Dial-up Telephone Lines Research Articles

Timing in telecommunications networks

I describe the statistical considerations used to design systems whose clocks are compared by the use of dial-up telephone lines or the Internet to exchange timing information. The comparison is usually used to synchronize the time of a client system to the time of a server, which is, in turn, synchronized to the time scale of a national timing laboratory. The design includes a dynamic estimate of the system performance and a comparison between the performance and a parameter that specifies the required stability based on external considerations. The algorithm adjusts the polling interval and other parameters of the algorithm to realize the specified performance at minimal cost, where the cost is taken to be proportional to the inverse of the interval between message exchanges using either the Internet or dial-up telephone calls.

Realizing UTC(NIST) at a remote location

I will describe the backup time scale system that I have constructed at the site of the NIST radio stations near Fort Collins, Colorado, and I will compare its performance with the primary ensemble in Boulder. The Fort Collins system is designed to be a backup for the Boulder system and is intended to support all of the NIST time services should the primary scale become unavailable for any reason. The backup time has a number of unique problems and requirements, and I will discuss the design considerations that I used to address these issues. The backup scale tracks UTC(NIST) in frequency with an uncertainty (measured by the Allan deviation of the difference) of about 1 × 10−14 by use of administrative steering that is applied not more often than once per week. The corresponding time deviation is less than 1 ns for all averaging times less than 1 week, and the peak time difference between UTC(NIST) and its backup realization is less than ±25 ns and is generally much better than this value. This is much better than would be needed for supporting the radio stations, the digital time services (ACTS, a time service that provides time in a digital format using dial-up telephone lines, and the Internet services) and the Frequency Measurement Service. Its frequency stability and time accuracy would not be adequate for the most demanding users of the Global Time Service and for international time and frequency coordination. The primary limitations to the performance of the backup time scale are caused by environmental perturbations, especially temperature and supply voltage, and the existing hardware could probably support all of the NIST services if the environment were improved.

Improvements to the NIST network time protocol servers

The National Institute of Standards and Technology (NIST) operates 22 network time servers at various locations. These servers respond to requests for time in a number of different formats and provide time stamps that are directly traceable to the NIST atomic clock ensemble in Boulder. The link between the servers at locations outside of the NIST Boulder Laboratories and the atomic clock ensemble is provided by the Automated Computer Time Service (ACTS) system, which has a direct connection to the clock ensemble and which transmits time information over dial-up telephone lines with a two-way protocol to measure the transmission delay. I will discuss improvements to the ACTS servers and to the time servers themselves. These improvements have resulted in an improvement of almost an order of magnitude in the performance of the system.

Obtaining digital files from radiographic films

Digitizing images with low cost off-the-shelf technology arises as an alternative to storage films in developing countries. The objective of this work is to determine a low-cost method to digitize Rx films for educative and remote-consult purposes. To this aim, different ways of digitizing were compared against conventional methods to determine the feasibility to have a simple-low cost method that is quality independent from facilities and operator. Different images from digital photo cameras and scanners (with or without transparency adapter at different resolutions and color depth) were analyzed. We present preliminary results for digitizing Rx films with a fast and simple inexpensive system that capture quality images, and optimal sizes for storage in basic PCs in hospitals, with the possibility of asking for a second opinion via e-mail or through a web-based service. It must be noted that in many cases the only way of communication is via dial-up telephone line.

Development of an automatic data collection system for a major box-girder bridge

The design and development of a computer controlled data acquisition system to monitor the behaviour of the Foyle Bridge, near Londonderry, Northern Ireland, is described. Details are given of the hardware used and of the software which has been purpose written. The system is designed to operate unattended and may be controlled remotely via modem and a dial-up telephone line. Examples of typical results obtained are given.

Network management issues in support of X.32 services

The X.32 recommendation was introduced to facilitate access to X.25-based packet switched public data networks by mobile packet-mode workstations via dial-up telephone lines. The provision of an X.32 service entails more than merely having a corresponding X.32 facility within the subnetwork to provide the connection to the data network. The subnetwork has to incorporate a number of additional facilities to handle X.32 mobility and security features, as well as other facilities relating to network administration and security management. The issues relating to such facilities are discussed in the paper, as they are essential in the provision of a supporting environment for a comprehensive X.32 service.

