130 research outputs found

    Quality aspects of Internet telephony

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    Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

    Analysing the characteristics of VoIP traffic

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    In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

    Multiple access protocols for multichannel communication systems

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    Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2000.Includes bibliographical references (leaves 108-111).System architecture design, evaluation, and optimization are key issues to developing communication systems that meet the requirements of today and expectations of the future. In this thesis, we introduce the concept of multiple access communication and the need to use efficient transmission techniques to expand both present and future wireless communication networks. We will study two areas regarding multiple access on multichannel communication systems. First, we describe fundamental multiplexing techniques that we can build upon and investigate the performance of different candidate architectures for the transmission of messages from bursty sources on multiple channels. We will consider traditional protocols such as Time Division Multiple Access (TDMA) and Slotted ALOHA (S-ALOHA) alongside a channelized architecture, which is based on the idea of multiplexing by dividing total transmission capacity into a fixed number of frequency channels. We develop mathematical models that describe the overall delay for sending large messages of a fixed length arriving from bursty sources and analyze their performances. We will make real-world parameter assumptions in the context of wireless networks and analyze the performance to develop intuition about the effectiveness of the different architectures. Second, we will investigate channel capacity allocation among mixed traffic, i.e., multiple classes of users. We will consider a first-come first-serve (FCFS) access strategy, a non-preemptive priority scheme, a preemptive resume priority scheme, and several channel capacity allocation schemes. We develop models that describe the overall delay for sending messages and analyze their performance. Our focus will concentrate on two classes of users. This scenario is typical of classes of users with small and large messages to transmit. present quantitative results by making real-world parameter assumptions in the context of wireless networks and analyze the performance to develop intuition about the effectiveness of each architecture.by Serena Chan.M.Eng

    Distributed call set-up algorithms in BISDN environment.

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    by Shum Kam Hong.Thesis (M.Phil.)--Chinese University of Hong Kong, 1992.Includes bibliographical references (leaves 125-131).Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Background --- p.1Chapter 1.2 --- Outline of the thesis --- p.6Chapter 1.3 --- Current Art in Packet Switching --- p.9Chapter 2 --- Management of Control Information --- p.17Chapter 2.1 --- Inter-node Exchange of Link Congestion Status --- p.21Chapter 2.2 --- Consistency of Control Information --- p.24Chapter 2.3 --- Alternate Format of Control Information --- p.26Chapter 3 --- Traffic Flow Control --- p.29Chapter 3.1 --- Control of Traffic Influx into the Network --- p.29Chapter 3.2 --- Control of Traffic Loading from the Node --- p.30Chapter 3.3 --- Flow Control for Connection Oriented Traffic --- p.32Chapter 3.4 --- Judgement of Link Status --- p.38Chapter 3.5 --- Starvation-free and Deadlock-free --- p.42Chapter 4 --- Call Set-up Algorithm Traffic Modelling --- p.47Chapter 4.1 --- Basic Algorithm --- p.47Chapter 4.2 --- Minimization of Bandwidth Overhead --- p.48Chapter 4.3 --- Two-way Transmission --- p.51Chapter 4.4 --- Traffic Modelling --- p.52Chapter 4.4.1 --- Aggregate Traffic Models --- p.53Chapter 4.4.2 --- Traffic Burstiness --- p.57Chapter 5 --- Parameters Tuning and Analysis --- p.76Chapter 5.1 --- Scheme I : Scout Pumping --- p.76Chapter 5.2 --- Scheme II : Speed-up Scout Pumping --- p.85Chapter 5.3 --- Blocking Probability --- p.90Chapter 5.4 --- Scout Stream Collision --- p.92Chapter 6 --- Simulation Modelling & Performance Evaluation --- p.96Chapter 6.1 --- The Network Simulator --- p.96Chapter 6.1.1 --- Simulation Event Scheduling --- p.97Chapter 6.1.2 --- Input Traffic Regulation --- p.100Chapter 6.1.3 --- Actual Offered Load --- p.101Chapter 6.1.4 --- Static and Dynamic Parameters --- p.103Chapter 6.2 --- Simulation Results --- p.107Chapter 7 --- Conclusions --- p.123Chapter A --- List of Symbols --- p.13

    Queueing Systems with Heavy Tails

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    Optimisation of stochastic networks with blocking: a functional-form approach

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    This paper introduces a class of stochastic networks with blocking, motivated by applications arising in cellular network planning, mobile cloud computing, and spare parts supply chains. Blocking results in lost revenue due to customers or jobs being permanently removed from the system. We are interested in striking a balance between mitigating blocking by increasing service capacity, and maintaining low costs for service capacity. This problem is further complicated by the stochastic nature of the system. Owing to the complexity of the system there are no analytical results available that formulate and solve the relevant optimization problem in closed form. Traditional simulation-based methods may work well for small instances, but the associated computational costs are prohibitive for networks of realistic size. We propose a hybrid functional-form based approach for finding the optimal resource allocation, combining the speed of an analytical approach with the accuracy of simulation-based optimisation. The key insight is to replace the computationally expensive gradient estimation in simulation optimisation with a closed-form analytical approximation that is calibrated using a single simulation run. We develop two implementations of this approach and conduct extensive computational experiments on complex examples to show that it is capable of substantially improving system performance. We also provide evidence that our approach has substantially lower computational costs compared to stochastic approximation
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