Showing papers on "Telephony published in 1986"

Channel Utilization and Blocking Probability in a Cellular Mobile Telephone System with Directed Retry

Abstract: A directed retry facility, which enables subscribers in a mobile telephone system to look for free radio channels in more than one cell, is investigated with respect to blocking probability and channel utilization. An iterative procedure is devised, by which the dependencies among cells can be illustrated. This procedure makes use of theories developed for overflow systems in classical telephony, and proves to be very accurate for the situations under study. Analytical results are compared with simulations and good agreement is observed. Results show that a substantial improvement, compared with systems without a directed retry facility, can be achieved as far as carried traffic is concerned. The improvement is accomplished at the expense of those subscribers who cannot make use of the directed retry facility due to variations in radio coverage.

The analog voice privacy system

Abstract: The Analog Voice Privacy System is based on individual sample permutation of the output samples of a sub-band coder analysis filterbank. The system has a large number of digital keys, giving it the strength of a digital encryption system, but also retains the good quality characteristics of analog scramblers. It has been implemented in a real-time hardware prototype designed for evaluation in the field. The units work with any modular telephone and standard 120 volts AC electricity. The device contains two circuitry boards, one for analog and one for digital processing which contain four digital signal processors. There are 125! possible permutation keys. These prototypes were designed to be tested in real telephone environments. To date, the device has been successfully tested over long distance telephone connections, several different analog and digital PBXs and telephone switches, and a channel simulator. The quality of the decrypted speech is considered very natural, and in particular, speaker recognition is retained. This is a significant advantage over digital vocoders. This paper describes the underlying principles of the algorithm, the details of its implementation and laboratory test results.

Methods and experiments for text-independent speaker recognition over telephone channels

Abstract: We consider methods for text-independent speaker identification that deal with the variability in the data introduced by unknown telephone channels. The methods investigated include probabilistic channel modeling, a channel-invariant model and a modified-Gaussian model. The methods are described and then evaluated with experiments conducted with a twenty speaker database of long distance telephone calls.

Signaling System No. 7 ISDN User Part

Abstract: The integrated services digital network (ISDN) user part of Signaling System No. 7 defines the signaling protocol which supports the establishment, supervision, and release of voice and nonvoice calls over circuit-switched connections between ISDN terminations of digital subscriber access lines. This paper gives an overview of the ISDN user part protocol, as defined in CCITT Recommendations Q.761-Q.766 [1], in terms of the signaling functions and procedures provided to support call and connection control in an ISDN, and in terms of the information elements and signaling messages that are used by the signaling functions in ISDN exchanges to communicate.

Multi-line telephone communications system

Abstract: There is disclosed a multi-line telephone communications system where the need for the hundreds of conductors that are necessary to connect each station to the control and switching equipment is eliminated. This is achieved by replacing the status indicating leads with analog signals impressed upon the voice leads. A specific signal level is assigned to each of the ring, busy, idle and hold functions so that the voice leads carry both the audio signal and status information signal for each line selectable at the telephone station. In response to activation of the telephone keys by the user a cross point switch establishes a voice signal connection between a selected voice line and a talk path associated with the selecting telephone station. A status signal representative of the status of at least the connected voice line and distinct from voice signals is impressed upon the voice line. This status signal is detected and processed for activating the status indicators at each telephone station. Assignment of lines to particular keys of a telephone station is achieved by storing the assignment in a programmable memory. Whenever a line assignment is to be changed from one key to a different key, it is necessary only to enter the change by reprogramming the memory.

CCITT Recommendations on the ISDN: A Review

Abstract: This paper gives a brief overview of the I-Series of Recommendations on the integrated services digital network (ISDN) developed by the International Telephone and Telegraph Consultative Committee (CCITT). This set of Recommendations is printed in the CCITT "Red Book." The I-Series Recommendations represent the first major step towards worldwide harmonization of the fast-growing digital network capabilities in support of multiservice (voice, data, and image) user applications. Over 25 different Recommendations set up standard guidelines and features for service capability, network architecture, and user-network interfaces in the ISDN.

Rapid prototyping and system development: examination of an interface toolkit for voice and telephony applications

Abstract: This paper discusses a set of tools supporting the rapid development of voice and telephony applications. The tools allow interfaces to be rapidly prototyped, tested and installed without impacting the underlying system. Used directly by behavioral specialists, they have played a key role in the building of two production systems. We review several essential features of this facility and then outline its role in the rapid development of a voice messaging system for the athletes and officials at the 1984 Summer Olympics in Los Angeles.

Mobile radio telephone

Abstract: Disclosed is a mobile radio telephone apparatus for use in data communication including a transmitter/receiver, a tone signal generator, a DTMF (dual tone multi-frequency) signal generator, a data terminal, and a data modem interposed between the data terminal, and a data modem interposed between the data terminal and the transmitter/receiver and having a central processor for controlling data transmission and/or reception, the processing means detecting completion of data transmission and/or reception thereby starting the tone signal generator.

Telecommunications exchanges allocating variable channel bandwidth

Abstract: In a time division multiplexed digital switching system intercommunication between telephony groups is by way of a serially operated data bus (2). Allocation of bandwidth on the data bus (2) is on a needs basis under control of a common control responsive (3) to requests from either applications processors or interface modules (10,20). The interface modules (10,20) receive addressing information from the common control (5) on a serially operated control bus (1) only when a change of status of a communication at least partly affecting that module (10,20) occurs. Such changes of status include new communication set-ups, communication clear-down, bandwidth increase and bandwidth decrease. Dynamic re-allocation of bandwidth of the data bus (2) occurs without affecting communications in progress.

Telecommunication system for alternatingly transmitting circuit-switched and packet-switched information

Abstract: The invention relates to a telecommunication system in which alternatingly circuit-switeched and packet-switched information is transmitted in time-divided form between transmitter and receiver equipments. A system is described for multiplexing telephony channels with static and dynamic capacity. A static channel occupies a time slot in a fixed frame structure. The dynamic channels utilize idle time slots. A status bit per time slot states whether the time slot is occupied or not. According to the invention, equipment for transmitting circuit-switched information is combined with equipment for transmitting packet-switched information, such that a hybrid system is obtained which, under the control of a status bit, alternatingly transfers information from the respective equipment in response to the polarity of the status bit. A circuit-switching memory (CS) is continuously scanned in a multiplexer (DS1). Data words are stored in the memory, each word corresponding to a time slot, which in turn corresponds to an information channel in a TDM system. A status bit of the mentioned kind is inserted in each data word. For an occupied time slot in the memory CS the word from the memory CS is transmitted to a circuit-switching receiver (CR) via a line (L) and a demultiplexer DS2. For an idle time slot in the memory CS sends the word which is first in the queue in a buffer memory PS included in a packet-switching connection and is transmitted to a packet-switching receiver PR via the line L.

Stereophonic speech teleconferencing

Abstract: Speech intelligibility in monophonic loudspeaking telephones has been subject to some limitations from both the transmission medium and room acoustics. The medium has a limited bandwidth in addition to inherent noise and distortions. Room acoustics usually add the undesired impurities of the "rain-barrel effect", reverberance and background noise to direct speech signals. Binaural listening offsets many of these effects, enhancing speech intelligility. In this work, a stereophonic conference telephone has been designed and tested. The test results showed the feasibility of talker localization and improvements in talker identification. If two or more persons are talking simultaneously, the stereo telephone makes it possible to concentrate with less strain on one of the talkers. Speech intelligibility was found to be higher with the stereophonic telephone.

New applications of digital signal processing in communications

Abstract: The evolution of telecommunications towards an Integrated Services Digital Network (ISDN) offers new opportunities for signal processing, applications. Recent progress in basic techniques, like perfect signal decomposition and reconstruction or Least Squares adaptive filtering, are crucial in that evolution. Beyond the emergence of new equipment, signal processing advances are paving the way for artificial intelligence techniques, which will accompany the "information age" promised by the ISDN.

Experiments in speech recognition over the telephone network

Abstract: Moving a speech recognition algorithm out of the laboratory and into a real-world setting introduces many variables that have the potential to degrade performance significantly. Specifically, algorithms that explicitly or implicitly measure the difference between the power spectra of frames of an unknown utterance and a reference template will be negatively affected by sources of spectral distortion unrelated to linguistic distinctions. We have performed a series of experiments aimed at a deeper understanding of several of the variables that come into play when implementing one of the best-performing algorithms, that of Itakura [1], in a particularly harsh real-world environment, that of the telephone network.

ISDN standards evolution

Abstract: The evolution of the Integrated Services Digital Network (ISDN) from a primarily analog telephony network includes the evolution of standards that define the ISDN framework and services, as well as signaling and management functions. The International Telegraph and Telephone Consultative Committee (CCITT) has already defined many of the necessary standards. For example, circuit-switched, packet-switched, and private-line capabilities have been standardized, as well as two types of interfaces to connect customer premises equipment to the network. Standards need to be extended to cover, for instance, the integration of video into the ISDN framework, including the definition of a new wideband customer interface. New technologies such as wideband packet and fiber also need to be considered. This article describes the existing ISDN standards and presents an evolutionary scenario for future standards.

Layers 2 and 3 ISDN Recommendations

Abstract: The main feature of an ISDN user-network interface is the support of a wide range of service capabilities, including voice and nonvoice applications in the same network by offering end-to-end digital connectivity to a user. To handle the wide range of applications, a number of new features were developed and incorporated in the ISDN user-network interface layers 2 and 3 Recommendations. This paper reviews the current status of the ISDN user-network interface layers 2 and 3 Recommendations, with a special emphasis on new features developed and incorporated in these Recommendations.

A Voice Activated Telephone

Abstract: Telephones are already widely used in many areas, and various services come into wide use by means of public telephone networks today. The importance of a telephone set has been increasing as an access device to those services. Some telephone have additional functions, for example, automatic dialing, redialing, timer, etc.

Software Protocol for Handling ISDN in the Packet-Switched Network

Abstract: The Integrated Services Digital Network (ISDN) provides basic architecture for existing, as well as future residential plus business communications. ISDN overlayed with CCS#7 of a digital PSTN (Public Switched Telephone Network) can be the ultimate, ubiquitous network for circuit switch (voice, data), packet switch (voice, data), and private line (voice, data) applications. Assuming that the present ISDN has to interwork in the present physically separate overlayed networks (voice and data), significant problems are expected to emerge for designing hardware and linking softwares for handling packet traffic. In this paper, the software-related problems, when ISDN packet distribution nodes have to handle an ISDN interface, will be outlined with an ISDN software protocol solution. An approximation of the delay involved in the telephone switching system which is part of ISDN processing as well as the delay for the interface gateways, the HOST computer nodes, and the LAN and WAN computer nodes will be identified and formulated to reflect the total performance measure defined. Major emphasis is given to flow and congestion control performance measures in the ISDN Gateways, which are analyzed and simulated with the assistance of the basic delay table transfer software model developed for the IMPS and gateways in the ARPANET, MILNET, and MINET. The performance evaluation of this basic ISDN interfacing software, which only involved one ISDN level, i.e., the HOST or gateway and its related subnetworks, is simulated on sections of these networks to illustrate its congestion control effectiveness. There are six mathematical software techniques to account for end-to-end delay, which form the basis for the solution to these ISDN software-hardware problems in the Interface Gateways connecting the electronic switch to the computer network components.

Character Pattern Recognition for a Telecommunication Aid for the Deaf

Abstract: To communicate via the telephone with the hearing impaired, systems that transmit characters that appear on standard telephone keypads to aids that decode telephone audio tones representing characters (dual tone multiple frequency (DTMF) systems) have been used. However, due to the existence of three letters on each key, several keystrokes are required to represent the desired character without ambiguity. This paper describes an improved alternative to systems that require multiple keystrokes. Our investigations have shown that a large percentage of English words have a unique word code (the keystroke sequence required to enter the letters of the word). Techniques were developed to recognize the pattern of characters entered to allow for one keystroke per character. Further economization, achieved by recognizing syllable-like letter groups (SLLG) is described. A microprocessor-based, portable device was developed to enable telecommunication with special equipment only on the hearing impaired side of the communication channel. The device currently has 1200 SLLG's, which allows recognition of more than 2000 of the most common words.

Test adapter tools for telephone cutover system

Abstract: A method and apparatus of converting or cutting over from an old or existing telephone company main distribution frame to a new digital/electronic main distribution frame. A specially designed tool apparatus, which enables practice of the method, results in a cutover system wherein multiple splicing operation into a main feeder cable are avoided, such multiple splicing operations having proven to be costly and service affecting in the telephony industry. With the relatively inexpensive tool of the invention, the cutover process is greatly simplified and accomplished more efficiently. The present invention further includes an apertured cover apparatus and a pair lifter tool adapted for use in the above system.

Resilient support with variable flexibility, particularly for motor vehicle suspensions

Abstract: This system can be included into the field of telecommuni­cation techniques. It resolves the technical-practical problem of transmitting and receiving over distances analog electrical signals of luminous and/or chromatic information of a still or moving image using all the usual means for telecommunications. The essence of the solution of this problem by this invention is that it has employed the systematic applica­tion of the characteristics of electron optics correlated with the fundamental relation of geometric optics. The principal uses of this invention are in bidirectional integral telephony, with sound and video.

Subscriber equipment for wireless digital telephony

Abstract: A base-band processor provides control of conversion of PCM signals at one bit rate to other bit rates. It converts received digital signals to voice signals and vice versa. It also provides echo cancellation where it uses dynamic memory to store received signals. PROM's store the echo cancellation program information as well as that for utilisation of the processor as a control processor.As a control processor it signals to the frequency synthesiser the frequency of operation for communication to the base station. The processor is directly coupled to a modem processor which is able to access the base band processor memory. This processor sends its signal at a predetermined sampling rate which are converted to analogue signals. The analogue signals are subjected to a cancellation process to reduce distortion. These signals are converted to an IF for addition to the synthesiser frequency to result in an HF signal for transmission

Arrangement for protection against voltage flash-over in telephony handsets

Abstract: The invention relates to an arrangement for protecting against voltage flash-over due to electrostatic charges in telephony handset bodies between an operating person and an electro-acoustic transducer arranged behind a receiver and/or transmitter capsule exhibiting sound transmission openings. For this purpose, the sound channels of the transducer holder(s) are provided with a cover(s) arranged at a distance therefrom. Applicable in telephony handsets (Figure 1).

Between Laboratory and Field Trial: Experience with a Communications Services Testbed:

Abstract: To attempt to bridge the gap between laboratory and field trial, we at the Network Services Research Division of Bellcore have designed and implemented the Modular Integrated Communications Environment (MICE). The primary motivation for MICE is to provide a means for research on the design and evaluation of advanced communications services and user interface technologies for the public communications network in a realistic environment, but without the expense and overhead of field trials. Thus, MICE is a real system, providing reliable communications services to a community of friendly users, but permitting convenient experimental control over services and interfaces. MICE has been constructed entirely of commercially-available subsystems, with controlling software executing in the UNIX™ environment. MICE has processed over 50, 000 communications attempts to date; capabilities range from basic telephony to voice paging, voice mail, and integrated voice/data mail.

A Digital Telephone Set for ISDN

Abstract: In addition to the present voice communications services, more diverse digital telephone services are in very strong demand. Among the critical system requirements for designing digital telephone sets are how to economically realize new telephone services and how to ensure the continuity of existing analog telephone services. This paper describes the system requirements from both the service and technical points of view in the design of telephone sets. Based on these requirements, it further describes a method of designing network interface and telephone function blocks. An outline of the configurations and characteristics of digital telephone sets which can be operated by remote power feeding is presented as a design example.

Bandwidth Compression of Voice and Modem Signals using Variable-Rate ATC Encoding

Abstract: We present a strategy based on local SNR for variable-rate encoding of a variety of telephony signals using Adaptive Transform Coding We first describe simple modifications of the speech version of ATC, involving chiefly the side information quantization and the bit allocation procedures, which substantially improve fixed-rate encoding performance on modem signals Then, using simple statistical models of bit-rate demand, we predict the compression performance of the variable-rate encoder in a typical telephony network We compare the performance of variable-rate and fixed-rate encoding of speech and modem signals, based on extensive listening tests and simulations, respectively We also report on the progress of an experimental implementation of the algorithm in 32-bit floating-point hardware