Multichannel physiological monitor plus simultaneous full-duplex speech channel using a dial-up telephone line

Because of bandwidth limitations, the Public Switched Telephone Network (PSTN) cannot normally accommodate simultaneous multichannel physiological signaling with speech. This communication describes work carried out to build a physiological monitoring system which can transmit up to six low-frequency data channels in the presence of a full-duplex speech channel using one telephone line.

Pessimistic quasipartitioning protocols for distributed database systems

A communication link failure can result in a network partitioning that fragments a distributed database system into isolated parts. If a severed high-speed link (e.g. satellite link) between the partitions can be replaced by a much slower backup link (e.g. a dial-up telephone line), the partitioning becomes a quasipartitioning. Two protocols for transaction processing in quasipartitioned databases are proposed. The protocols are pessimistic in that they permit transactions to be updated in exactly one partition. The first protocol is defined for a fully partition-replicated database in which every partition contains a copy of every data object. The second protocol is defined for a partially partition-replicated database in which some objects have no copies in some partitions. Both protocols improve their major performance measures linearly with the backup link speed but are not visibly affected by duration of the partitioning or the database size. >

High performance connected digit recognition using hidden Markov models

The authors use an enhanced analysis feature set consisting of both instantaneous and transitional spectral information and test the hidden-Markov-model (HMM)-based connected-digit recognizer in speaker-trained, multispeaker, and speaker-independent modes. For the evaluation, both a 50-talker connected-digit database recorded over local, dialed-up telephone lines, and the Texas Instruments, 225-adult-talker, connected-digits database are used. Using these databases, the performance achieved was 0.35, 1.65, and 1.75% string error rates for known-length strings, for speaker-trained, multispeaker, and speaker-independent modes, respectively, and 0.78, 2.85, and 2.94% string error rates for unknown-length strings of up to seven digits in length for the three modes. Several experiments were carried out to determine the best set of conditions (e.g., training, recognition, parameters, etc.) for recognition of digits. The results and the interpretation of these experiments are described. >

Remote vital-signs monitor using a dial-up telephone line.

Remote vital-signs monitor using a dial-up telephone line.

On the use of instantaneous and transitional spectral information in speaker recognition

The use of instantaneous and transitional spectral representations of spoken utterances for speaker recognition is investigated. Linear-predictive-coding (LPC)-derived cepstral coefficients are used to represent instantaneous spectral information, and best linear fits of each cepstral coefficient over a specified time window are used to represent transitional information. An evaluation has been carried out using a database of isolated digit utterances over dialed-up telephone lines by 10 talkers. Two vector quantization (VQ) codebooks, instantaneous and transitional, were constructed from each speaker's training utterances. The experimental results show that the instantaneous and transitional representations are relatively uncorrelated, thus providing complementary information for speaker recognition. A rectangular window of approximately 100 ms duration provides an effective estimate of the transitional spectral features for speaker recognition. Also, simple transmission channel variations are shown to affect both the instantaneous spectral representations and the corresponding recognition performance significantly, while the transitional representations and performance are relatively resistant.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

A model-based connected-digit recognition system using either hidden Markov models or templates

