Showing papers on "Telephony published in 1985"

Computer-Based Real-Time Conferencing Systems

Abstract: Multimedia Communications r 1aI-Time 'stems of America r support for group including electronic mail, nferencing,I form man-a ,2 and coordination sup-primarily addresses asyn-us interaction among users. systems are most useful when user can work at times of his own sing. However, although relative-little work has been done on com-uter support for people working together simultaneously, for certain group tasks, such as crisis handling, simultaneous (or real-time) interaction is essential. In a real-time conference, for example, each participant can be seated in his own office at a workstation that might include a high-resolution screen for computer output, a keyboard and a pointing device, a microphone and a speaker, and possibly a camera and video monitor. Parts of each participant's screen can be dedicated to displaying a shared space in which everyone sees the same information. The voice communication equipment can be used by the conference participants for discussion and negotiation; video communication can add an illusion of physical presence by simulating a face-to-face meeting; and conversational references ("this num-ber" or "that sentence") can be clarified by pointing at displayed information. The displayed information can be dynamically edited and processed, permanent records can be saved, and new information that is relevant to the discussion can be retrieved for display at any time. Participants can, in addition , have private spaces on their screens that allow them to view relevant private data or to compose and review information before submitting it to the shared space. Systems that provide some of the above features already exist. As early as 1968, the NLS system5 provided a shared-screen mode for simultaneous collaborative authoring of structured documents. This facility, which can be used to access any interactive program from multiple terminals, is now available in many time-shared operating systems in the form of terminal linking. Terminal linking on most systems does not work correctly unless all linked terminals are of the same type. A notable exception to this is Tym-share's Augment6 system (the commercial successor to NLS), which supports "virtual" terminal linking across dissimilar terminal types. Real-time conferencing can be used to support joint work in many different applications.

Microprocessor controlled interface for cellular system

Abstract: A microprocessor-controlled interface for a cellular radio telephone system enables the user to communicate with such system by means of conventional telephony devices. A standard telephone outlet communicates with the transceiver logic bus through a microprocessor. Switches are provided for controlling the audio path between the handset and the transceiver of the mobile, vehicular-mounted unit in response to the status of the interface outlet. The microprocessor interprets both the transceiver bus and various signals generated by the operation of a device that is plugged into the interface so that the operation of the device is rendered fully compatible with the cellular system. The microprocessor is programmed to decode known manufacturer protocols whereby the interface is able to render itself compatible with various equipment types.

Method and apparatus for controlling digital voice recording and playback over telephone lines and adapted for use with any host computer

Abstract: A method of enabling the efficient use of any host computer with a voice message processing apparatus using a telephone interface and voice processing system, that comprises, providing a universal supervisory circuit between the host computer and the telephone interface and voice processing system, said circuit having a memory that stores data of a type useful to interface with said telephone and voice processing interface system, said circuit performing the steps of receiving standard serial data from any host computer such as telephone answering, voice message storing and other commands; storing in said supervisory circuit memory particular serial data commands tailored to command and control the telephone interface and voice processing system; using said data stored in the memory to translate said received standard serial data from the host computer into command and control language recognizable by the telephone interface and voice processing system; converting the translated language into parallel data; and prioritizing the application of said parallel data to said telephone interface and voice processing system for efficient transmission to the telephone interface system; the said telephone interface and voice processing system generating parallel data, such as line-ringing and other status data, translated by said supervisory circuit into standard serial data recognizable by the host computer; and prioritizing the transmission of the last-named standard serial data to the host computer.

A Metropolitan Area Network

Abstract: Most high-speed multiple-access networks on the market, or those currently under design, operate in a limited geographical environment and are devoted either to data-only or voice-only transmission. However, challenges issued by the marketplace require the full integration of voice, data, and video in a real-time environment. This paper proposes a metropolitan area network that covers a campus environment and truly integrates voice, data, and video under a single transmission medium. Furthermore, the transmission is achieved through a single media-access schema. This paper also presents marketing issues, network requirements, and technology constraints. The proposed mediaaccess schema is also being considered by the IEEE 802.6 MAN Committee for Standardization.

Digital telephony and network integration

Abstract: A network in transition. Speech digitalization fundamentals. Waveform coding. Parametric and hybrid coding. Digital techniques in the telephone network. Digital transmission. Digital cellular radio. Microwave transmission. Satellite transmission. Fiber-optic transmission. Digital switching architecture. Operational switching systems. The evolving switched digital network. The integrated services digital network (ISDN). Closing the loop. Appendices. References. Glossary. Index.

