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International Telemetering Conference Proceedings / October 17-20, 1994 / Town & Country Hotel and Conference Center, San Diego, California / The area of range telemetry and communications has been under budget constraints and interoperability enhancement requirements for some time. The near-term onslaught of multimedia communications offerings by telephony and communications companies is certain to cause range engineering personnel to conduct extensive research and possibly make numerous decisions on procurements and technologies before standards are finalized. This paper will address a low-risk migration path for range telemetry to the new multimedia communications for ranges based on current capabilities. This migration path has an end goal of positioning the ranges to take advantage of future multimedia communications as they become available, while leveraging off of current products and procurements, without a major investment.

Der Asynchronous Transfer Mode (ATM) stellt die zukünftige und einheitliche Basistechnologie für das Breitband-ISDN dar. Da nahezu alle wesentlichen Protokollfunktionen in Hardware realisierbar sind, soll nachfolgend ein Überblick über bereits angebotene VLSI-Schaltkreise gegeben werden. Eine Systematisierung und Einordnung vorhandener ATM-Chips hinsichtlich ihrer Leistungsfähigkeit und ihres Funktionsumfangs erfolgt in Hinblick auf das sogenannte B-ISDN-Referenzmodell. Dieses Schichtenmodell definiert die notwendigen Protokolle und Schnittstellen für den Asynchronous Transfer Mode. Zum grundlegenden Verständnis wird einleitend eine kurze Einführung in die Basisprinzipien von ATM gegeben.

Future wireless networks will have to handle varying combinations of multimedia traffic that present the network with numerous quality of service (QoS) requirements. The continuously growing demand for mobile phones has resulted in radio spectrum becoming a precious resource that cannot be wasted. The current second-generation mobile networks are designed for voice communication and, even with the enhancements being implemented to accommodate data, they cannot efficiently handle the multimedia traffic demands that will be introduced in the near future. This thesis begins with a survey of existing wireless ATM (WATM) protocols, followed by an examination of some medium access control (MAC) protocols, supporting multimedia traffic, and based on code division multiple access (CDMA) physical layers. A WATM link layer protocol based on a CDMA physical layer, and incorporating techniques from some of the surveyed protocols, is then proposed. The MAC protocol supports a wide range of service requirements by utilising a flexible scheduling algorithm that takes advantage of the graceful degradation of CDMA with increasing user interference to schedule cells for transmission according to their maximum bit error rate (BER) requirements. The data link control (DLC) accommodates the various traffic types by allowing virtual channels (VCs) to make use of forward error correction (FEc) or retransmission techniques. The proposed link layer protocol has been implemented on a Blue Wave Systems DSP board that forms part of Alcatel Altech Telecoms' software radio platform. The details and practicality of the implementation are presented. A simulation model for the protocol has been developed using MIL3 's Opnet Modeler. Hence, both simulated and measured performance results are presented before the thesis concludes with suggestions for improvements and future work. / Thesis (M.Sc.)-University of Natal, Durban, 2000.

The characteristics of service independence and flexibility of ATM networks make the control problems of such networks very critical. One of the main challenges in ATM networks is to design traffic control mechanisms that enable both economically efficient use of the network resources and desired quality of service to higher layer applications. Window flow control mechanisms of traditional packet switched networks are not well suited to real time services, at the speeds envisaged for the future networks. In this work, the utilisation of the Probability of Congestion (PC) as a bandwidth decision parameter is presented. The validity of PC utilisation is compared with QOS parameters in buffer-less environments when only the cell loss ratio (CLR) parameter is relevant. The convolution algorithm is a good solution for CAC in ATM networks with small buffers. If the source characteristics are known, the actual CLR can be very well estimated. Furthermore, this estimation is always conservative, allowing the retention of the network performance guarantees. Several experiments have been carried out and investigated to explain the deviation between the proposed method and the simulation. Time parameters for burst length and different buffer sizes have been considered. Experiments to confine the limits of the burst length with respect to the buffer size conclude that a minimum buffer size is necessary to achieve adequate cell contention. Note that propagation delay is a no dismiss limit for long distance and interactive communications, then small buffer must be used in order to minimise delay. Under previous premises, the convolution approach is the most accurate method used in bandwidth allocation. This method gives enough accuracy in both homogeneous and heterogeneous networks. But, the convolution approach has a considerable computation cost and a high number of accumulated calculations. To overcome this drawbacks, a new method of evaluation is analysed: the Enhanced Convolution Approach (ECA). In ECA, traffic is grouped in classes of identical parameters. By using the multinomial distribution function instead of the formula-based convolution, a partial state corresponding to each class of traffic is obtained. Finally, the global state probabilities are evaluated by multi-convolution of the partial results. This method avoids accumulated calculations and saves storage requirements, specially in complex scenarios. Sorting is the dominant factor for the formula-based convolution, whereas cost evaluation is the dominant factor for the enhanced convolution. A set of cut-off mechanisms are introduced to reduce the complexity of the ECA evaluation. The ECA also computes the CLR for each j-class of traffic (CLRj), an expression for the CLRj evaluation is also presented. We can conclude that by combining the ECA method with cut-off mechanisms, utilisation of ECA in real-time CAC environments as a single level scheme is always possible.

