Objective Measurement of Speech Quality in VoIP over Wireless LAN during Handoff

Quality of Service is a very important factor to determine the quality of a VoIP call. Different subjective and objective models exist for evaluating the speech quality in VoIP. E-model is one of the objective methods of measuring the speech quality; it considers various factors like packet loss, delay and codec impairments. The calculations of Emodel are not very accurate in case of handovers – when a VoIP call moves from one wireless LAN to another. This project conducted experimental evaluation of performance of E-model during handovers and proposes a new approach to accurately calculate the speech quality of VoIP during handovers. A detailed description of the experimental setup and the comparison of the new approach with E-model is presented in this report

VOIP weathermap - a VOIP QOS collection analysis and dissemination system

 Current trends point to VoIP as a cheaper and more effective long term solution than possible future PSTN upgrades. To move towards greater adoption of VoIP the future converged digital network is moving towards a service level management and control regime. To ensure that VoIP services provide an acceptable quality of service (QoS) a measurement solution would be helpful. The research outcome presented in this thesis is a new system for testing, analysing and presenting the call quality of Voice over Internet Protocol (VoIP). The system is called VoIP WeatherMap. Information about the current status of the Internet for VoIP calls is currently limited and a recognised approach to identifying the network status has not been adopted. An important consideration is the difficulty of assessing network conditions across links including network segments belonging to different telecommunication companies and Internet Service Providers. The VoIP WeatherMap includes the use of probes to simulate voice calls by implementing RTP/RTCP stacks. VoIP packets are sent from a probe to a server over the Internet. The important characteristics of VoIP calls such as delay and packet loss rate are collected by the server, analysed, stored in a database and presented through a web based interface. The collected voice call session data is analysed using the E-model algorithm described in ITU-T G.107. The VoIP WeatherMap presentation system includes a geographic display and internet connection links are coloured to represent the Quality of Service rank

On Design and Realization of New Generation Misson-critial Application Systems

Mission-critical system typically refers to a project or system for which the success is vital to the mission of the underlying organization. The failure or delayed completion of the tasks in mission-critical systems may cause severe financial loss, even human casualties. For example, failure of an accurate and timely forecast of Hurricane Rita in September 2005 caused enormous financial loss and several deaths. As such, real-time guarantee and reliability have always been two key foci of mission-critical system design. Many factors affect real-time guarantee and reliability. From the software design perspective, which is the focus of this paper, three aspects are most important. The first of these is how to design a single application to effectively support real-time requirement and improve reliability, the second is how to integrate different applications in a cluster environment to guarantee real-time requirement and improve reliability, and the third is how to effectively coordinate distributed applications to support real-time requirements and improve reliability. Following these three aspects, this dissertation proposes and implements three novel methodologies: real-time component based single node application development, real-time workflow-based cluster application integration, and real-time distributed admission control. For ease of understanding, we introduce these three methodologies and implementations in three real-world mission-critical application systems: single node mission-critical system, cluster environment mission-critical system, and wide-area network mission-critical system. We study full-scale design and implementation of these mission-critical systems, more specifically: 1) For the single node system, we introduce a real-time component based application model, a novel design methodology, and based on the model and methodology, we implement a real-time component based Enterprise JavaBean (EJB) System. Through component based design, efficient resource management and scheduling, we show that our model and design methodology can effectively improve system reliability and guarantee real-time requirement. 2) For the system in a cluster environment, we introduce a new application model, a real-time workflow-based application integration methodology, and based on the model and methodology, we implement a data center management system for the Southeastern Universities Research Association (SURA) project. We show that our methodology can greatly simplify the design of such a system and make it easier to meet deadline requirements, while improving system reliability through the reuse of fully tested legacy models. 3) For the system in a wide area network, we narrow our focus to a representative VoIP system and introduce a general distributed real-time VoIP system model, a novel system design methodology, and an implementation. We show that with our new model and architectural design mechanism, we can provide effective real-time requirement for Voice over Internet Protocol (VoIP)

