Integrating Networks Measurements and Speech Quality Subjective Scores for Control Purposes

network measurements (e.g., loss rates and delays), and little attention has been paid to the quality perceived by end-users of the applications running over the network. Here, we address the issue of integrating speech quality subjective scores and network parameters measurements, for designing control algorithms that would yield the best QoS that could be delivered under a given communications network situation. First, we build a neural network based automaton to measure speech quality in real time, at the style of a group of human subjects when participating in an MOS test. We consider the effects of changes in network parameters (e.g., packetization interval, packet loss rate and their pattern distribution) and encoding on speech signals transmitted over the network. Our database includes transmitted speech signals in different languages. Then, we outline a control mechanism which, based on the application performance within a session (i.e., MOS speech quality scores generated by the neural networks), dynamically adjusts parameters (codec and packetization interval). Finally, we analyze preliminary results to show two main benefits: first, a better use of bandwidth, and second, delivery of the best possible speech quality given the network current situation

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

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

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Sync & Sense Enabled Adaptive Packetization VoIP

The quality and reliability problem of VoIP comes from the fact that VoIP relies on the network to transport the voice packets. The inherent problem of VoIP is that there is a mismatch between VoIP and the network. Namely, VoIP has a strict requirement of bandwidth, delay, and loss, but the network (particularly best-effort service networks) cannot guarantee such a requirement. A solution to deal with this problem is to enhance VoIP with an adaptive-rate control, called adaptive-rate VoIP. Adaptive-rate VoIP has the ability to detect the state of the network and adjust the transmission accordingly. Therefore, it gives VoIP the intelligence to optimize its performance, and making it resilient and robust to the service offered by the network. The objective of this dissertation is to develop an adaptive-rate VoIP system. We take a comprehensive approach in the study and development. Adaptive-rate VoIP is generally composed of three components: rate adaptation, network state detection, and adaptive-rate control. In the rate adaptation component, we study optimizing packetization, which can be used as an alternative means for rate adaptation. An advantage is that rate adaptation is independent of the speech coder. With this method, an adaptive-rate VoIP can be based on any constant bitrate speech coder. The study shows that the VoIP performance is primarily affected by three factors: packetization, network load, and significance of VoIP traffic; and, optimizing packetization allows us to ensure the highest possible performance. In the network state detection component, we propose a novel measurement methodology called Sync & Sense of periodic stream. Sync & Sense is unique in that it can virtually synchronize the transmission and reception timing of the VoIP session without requiring a synchronized clock. The simulation result shows that Sync & Sense can accurately measure one-way network delay. Other benefits of Sync & Sense include the ability to estimate the available network bandwidth and the full spectrum of the delays of the VoIP session. In the adaptive-rate control component, we consider the design choices and develop an adaptive-rate control that makes use of the first two components. The integration of the three components is a novel and unique adaptive-rate VoIP called Sync & Sense Enabled Adaptive Packetization VoIP. The simulation result shows that our adaptive VoIP can optimize the performance under any given network condition, and deliver a better performance than traditional VoIP. The simulation result also demonstrates that our adaptive VoIP possesses the desirable properties, which include fast response to network condition, aggressiveness to compete for the needed share of bandwidth, TCP-friendliness, and fair bandwidth allocation

Quality of service assessment and analysis of wireless multimedia networks.

Recent years have witnessed a vast technological progress in the area of Quality of Service (QoS), mainly due to the emergence of multimedia networking and computing. QoS measurement and analysis have long been of interest to the networking research community. The major goals of this thesis are of two fold: Firstly, to investigate the effect of the QoS parameters on the overall QoS experienced by wireless networks. Secondly, to utilise the results in developing efficient mechanisms for intrusive and non-intrusive assessments of the performance of wireless ad hoc networks as well as the measurement of the available QoS for audio and videoconferencing applications over the IEEE 802.11 standard. To evaluate the network performance and the overall QoS of multimedia applications, new fuzzy logic and distance measure assessment approaches were developed taking into account the QoS parameters requirements of each application. The developed approaches essentially include measuring the main QoS parameters (delay, jitter and packet loss) and use them as input to the measurement systems, which combine them and produce an output that represents the instantaneous QoS. The devised approaches showed how the QoS can be measured without a need for complicated analytical mathematical models.In this study, several techniques were devised for estimating QoS. Firstly, a probe-based assessment method (active technique) was developed. In this method, special artificial monitoring packets were injected into the network. The overall QoS and its parameters were estimated by collecting statistics from these packets. It was possible to make reasonable inferences about the delay, throughput, packet losses and the overall average QoS using different probe rates. This technique showed some limitations for measuring the jitter. In addition, the rate of the monitoring packets played an essential role in the precision, level of resolution of estimated results and negatively impacted the network performance. Secondly, to overcome some of the drawbacks of the probing-based method, a new assessment technique was, subsequently, devised based on passive monitoring standard sampling methods. Unlike the active technique, the new method has the advantage of not adding an extra load to the network. In addition, it is not like the typical passive methods, which require the transfer and calculations of the whole captured data. Generally, all sampling schemes provided satisfactory measures of the overall QoS and its parameters and produced very acceptable bias and Relative Standard Error (RSE) result. Systematic sampling provided the most accurate estimates compared to the stratified and random approaches. In addition, after sample fraction of 2%, the estimated overall QoS bias from the actual QoS became constant and equal to -0.5% and RSE was less than 0.005 using both fuzzy and distance assessment systems. Thirdly, in order to overcome some negative aspects of inaccuracy and biasness caused by sampling techniques, a new scheme was proposed to correct these results to be closer to the actual traffic measurements. The new approach does not disturb the network performance (as in active methods), neither depends on the whole traffic (as in passive methods), nor bias the actual results (as in the standard sampling technique). Similarly, systematic sampling showed the best performance. Sample fractions, using the systematic sampling, greater than 2% gave an overall estimated QoS identical to the actual QoS because the obtained relative error was nearly constant and approximately close to zero using both assessment systems. The measured QoS can be used to optimise the received quality of the multimedia services along with the changing network conditions and to manage the utilisation of the network available resources especially for ad hoc networks. Overall, the findings of this study contribute to a method for drawing a realistic picture of the wireless multimedia networks QoS and provide a firm basis and useful insights on how to effectively design future QoS solutions