Evaluación del impacto de los mecanismos de control de error en la calidad de servicio de telefonía IP basado en Asterisk sobre una red inalámbrica de banda ancha en la provincia de Tayacaja - Huancavelica

En la presente tesis de maestría de telecomunicaciones titulada “Evaluación del Impacto de los Mecanismos de Control de Error en la Calidad de Servicio de Telefonía IP basado en Asterisk sobre una Red Inalámbrica de Banda Ancha en la Provincia de Tayacaja - Huancavelica” se describe el proceso y los elementos de la transmisión de VoIP en las redes LAN inalámbricas, se evalúa los diversos mecanismos de control de error existentes y particularmente se indica el más adecuado para la red LAN inalámbrica. Por otro lado, se analizan los parámetros de calidad de servicio y su relación en el servicio de VoIP, identificando los elementos que degradan la calidad de la voz que un usuario percibe en una comunicación como el retardo, la pérdida de paquetes, variación de retardo, latencia extremo a extremo, errores binarios en el canal de transmisión. En ese sentido, también se describe el proceso de medición de calidad de la VoIP basado en los estándares de la UIT. Se propone unas estrategias de mejora del sistema de comunicaciones para VoIP en entorno WLAN bajo la operación mayoritaria DCF restringiendo la operación del mecanismo de retransmisión WLAN al tiempo máximo permitido por la aplicación y la memoria de reproducción del receptor. Se investigan estos mecanismos de retransmisión básicos de WLAN y los mismos interactuando con la memoria de reproducción y la limitación temporal de la aplicación. Para ello, se realiza una definición de un modelo de simulación para encontrar los resultados; los cuales indican una mejora en el rendimiento ya que mantienen el nivel de calidad en entornos inalámbricos hostiles como los de una tasa de error peor de 10-4.Tesi

On modeling and mitigating new breed of dos attacks

Denial of Service (DoS) attacks pose serious threats to the Internet, exerting in tremendous impact on our daily lives that are heavily dependent on the good health of the Internet. This dissertation aims to achieve two objectives:1) to model new possibilities of the low rate DoS attacks; 2) to develop effective mitigation mechanisms to counter the threat from low rate DoS attacks. A new stealthy DDoS attack model referred to as the quiet attack is proposed in this dissertation. The attack traffic consists of TCP traffic only. Widely used botnets in today\u27s various attacks and newly introduced network feedback control are integral part of the quiet attack model. The quiet attack shows that short-lived TCP flows used as attack flows can be intentionally misused. This dissertation proposes another attack model referred to as the perfect storm which uses a combination of UDP and TCP. Better CAPTCHAs are highlighted as current defense against botnets to mitigate the quiet attack and the perfect storm. A novel time domain technique is proposed that relies on the time difference between subsequent packets of each flow to detect periodicity of the low rate DoS attack flow. An attacker can easily use different IP address spoofing techniques or botnets to launch a low rate DoS attack and fool the detection system. To mitigate such a threat, this dissertation proposes a second detection algorithm that detects the sudden increase in the traffic load of all the expired flows within a short period. In a network rate DoS attacks, it is shown that the traffic load of all the expired flows is less than certain thresholds, which are derived from real Internet traffic analysis. A novel filtering scheme is proposed to drop the low rate DoS attack packets. The simulation results confirm attack mitigation by using proposed technique. Future research directions will be briefly discussed

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

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

Delay aspects in Internet telephony

In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation