Modeling call control for distributed applications in telephony

The advanced intelligent network (AIN) architecture, an environment in which distribution applications are designed, developed, and deployed independent of the network implementation by different service developers, on possibly different platforms, at different times, is considered. The focus is primarily on support for call-processing applications through call-modeling techniques. However, the entities of NECs application service processor (ASP) architecture as a whole are briefly described to provide a basic understanding of the scope of the call model. The call-modeling concepts discussed at the Bellcore-sponsored Multi-Vendor Interaction forum are compared with NEC's call-modeling concepts. A simple approach (i.e. the logical path) for controlling the execution of FCs (function components) when multiple AIN services are active on a call is suggested. An object-oriented implementation based on the NEC call model is described     

A low-power integrated tuner for cable telephony applications

This paper describes an integrated tuner for cable telephony in a 0.35 /spl mu/m, 27 GHz SOI BiCMOS technology. The IC integrates a complete dual-conversion signal path including upconverter, downconverter, variable-gain amplifier, LO synthesizers with fully integrated voltage-controlled oscillators, gain control circuitry, as well as digital calibration and interface circuits. It accepts signals in the 200-880 MHz band and produces a 44 MHz IF. Drawing 168 mA from a 3 V supply, the tuner system has a worst case noise factor of 7.3 dB, system phase noise below -78 dBc/Hz at a 10 kHz offset, spurs below -42 dBc for 137 5 dBmV input channels, a gain of 60 dB, and gain control range of 68 dB. The 13 mm/sup 2/ IC meets specifications across an outdoor temperature range of -40/spl deg/C to 100/spl deg/C in production lots.     

Passive optical local networks for telephony applications and beyond

A new digital telephone system is described using a passive optical network. Downstream transmission is broadcast and optical time-division, multiple-access (OTDMA) is used in the upstream path. The principle advantage of this approach is that exchange resources and fibres are shared between a large number of customers     

A single-chip NMOS dual channel filter for PCM telephony applications

The implementation of a completely monolithic channel filter containing all frequency selective functions associated with a PCM line interface is described. The circuit utilizes switched capacitor techniques. Design of the overall architecture, the individual filter sections, and the operational amplifiers in NMOS technology is described. Experimental results are presented.     

Internet telephony in India

Internet telephony is a novel and cheaper method of communication and conducting business over the Internet. The paper presents an overview of Internet telephony, its methods, viz. PC-to-PC, PC-to-telephone, telephone-to-telephone and telephone-to-PC; benefits in cost advantage, simplification, consolidation, higher efficiency and reliability, etc., quality issues, protocols and drivers; challenges and regulatory framework; and status of Internet telephony in Asia Pacific region. Further, highlights its potentiality for India, implications of guidelines of Internet telephony, issues of concern, etc. Concludes that Internet telephony cannot make compromises in voice quality, reliability, scalability and manageability, and work seamlessly with telephone systems all over the world. Internet telephony will prove to be a boon for a price-sensitive market like India and rural telephony will receive an impetus. The Government of India may further deregulate the market and allow phone-to-phone telephony through the Internet and open long distance calling within the country for ISPs to realize “telecom for the common man” or “telecom for all” a reality.     

Screen-based telephony

Experimentation with screen-based telephony is discussed in the framework of Apple Computer's user-centered design philosophy. The approach described enables programs to be developed for the most ubiquitous technologies today-plain old telephone service (POTS) and proprietary private branch exchanges (PBXs)-but also allows for an easy, logical upgrade to future technologies like ISDN. The model has been built in three layers, which reflect the three constituencies who must cooperate to bring ISDN applications (and their forerunners) to market, in the volume required to spark the growth of this area: telecommunications vendors, personal computer vendors, and independent software developers. The Call Manager Application Programmers Interface is the code/specification from Apple that enables voice and data applications to be written independently of the vendor or technology. Telecommunications vendors can write drivers to the Application Program Interface (API) and build the appropriate hardware interfaces, without having to support thousands of software developers. Some of the features of Apple's prototype human interface to the voice network on this API are described     

ISDN computer-aided telephony

It is argued that ISDN computer-aided telephony requires properly architected platforms to satisfy changing application needs during the 1990s. Proper architecting necessitates the use of functionally rich and consistent telephony application programming interfaces (APIs). Other APIs are also needed to support integrated applications. The coexistence of telephony and other APIs must be accommodated in the ISDN driver architecture to make efficient use of D-channel signaling and voice, data, or image communications on the associated B/H channels. This driver may support Open Systems Interconnection (OSI), Systems Network Architecture (SNA), X.25, or other protocol stacks in the same computer using a single ISDN access link. Applications being currently explored show that significant benefits can be realized using incoming call management and LAN-based image server access by means of ISDN. It is envisioned that by the year 2001, a common API will facilitate multimedia applications on multivendor platforms architected within the OSI framework. These platforms will support interconnections of public and private ISDNs and bridging to BISDNs     

Internet telephony signaling

The term “multimedia session” refers to the integration of data coming from various sources, such as sound, video and text, within a computer application. Telephony over the Internet is among the more exciting current developments. The signaling of a telephone call consists of the set of messages and procedures used to establish a connection, to request changes in communication bandwidth, to obtain the message status for the end points participating in the conversation, and to close the link. At present there exist two competing signaling protocols for Internet telephony, viz., the H.323 protocol sponsored by the ITU and the Session Invitation Protocol (SIP) sponsored by the IETF. Each of them supplies its own signaling mechanisms.

