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

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

A novel non-intrusive objective method to predict voice quality of service in LTE networks.

This research aimed to introduce a novel approach for non-intrusive objective measurement of voice Quality of Service (QoS) in LTE networks. While achieving this aim, the thesis established a thorough knowledge of how voice traffic is handled in LTE networks, the LTE network architecture and its similarities and differences to its predecessors and traditional ground IP networks and most importantly those QoS affecting parameters which are exclusive to LTE environments. Mean Opinion Score (MOS) is the scoring system used to measure the QoS of voice traffic which can be measured subjectively (as originally intended). Subjective QoS measurement methods are costly and time-consuming, therefore, objective methods such as Perceptual Evaluation of Speech Quality (PESQ) were developed to address these limitations. These objective methods have a high correlation with subjective MOS scores. However, they either require individual calculation of many network parameters or have an intrusive nature that requires access to both the reference signal and the degraded signal for comparison by software. Therefore, the current objective methods are not suitable for application in real-time measurement and prediction scenarios. A major contribution of the research was identifying LTE-specific QoS affecting parameters. There is no previous work that combines these parameters to assess their impacts on QoS. The experiment was configured in a hardware in the loop environment. This configuration could serve as a platform for future research which requires simulation of voice traffic in LTE environments. The key contribution of this research is a novel non-intrusive objective method for QoS measurement and prediction using neural networks. A comparative analysis is presented that examines the performance of four neural network algorithms for non-intrusive measurement and prediction of voice quality over LTE networks. In conclusion, the Bayesian Regularization algorithm with 4 neurons in the hidden layer and sigmoid symmetric transfer function was identified as the best solution with a Mean Square Error (MSE) rate of 0.001 and regression value of 0.998 measured for the testing data set

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated