide application rather than having each national network go its own way. While national networks still are interconnected through gateways, primarily for billing control, international signaling systems are becoming standardized with only minor national differences.

A report is given on an expert system called NOAA, Network Operations Analyzer and Assistant, that manages the Pacific Bell Californian telephone network. Progress towards automatic implementation of expansive controls is complete. Progress towards restrictive controls is partially complete. Comments are made on current research including the use of neural networks for Time Series Prediction.

It is difficult to implement talker recognition on the telephone network because of normal variation in the channel characteristics. The primary component of variation is due to the different telephone handsets or microphone frequency characteristics (Rosenberg and Soong, 1992). Lack of availability of telephone speech databases has also contributed to slow progress in the solution of these problems, though clean speech databases such as TIMIT (Garofolo et al., 1988) have been available. A telephone speech database suitable for talker identification research (Godfrey, 1992) was not generally available at the time of this research.

Mobile radio telephony is becoming a driving application for analog circuit design and silicon CMOS and RF bipolar technology. New telecom systems like GSM, PCN, DECT, Wireless in the loop ... are all developing very rapidly and will enable us very soon to organise a complete telephone network with full coverage for your mobile car phone, as well as in your kitchen and on your office desk. In Europe the major telecom companies have worked together to establish one common standard for cellular mobile radio communications at 900MHz. Similar things are happening for other wireless personal communication systems. Basically the cellular radio telephone, the wireless PABX and the wireless SLIC are bringing the same challenges to analog circuit design: maximum integration of the basic radio functions into 1 or 2 silicon chips, CMOS, Bipolar or BICMOS or GaAs.

A new generation of High Speed Switching equipment that can handle multiple types of information — such as voice, video and data is being developed. Because voice services dominate current telephone network applications, bandwidth for network access is based on Grade-of-Service (GOS) requirements for it. This paper provides a method for allocating access capacity for High Speed networks that carry multiple types of information with different GOS requirements. The GOS is assumed to be given in terms of mean waiting time until access to the network is provided. We will develop an algorithm that calculates priority factors which will meet given Grade-of-Service requirements for different types of information.

For low-delay speech coders, the research challenge is to obtain higher compression rates while maintaining very high speech quality and meeting stringent low delay requirements. Such coders have applications in telephone networks, mobile radio, and increasingly for in-building wireless telephony.

The transport protocols TCP and TP4 have been designed in the late seventies at a time when the network environment was essentially based on the switched telephone network and on leased-lines and when the packet switching was an emerging concept.

Standardization activities in digital speech coding over the past few years have resulted in an increasing need to develop and understand the methodologies used to subjectively assess new voice transmission systems before they are introduced into a telephone network. In this chapter a review of subjective methodologies for the assessment of telephone or good communications quality digital speech coding systems is provided. Technical aspects concerning the network applications and other characteristics relevant to the type of system under evaluation are briefly considered first, since these factors influence the selection of a suitable assessment methodology. Next, listener opinion tests are described. Finally, articulation and diagnostic tests as well as conversational opinion and field tests are briefly addressed.

This article surveys the post-divestiture trend of service quality in the public telephone network and proposes an incentive system for assuring such service quality, while providing greater flexibility to telephone companies in reaching high quality standards. The approach could be part of a price formula involving inflation and productivity; it could also be applied under different regulatory arrangements.

There is an increasing demand for providing voice communications to a person away from his/her wireline telephone. The popularity of cordless telephones, mobile radio telephones, radio paging, and other emerging portable communications technologies clearly demonstrates this demand. By extending the capabilities of the existing telephone network, portable communications can be provided to roving users.

While the telephone network has progressed in the process of transmitting and switching information based on the digitization of voice, leading to the integrated digital network (IDN), the rapid advances in computer and data communication technologies have evolved and have generated their own need for network capability to meet data needs. This has led to the realization that the same digital techniques can be used for data or voice transmission and switching. With ever-increasing demand for efficient interconnection of voice and data customers, the differences between voice and data switching and transmission have led to the study and development of integrated systems to transmit, switch, and process all types of digital signals. The final evolution of these systems is the integrated service digital network (ISDN). With the IDN having the inherent capability of handling both voice and data in digital form, the ISDN provides end-to-end digital connectivity, supporting a full range of services by means of standardized local access arrangements to a “transparent” transport medium.

A high level system design for GENESIS, a GEneric Network Expert System with Intelligent Simulation, is presented. GENESIS is a real-time expert system which will monitor and control the public voice telephone network. The design includes a human interface for display and user input, a realistic simulation of the network with expert heuristics to drive and test the expert system in real-time, an interface to required data bases and a blackboard for intermodule communications and control. The system is generic in that it was designed to reside above any network monitoring or data-collection system and to handle heterogeneous networks consisting of switching equipment from different manufacturers and a mix of product release levels. Key features of the design are flexibility and expandability to handle a growing network and changing needs.