In this paper, these two protocols in terms of their main functionalities are compared. Based on the results of this comparison, a Client/Server architecture for the development of an application that supports a basic SIP implementation, as well as the formulation of requests allowing the establishment and the disconnection of communications between a number of users in a multimedia session are then defined.     

Integrated circuits for telephony

Many integrated circuits for large-scale application to telephone networks have been designed, including subscriber line interfaces, antialiasing filters, analog-to-digital-to-analog converters, and tone receivers. This paper summarizes in a unified fashion both interfacing and functional requirements for these devices, as well as the related circuit and technology approaches which have been utilized.     

Programming Internet telephony services

Internet telephony enables a wealth of new service possibilities. Traditional telephony services such as call forwarding, transfer, and 800 number services, can be enhanced by interaction with e-mail, Web, and directory services. Additional media types, like video and interactive chat, can be added as well. One of the challenges in providing these services is how to effectively program them. Programming these services requires decisions regarding where the code executes, how it interfaces with the protocols that deliver the services, and what level of control the code has. In this article we consider this problem in detail. We develop requirements for programming Internet telephony services, and we show that at least two solutions are required-one geared for service creation by trusted users (such as administrators), and one geared for service creation by untrusted users (such as consumers). We review existing techniques for service programmability in the Internet and in the telephone network, and extract the best components of both. The result is a common gateway interface that allows trusted users to develop services, and the call processing language that allows untrusted users to develop services     

The missing link-computer-supported telephony

From an engineering viewpoint, the physical realisation of a computer-supported telephony application involves connecting a computer with a telephone system by a signalling or data link which supports a protocol to carry command and status information between the two systems. In principle, any computer can support the computer supported telephony (CST) protocol, and the telephone system may be a PABX, an automatic call distributor, a key system or any other telephone switch. Existing or new systems can be linked by CST. The author discusses applications of CST the main one being telemarketing. The technology to support CST is not very advanced. At the heart of a modern telephone system is a computer. What CST does is link that embedded computer to a less specialised one. This technology is discussed by the author. The role of CST within the wide telecommunications and IT environment is also discussed     

Europe's future mobile telephony system

Building on the huge success of its homegrown cellular system, Europe is planning a next-generation wireless system to handle data as well as voice, and-it is hoped-lay the foundation for universal roaming. The system being developed in the framework of the European Telecommunications Standards Institute (ETSI) is called UMTS (Universal Mobile Telecommunication System). In parallel, the International Telecommunication Union, based in Geneva, is formulating IMT (International Mobile Telecommunications) 2000, which is to be a family of systems that will let users roam worldwide with the same handset, and which will include UMTS as a subset     

Adaptive delta modulator for telephony

An adaptive version of the basic delta modulator employing full-width pulses and RC integration has been designed. A digital technique is used to sense the level of the input signal and to control the amplitude of the pulses supplied to the RC integrator in the feedback circuit. For an 800 Hz sinewave input a signal/quantisation-noise ratio of 32 dB has been obtained over a dynamic input range of 30 dB.     

Optimum frequency reuse in mobile telephony systems

The reason why cellular mobile telephony systems exist is that they allow frequency reuse. Dynamic allocation algorithms improve the network efficiency and the service quality. Dynamic allocation is closely related to the following instantaneous allocation problem: given a number of channels (frequencies), a cell structure and the number of calls to be accommodated in each cell, find the optimum allocation of channels to cells subject to restrictions concerning the distance of cells, where the same frequency can be reused. In this paper we formulate and solve this problem by showing that it can be reduced to a 0–1 programming problem. Finally we present results, and draw subsequent conclusions.     

Channel-adaptive wideband wireless video telephony

The fundamental advantage of burst-by-burst (BbB) adaptive intelligent multimode multimedia transceivers (IMMTs) is that-irrespective of the propagation environment encountered-when the mobile roams across different environments subject to path loss; shadow- and fast-fading; co-channel-, intersymbol-, and multiuser interference, while experiencing power control errors, the system will always be able to configure itself in the highest possible throughput mode, while maintaining the required transmission integrity. Finding a specific solution to a distributive or interactive video communications problem has to be based on a compromise in terms of the inherently contradictory constraints of video quality, bit rate, delay, robustness against channel errors, and the associated implementational complexity. Considering some of these tradeoffs and proposing a range of attractive solutions to various video communications problems is the basic aim of this overview. The article portrays a range of proprietary video codecs and compares them to some of the existing standard video codecs. A number of multimode video transceivers are also characterized. Systems employing the standard H.263 video codec in the context of wideband BbB adaptive video transceivers are examined, and the concept of BbB-adaptive video transceivers is then extended to CDMA-based systems     

Speech enhancement for mobile telephony

The properties of noise fields in automobile interiors are discussed with a view toward speech enhancement for voice-activated mobile telephony. The limitations on performance of adaptive noise cancellation are explained in the context of the spatial correlation properties of the noise field. A simple delay-equalized near-field array of directional microphones is analyzed and found to be effective for increasing the signal-to-noise density ratio (SNR) and reducing the reverberant distortion of the speech, without introducing any further distortion. An array of N microphones, each with a delay and weight chosen according to its distance from the speech source, is a viable solution. Such an array gives gains on the order of N over the speech band, reduces reverberation, and does not introduce waveform distortion. Experimental results confirming the predicted performance are presented     

Internet protocol support for telephony

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.