A discrete-time Markov modulated queuing system with batched arrivals

This paper examines a discrete-time queuing system with applications to telecommunications traffic. The arrival process is a particular Markov modulated process which belongs to the class of discrete batched Markovian arrival processes. The server process is a single server deterministic queue. A closed form exact solution is given for the expected queue length and delay. A simple system of equations is given for the probability of the queue exceeding a given length.Comment: to appear Performance Evaluatio

STOCHASTIC MODELING AND TIME-TO-EVENT ANALYSIS OF VOIP TRAFFIC

Voice over IP (VoIP) systems are gaining increased popularity due to the cost effectiveness, ease of management, and enhanced features and capabilities. Both enterprises and carriers are deploying VoIP systems to replace their TDM-based legacy voice networks. However, the lack of engineering models for VoIP systems has been realized by many researchers, especially for large-scale networks. The purpose of traffic engineering is to minimize call blocking probability and maximize resource utilization. The current traffic engineering models are inherited from the legacy PSTN world, and these models fall short from capturing the characteristics of new traffic patterns. The objective of this research is to develop a traffic engineering model for modern VoIP networks. We studied the traffic on a large-scale VoIP network and collected several billions of call information. Our analysis shows that the traditional traffic engineering approach based on the Poisson call arrival process and exponential holding time fails to capture the modern telecommunication systems accurately. We developed a new framework for modeling call arrivals as a non-homogeneous Poisson process, and we further enhanced the model by providing a Gaussian approximation for the cases of heavy traffic condition on large-scale networks. In the second phase of the research, we followed a new time-to-event survival analysis approach to model call holding time as a generalized gamma distribution and we introduced a Call Cease Rate function to model the call durations. The modeling and statistical work of the Call Arrival model and the Call Holding Time model is constructed, verified and validated using hundreds of millions of real call information collected from an operational VoIP carrier network. The traffic data is a mixture of residential, business, and wireless traffic. Therefore, our proposed models can be applied to any modern telecommunication system. We also conducted sensitivity analysis of model parameters and performed statistical tests on the robustness of the models’ assumptions. We implemented the models in a new simulation-based traffic engineering system called VoIP Traffic Engineering Simulator (VSIM). Advanced statistical and stochastic techniques were used in building VSIM system. The core of VSIM is a simulation system that consists of two different simulation engines: the NHPP parametric simulation engine and the non-parametric simulation engine. In addition, VSIM provides several subsystems for traffic data collection, processing, statistical modeling, model parameter estimation, graph generation, and traffic prediction. VSIM is capable of extracting traffic data from a live VoIP network, processing and storing the extracted information, and then feeding it into one of the simulation engines which in turn provides resource optimization and quality of service reports

Performance Evaluation of Transition-based Systems with Applications to Communication Networks

Since the beginning of the twenty-first century, communication systems have witnessed a revolution in terms of their hardware capabilities. This transformation has enabled modern networks to stand up to the diversity and the scale of the requirements of the applications that they support. Compared to their predecessors that primarily consisted of a handful of homogeneous devices communicating via a single communication technology, today's networks connect myriads of systems that are intrinsically different in their functioning and purpose. In addition, many of these devices communicate via different technologies or a combination of them at a time. All these developments, coupled with the geographical disparity of the physical infrastructure, give rise to network environments that are inherently dynamic and unpredictable. To cope with heterogeneous environments and the growing demands, network units have taken a leap from the paradigm of static functioning to that of adaptivity. In this thesis, we refer to adaptive network units as transition-based systems (TBSs) and the act of adapting is termed as transition. We note that TBSs not only reside in diverse environment conditions, their need to adapt also arises following different phenomena. Such phenomena are referred to as triggers and they can occur at different time scales. We additionally observe that the nature of a transition is dictated by the specified performance objective of the relevant TBS and we seek to build an analytical framework that helps us derive a policy for performance optimization. As the state of the art lacks a unified approach to modelling the diverse functioning of the TBSs and their varied performance objectives, we first propose a general framework based on the theory of Markov Decision Processes. This framework facilitates optimal policy derivation in TBSs in a principled manner. In addition, we note the importance of bespoke analyses in specific classes of TBSs where the general formulation leads to a high-dimensional optimization problem. Specifically, we consider performance optimization in open systems employing parallelism and closed systems exploiting the benefits of service batching. In these examples, we resort to approximation techniques such as a mean-field limit for the state evolution whenever the underlying TBS deals with a large number of entities. Our formulation enables calculation of optimal policies and provides tangible alternatives to existing frameworks for Quality of Service evaluation. Compared to the state of the art, the derived policies facilitate transitions in Communication Systems that yield superior performance as shown through extensive evaluations in this thesis

Packet level measurement over wireless access

PhDPerformance Measurement of the IP packet networks mainly comprise of monitoring the network performance in terms of packet losses and delays. If used appropriately, these network parameters (i.e. delay, loss and bandwidth etc) can indicate the performance status of the network and they can be used in fault and performance monitoring, network provisioning, and traffic engineering. Globally, there is a growing need for accurate network measurement to support the commercial use of IP networks. In wireless networks, transmission losses and communication delays strongly affect the performance of the network. Compared to wired networks, wireless networks experience higher levels of data dropouts, and corruption due to issues of channel fading, noise, interference and mobility. Performance monitoring is a vital element in the commercial future of broadband packet networking and the ability to guarantee quality of service in such networks is implicit in Service Level Agreements. Active measurements are performed by injecting probes, and this is widely used to determine the end to end performance. End to end delay in wired networks has been extensively investigated, and in this thesis we report on the accuracy achieved by probing for end to end delay over a wireless scenario. We have compared two probing techniques i.e. Periodic and Poisson probing, and estimated the absolute error for both. The simulations have been performed for single hop and multi- hop wireless networks. In addition to end to end latency, Active measurements have also been performed for packet loss rate. The simulation based analysis has been tried under different traffic scenarios using Poisson Traffic Models. We have sampled the user traffic using Periodic probing at different rates for single hop and multiple hop wireless scenarios. 5 Active probing becomes critical at higher values of load forcing the network to saturation much earlier. We have evaluated the impact of monitoring overheads on the user traffic, and show that even small amount of probing overhead in a wireless medium can cause large degradation in network performance. Although probing at high rate provides a good estimation of delay distribution of user traffic with large variance yet there is a critical tradeoff between the accuracy of measurement and the packet probing overhead. Our results suggest that active probing is highly affected by probe size, rate, pattern, traffic load, and nature of shared medium, available bandwidth and the burstiness of the traffic

