14 research outputs found

    Synergy between adaptive channel coding and media access control for wireless ATM

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    In this paper, we propose mechanisms to exploit the synergy between the Media Access Control (MAC) layer and the physical layer for wireless ATM applications. For simplicity, the system considered consists of a single server and a single wireless ATM terminal. A number of virtual circuit connections (VC), with varying Quality of Service (QoS) requirements, are supported. We focus on two components of the wireless ATM system, namely the channel encoder and the ATM scheduler in the MAC layer. The channel encoder is responsible for protecting the ATM cells over the hostile radio channel while the ATM scheduler is responsible for allocating limited resources to the ATM cells for each virtual connection so as to meet the specified QoS requirements. We consider two configurations, namely System-I with isolated adaptive channel encoder and ATM scheduler, and System-II with bi-directional information exchange. It is found that significant performance improvement on the Cell Loss Rate (CLR) and the Mean Cell Delay (MCD) could be achieved for systems exploiting the synergy. © 1999 IEEE.published_or_final_versio

    Statistical multiplexing of identical bursty sources in an ATM network

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    Load control in atm networks

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    An Adaptive Scheme for Admission Control in ATM Networks

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    This paper presents a real time front-end admission control scheme for ATM networks. A call management scheme which uses the burstiness associated with traffic sources in a heterogeneous ATM environment to effect dynamic assignment of bandwidth is presented. In the proposed scheme, call acceptance is based on an on-line evaluation of the upper bound on cell loss probability which is derived from the estimated distribution of the number of calls arriving. Using this scheme, the negotiated quality of service will be assured when there is no estimation error. The control mechanism is effective when the number of calls is large, and tolerates loose bandwidth enforcement and loose policing control. The proposed approach is very effective in the connection oriented transport of ATM networks where the decision to admit new traffic is based on thea priori knowledge of the state of the route taken by the traffic

    Rethinking Mobile Telephony with the IMP

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    Design, analysis and implementation of integrated services networks.

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    by Wong, Chan-foon.Thesis (M.Phil.)--Chinese University of Hong Kong, 1993.Includes bibliographical references (leaves 59-67 (1st gp.)).Chapter Chapter I --- Introduction --- p.1Chapter 1.1 --- Background --- p.1Chapter 1.2 --- Traffic Characteristics --- p.2Chapter 1.3 --- Related Works --- p.5Chapter Chapter II --- Integrated Services Protocol (ISP) --- p.7Chapter 2.1 --- Ethernet --- p.7Chapter 2.2 --- ISP Description --- p.9Chapter 2.2.1 --- Voice Communications Characteristics --- p.9Chapter 2.2.2 --- Voice Packet Format --- p.12Chapter 2.2.3 --- Call Management --- p.13Chapter 2.4.4 --- Voice Packet Transmission Protocol --- p.14Chapter 2.4.5 --- Error Handling --- p.16Chapter Chapter III --- Protocol Studies --- p.17Chapter 3.1 --- Simulation Model And Parameters --- p.17Chapter 3.2 --- Voice Loss --- p.18Chapter 3.3 --- Data Delay --- p.20Chapter 3.4 --- Maximum Number Of Active Voice Stations --- p.22Chapter 3.5 --- Summary --- p.23Chapter Chapter IV --- Implementation --- p.24Chapter 4.1 --- System Platform --- p.24Chapter 4.2 --- Integrated Services Adapter (ISA) --- p.25Chapter 4.2.1 --- Hardware Design --- p.26Chapter 4.3 --- Voice on Ethernet Adapter (VEA) --- p.29Chapter 4.3.1 --- Hardware Design --- p.29Chapter 4.3.2 --- Software Design --- p.31Chapter 4.3.2.1 --- Programming The VEA --- p.32Chapter 4.3.2.2 --- Software Development Under DOS --- p.35Chapter 4.3.2.3 --- Software Development Under Linux --- p.37Chapter 4.4 --- Summary --- p.41Chapter Chapter V --- Implementation Results --- p.42Chapter 5.1 --- Frequency Response --- p.43Chapter 5.2 --- Distortion --- p.44Chapter 5.3 --- Amplification and Linearity --- p.45Chapter 5.4 --- Voice Quality With Different Voice Packet Sizes --- p.46Chapter 5.5 --- Voice Loss Under Various Data Loadings --- p.47Chapter Chapter VI --- Implementation Experiences --- p.49Chapter 6.1 --- CPU Bottle-neck --- p.49Chapter 6.2 --- Data Bus Bottle-neck --- p.50Chapter 6.3 --- Operating System --- p.50Chapter Chapter VII --- Future Works --- p.52Chapter 7.1 --- Enhancement of ISA --- p.52Chapter 7.2 --- Extensions To Other Networks --- p.53Chapter 7.3 --- A New Architecture For Future Multimedia Workstation --- p.54Chapter Chapter VIII --- Conclusions --- p.57Bibliography --- p.59Appendices --- p.A.lAppendix A: Detailed Circuit Designs --- p.A.2Appendix B: Detailed Software Designs --- p.A.5Appendix C: Schematic Diagrams --- p.A.15Appendix D: Program Listings --- p.A.2

    Network resource allocation for bursty ATM traffic sources

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    Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1990.Title as it appears in the M.I.T. Graduate List, June 1990: Resource allocation for bursty traffic sources in Asynchronous Transfer Mode networks.Includes bibliographical references (leaves 88-90).by Anuradha Vedantham.M.S

    VOICE OVER IP NETWORKS: QUALITY OF SERVICE, PRICING AND SECURITY

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    The growth of the Internet over the past decade together with the promise of lower costs to the customer has led to the rapid emergence of Voice over Internet Protocol (VoIP). As a real-time application in large scale packet switched networks, VoIP networks face many challenges such as availability, voice quality and network security. This dissertation addresses three important issues in VoIP networks: Quality of Service, pricing and security.In addressing Quality of Service (QoS), this dissertation introduces the notion of delay not exceeding an upper limit, termed the bounded delay (rather than the average delay), to measure the Quality of Service in VoIP networks. Queuing models are introduced to measure performance in terms of bounded delays. Closed form solutions relating the impact of bounding delays on throughput of VoIP traffic are provided. Traffic that exceeds the delay threshold is treated as lost throughput. The results addressed can be used in scaling resources in a VoIP network for different thresholds of acceptable delays. Both single and multiple switching points are addressed. The same notion and analysis are also applied on jitter, another important indicator of the VoIP QoSThis dissertation also develops a pricing model based on the Quality of Service provided in VoIP networks. It presents the impact of quality of VoIP service demanded by the customer on the transmission resources required by the network using an analytical approach. The price to be paid by the customer is based on the throughput meeting this criterion and the network transmission resources required. In particular, the impact of Quality of Service presented can be used in the design of VoIP networks in a way that would provide fairness to the user in terms of quality of service and price while optimizing the resources of the network at the same time.This dissertation also extends and applies the delay throughput analysis developed for VoIP networks in assessing the impact of risks constituted by a number of transportation channels, where the risk associated with each channel can be quantified by a known distribution. For VoIP security, this dissertation mainly focuses on the signaling authentication. It presents a networking solution that incorporates network-based authentication as an inherent feature. The authentication feature that we propose introduces a range of flexibilities not available in the PSTN. Since most calls will likely terminate on the network of another service provider, we also present a mechanism using which networks can mutually authenticate each other to afford the possibility of authentication across networks. Finally, this dissertation explores areas for future research that can be built on the foundation of research presented
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