ISDN protocol architecture

Abstract: IT IS NOW widely recognized that the Integrated Services Digital Network (ISDN) concept is providing a useful framework for the development of future telecommunications networks and services. An ISDN can be regarded as a generalpurpose digital network capable of supporting (or integrating) a wide range of services (voice and non-voice) using a small set of standard multipurpose user-network interfaces. Relevant CCITT Recommendations (or standards) on ISDN interfaces were expected to be available in 1984, the final year of the current study period. This paper reviews CCITT progress on the architectural studies of protocols associated with ISDN's, in particular, the ISDN Protocol Reference Model being developed by Study Group XVIII. This Model, based on the concepts and principles of the ISO/CCITT Open Systems Interconnection (OSI) Reference Model, aims at providing a unified framework for modeling ISDN multiservice communications and capabilities. The new Model is applied to several possible ISDN communications configurations, enabling a number of technical issues to be easily identified.

Time independent data transmission system having capability for accessing a plurality of data devices

Abstract: A time independent data transmission system which allows a computer to access a selected one of a plurality of data devices at a location remote from the computer at any time via a non-dedicated telephone line is disclosed. The system is coupled to a telephone instrument, the plurality of data devices and the telephone transmission line and normally connects a data set coupled to the data devices to the telephone transmission line. The telephone instrument is normally disconnected. A counting circuit senses the number of ring bursts in a ringing signal from a caller and a central computer is programmed to transmit a ringing signal having only one ring burst. When only one ring burst is sensed, the data set remains connected to the telephone transmission line and the data set dials up the central computer when the ringing signal terminates. The data set comprises a decoder circuit which receives a coded signal from the central computer designating which of the data devices is to be accessed. The decoder controls a switch to select one of the data devices. If more than one ring burst occurs, a non-computer call is being sensed and the data set is disconnected and the telephone connected to the telephone transmission line so that normal telephone usage is allowed. Communication between the computer and the data set via the telephone line occurs without ringing the telephone instrument and without otherwise interfering with normal telephone usage.

A simulation study of advanced routing methods in a multipriority telephone network

Abstract: Traffic and load control in multipriority telephone networks using advanced routing methods are analyzed using a special-purpose simulation program. Nonadaptive schemes as well as local and centralized adaptive schemes are studied, and a new centralized adaptive method is proposed. The grade of service, probability of preemption, and average number of trunks used per call serve as performance criteria and are compared for different load conditions and routing methods. It is shown that the centralized adaptive methods are superior to all others, with the newly proposed centralized adaptive method allowing for greater flexibility in network management without degradation in performance.

Voiceband Quality–Of–Service Issues in the Post Divestiture Environment

Abstract: This report discusses quality–of–service (QOS) issues for telephone networks. Deregulation and divestiture have fostered increased competition in the United States in the telephone equipment and service industries. There are many economic, policy, and technical issues that remain to be solved as the result of the plethora of equipment and services now available. This report addresses the technical problems associated with the interconnection of equipment from many vendors. In order to maintain a satisfactory quality of service to the end user, performance standards must be developed, approved, and implemented. The work of IEEE, CCITT, and ANSI–accredited standards groups responsible for telephone QOS is reviewed. The problem of interconnecting different national networks in the international community is seen to be analogous, in part, to the problem of interconnecting the numerous public and private networks within the United States. Although progress has been made by both national and international telephone QOS standards groups, unsolved issues remain. Principal among these are the development of objective measures of voice quality, the mapping of these objective measures into five levels of quality, enhancement of IEEE and CCITT telephony QOS standards (including the development of standards for the transmission of data on voiceband networks), and the development of QOS standards for Integrated Services Digital Networks. These are discussed in this report along with recommendations for new programs that would contribute to their resolution.

Life safety audio system having a voice synthesizer and a constant volume telephone network

Abstract: A life safety system for monitoring a building has a plurality of detectors for detecting alarm conditions located throughout the building, a plurality of speakers located throughout the building, a source of pre-stored voice messages which can be broadcast over the speakers, and a telephone network. The improvement is a voice synthesizer for providing the source of pre-stored voice messages and a constant volume telephone system. The constant volume telephone system has a telephone line and a plurality of telephones which can be connected to the telephone line, and a power supply having an adjustable output voltage which provides the voltage for the telephone line. A sensing and adjusting circuit senses the number of telephones off-hook and causes the power supply to be adjusted in response to the number of telephones off-hook.

