An improved medium access control protocol for real-time applications in WLANs and its firmware development

The IEEE 802.11 Wireless Local Area Network (WLAN), commonly known as Wi-Fi, has emerged as a popular internet access technology and researchers are continuously working on improvement of the quality of service (QoS) in WLAN by proposing new and efficient schemes. Voice and video over Internet Protocol (VVoIP) applications are becoming very popular in Wi-Fi enabled portable/handheld devices because of recent technological advancements and lower service costs. Different from normal voice and video streaming, these applications demand symmetric throughput for the upstream and downstream. Existing Wi-Fi standards are optimised for generic internet applications and fail to provide symmetric throughput due to traffic bottleneck at access points. Performance analysis and benchmarking is an integral part of WLAN research, and in the majority of the cases, this is done through computer simulation using popular network simulators such as Network Simulator ff 2 (NS-2) or OPNET. While computer simulation is an excellent approach for saving time and money, results generated from computer simulations do not always match practical observations. This is why, for proper assessment of the merits of a proposed system in WLAN, a trial on a practical hardware platform is highly recommended and is often a requirement. In this thesis work, with a view to address the abovementioned challenges for facilitating VoIP and VVoIP services over Wi-Fi, two key contributions are made: i) formulating a suitable medium access control (MAC) protocol to address symmetric traffic scenario and ii) firmware development of this newly devised MAC protocol for real WLAN hardware. The proposed solution shows signifocant improvements over existing standards by supporting higher number of stations with strict QoS criteria. The proposed hardware platform is available off-the-shelf in the market and is a cost effective way of generating and evaluating performance results on a hardware system

Implementing a wireless base station for a sensor network

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Civil and Environmental Engineering, 2004.Includes bibliographical references (leaves 68-69).Using wireless sensor networks for monitoring infrastructure is a new trend in civil engineering. Compared with traditional ways to monitor infrastructure, wireless sensor networks are cheap, safe, and compact. However, there are many available wireless communication techniques and hardware for a wireless sensor network. Therefore, it is an important step to choose the best communication method and hardware to construct a wireless sensor network for a particular infrastructure. The London Underground project, which is described in this thesis as a reference case study, demands real-time data transmission, low-power network, and wireless network communication, and also a hardware/software system to collect, archive and display data from the monitoring activity. We consider the trade-offs in choosing 802.1 lb as a communication method. A web service architecture for data visualization is then described. Finally we discuss the appropriate selection of a computer device to serve as the base station.by Heewon Song.M.Eng

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

An Experimental Analysis of the Call Capacity of IEEE 802.11b Wireless Local Area Networks for VoIP Telephony

The use of the Internet to make phone calls is growing in popularity as the Voice over Internet protocol (VoIP) allows users to make phone calls virtually free of charge. The increased uptake of broadband services by domestic users will further increase the use of VoIP telephony. Furthermore, the emergence of low cost wireless networks (namely IEEE 802.11a/b/g WLANs) is expected to bring wireless VoIP into the mainstream. As the number of wireless hotspots increases more users will want to use VoIP calls wherever possible by connecting to open access points (AP). A major concern with VoIP is Quality of Service (QoS). In order for VoIP to be truly successful users must enjoy a similar perceived QoS as a call made over a traditional telephone network. There are many factors that influence QoS which include: throughput, packet delay, delay variation (or jitter), and packet loss. This thesis is an experimental study of the call capacity of an IEEE 802.11b network when using VoIP telephony. Experiments included increasing the number of VoIP stations and also increasing the level of background traffic until network saturation occurs. Results show that the network is capable of supporting at least 16 VoIP stations. Due to the operation of the IEEE 802.11 medium access control (MAC) mechanism, the AP acts as a bottleneck for all traffic destined for wireless stations, in that significant delays can be incurred by VoIP packets which can lead to a poor perceived QoS by users. Consequently the performance of the AP downlink is the critical component in determining VoIP call capacity