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

Traffic integration in personal, local and geograhical wireless networks

Currently, users identify wireless networks with the first and second generation of cellular-telephony networks. Although voice and short messaging have driven the success of these networks so far, data and more sophisticated applications are emerging as the future driving forces for the extensive deployment of new wireless technologies. In this chapter we will consider future wireless technologies that will provide support to different types of traffic including legacy voice applications, Internet data traffic, and sophisticated multimedia applications. In the near future, wireless technologies will span from broadband wide-area technologies (such as satellite-based network and cellular networks) to local and personal area networks. Hereafter, for each class of networks, we will present the emerging wireless technologies for supporting service integration. Our overview will start by analyzing the Bluetooth technology that is the de-facto standard for Wireless Personal Area Networks (WPANs), i.e. networks that connect devices placed inside a circle with radius of 10 meters. Two main standards exist for Wireless Local Area Networks (WLANs): IEEE 802. and HiperLAN. In this chapter we focus on the IEEE 802.11 technology, as it is the technology currently available on the market. In this chapter, after a brief description of the IEEE 802.11 architecture, we will focus on the mechanisms that have been specifically designed to support delay sensitive traffics

Improved multi-point communication for data and voice over IEEE 802.11b

There is a growing demand for faster, improved data and voice services in rural areas without modern telecom infrastructure. A wireless network is often the only feasible solution for providing network access in this environment, due to the sparse populations and difficult natural conditions. A system solution that incorporates the Multipoint Communication System (MCS) algorithm created by TRLabs into the available IEEE 802.11b Wireless Local Area Network (WLAN) devices was proposed and studied in this thesis. It combines the advantages of both systems, that is, the MCS’ capability of integrating Voice over Internet Protocol (VoIP) and data services and the IEEE 802.11b standard, currently the most widely used in WLAN products. A system test bed was set up inside Network Simulator-2 (NS-2). The data and VoIP performance was tested. Modifications to the original MCS algorithm to improve system performance were made throughout this thesis. In a constant rate radio channel, data performance (throughput and transmission efficiency) was measured using the original MCS algorithm, which was comparable to the standard Distribution Coordination Function (DCF) operation of IEEE 802.11b when both were simulated at similar conditions. On an 802.11b platform, the Automatic Rate Fallback (ARF) feature was incorporated into the original MCS algorithm. However, when clients with different data rates were present in the same channel, all the clients involved received unacceptably low and equal data throughput, dragged down by the low rate clients. A modified MCS data polling algorithm was proposed with the capability of repeated polling, which eliminated the negative effect of low rate clients in a multi-rate channel. In addition, the original MCS algorithm was modified to be more efficient in the voice polling process. The voice performance and data throughput were tested at various conditions. However, the one-by-one polling still resulted in very low voice transmission efficiency. The time wasted became more severe with increasing relay distance and channel rate (only 8.5% in an 11 Mbps channel at 30 km). A new voice handling process similar to Time Division Multiple Access (TDMA) mode was implemented and simulated. Its voice efficiency can be kept at 25% at any setting of relay distance and channel rate. Data transmission in the same channel can also benefit from using the new voice scheme. The normalized saturation throughput could be improved by 13.5% if there were 40 voice clients involved in an 11 Mbps channel at the relay distance of 15 km, compared to the original MCS algorithm. More improvement in voice efficiency, voice capacity, and data throughput can be achieved at longer relay distance, or with more voice calls set up

Improving the Performance of Wireless LANs

This book quantifies the key factors of WLAN performance and describes methods for improvement. It provides theoretical background and empirical results for the optimum planning and deployment of indoor WLAN systems, explaining the fundamentals while supplying guidelines for design, modeling, and performance evaluation. It discusses environmental effects on WLAN systems, protocol redesign for routing and MAC, and traffic distribution; examines emerging and future network technologies; and includes radio propagation and site measurements, simulations for various network design scenarios, numerous illustrations, practical examples, and learning aids

Analysis and Performance Evaluation of IEEE 802.11 WLAN

With fast deployment of wireless local area networks VoIP over IEEE 802.11 wireless local area network (WLAN) is growing very fast and is providing a cost effective alternative for voice communications. WLANs were initially set up to handle bursty nonreal time type of data traffic. Therefore, the wireless access protocols initially defined are not suitable for voice traffic. Subsequently, updates in the standard have been made to provision for QoS requirements of data, especially the real time traffic of the type voice and video. Despite these updates, however, transmitting voice traffic over WLAN does not utilize the available bandwidth (BW) efficiently, and the number of simultaneous calls supported in practice is significantly lower than what the BW figures would suggest. Several modifications have been proposed to improve the call capacity, and recently isochronous coordination function (ICF) was introduced to mitigate the problem of low call capacity. The proposed modified ICF which further improves the performance in terms of the call capacity. The proposed scheme uses multiplexing and multicasting in the downlink to substantially increase the call capacity