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

SMA aplicado a la gestión de tráfico de voz en redes LAN

El objetivo principal de este proyecto es el desarrollo de una metodología para el diseño de redes de datos que soporten de una forma efectiva, aplicaciones en tiempo real como es el caso de la VoIP. Así mismo se plantea la implementación de un sistema multiagente para la gestión del tráfico en la red, mejorando el desempeño de ésta, permitiendo la priorización efectiva del tráfico. Este proyecto utiliza los conceptos básicos para diseño de redes LAN como son la segmentación de tráfico, los dominios de broadcast y de colisiones para proponer un diseño físico y lógico de la red. Se analizan las características de una red de datos existente como la capacidad de procesamiento y conmutación de los equipos activos de red, la función de distribución de probabilidad de tráfico, el sistema operativo, la velocidad de transmisión y los servicios utilizados por las estaciones de trabajo. Se realiza la simulación de la red de datos utilizando software especializado como el OP-NET, proceso que permite obtener información de la capacidad y la utilización de la red, variables requeridas por los algoritmos como SLoPS y TOPP, para determinar el ancho de banda disponible para el tráfico de VoIP. La inserción del tráfico de VoIp tambien es simulado para determinar si es necesario un replanteamiento de la estructura de la red. Se utiliza la metodología MAS CommonKads [25] para el diseño y la implantación del SMA, sistema encargado de mantener una calidad del servicio adecuada en los momentos de tráfico pico de la red / Abstract: The main objective of this project is the development of a methodology in order to design data networks that can support in a effective way real time applications such as VoIP. Also it is proposed the implementation of a Multi Agent System for the management of the network traffic, enhancing the performance of the network, and allowing the effective scheduling of the traffic. This project uses basic concepts for the design of LAN networks such as traffic segmentation, broadcast and collision domains to propose a physical and logical design of the network. The characteristics of an existing network are analyzed, including the process and switching capacities of active network equipment, the traffic probability function, the operating system, the transmission rate and the services utilized by the workstations. The simulation of the data network is done using specialized software such as OPNET. This process allows to get information about the capacity and the utilization of the network, which are important variables required by the algorithms such as SLoPS and TOPP in order to calculate the available bandwidth for the VoIP traffic. The VoIP traffic insertion is also simulated to calculate if it is necessary to change the network structure. The methodology MAS CommonKads [25] is used in order to design and implement the MAS, which is the system that has to maintain an adequate QoS at the peak traffic hours of the network.Maestrí

ボイスIPネットワークのモデリングと高性能化

Tohoku University堀口 進課

Diseño y configuración de un sistema de VoIP para la ETSIT de la Universidad Politécnica de Cartagena

Escueal Técnica Superior de Ingeniería de Telecomunicació

Implementation of QoS-Provisioning System for Voice over IP

In this paper, we address issues related to implementing Voice-over-IP (VoIP) services in packet switching networks. VoIP has been identified as a critical real-time application in the network QoS research community and has been implemented in commercial products. To provide competent quality of service for VoIP, the call admission control (CAC) mechanism has to be introduced to prevent packet losing and over-queuing. Several well-designed CAC mechanisms, such as the Site-Utilization-Based CAC and the linkutilization-based CAC mechanisms, are in place. However, the existing commercial VoIP systems have not been able to adequately apply and support these CAC mechanisms, and hence unable to provide QoS guarantees to VoIP. We have designed and implemented a QoS-Provisioning system that can be seamlessly integrated to the existing VoIP system to overcome its weakness in offering QoS guarantees. As a result, our system has been realized at Internet2 Voice Over IP Testbed in Texas A&M University.

Design and Implementation of QoS-Provisioning System for Voice over IP

Abstract—In this paper, we address issues in implementing Voice over IP (VoIP) services in packet switching networks. VoIP has been identified as a critical real-time application in the network QoS research community and has been implemented in commercial products. To provide competent quality of service for VoIP systems comparable to traditional PSTN systems, a call admission control (CAC) mechanism has to be introduced to prevent packet loss and over-queuing. Several well-designed CAC mechanisms, such as the Site-Utilization-based CAC and the Link-Utilization-based CAC mechanisms have been in place. However, the existing commercial VoIP systems have not been able to adequately apply and support these CAC mechanisms and, hence, have been unable to provide QoS guarantees to voice over IP networks. We have designed and implemented a QoS-provisioning system that can be seamlessly integrated with the existing VoIP systems to overcome their weakness in offering QoS guarantees. A practical implementation of our QoS-provisioning system has been realized. Index Terms—Voice over IP, real-time, delay, admission control.