Lindley-type recursions

In dit proefschrift staat de volgende Lindley-achtige recursie centraal: Wn+1 = max{0,Bn+1 - An -Wn}. (1) Deze "niet-stijgende" recursie is belangrijk in de analyse van systemen waarbij een bediende alterneert tussen twee bedieningsstations. Een station biedt ruimte voor ´e´en klant. De bediende alterneert tussen beide stations en bediend ´e´en klant per keer. Aangenomen wordt dat voortdurend bij beide stations klanten staan te wachten. Zodra een wachtende klant een station betreed, begint de eerste fase van zijn bediening, die bestaat uit een voorbereidende fase. De bediende is hier niet bij betrokken: pas nadat de voorbereidende fase is afgerond kan een klant aan de tweede fase van zijn bediening beginnen, welke wordt uitgevoerd door de bediende. Dus de eigenlijke bediening bestaat alleen uit de tweede fase. Het kan voorkomen dat de bediende moet wachten totdat de voorbereiding van de volgende klant is afgelopen. We zijn dan ook ge¨interesseerd in de wachttijd van de bediende. Als Bn de voorbereidingstijd is voor de n-de klant en An de bedieningstijd is van de n-de klant, dan kan de wachttijd van de bediende voor de (n + 1)-ste klant beschreven worden door middel van Recursie (1). Een belangrijke observatie is dat deze recursie vrijwel identiek is aan Lindley’s recursie. Het enige verschil is het min-teken voor Wn. Dit model is gemotiveerd door diverse toepassingen waarvan er twee worden besproken in Hoofdstuk 1. De eerste toepassing betreft oog-operaties. De tweede toepassing is gerelateerd aan carousel systemen. Dit soort systemen zijn uitgebreid bestudeerd; Sectie 1.3 geeft een literatuuroverzicht. Verderop in dit hoofdstuk geven we een gedetailleerde modelbeschrijving en noemen we enkele verschillen tussen de analyse van dit model en het standaard wachtrijmodel. Hoofdstuk 2 bestudeert enkele algemene eigenschappen van Recursie (1), zoals de stabiliteit van het systeem, existentie van een evenwichtsverdeling, convergentie naar deze verdeling als n naar oneindig gaat en het staartgedrag en de covariantie functie van de verdeling van de wachttijd van de bediende. Een rode draad in dit proefschrift is de afleiding van de evenwichtsverdeling van de wachttijd van de bediende. In de volgende drie hoofdstukken leiden we deze verdeling af onder diverse aannames over de verdeling van de voorbereidingstijd en bedieningstijd van een generieke klant. We bestuderen gevallen die analoog zijn aan de klassieke M/G/1, G/PH/1 en PH/P/1 wachtrijmodellen, waarbij "P" staat voor polynomiale verdelingen. Ge¨inspireerd door de toepassingen van ons model, bekijken we enkele prestatiematen voor dit systeem, zoals de doorzet. Dit maakt een vergelijk met de prestatie van niet-alternerende systemen mogelijk. In Hoofdstuk 6 onderzoeken we methoden om de wachttijdverdeling te benaderen door de verdeling van de voorbereidingstijd of bedieningstijd te benaderen met een verdeling die exacte berekeningen mogelijk maakt. We beschrijven hoe zo’n verdel- ing kan worden gevonden en we geven een bovengrens voor de fout tussen de werkelijke wachttijdverdeling en zijn benadering. In alle voorgaande hoofdstukken hebben we aangenomen dat alle voorbereidingstijden en bedieningstijden onafhankelijk van elkaar zijn. In Hoofdstuk 7 laten we deze aanname vallen. We onderzoeken twee specifieke vormen van afhankelijkheid tussen deze variabelen. Voor beide vormen leiden we opnieuw de limietverdeling af van de wachttijd van de bediende. Hoofdstuk 8 analyseert een recursie welke een uitbreiding is van zowel Lindley’s recursie als (1). We bekijken, namelijk, de recursie Wn+1 = max{0,Bn+1 - An + YnWn}, met Yn een stochastische variabele die zowel de waarde 1 als -1 kan aannemen. Voor deze recursie onderzoeken we stabiliteit, en we berekenen de limietverdeling in twee specifieke gevallen, waarmee we de bestaande theorie voor Lindley’s recursie en Recursie (1) generaliseren. De analyse maakt duidelijk dat de technieken voor het analyseren van (1) en voor het analyseren Lindley’s recursie moeten worden gecombineerd. Diverse methoden om Lindley’s recursie te analyseren zijn ook nuttig voor de analyse van (1). Wanneer we aannemen dat de voorbereidingstijd een fase-type verdeling heeft, dan reduceert de analyse van (1) tot de analyse van een Markovketen met eindige toestandsruimte. Ook kunnen Laplace-transformaties of Wiener- Hopf technieken in diverse gevallen worden toegepast (cf. Sectie 1.6). In andere gevallen moet een niet-standaard differentiaalvergelijking worden opgelost, of moet uitgeweken worden naar een iteratieve benadering van de wachttijdverdeling. In Hoofdstuk 5 dient ook een speciale klasse van verdelingen ge¨introduceerd te worden die het mogelijk maakt om een Fredholm vergelijking op te lossen. In de meeste gevallen zijn de resultaten expliciet of kunnen worden weergegeven in termen van de oplossing van een lineair stelsel vergelijkingen, zie bijvoorbeeld Stelling 4.8. Het proefschrift wordt afgesloten met enkele afsluitende opmerkingen en diverse suggesties voor verder onderzoek

