Dimensioning of Telecommunication Network Based on Quality of Services Demand and Detailed Behaviour of Users

The aim of this paper is to be determined the network capacity (number of necessary internal switching lines) based on detailed users’ behaviour and demanded quality of service parameters in an overall telecommunication system. We consider detailed conceptual and its corresponded analytical traffic model of telecommunication system with (virtual) circuit switching, in stationary state with generalized input flow, repeated calls, limited number of homogeneous terminals and losses due to abandoned and interrupted dialing, blocked and interrupted switching, not available intent terminal, blocked and abandoned ringing (absent called user) and abandoned conversation. We propose an analytical - numerical solution for finding the number of internal switching lines and values of the some basic traffic parameters as a function of telecommunication system state. These parameters are requisite for maintenance demand level of network quality of service (QoS). Dependencies, based on the numericalanalytical results are shown graphically. For proposed conceptual and its corresponding analytical model a network dimensioning task (NDT) is formulated, solvability of the NDT and the necessary conditions for analytical solution are researched as well. It is proposed a rule (algorithm) and computer program for calculation of the corresponded number of the internal switching lines, as well as corresponded values of traffic parameters, making the management of QoS easily

A study of teletraffic problems in multicast networks

This dissertation studies teletraffic engineering of dynamic multicast connections. The traditional models in teletraffic engineering do not handle multicast connections properly, since in a dynamic multicast tree, users may join and leave the connection freely, and thus the multicast tree evolves in time. A model called multicast loss system is used to calculate blocking probabilities in a single link and in tree-type networks. In a single link case, the problem is a generalised Engset problem, and a method for calculating call blocking probabilities for users is presented. Application of the reduced load approximation for multicast connections is studied. Blocking probabilities in a cellular system are studied by means of simulation. The analysis is mainly concentrated on tree type networks, where convolution-truncation algorithms and simulation methods for solving the blocking probabilities exactly are derived. Both single layer and hierarchically coded streams are treated. The presented algorithms reduce significantly the computational complexity of the problem, compared to direct calculation from the system state space. An approximative method is given for background traffic. The simulation method presented is an application of the Inverse Convolution Monte-Carlo method, and it gives a considerable variance reduction, and thus allows simulation with smaller sample sizes than with traditional simulation methods. Signalling load for dynamic multicast connections in a node depends on the shape of the tree as well as the location of the node in the tree. This dissertation presents a method for calculating the portion of signalling load that is caused by call establishments and tear-downs.reviewe

Study of Queuing Systems with a Generalized Departure Process

This work was supported by the Bulgarian National Science Fund under grant BY-TH-105/2005.This paper deals with a full accessibility loss system and a single server delay system with a Poisson arrival process and state dependent exponentially distributed service time. We use the generalized service flow with nonlinear state dependence mean service time. The idea is based on the analytical continuation of the Binomial distribution and the classic M/M/n/0 and M/M/1/k system. We apply techniques based on birth and death processes and state-dependent service rates. We consider the system M/M(g)/n/0 and M/M(g)/1/k (in Kendal notation) with a generalized departure process Mg. The output intensity depends nonlinearly on the system state with a defined parameter: “peaked factor p”. We obtain the state probabilities of the system using the general solution of the birth and death processes. The influence of the peaked factor on the state probability distribution, the congestion probability and the mean system time are studied. It is shown that the state-dependent service rates changes significantly the characteristics of the queueing systems. The advantages of simplicity and uniformity in representing both peaked and smooth behaviour make this queue attractive in network analysis and synthesis

Introduction to Queueing Theory and Stochastic Teletraffic Models

The aim of this textbook is to provide students with basic knowledge of stochastic models that may apply to telecommunications research areas, such as traffic modelling, resource provisioning and traffic management. These study areas are often collectively called teletraffic. This book assumes prior knowledge of a programming language, mathematics, probability and stochastic processes normally taught in an electrical engineering course. For students who have some but not sufficiently strong background in probability and stochastic processes, we provide, in the first few chapters, background on the relevant concepts in these areas.Comment: 298 page

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

State-Dependent Bandwidth Sharing Policies for Wireless Multirate Loss Networks

We consider a reference cell of fixed capacity in a wireless cellular network while concentrating on next-generation network architectures. The cell accommodates new and handover calls from different service-classes. Arriving calls follow a random or quasi-random process and compete for service in the cell under two bandwidth sharing policies: 1) a probabilistic threshold (PrTH) policy or 2) the multiple fractional channel reservation (MFCR) policy. In the PrTH policy, if the number of in-service calls (new or handover) of a service-class exceeds a threshold (difference between new and handover calls), then an arriving call of the same service-class is accepted in the cell with a predefined state-dependent probability. In the MFCR policy, a real number of channels is reserved to benefit calls of certain service-classes; thus, a service priority is introduced. The cell is modeled as a multirate loss system. Under the PrTH policy, call-level performance measures are determined via accurate convolution algorithms, while under the MFCR policy, via approximate but efficient models. Furthermore, we discuss the applicability of the proposed models in 4G/5G networks. The accuracy of the proposed models is verified through simulation. Comparison against other models reveals the necessity of the new models and policies

On a Bicriterion Server Allocation Problem for a Multidimensional Erlang Loss System

In this work an optimization problem on a classical elementary stochastic system system, modeled as an Erlang-B (M/M/x) loss system, is formulated by using a bicriteria approach. The problem is focused on the allocation of a given total of k servers to a number of groups of servers capable of carrying certain offered traffic processes assumed as Poissonian in nature. Two main objectives are present in this formulation. Firstly a criterion of equity in the grade of service, measured by the call blocking probabilities, entails that the absolute difference between the blocking probabilities experienced by the calls in the different service groups must be as small as possible. Secondly a criterion of system economic performance optimization requires the total traffic carried by the system, to be maximized. Relevant mathematical results characterizing the two objective functions and the set N of the non-dominated solutions, are presented. An algorithm for traveling on N based on the resolution of single criterion convex problems, using a Newton-Raphson method, is also proposed. In each iteration the two first derivatives of the Erlang-B function in the number of circuits (a difficult numerical problem) are calculated using a method earlier proposed. Some computational results are also presented

LTE 네트워크에서 비디오 전달 서비스의 성능 향상

학위논문 (박사)-- 서울대학교 대학원 : 전기·컴퓨터공학부, 2015. 2. 권태경.LTE includes an enhanced multimedia broadcast/multicast service(eMBMS)but delay-sensitive real-time video streaming requires the combination of efficient handling of wireless link bandwidth and reduced handover delays, which remains a challenge. The 3GPP standard introduces a Multimedia Broadcast and multicast service over a Single Frequency Network (MBSFN) area which is a group of base stations broadcasting the same multicast packets. It can reduce the handover delay within MBSFN areas, but raises the traffic load on LTE networks. In this dissertation, we first presents an MBSFN architecture based on location management areas (LMAs) which can increase the sizes of MBSFN areas to reduce the average handover delay without too much bandwidth waste. An analytical model is developed to quantify service disruption time, bandwidth usage, and blocking probability for different sizes of MBSFN areas and LMAs while considering user mobility, user distribution, and eMBMS session popularity. Using this model, we also propose how to determine the best sizes of MBSFN areas and LMAs along with performance guarantees. Analytical and simulation results demonstrate that our LMA-based MBSFN scheme can achieve bandwidth-efficient multicast delivery while retaining an acceptable service disruption time. We next propose to transmit the real-time video streaming packets of eMBMSs proactively and probabilistically, so that the average handover delay perceived by a user is stochastically guaranteed. To quantify the tradeoff between the perceived handover delay and the bandwidth overhead of proactive transmissions, we develop an analytical model considering user mobility, user distribution, and session popularity. Comprehensive simulation is carried out to verify the analysis. On the other hand, hypertext transfer protocol (HTTP) based adaptive streaming (HAS) is expected to be a dominant technique for non-real-time video delivery in LTE networks. In this dissertation, we first analyze the root causes of the problems of the existing HAS techniques. Based on the insights gained from our analysis, we propose a network-side HAS solution to provide a fair, efficient, and stable video streaming service. The key characteristics of our solution are: (i) unification of video- and data-users into a single utility framework, (ii) direct rate control conveying the assigned rates to the video client through overwritten HTTP Response messages, and (iii) rate allocation for stability by a stateful approach. By the experiments conducted in a real LTE femtocell network, we compare the proposed solution with state-of-the-art HAS solutions. We reveal that our solution (i) enhances the average video bitrates, (ii) achieves the stability of video quality, and (iii) supports the control of the balance between video- and data-users.Abstract i I. Introduction 1 II. Performance Improvements on Real-time Multicast Video Delivery 4 2.1 Introduction 4 2.2 Related Work 7 2.3 Location Management Area Based MBSFN 9 2.3.1 Location Management Area (LMA) 10 2.3.2 Handover Delays 12 2.3.3 LMA-based MBSFN Area Planning 12 2.4 Performance Analysis 14 2.4.1 Disruption Time 17 2.4.2 Bandwidth Usage 20 2.4.3 Blocking Probability 21 2.5 Numerical Results 23 2.5.1 Effect of NZ and NL 24 2.5.2 Deciding NZ and NL 27 2.5.3 Effects of v and rho* 31 2.5.4 Effect of alpha 32 2.6 Simulation Results 35 2.7 Conclusion 37 III. Proactive Approach for LMA-based MBSFN 39 3.1 Introduction 39 3.2 Network and MBSFN Modeling 41 3.3 Proactive LMA-based MBSFN 44 3.3.1 Problem Formulation 45 3.3.2 Overall procedure 47 3.4 Performance Evaluation 48 3.4.1 Simulation Setup 48 3.4.2 Computation of pi 50 3.4.3 Simulation Results 51 3.5 Conclusions 53 IV. Performance Improvements on HTTP Adaptive Video Streaming 55 4.1 Introduction 55 4.2 Related Work 57 4.3 Problem Definition 59 4.4 Utility-aware Network-side Streaming Approach 62 4.4.1 Streaming Proxy (SP) 63 4.4.2 Message Flows 65 4.4.3 Characteristics 67 4.5 Bitrate Assignment 68 4.5.1 Bitrate Calculation 69 4.5.2 Enhancing Stability 70 4.5.3 Algorithm for Continuous Bitrates 71 4.5.4 Handling the Bottleneck of Wired Networks 71 4.6 Simulation 73 4.6.1 Static Scenario 73 4.6.2 Mobile Scenarios 75 4.6.3 Algorithm for Continuous Bitrates 77 4.7 Experiments 78 4.7.1 Implementation of DASH Player 79 4.7.2 Implementation of eNB 80 4.7.3 Implementation of Streaming Proxy 83 4.7.4 Experimental Results 83 4.8 Conclusion 87 V. Summary & FutureWork 89 Bibliography 92Docto