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

Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

Quality of service differentiation for multimedia delivery in wireless LANs

Delivering multimedia content to heterogeneous devices over a variable networking environment while maintaining high quality levels involves many technical challenges. The research reported in this thesis presents a solution for Quality of Service (QoS)-based service differentiation when delivering multimedia content over the wireless LANs. This thesis has three major contributions outlined below: 1. A Model-based Bandwidth Estimation algorithm (MBE), which estimates the available bandwidth based on novel TCP and UDP throughput models over IEEE 802.11 WLANs. MBE has been modelled, implemented, and tested through simulations and real life testing. In comparison with other bandwidth estimation techniques, MBE shows better performance in terms of error rate, overhead, and loss. 2. An intelligent Prioritized Adaptive Scheme (iPAS), which provides QoS service differentiation for multimedia delivery in wireless networks. iPAS assigns dynamic priorities to various streams and determines their bandwidth share by employing a probabilistic approach-which makes use of stereotypes. The total bandwidth to be allocated is estimated using MBE. The priority level of individual stream is variable and dependent on stream-related characteristics and delivery QoS parameters. iPAS can be deployed seamlessly over the original IEEE 802.11 protocols and can be included in the IEEE 802.21 framework in order to optimize the control signal communication. iPAS has been modelled, implemented, and evaluated via simulations. The results demonstrate that iPAS achieves better performance than the equal channel access mechanism over IEEE 802.11 DCF and a service differentiation scheme on top of IEEE 802.11e EDCA, in terms of fairness, throughput, delay, loss, and estimated PSNR. Additionally, both objective and subjective video quality assessment have been performed using a prototype system. 3. A QoS-based Downlink/Uplink Fairness Scheme, which uses the stereotypes-based structure to balance the QoS parameters (i.e. throughput, delay, and loss) between downlink and uplink VoIP traffic. The proposed scheme has been modelled and tested through simulations. The results show that, in comparison with other downlink/uplink fairness-oriented solutions, the proposed scheme performs better in terms of VoIP capacity and fairness level between downlink and uplink traffic

Performance Modelling and Analysis of Handover and Call Admission Control Algorithm for Next Generation Wireless Networks

The next generation wireless system (NGWS) has been conceived as a ubiquitous wireless environment. It integrates existing heterogeneous access networks, as well as future networks, and will offer high speed data, real-time applications (e.g. Voice over IP, videoconference ) and real-time multimedia (e.g. real-time audio and video) support with a certain Quality of Service (QoS) level to mobile users. It is required that the mobile nodes have the capability of selecting services that are offered by each provider and determining the best path through the various networks. Efficient radio resource management (RRM) is one of the key issues required to support global roaming of the mobile users among different network architectures of the NGWS and a precise call admission control (CAC) scheme satisfies the requirements of high network utilization, cost reduction, minimum handover latency and high-level QoS of all the connections. This thesis is going to describe an adaptive class-based CAC algorithm, which is expected to prioritize the arriving channel resource requests, based on user¿s classification and channel allocation policy. The proposed CAC algorithm couples with Fuzzy Logic (FL) and Pre-emptive Resume (PR) theories to manage and improve the performance of the integrated wireless network system. The novel algorithm is assessed using a mathematical analytic method to measure the performance by evaluating the handover dropping probability and the system utilization

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

Optimización en el despliegue de servicios de Voz sobre IP (VoIP) sobre redes WiFi con restricciones de calidad de servicio