Although a great deal of effort has gone into studying large-vocabulary speech-recognition problems, there remains a number of interesting, and potentially exceedingly important, problems which do not require the complexity of these large systems. One such problem is connected-digit recognition, which has applications to telecommunications, order entry, credit-card entry, forms automation, and data-base management, among others. Connected-digit recognition is also an interesting problem for another reason, namely that it is one in which whole-word training patterns are applicable as the basic speech-recognition unit. Thus one can bring to bear all the fundamental speech recognition technology associated with whole-word recognition to solve this problem. As such, several connected digit recognizers have been proposed in the past few years. The performance of these systems has steadily improved to the point where high digit-recognition accuracy is achievable in a speaker-trained mode. In this paper we present a unified system for automatically recognizing fluently spoken digit strings based on whole-word reference units. The system that we will describe can use either hidden Markov model (HMM) technology or template-based technology. In fact the overall system contains features from both approaches. A key factor in the success of the various connected digit recognizers is the ability to derive, via a training procedure, a good set of representations of the behavior of the individual digits in actual connected digit strings. For most applications, isolated digit training does not provide a good enough characterization of the variability of the digits in strings. The “best” training procedure is to derive the digit reference patterns (either templates or statistical models) from connected digit strings. Such a connected word training procedure, based on a segmental k-means loop, has been proposed and was tested on seven experienced users of speech recognizers. For these seven talkers, average string accuracies of greater than 98% for unknown length strings, and greater than 99% for known length strings were obtained on an independent test set of 525 variable length strings (1–7 digits) recorded over local dialed-up telephone lines. To evaluate the performance of the overall connected digit recognizer under more difficult conditions, a set of 50 people (25 men, 25 women), from the non-technical local population, was each asked to record 1200 random digit strings over local dialed-up telephone lines. Both a speaker-trained and a multi-speaker training set was created, and a full performance evaluation was made. Results show that the average string accuracy for unknown- and known-length strings, in the speaker-trained mode, was 98% and 99% respectively; in the multi-speaker mode the average string accuracies were 94% and 96.6% respectively. A complete analysis of these results is given in this paper.

Report: A Voice Password System for Access Security

A voice password system for access security using speaker verification technology has been designed for use over dial-up telephone lines. The voice password system (VPS) can provide secure access to telephone networks, computers, rooms, and buildings. It also has application in office automation systems, electronic funds transfer, and “smart cards” (interactive computers embedded in credit-card-sized packages). As increasing attention is focused on access security in the public, private, and government sectors, the voice password system (Figure 1) can provide a timely solution to the security dilemma.

Gray-Scale Image Coding for Freeze-Frame Videoconferencing

A new gray-scale image coding technique has been developed, in which an extended DPCM approach has been combined with entropy coding. This technique has been implemented in a freeze-frame videoconferencing system which is now operational at IBM sites throughout the world. Following image preprocessing, the two fields of the interlaced 512 x 480 pixel video frame are compressed sequentially with different algorithms. The reconstructed image quality is improved by subsequent image postprocessing, the final reconstructed image being almost indistinguishable from the original image. Typical gray-scale video images compress to about a half bit per pixel and transmit over 4.8 kbit/s dial-up telephone lines in about a half minute. The gray-scale image processing and compression algorithms are described in this paper.

Series/1-based videoconferencing system

Discussed is a new videoconferencing system that has been developed and deployed at several IBM locations. This system transmits high-quality monochrome, freeze-frame images over dial-up telephone lines between two (or three) dedicated video-conferencing rooms. There are two main system components. An IBM Series/1 provides control, communication, data compression, and storage, and Grinnell GMR-270 image processing display system implements image acquisition, processing, and video buffering functions. Conference participants may choose either a basically black and white rendering of an image for fast transmission or a continuous-tone rendering with a longer transmission time. Details are given regarding the system configuration, function, and operation.

Speaker independent connected word recognition using a syntax-directed dynamic programming procedure

A method for speaker independent connected word recognition is described. Speaker independence is achieved by clustering isolated word utterances of a 100 speaker population. Connected word recognition is based on a syntax-directed dynamic programming algorithm which matches the isolated word templates to sentence length utterances. The method has been tested on an artificial task-oriented language based on a 127 word vocabulary. Four subjects, two men and two women, spoke a total of 209 sentences comprising 1750 words. At an average speaking rate of 171 words/min over dialed-up telephone lines, a correct word recognition rate of 97 percent was observed.

