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Nguyen Le

Bio: Nguyen Le is an academic researcher. The author has contributed to research in topics: Integrated Services Digital Network & User requirements document. The author has an hindex of 2, co-authored 2 publications receiving 125 citations.

Papers
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Journal ArticleDOI
TL;DR: In this paper, the authors address top-down end-to-end user-oriented performance requirements pertaining primarily to voice and digital data services, and consider both traditional and contemporary parameters associated with new and evolving systems.
Abstract: This paper addresses top-down end-to-end user-oriented performance requirements pertaining primarily to voice and digital data services. The discussion of requirements for voice parameters accounts for the performance of existing analog and mixed analog/digital networks, as well as the likely effects on performance of short, medium, and long term evolution toward the ultimate all digital ISDN. The requirements for digital data parameters necessarily reflect an evolutionary process which is less consistent than for voice, and therefore these requirements are less definitive in nature. The discussions of voice and digital data performance apply largely to a wide variety of appropriate network designs, transmission schemes, and switching architectures. Both traditional parameters, as well as contemporary parameters associated with new and evolving systems, are considered. The emphasis is on the performance of nation-wide public and private networks, but the paper also considers the constraints of international connections.

121 citations

01 Dec 1983
TL;DR: Top-down end-to-end user-oriented performance requirements pertaining primarily to voice and digital data services are addressed, and both traditional parameters, as well as contemporary parameters associated with new and evolving systems are considered.
Abstract: This paper addresses top-down end-to-end user-oriented performance requirements pertaining primarily to voice and digital data services. The discussion of requirements for voice parameters accounts for the performance of existing analog and mixed analog/digital networks, as well as the likely effects on performance of short, medium, and long term evolution toward the ultimate all digital ISDN. The requirements for digital data parameters necessarily reflect an evolutionary process which is less consistent than for voice, and therefore these requirements are less definitive in nature. The discussions of voice and digital data performance apply largely to a wide variety of appropriate network designs, transmission schemes, and switching architectures. Both traditional parameters, as well as contemporary parameters associated with new and evolving systems, are considered. The emphasis is on the performance of nation-wide public and private networks, but the paper also considers the constraints of international connections.

4 citations


Cited by
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Journal ArticleDOI
TL;DR: An architectural framework for resilience and survivability in communication networks is provided and a survey of the disciplines that resilience encompasses is provided, along with significant past failures of the network infrastructure.

698 citations

Journal ArticleDOI
TL;DR: This paper presents an ATM-based transport architecture for next-generation multiservices personal communication networks (PCN) that uses a hierarchical ATM switching network for interconnection of PCN microcells based on ATM-compatible cell, relay principles.
Abstract: This paper presents an ATM-based transport architecture for next-generation multiservices personal communication networks (PCN). Such "multimedia capable" integrated services wireless networks are motivated by an anticipated demand for wireless extensions to future broadband networks. An ATM compatible wireless network concept capable of supporting a mix of broadband ISDN services including constant bit-rate (CBR), variable bit-rate (VBR), and packet data transport is explored from an architectural viewpoint. The proposed system uses a hierarchical ATM switching network for interconnection of PCN microcells, each of which is serviced by high-speed, shared-access radio links based on ATM-compatible cell, relay principles. Design issues related to the physical (modulation), media access control (MAC), and data-link layers of the ATM-based radio link are discussed, and preliminary technical approaches are identified in each case. An example multiservice dynamic reservation (MDR) TDMA media access protocol is then considered in further detail, and simulation results are presented for an example voice/data scenario with a proportion of time-critical (i.e., multimedia) packet data. Time-of-expiry (TOE) based queue service disciplines are also investigated as a mechanism for improving the quality-of-service (QoS) in this scenario. >

621 citations

Proceedings ArticleDOI
12 Jun 1994
TL;DR: The authors investigate the performance of four different algorithms for adaptively adjusting the playout delay of audio packets in an interactive packet-audio terminal application, and indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay can achieve a lower rate of lost packets.
Abstract: Recent interest in supporting packet-audio applications over wide area networks has been fueled by the availability of low-cost, toll-quality workstation audio and the demonstration that limited amounts of interactive audio can be supported by today's Internet. In such applications, received audio packets are buffered, and their playout delayed at the destination host in order to compensate for the variable network delays. The authors investigate the performance of four different algorithms for adaptively adjusting the playout delay of audio packets in an interactive packet-audio terminal application, in the face of such varying network delays. They evaluate the playout algorithms using experimentally-obtained delay measurements of audio traffic between several different Internet sites. Their results indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay which were observed in the traces can achieve a lower rate of lost packets for both a given average playout delay and a given maximum buffer size. >

567 citations

Journal ArticleDOI
TL;DR: SpeakSkimmer as discussed by the authors uses speech processing techniques to allow a user to hear recorded sounds quickly, and at several levels of detail, and provides continuous real-time control of the speed and detail level of the audio presentation.
Abstract: Listening to a speech recording is much more difficult than visually scanning a document because of the transient and temporal nature of audio. Audio recordings capture the richness of speech, yet it is difficult to directly browse the stored information. This article describes techniques for structuring, filtering, and presenting recorded speech, allowing a user to navigate and interactively find information in the audio domain. This article describes the SpeechSkimmer system for interactively skimming speech recordings. SpeechSkimmer uses speech-processing techniques to allow a user to hear recorded sounds quickly, and at several levels of detail. User interaction, through a manual input device, provides continuous real-time control of the speed and detail level of the audio presentation. SpeechSkimmer reduces the time needed to listen by incorporating time-compressed speech, pause shortening, automatic emphasis detection, and nonspeech audio feedback. This article also presents a multilevel structural approach to auditory skimming and user interface techniques for interacting with recorded speech. An observational usability test of SpeechSkimmer is discussed, as well as a redesign and reimplementation of the user interface based on the results of this usability test.

253 citations

Journal ArticleDOI
TL;DR: In this article, the authors explore techniques for replacing missing speech with wave-form segments from correctly received packets in order to increase the maximum tolerable missing packet rate in voice communications.
Abstract: Packet communication systems cannot, in general, guarantee accurate and prompt delivery of every packet. The effect of network congestion and transmission impairments on data packets is extended delay; in voice communications these problems lead to lost packets. When some speech packets are not available, the simplest response of a receiving terminal is to substitute silence for the missing speech. Here, we explore techniques for replacing missing speech with wave-form segments from correctly received packets in order to increase the maximum tolerable missing packet rate. After presenting a simple formula for predicting the probability of waveform substitution failure as a function of packet duration and packet loss rate, we introduce two techniques for selecting substitution waveforms. One method is based on pattern matching and the other technique explicitly estimates voicing and pitch. Both approaches achieve substantial improvements in speech quality relative to silence substitution. After waveform substitution, a significant component of the perceived distortion is due to discontinuities at packet boundaries. To reduce this distortion, we introduce a simple smoothing procedure.

247 citations