Simulation Of A Composite Queuing Model Of The Entrance And Exit Of Pilgrims From Al-Masjid Al-Haram

Mathematics has always provided solutions for practical life problems. This study uses mathematics to suggest solutions to the problem of overcrowd during ‘Tawaf’ ritual of Hajj. One of the rituals during Hajj is the ‘Tawaf”, that is the circling seven times around Ka'aba. This ritual is among the ‘problems’ faced by pilgrims in the Hajj season due to the overcrowd occurring when individuals or groups of pilgrims arrive to perform the Tawaf in random order. All arrivals are allowed to perform the Tawaf at anytime. The density of pilgrims circulating close around Ka’aba reaches the maximum at certain times of the day

Discrete-time queueing model for responsive network traffic and bottleneck queues

The Internet has been more and more intensively used in recent years. Although network infrastructure has been regularly upgraded, and the ability to manage heavy traffic greatly increased, especially on the core networks, congestion never ceases to appear, as the amount of traffic that flow on the Internet seems to be increasing at an even faster rate. Thus, congestion control mechanisms play a vital role in the functioning of the Internet. Active Queue Management (AQM) is a popular type of congestion control mechanism that is implemented on gateways (most notably routers), which can predict and avoid the congestion before it happens. When properly configured, AQMs can effectively reduce the congestion, and alleviate some of the problems such as global synchronisation and unfairness to bursty traffic. However, there are still many problems regarding AQMs. Most of the AQM schemes are quite sensitive to their parameters setting, and these parameters may be heavily dependent on the network traffic profile, which the administrator may not have intensive knowledge of, and is likely to change over time. When poorly configured, many AQMs perform no better than the basic drop-tail queue. There is currently no effective method to compare the performance of these AQM algorithms, caused by the parameter configuration problem. In this research, the aim is to propose a new analytical model, which mainly uses discrete-time queueing theory. A novel transient modification to the conventional equilibrium-based method is proposed, and it is utilised to further develop a dynamic interactive model of responsive traffic and bottleneck queues. Using step-by-step analysis, it represents the bursty traffic and oscillating queue length behaviour in practical network more accurately. It also provides an effective way of predicting the behaviour of a TCP-AQM system, allowing easier parameter optimisation for AQM schemes. Numerical solution using MATLAB and software simulation using NS-2 are used to extensively validate the proposed models, theories and conclusions

Feedback fluid queues with multiple tresholds

Cataloged from PDF version of article.Unlike discrete or continuous time queuing systems fed with point processes, workload in fluid queues arrives at the system as a fluid flow rather than jobs or packets. The rate of the fluid flow is governed by a continuous time Markov chain in Markov fluid queues. In first order fluid queues, rates are deterministically determined by a background Markov chain whereas in second order fluid queues, a Brownian motion is additionally inserted to the queue content process. Each of those queues can either accommodate a single regime or multiple regimes (equivalently multiple thresholds) in which the rates and the infinitesimal generator might be different in different regimes but they should be fixed within a single regime. In this thesis, we first generalize the existing solution of first order feedback fluid queues with multiple thresholds for the steady state distribution function of queue occupancy by also allowing the existence of repulsive type boundaries and states with zero rates. Secondly, we complete the boundary conditions for not only the transient but also the steady state solution of second order feedback fluid queues with multiple thresholds. Finally, we apply the theory of feedback fluid queues with multiple thresholds as an effective approximation to the Markov modulated discrete time queueing model that arises in the performance evaluation of an adaptive MPEG video streaming system in UMTS environment. By doing so, we eliminate the state space explosion problem that arises in the original discrete model.Kankaya, Hüseyin EmreM.S

Quality aspects of Internet telephony

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