Automatic speaker recognition using time alignment of spectrograms

New techniques for automatic speaker recognition from telephone speech are described. The recognition is based on spectral analysis of fixed sentence-long utterances. From the whole utterance the form the time-dependent spectrogram. Normalization procedures are applied to the spectrogram to take account of spectral distortions introduced by the telephone transmission system and of amplitude variations in the utterance. The decision on the speaker's identity is arrived at by comparing the spectrogram of a sample utterance with stored reference spectrograms and calculating a measure of dissimilarity between the spectrograms. To perform this spectrogram comparison, it is necessary to take into account differences in speaking rate and to bring corresponding speech events of the utterances into exact time synchronisation. The details of this so-called time alignment are described with respect to the requirements of speaker recognition. Time alignment is based on a measure of dissimilarity between speech events and is carried out by a dynamic programming algorithm which minimizes timing differences between corresponding speech events. A set of utterances spoken by cooperative speakers and transmitted via conventional dialed-up telephone lines was used for the evaluation of the system. Different versions of the recognition procedure were evaluated and compared. For identification as well as for verification, error rates of 2% or less have been obtained.

Application of dynamic time warping to connected digit recognition

A connected digit recognizer is proposed in which a set of isolated word templates is used as reference patterns and an unconstrained dynamic time warping (DTW) algorithm is used to literally spot the digits in the string. Segmentation boundaries between digits are obtained as the termination point of the dynamic path from the previous time warp. A region around the boundary is searched for the optimum starting point for the succeeding digit. At each stage the recognizer keeps track of a set of candidate digit strings for each test string. The string with the smallest accumulated distance is used as the preliminary string estimate. To help improve the recognition accuracy, two post-correction techniques were applied to the entire set of hypothesized digit strings. One technique creates a reference string by concatenating reference contours of the digits of the string, and comparing this to the test string using a constrained dynamic time warping algorithm. The second technique performs a similar comparison using voiced-unvoiced-silence contours instead of the measured features. Small but consistent improvements in recognition accuracy have been obtained using these techniques for both speaker-trained and speaker-independent systems with digit strings recorded over dialed-up telephone lines. For variable length digit strings of from 2 to 5 digits (where the recognizer was not told the length of the string), word error rates of about 2-3 percent and string error rates on the order of 8 percent were obtained for both speaker-dependent and speaker-independent systems.

The use of slow-scan video for CME in a remote area.

Since August 1977 a slow-scan video system has been operating in a remote area of northwestern Ontario, Canada. This system, using regular dial-up telephone lines, interconnects the local hospital, five remote communities, and two teaching hospitals in Toronto, 1,000 miles distant. Since August 1978 the system has been used on a regular basis for continuing medical education programs, graduate medical education including X-ray rounds, medical rounds, nursing rounds, in-service education, and patient education. These CME programs for physicians have been accepted for study credits by the College of Family Physicians of Canada.

An intelligent terminal for access to a medical data base

Very powerful data base management systems (DBMS) now exist which allow medical personnel access to patient record data bases. DBMSs make it easy to retrieve either complete or abbreviated records of patients with similar characteristics. In addition, statistics on data base records are immediately accessible. However, the price of this power is a large computer with the inherent problems of access, response time, and reliability. If a general purpose, time-shared computer is used to get this power, the response time to a request can be either rapid or slow, depending upon loading by other uses. Furthermore, if the computer is accessed via dial-up telephone lines, there is competition with other users for telephone ports. If either the DBMS or the host machine is replaced, the medical users, who are typically not sophisticated in computer usage, are forced to learn the new system. Microcomputers, because of their low cost and adaptability, lend themselves to a solution of these problems. A microprocessor-based intelligent terminal has been designed and implemented at the USAF School of Aerospace Medicine to provide a transparent interface between the user and his data base. The intelligent terminal system includes multiple microprocessors, floppy disks, a CRT terminal, and a printer. Users interact with the system at the CRT terminal using menu selection (framing). The system translates the menu selections into the query language of the DBMS and handles all actual communication with the DBMS and its host computer, including telephone dialing and sign on procedures, as well as the actual data base query and response. Retrieved information is stored locally for CRT display, hard copy production, and/or permanent retention. Microprocessor-based communication units provide security for sensitive medical data through encryption/decryption algorithms and high reliability error detection transmission schemes. Highly modular software design permits adaptation to a different DBMS and/or host computer with only minor localized software changes. Importantly, this portability is completely transparent to system users. Although the terminal system is independent of the host computer and its DBMS, it has been linked to a UNIVAC 1108 computer supporting MRIs SYSTEM 2000 DBMS.