Las tecnologías de Voz sobre IP (VoIP) han permitido el despliegue de nuevos servicios de voz a través de Internet durante las dos últimas décadas. Por otro lado, las redes inalámbricas de área local (WLAN) basadas en el estándar IEEE 802.11 (i.e., WiFi) han experimentado un crecimiento de popularidad debido a su bajo coste y flexibilidad. Sin embargo, el despliegue de comunicaciones de VoIP con garantías de calidad sobre redes IEEE 802.11 implica una serie de dificultades (i.e., los paquetes pueden sufrir pérdidas, colisiones, y retardos variables) que no han sido satisfactoriamente resueltas con las técnicas y modelos disponibles en la actualidad. En esta tesis se desarrolla un nuevo modelo analítico de la sub-capa MAC de IEEE 802.11 que permite estimar la calidad y consumo energético de las conversaciones en un escenario realista de VoIP sobre WiFi (VoWiFi). Además, el modelo anterior se utiliza para plantear y resolver dos nuevas aplicaciones de despliegue y optimización de servicios VoWiFi: (a) el despliegue de vehículos aéreos no tripulados (UAVs) para proveer de un servicio de VoWiFi con garantías de calidad a un conjunto de usuarios y, (b) un nuevo mecanismo de control de admisión de llamadas en la red WiFi corporativa y unifica el acceso al servicio tanto para usuarios de terminales cableados como inalámbricos. Validamos el modelo analítico propuesto frente a simulaciones realizadas con el simulador de red ns-3. Los resultados muestran la utilidad del modelo propuesto para predecir las prestaciones (e.g., retardo, pérdidas) y el consumo energético en la tarjeta de red cuando se transmiten flujos de voz sobre IEEE 802.11 en condiciones no ideales. Esta capacidad de predicción ha sido clave en las propuestas realizadas de nuevas aplicaciones. En el caso del despliegue de drones, nos ha permitido definir un nuevo problema de posicionamiento inicial que puede resultar muy práctico en situaciones de rescate al aire libre. En el caso del control de admisión en entornos corporativos, el modelo nos ha permitido predecir la capacidad máxima de flujos de voz que puede ser admitida en la organización para garantizar calidad a las conversaciones existentes. Usando esta capacidad, hemos planteado un algoritmo nuevo que puede ser utilizado para unificar el control de acceso para usuarios WiFi y usuarios de terminales cableados y que aumenta el número de usuarios concurrentes respecto a los algoritmos existentes.Voice over IP (VoIP) technologies have enabled the deployment of new voice services over the Internet during the last two decades. Meanwhile, wireless local area networks (WLAN) based on the IEEE 802.11 standard (i.e., WiFi) have grown in popularity due to their low cost and flexibility. However, the deployment of quality-guaranteed VoIP communications over IEEE 802.11 networks implies a series of technical difficulties (i.e. lost packets, collisions, and delays) that have not been successfully addressed by the techniques and models available today. In this thesis, we develop a new analytical model for the IEEE 802.11 MAC sub-layer that allows one to estimate quality and energy consumption in a realistic VoIP over WiFi (VoWiFi) scenario. In addition, the previous model is used to propose and solve two new applications for the deployment and optimization of VoWiFi services: (a) deploying unmanned aerial vehicles (UAVs) to provide a VoWiFi service under guaranteed quality to a group of ground users and, (b) a new call admission control mechanism for WiFi corporate networks, which unifies the access to the voice service for both wired and wireless terminals. We validate the proposed analytical model against simulation results obtained with the ns-3 network simulator. Results show the accuracy of the proposed model for the prediction of the performance (e.g. delay, losses) and energy consumption of network interfaces when voice flows are transmitted over IEEE 802.11 under non-ideal conditions. This prediction capability has been a key component of the two VoWiFi applications developed. In the UAV deployment, it has allowed us to define a new initial positioning problem that can be very practical in outdoor rescue situations. Regarding admission control in corporate environments, the model has allowed us to predict the maximum capacity of voice flows that can be admitted in the organization to guarantee quality to existing conversations. Using this capability, we have proposed a new algorithm that can be used to unify access control for wireless and wired users, and that increases the number of concurrent users with respect to existing algorithm

Vertical Handoff between 802.11 and 802.16 Wireless Access Networks

Heterogeneous wireless networks will be dominant in the next-generation wireless networks with the integration of various wireless access networks. Wireless mesh networks will become to a key technology as an economically viable solution for wide deployment of high speed, scalable and ubiquitous wireless Internet services. In this thesis, we consider an interworking architecture of wireless mesh backbone and propose an effective vertical handoff scheme between 802.11 and 802.16 wireless access networks. The proposed vertical handoff scheme aims at reducing handoff signaling overhead on the wireless backbone and providing a low handoff delay to mobile nodes. The handoff signaling procedure in different scenarios is discussed. Together with call admission control, the vertical handoff scheme directs a new call request in the 802.11 network to the 802.16 network, if the admission of the new call in the 802.11 network can degrade quality-of-service (QoS) of the existing real-time traffic flows. Simulation results demonstrate the performance of the handoff scheme with respect to signaling cost, handoff delay, and QoS support