Asynchronous Transfer Mode (ATM) Adaptation Layer Type 2 (AAL2) has been developed to carry low and variable bit rate traffic. It provides high bandwidth efficiency with low packing delay by allowing voice traffic from different AAL2 channels to be multiplexed onto a single ATM virtual channel connection. Examples of where AAL2 are used include the Code Division Multiple Access and the Third Generation mobile telephony networks. The main objective of this thesis is to study traditional and novel AAL2 multiplexing methods and to characterise their performance when carrying low and variable bit rate (VBR) voice traffic. This work develops a comprehensive QoS framework which is used as a basis to study the performance of the AAL2 multiplexer system. In this QoS framework the effects of packet delay, delay variation, subjective voice quality and bandwidth utilisation are all used to determine the overall performance of the end-to-end system for the support of real time voice communications. Extensions to existing AAL2 voice multiplexers are proposed and characterised. In the case where different types of voice applications are presented to the AAL2 multiplexer, it was observed that increased efficiency gains are possible when a priority queuing scheme is introduced into the traditional AAL2 multiplexer system. Studies of the voice traffic characteristics and their effects on the performance of the AAL2 multiplexer are also investigated. It is shown that particular source behaviours can have deleterious effect on the performance of the AAL2 multiplexer. Methods of isolating these voice sources are examined and the performance of the AAL2 multiplexer re-evaluated to show the beneficial effects of a particular source isolation technique. The extent to which statistical multiplexing is possible for real time variable VBR sources is theoretically examined. These calculations highlight the difficulties in multiplexing VBR real time traffic while maintaining guaranteed delay bounds for these sources. Based on these calculations, multiplexing schemes that incorporate data transfers within the real time traffic transfer are proposed as alternatives for utilising unused bandwidth caused by the VBR nature of the voice traffic.

Thesis (Ph.D.) - Carleton University, 2006. / Includes bibliographical references (p. 139-144). Also available in electronic format on the Internet.

Thesis (M. Eng.), Memorial University of Newfoundland, 1998. / Bibliography: leaves 107-112.

Thesis (M. Eng.)--Carleton University, 1999. / Also available in electronic format on the Internet.

D.Ing. / Conventional traffic models used for the analysis of packet-switched data are Markovian in nature and are based on assumptions, such as Poissonian arrivals. The introduction of packet oriented networks has resulted in an influx of information highlighting numerous discrepancies from these assumptions. Several studies have shown that traffic patterns from diverse packet-switched networks and services exhibit the presence of properties such as self-similarity, long-range dependencies, slowly decaying variances, "heavy tailed" or power law distributions, and fractal structures. Heavy Tailed distributions decay slower than predicted by conventional exponential assumptions and lead to significant underestimation of network traffic variables. Furthermore, it was shown that the statistical multiplexing of multiple packet-switched sources do not give rise to a more homogenous aggregate, but that properties such as burstiness are conserved. The results of the above mentioned studies have shown that none of the commonly used traffic models and assumptions are able to completely capture the bursty behaviour of packet- and cellbased networks. Artificial Intelligent methods provide the capability to extract the inherent characteristics of a system and include soft decision-making approaches such as Fuzzy Logic. Adaptive methods such as Fuzzy Logic Self-learning algorithms have the potential to solve some of the most pressing problems of traffic Modelling and Management in modern packet-switched networks. This dissertation is concerned with providing alternative solutions to the mentioned problems, in the following three sub-sections; the Description of Heavy Tailed Arrival Distributions, Timeseries Forecasting of bursty Traffic Intensities, and Management related Soft Decision-Making. Although several alternative methods, such as Kalman Filters, Bayesian Distributions, Fractal Analysis and Neural Networks are considered, the main emphasis of this work is on Fuzzy Logic applications.