5 research outputs found

    Asymptotic Behavior of the Number of Lost Messages

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    The goal of the paper is to study asymptotic behavior of the number of lost messages. Long messages are assumed to be divided into a random number of packets which are transmitted independently of one another. An error in transmission of a packet results in the loss of the entire message. Messages arrive to the M/GI/1M/GI/1 finite buffer model and can be lost in two cases as either at least one of its packets is corrupted or the buffer is overflowed. With the parameters of the system typical for models of information transmission in real networks, we obtain theorems on asymptotic behavior of the number of lost messages. We also study how the loss probability changes if redundant packets are added. Our asymptotic analysis approach is based on Tauberian theorems with remainder.Comment: 18 pages, The list of references and citations slightly differ from these appearing in the journa

    Packet loss characteristics for M/G/1/N queueing systems

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    In this contribution we investigate higher-order loss characteristics for M/G/1/N queueing systems. We focus on the lengths of the loss and non-loss periods as well as on the number of arrivals during these periods. For the analysis, we extend the Markovian state of the queueing system with the time and number of admitted arrivals since the instant where the last loss occurred. By combining transform and matrix techniques, expressions for the various moments of these loss characteristics are found. The approach also yields expressions for the loss probability and the conditional loss probability. Some numerical examples then illustrate our results

    A system for improving the quality of real-time services on the internet

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    Real-time Internet services are becoming more popular every day, and Voice over Internet Protocol (VOIP) is arguably the most popular of these, despite the quality and reliability problems that are so characteristic of VOIP. This thesis proposes to apply a routing technique called Fully Redundant Dispersity Routing to VOIP and shows how this mitigates these problems to deliver a premium service that is more equal to traditional telephony than VOIP is currently.Fully Redundant Dispersity Routing uses the path diversity readily available in the Internet to route complete copies of the data to be communicated over multiple paths. This allows the effect of a failure on a path to be reduced, and possibly even masked completely, by the other paths. Significantly, rather than expecting changes of the Internet that will improve real-time service quality, this approach simply changes the manner in which real-time services use the Internet, leaving the Internet itself to stay the way it is.First, real VOIP traffic in a commercial call centre is measured (1) to establish a baseline of current quality characteristics against which the effects of Fully Redundant Dispersity Routing may be measured, and (2) as a source of realistic path characteristics. Simulations of various Fully Redundant Dispersity Routing systems that adopt the measured VOIP traffic characteristics then (1) show how this routing technique mitigates quality and reliability problems, and (2) quantify the quality deliverable with the VOIP traffic characteristics measured. For example, quantifying quality as a Mean Opinion Score (MOS) estimated from the measurements with the International Telecommunication Union’s E-model, slightly more than 1 in every 23 of the VOIP telephone calls measured in the call centre is likely to be perceived to be of a quality with which humans would be less than very satisfied. Simulations carried out for this thesis show that using just two paths adopting the same measurements, Fully Redundant Dispersity Routing may increase quality to reduce that proportion to slightly less than 1 in every 10 000 VOIP telephone calls

    Dynamic information and constraints in source and channel coding

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    Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2004.This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.Includes bibliographical references (p. 237-251).This thesis explore dynamics in source coding and channel coding. We begin by introducing the idea of distortion side information, which does not directly depend on the source but instead affects the distortion measure. Such distortion side information is not only useful at the encoder but under certain conditions knowing it at the encoder is optimal and knowing it at the decoder is useless. Thus distortion side information is a natural complement to Wyner-Ziv side information and may be useful in exploiting properties of the human perceptual system as well as in sensor or control applications. In addition to developing the theoretical limits of source coding with distortion side information, we also construct practical quantizers based on lattices and codes on graphs. Our use of codes on graphs is also of independent interest since it highlights some issues in translating the success of turbo and LDPC codes into the realm of source coding. Finally, to explore the dynamics of side information correlated with the source, we consider fixed lag side information at the decoder. We focus on the special case of perfect side information with unit lag corresponding to source coding with feedforward (the dual of channel coding with feedback).(cont.) Using duality, we develop a linear complexity algorithm which exploits the feedforward information to achieve the rate-distortion bound. The second part of the thesis focuses on channel dynamics in communication by introducing a new system model to study delay in streaming applications. We first consider an adversarial channel model where at any time the channel may suffer a burst of degraded performance (e.g., due to signal fading, interference, or congestion) and prove a coding theorem for the minimum decoding delay required to recover from such a burst. Our coding theorem illustrates the relationship between the structure of a code, the dynamics of the channel, and the resulting decoding delay. We also consider more general channel dynamics. Specifically, we prove a coding theorem establishing that, for certain collections of channel ensembles, delay-universal codes exist that simultaneously achieve the best delay for any channel in the collection. Practical constructions with low encoding and decoding complexity are described for both cases.(cont.) Finally, we also consider architectures consisting of both source and channel coding which deal with channel dynamics by spreading information over space, frequency, multiple antennas, or alternate transmission paths in a network to avoid coding delays. Specifically, we explore whether the inherent diversity in such parallel channels should be exploited at the application layer via multiple description source coding, at the physical layer via parallel channel coding, or through some combination of joint source-channel coding. For on-off channel models application layer diversity architectures achieve better performance while for channels with a continuous range of reception quality (e.g., additive Gaussian noise channels with Rayleigh fading), the reverse is true. Joint source-channel coding achieves the best of both by performing as well as application layer diversity for on-off channels and as well as physical layer diversity for continuous channels.by Emin Martinian.Ph.D
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