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

A novel method to assess human population exposure induced by a wireless cellular network

<p>This paper presents a new metric to evaluate electromagnetic exposure induced by wireless cellular networks. This metric takes into account the exposure induced by base station antennas as well as exposure induced by wireless devices to evaluate average global exposure of the population in a specific geographical area. The paper first explains the concept and gives the formulation of the Exposure Index (EI). Then, the EI computation is illustrated through simple phone call scenarios (indoor office, in train) and a complete macro urban data long-term evolution scenario showing how, based on simulations, radio-planning predictions, realistic population statistics, user traffic data, and specific absorption rate calculations can be combined to assess the index.</p

Analysing the characteristics of VoIP traffic

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

Performance study of FMIPv6-based cross-layer WiMAX handover scheme for supporting VoIP service

This report presents performance study of the FMIPv6-based cross-layer handover scheme for VoIP supports over mobile WiMAX network. For this performance validation and evaluation, the handover delays for four different handover mechanisms are formulated and ns2 based simulation module has been developed. The handover delay, the total delay, and the R factor representing VoIP quality are measured to evaluate the VoIP support characteristics of the FMIPv6-based cross-layer scheme. Simulation results verified that the proposed FMIPv6-based cross-layer handover scheme, compared to the non-cross-layer scheme, successfully reduces total handover delay by almost 50% for the case of layer-3 handover. Further, simulation was also evaluated in terms of R factor indicating voice quality level, of which 70 is a minimum value of a traditional PSTN call to be considered as the lower limit of a VoIP call quality [6]. Through the simulation in this study, the result revealed that the proposed scheme effectively improves VoIP call quality from unacceptable quality to acceptable quality (R factor of 75). Based on these simulation results, it was found that the proposed FMIPv6-based cross-layer handover scheme is an adequate protocol for supporting VoIP services in mobile WiMAX environment

Role of Wireless technology in mobile augmented reality system (MARS)

Not availabl

On-the-Fly Establishment of Multi-hop D2D Communication based on Android Smartphones and Embedded Platforms: Implementation and Real-Life Experiments

Masteroppgave informasjons- og kommunikasjonsteknologi - Universitetet i Agder, 2015(Konfidensiell til/confidential until 01.07.2020

Streaming Video Performance and Enhancements in Resource-Constrained Wireless Networks

Streaming video is an increasingly popular application in wireless networks. The concept of a live streaming video yields several enticing possibilities: real-time video conferencing, television broadcasting, pay-per-view movie streaming, and more. These ideas have already been explored via the internet and have met with mixed success, largely due to the shortcomings of the underlying network. Taking streaming video to wireless networks, then, poses several significant challenges. Wireless networks are inherently more susceptible to failures and data corruption due to their unstable communications medium. This volatility suggests serious drawbacks for any implementation of streaming video. Video frame errors, jitter, and even complete sync loss are entirely conceivable in a wireless environment. Many of these issues have been undertaken and several approaches to mediation or even solution of these problems are underway. This thesis proposes to use advanced simulation techniques to properly exhaustively permute many vital parameters within a UMTS network and uncover, if they exist, bottlenecks in UMTS performance under considerable network load. This is accomplished via a described testing plan with simulation environment. Additionally this thesis proposes a new UDP-like transport layer specially optimized for streaming media over resource-constrained networks, tested to work with significant improvements under the UMTS cellular networking system. Finally this thesis provides several innovative new methods in the furtherance of the field of streaming media research in resourceconstrained and cellular environments. Overall this thesis makes several important contributes to an exciting and ever-growing field of active research and discussion

Literature review of recent, practical - oriented computer networks papers

This thesis report is submitted in partial fulfillment of the requirements for the degree of Bachelor of Science in Electrical and Electronic Engineering, 2009.Cataloged from PDF version of thesis report.Includes bibliographical references (page 61).This thesis will include a literature review of recent practical oriented computer networks papers in premier IEEE conferences. The objective of the thesis is to provide detailed plans for a number of computer network related projects that may be implemented by subsequent BRAC University students. We will do an in-depth study of the selected topics and provide a step by step implementation process.Hasibul IslamHassan SameerIrtiza Ahmad FarooqB. Electrical and Electronic Engineerin