Performance Analysis of Tandem Burst-Error Links with Applications to ISDN

Abstract: In this paper, tandem links with burst errors are analyzed using the Gilbert model for individual links. The results, apart from being of general interest, are of particular relevance to the Integrated Services Digital Networks (ISDN) for which the International Telegraph and Telephone Consultative Committee (CCITT) is currently developing performance specifications based on hypothetical reference connections (HRX's), i.e., composite channels consisting of several tandem reference circuits. The average error rate, the error gap distribution, its mean, and a tight upper bound on its standard deviation are analytically derived for the composite channel (equivalent to an HRX) with N tandem links (circuits). These statistics are then related to the performance measures proposed for ISDN. Some numerical results are presented to highlight the large differences in the estimated performance with a burst-error model approach and with an independent-error approach (having identical average error probabilities).

Voice-excited predictive coder (VEPC) implementation on a high-performance signal processor

Abstract: In this paper, we discuss the implementation of a medium-bit-rate linear prediction baseband coder on an IBM bipolar signal processor prototype having a high processing capacity. We show that the implementation of our algorithm requires a processing load of 5 MIPS, with a program size of 5K instructions. We then discuss the application of our coder in a normal telephone environment, which requires mu-law to linear PCM conversion and other signal processing functions such as voice activity detection, automatic gain control, echo control, and error recovery. Quality evaluation tests are also reported which show that this type of coder, operating at 7.2 kbps, allows the transmission of telephone speech with communications quality. Moreover, obtained intelligibility scores and speaker recognition levels are high enough to demonstrate that this coder is a good candidate for telephony applications such as digital trunk transmissions, satellite speech communications, secure voice communications, and audio distribution systems.

900 MHz digital cordless telephone

Abstract: The paper discusses several aspects of cordless telephony which are relevant to systems being discussed for use in the UK in the future. A variety of methods of arriving at traffic estimates are considered, including studies of population density in urban and suburban London. From these estimates it is shown that an allocation of at least forty channels will be needed to satisfy the demand of domestic and small business users. Three analogue and one digital modulation systems are considered, and it is shown that the digital system offers better use of the spectrum than the analogue systems. Additionally, the digital scheme allows convenient implementation of time-division duplex transmission, as well as ongoing call-quality monitoring. The paper looks at the merits of allocating one common channel for signalling purposes and compares these with the advantages of a simpler system, where the signalling information is carried on the speech channels. It is shown that significant extensions of battery life can be achieved by using a common signalling channel. Finally, the paper considers some of the implications of the time-division duplex system on receiver design.

Optimal design for cellular mobile systems

Abstract: Cellular zone design parameters for a land mobile communication system are clarified. System parameters having a strong influence on cellular system performance are discussed and some problems encountered when system parameters are changed are examined.

Integrated circuit delivering an output current whose variation is inverse to that of the input current.

Abstract: The invention relates to an integrated circuit delivering an output current whose variation is inverse to that of the input current, more particularly used in the field of telephony. According to the invention, part IL of the current in the telephone line flows through a resistor RA placed across the terminals of the set and is transformed into a potential at the input A of a differential amplifier T3, T6, and then compared with a reference potential Vref. The supply current IR to the differential amplifier is shared between the two branches of the amplifier, and the current in the input branch of the potential to be compared is then forwarded to the output transistor T5 by a current mirror T4, T5. The invention applies to telephone installations.

The Power Supply of Supplementary Equipment at the ISDN-Telephoneset

Abstract: 1. Definitions of "supplementary equipment" To define the "supplementary equipment" at the telephoneset in a way that meets the world-wide existing regulations, would be very complicated, abstract and time-vasting. So I prefer to draw your attention to table 1, that presents the most common and important 'SEs'. Supplementary equipment in German called "Zusatzeinrichtungen" increases the features of the telephone at the subscriber's premises without increasing cost and size of the basic telephoneset itself. Normally the SE is privately owned but may also be the property of the PTT. 2. Interfaces for "supplementary equipment" within tne suoscriber's network

The integrated services digital network. Bringing home the bits

Abstract: A revolution is in the offing for telephony. Over the next 20 years, the public switched telephone network will be integrated with specialist data-transmission networks to create a common universal network for speech and data ? the ISDN

Circuit for conference telephony

Abstract: A circuit for conference telephony using a conventional system for transmission of voice data and having no use for the conventional hybrid station is materialized by this invention using at least two buffer memory circuits provided for each of the frames and adapted to be switched to each other one for input and the other for output, enabling one of the buffer memory circuits to memorize voice data transmitted thereto and, in the meantime, permitting the voice data of the preceding frame memorized in the other buffer memory circuit to be read out, converting the voice data so read out to be converted into analog signals, and synthesizing the analog signals one another.

ASBU 501 - A Digital SPC Voice/Data Switching System

Abstract: The ASBU 501 is a fourth-generation all-digital switch employing distributed control architecture and a functional top-down design. This paper is organized in three major sections describing the structured system design, the basic equipment, and the software design principles for distributed control. The first section diagrams the layered functional structure and describes the subsystems, function blocks, and unit level (hardware and software). The second section on distributed digital equipment gives a description of the components for both telephony and data switching. The basic switching units which house and distribute the hardware devices are explained and a description and function of all of the equipment is provided. The third section discusses the software design principles utilized in the ASBU 501 distributed control system.

Personal Computer Communications Via Telephone Facilities

Abstract: Communication from terminals to other terminals and to computing service systems has been an important aspect of data communications since the early 1960's. When personal computers arrived on the scene, in the 1970's, the use of telephone facilities and modems for communications was a natural extension of existing technology. Now, however, many modems have "intelligence" and can be controlled from their electrical interface, communication software is available to facilitate the use of a modem and coordinate the communication process, and there are many uses for personal computer communications. Some of the hardware, software, and computing services for personal computer communications are discussed in this paper.

Radio telephony exchange system

Abstract: PURPOSE:To call back a caller during stanby by allowing a subscriber possessing a pager to dial a special number and an own pager number from a conventional telephone set placed nearby when the subscriber recognizes an incoming call. CONSTITUTION:When the caller dials a telephone number of a pager subscriber from a conventional telephone set TA2, the TA2 is connected to a radio telephony exchange MTS5, which makes connection to a paging radio equipment TX4 and transmits a call signal 100. The TX4 transmits a call code 101 based thereupon. A call tone is transmitted from a callback tone trunk RBT51 to the TA2 to allow the caller to recognize the calling state. On the other hand, when the pager subscriber recognizing the incoming call dials a connection special number from a telephone set TB3 placed nearby, the MTS5 connects an incoming trunk ICT54 to a voice trunk ANT53 and a numeral number reception trunk REC55. The ANT53 transmits a message to the TB3 to give commands of transmission of a pager number and a secret number.

Digital telephone and data transmission system

Abstract: 1. A digital telephony and data transmission system comprising a plurality of satellite stations (13, 14, and 15) coupled with a central station (1 through 11), the satellite stations comprising means 55, 56 and 57) in order to transmit, and to receive, voice and data signals, and to transmit toward the central station (1 through 11) telecontrol signals ; the said central station (1 through 11) comprising : - means (9) for decoding the telecontrol signals ; - time multiplexing and demultiplexing means (18, 23 and 25) controlled by the decoding means (9) and of the telecontrol signals, in order to direct the voice and data signals towards the satellite stations 13 through 15) in accordance with the telecontrol signals ; characterized in that the telecontrol signals, the voice signals and the data signals are binary signals transmitted in series in accordance with a standardized format for telephonic transmission by pulse code modulation (PCM).

Development of a medium-/small-scale local area optical fiber loop network

Abstract: This report describes the outline of an optical loop network which is the most suitable for a system consisting of about 200 terminals. Upon development of an optical local-area network (LAN) to be used within this range, the compact, low cost, and general purpose properties, as well as the reliability and expandability of the system should be considered first. Especially, determination of the transmission rate and selection of the microprocessor to be used are important items to realize the compact and low cost network as desired. There are two examples for such systems which are used actually now. One is the HLN-8140 Loop, which is suitable for small-sized systems. It is a packet switching loop network using the token-passing method with a transmission rate of 1 Mbit/s. Another example is the HLN-8380 Loop which can cover up to medium-sized systems. It is a circuit-switching loop network with a transmission rate of 10 Mbit/s, and it allows transparent communications to be made via telephones and facsimiles.

Progress in the 800MHz land mobile telephone system in Japan

Abstract: NIPPON TELEGRAPH & TELEPHONE CORPORATION (NTT) developed an 800MHz land mobile telephone system and put it into commercial service in the Tokyo area in 1979.(1) This service has now been extended to cover most of the country. At present, it accommodates approximately 40,000 subscribers with service areas covering 500 cities. This paper describes system development over the past five years as well as various new services which NTT plans to offer in the near future.

Wide-band communications system

Abstract: In an optical wide-band network with subscriber-lines, each comprising two optical wave-guides running between an exchange and the subscriber-station, only TV and/or stereo-sound wide-band signals are transmitted, through one optical wave-guide, from the exchange to the subscriber-station, whereas through the other optical wave-guide, in both directions, are transmitted all digital communications narrow-band signals - for 64 kbit/s digital telephony, facsimile, 8 kbit/s signalling, synchronizing, data-transmission-, also possibly bi-directional video-telephony digital signals, between the exchange and the subscriber station. The digital communications signals in the two transmission directions are each transmitted in time-division multiplex in the separate-time-position process; the wide-band signals are transmitted on analog-modulated carriers.