Gráficos de control multivariables aplicados a la estimación de la calidad percibida de la voz sobre ip

Con el desarrollo del protocolo de red IP se ha generado un crecimiento explosivo de Internet en los últimos diez o quince años, que ha llevado a la proliferación de múltiples servicios de comunicación, dada la naturaleza de este protocolo se hace necesario implementar mecanismos que permitan asegurar la calidad  de servicio a los usuarios finales. El presente artículo presenta un sistema de evaluación en línea efectiva en la vigilancia y medición de la calidad percibida de la voz sobre IP consistente en el tiempo. El modelo presentado combina el control estadístico de procesos y los métodos objetivos de estimación de la calidad percibida de voz. Para esto se ha implementado el grafico de control multivariable T_ de Hotteling y el método objetivo no intrusivo de estimación de la calidad percibida de voz “Modelo-E”.The development of the Internet Protocol ( IP) it´s  generating an explosive growth of Internet and that have taken the proliferation of multiples services of communication. But  this protocol needs to implement mechanisms that allow to assure the quality of service to the end users. The present article presents a system of effective on-line evaluation in the surveillance and measurements  of the perceived quality of the voice on IP consistent in the time. The presented model combines the statistical control of processes and the objective methods of estimate of the perceived quality of voice. For this it has been implemented Multivariable Control Chart T_ de Hotteling and the perceived method for voice quality prediction to voice over IP

Call admission control for adaptive bit-rate VoIP over 802.11 WLAN.

Cui, Yuanyuan.Thesis (M.Phil.)--Chinese University of Hong Kong, 2009.Includes bibliographical references (p. 64-68).Abstract also in Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1 .1 --- Motivations and Contributions --- p.1Chapter 1.2 --- Related Works --- p.3Chapter 1.3 --- Organization of the Thesis --- p.4Chapter Chapter 2 --- Background --- p.5Chapter 2.1 --- IEEE 802.11 --- p.5Chapter 2.1.1 --- IEEE 802.11 Topologies --- p.5Chapter 2.1.2 --- IEEE 802.11 MAC --- p.8Chapter 2.2 --- Voice over Internet Protocol (VoIP) --- p.11Chapter 2.2.1 --- A VoIP system --- p.11Chapter 2.2.2 --- QoS requirements for VoIP --- p.11Chapter 2.2.3 --- VoIP speech codecs --- p.12Chapter 2.3 --- VoIP over WLAN --- p.13Chapter 2.3.1 --- System Architecture of VoIP over WLAN --- p.14Chapter 2.3.2 --- VoIP Capacity over WLAN --- p.15Chapter 2.4 --- Skype --- p.16Chapter Chapter 3 --- Skype Rate Adaptation Mechanism --- p.17Chapter 3.1 --- Experimental Setting --- p.17Chapter 3.2 --- Overview --- p.19Chapter 3.3 --- Flow Rate Region --- p.20Chapter 3.4 --- Feedback: Receiver Report (RR) --- p.21Chapter 3.5 --- Bandwidth Usage Target (BM) --- p.24Chapter 3.6 --- Summary of Skype Rate Adaptation Mechanism --- p.28Chapter 3.7 --- Skype-emulating Traffic Generator --- p.28Chapter Chapter 4 --- "Call Admission, Fairness and Stability Control" --- p.32Chapter 4.1 --- Unfair and Instability problems for AVoIP --- p.32Chapter 4.1.1 --- Analysis --- p.32Chapter 4.1.2 --- Simulation Evaluation --- p.34Chapter 4.2 --- CFSC scheme --- p.37Chapter 4.2.1 --- Pre-admission Bandwidth-reallocation Call Admission Control (PBCAC) --- p.39Chapter 4.2.2 --- Fairness Control --- p.42Chapter 4.2.3 --- Stability Control --- p.43Chapter Chapter 5 --- Performance Evaluation of CFSC --- p.44Chapter 5.1 --- Evaluation of Fairness Control --- p.44Chapter 5.2 --- Evaluation of Stability Control --- p.46Chapter 5.3 --- Evaluation of PBCAC --- p.46Chapter 5.4 --- Evaluation of complete CFSC --- p.49Chapter Chapter 6 --- Conclusion --- p.51Appendices --- p.53References --- p.6

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

IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav