Towards the Quality of Service for VoIP traffic in IEEE 802.11 Wireless Networks

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic

Voice Call Capacity Over Wireless Mesh Networks

The goal of this thesis is to understand the voice call carrying capacity of an IEEE 802.11b/e based ad hoc network. We begin with the modelling of conversational speech and define a six state semi-Markov voice model based on ITU-T P59 recommendation. We perform a theoretical analysis of the voice model and compare it with results obtained via simulations. Using a Java based IEEE 802.11 medium access layer simulator, we determine the upper-bound for the number of voice calls carried by an ad hoc network. We use a linear topology with the ideal carrier sensing range and evaluate the number of calls carried using packet loss and packet delay as metrics. We observe that, for one, two, three and four hop, 5.5 Mbps IEEE 802.11 wireless links have an upper-bound of eight, six, five, and three voice calls respectively. We then consider a carrier sensing range and a path loss model and compare them with the ideal case. We observe, after considering a carrier sensing range with path loss model, there is a reduction in the number of calls carried by the linear networks. One, two, three and four hop 5.5 Mbps IEEE 802.11 wireless links support eight, five, four, and two voice calls respectively, when a carrier sensing range and a path loss model is considered. We also find that by adopting packet dropping policies at the nodes, we improve the call carrying capacity and quality of service on the network. In our simulations of a two hop network in path loss conditions, we find that, by adopting a time delay based packet dropping policy at the nodes, the number of calls supported simultaneously increased from five to six. In a four hop linear network we find that by total packet loss is reduced by 20%, adopting a random packet dropping policy and by 50% adopting a time delay based packet dropping policy. Although there is no change in number of calls supported, load on the network is reduced

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

A Dynamic Multimedia User-Weight Classification Scheme for IEEE_802.11 WLANs

In this paper we expose a dynamic traffic-classification scheme to support multimedia applications such as voice and broadband video transmissions over IEEE 802.11 Wireless Local Area Networks (WLANs). Obviously, over a Wi-Fi link and to better serve these applications - which normally have strict bounded transmission delay or minimum link rate requirement - a service differentiation technique can be applied to the media traffic transmitted by the same mobile node using the well-known 802.11e Enhanced Distributed Channel Access (EDCA) protocol. However, the given EDCA mode does not offer user differentiation, which can be viewed as a deficiency in multi-access wireless networks. Accordingly, we propose a new inter-node priority access scheme for IEEE 802.11e networks which is compatible with the EDCA scheme. The proposed scheme joins a dynamic user-weight to each mobile station depending on its outgoing data, and therefore deploys inter-node priority for the channel access to complement the existing EDCA inter-frame priority. This provides efficient quality of service control across multiple users within the same coverage area of an access point. We provide performance evaluations to compare the proposed access model with the basic EDCA 802.11 MAC protocol mode to elucidate the quality improvement achieved for multimedia communication over 802.11 WLANs.Comment: 15 pages, 8 figures, 3 tables, International Journal of Computer Networks & Communications (IJCNC

Adaptive medium access control for VoIP services in IEEE 802.11 WLANs

Abstract- Voice over Internet Protocol (VoIP) is an important service with strict Quality-of-Service (QoS) requirements in Wireless Local Area Networks (WLANs). The popular Distributed Coordination Function (DCF) of IEEE 802.11 Medium Access Control (MAC) protocol adopts a Binary Exponential Back-off (BEB) procedure to reduce the packet collision probability in WLANs. In DCF, the size of contention window is doubled upon a collision regardless of the network loads. This paper presents an adaptive MAC scheme to improve the QoS of VoIP in WLANs. This scheme applies a threshold of the collision rate to switch between two different functions for increasing the size of contention window based on the status of network loads. The performance of this scheme is investigated and compared to the original DCF using the network simulator NS-2. The performance results reveal that the adaptive scheme is able to achieve the higher throughput and medium utilization as well as lower access delay and packet loss probability than the original DCF

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

Dynamic Queue Utilization Based MAC for multi-hop Ad Hoc networks

The end-to-end throughput in single flow multi-hop Ad Hoc networks decays rapidly with path length. Along the path, the success rate of delivering packets towards the destination decreases due to higher contention, interference, limited buffer size and limited shared bandwidth constraints. In such environments the queues fill up faster in nodes closer to the source than in the nodes nearer the destination. In order to reduce buffer overflow and improve throughput for a saturated network, this paper introduces a new MAC protocol named Dynamic Queue Utilization Based Medium Access Control (DQUB-MAC). The protocol aims to prioritise access to the channel for queues with higher utilization and helps in achieving higher throughput by rapidly draining packets towards the destination. The proposed MAC enhances the performance of an end-to-end data flow by up to 30% for a six hop transmission in a chain topology and is demonstrated to remain competitive for other network topologies and for a variety of packet sizes