Modelling the IEEE 802.11 wireless MAC layer under heterogeneous VoIP traffic to evaluate and dimension QoE

PhDAs computers become more popular in the home and workplace, sharing resources and Internet access locally is a necessity. The simplest method of choice is by deploying a Wireless Local Area Network; they are inexpensive, easy to configure and require minimal infrastructure. The wireless local area network of choice is the IEEE 802.11 standard; IEEE 802.11, however, is now being implemented on larger scales outside of the original scope of usage. The realistic usage spans from small scale home solutions to commercial ‘hot spots,’ providing access within medium size areas such as cafés, and more recently blanket coverage in metropolitan. Due to increasing Internet availability and faster network access, in both wireless and wired, the concept of using such networks for real-time services such as internet telephony is also becoming popular. IEEE 802.11 wireless access is shared with many clients on a single channel and there are three non-overlapping channels available. As more stations communicate on a single channel there is increased contention resulting in longer delays due to the backoff overhead of the IEEE 802.11 protocol and hence loss and delay variation; not desirable for time critical traffic. Simulation of such networks demands super-computing resource, particularly where there are over a dozen clients on a given. Fortunately, the author has access to the UK’s super computers and therefore a clear motivation to develop a state of the art analytical model with the required resources to validate. The goal was to develop an analytical model to deal with realistic IEEE 802.11 deployments and derive results without the need for super computers. A network analytical model is derived to model the characteristics of the IEEE 802.11 protocol from a given scenario, including the number of clients and the traffic load of each. The model is augmented from an existing published saturated case, where each client is assumed to always have traffic to transmit. The nature of the analytical model is to allow stations to have a variable load, which is achieved by modifying the existing models and then to allow stations to operate with different traffic profiles. The different traffic profiles, for each station, is achieved by using the augmented model state machine per station and distributing the probabilities to each station’s state machine accordingly. To address the gap between the analytical models medium access delay and standard network metrics which include the effects of buffering traffic, a queueing model is identified and augmented which transforms the medium access delay into standard network metrics; delay, loss and jitter. A Quality of Experience framework, for both computational and analytical results, is investigated to allow the results to be represented as user perception scores and the acceptable voice call carrying capacity found. To find the acceptable call carrying capacity, the ITU-T G.107 E-Model is employed which can be used to give each client a perception rating in terms of user satisfaction. PAGE 4 OF 162 QUEEN MARY, UNIVERSITY OF LONDON OLIVER SHEPHERD With the use of a novel framework, benchmarking results show that there is potential to maximise the number of calls carried by the network with an acceptable user perception rating. Dimensioning of the network is undertaken, again compared with simulation from the super computers, to highlight the usefulness of the analytical model and framework and provides recommendations for network configurations, particularly for the latest Wireless Multimedia extensions available in IEEE 802.11. Dimensioning shows an overall increase of acceptable capacity of 43%; from 7 to 10 bidirectional calls per Access Point by using a tuned transmission opportunity to allow each station to send 4 packets per transmission. It is found that, although the accuracy of the results from the analytical model is not precise, the model achieves a 1 in 13,000 speed up compared to simulation. Results show that the point of maximum calls comes close to simulation with the analytical model and framework and can be used as a guide to configure the network. Alternatively, for specific capacity figures, the model can be used to home-in on the optimal region for further experiments and therefore achievable with standard computational resource, i.e. desktop machines

MIPv6 Experimental Evaluation using Overlay Networks

The commercial deployment of Mobile IPv6 has been hastened by the concepts of Integrated Wireless Networks and Overlay Networks, which are present in the notion of the forthcoming generation of wireless communications. Individual wireless access networks show limitations that can be overcome through the integration of different technologies into a single unified platform (i.e., 4G systems). This paper summarises practical experiments performed to evaluate the impact of inter-networking (i.e. vertical handovers) on the Network and Transport layers. Based on our observations, we propose and evaluate a number of inter-technology handover optimisation techniques, e.g., Router Advertisements frequency values, Binding Update simulcasting, Router Advertisement caching, and Soft Handovers. The paper concludes with the description of a policy-based mobility support middleware (PROTON) that hides 4G networking complexities from mobile users, provides informed handover-related decisions, and enables the application of different vertical handover methods and optimisations according to context.Publicad

Optimization and Performance Analysis of High Speed Mobile Access Networks

The end-to-end performance evaluation of high speed broadband mobile access networks is the main focus of this work. Novel transport network adaptive flow control and enhanced congestion control algorithms are proposed, implemented, tested and validated using a comprehensive High speed packet Access (HSPA) system simulator. The simulation analysis confirms that the aforementioned algorithms are able to provide reliable and guaranteed services for both network operators and end users cost-effectively. Further, two novel analytical models one for congestion control and the other for the combined flow control and congestion control which are based on Markov chains are designed and developed to perform the aforementioned analysis efficiently compared to time consuming detailed system simulations. In addition, the effects of the Long Term Evolution (LTE) transport network (S1and X2 interfaces) on the end user performance are investigated and analysed by introducing a novel comprehensive MAC scheduling scheme and a novel transport service differentiation model

Quality of service optimization of multimedia traffic in mobile networks

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes

QoS in LEO satellite networks with multipacket reception

Dissertação apresentada para obtenção do Grau de Mestre em Engenharia Electrotécnica e de Computadores, pela Universidade Nova de Lisboa, Faculdade de Ciências e TecnologiaLow Earth Orbit (LEO) satellite networks can improve terrestrial wireless networks to allow global broadband services for Mobile Terminals (MT), regardless of the users' location. In this context, hybrid telecommunication systems combining satellites with Long Term Evolution (LTE) networks, like the LightSquared technology, are intended to provide ubiquitous high-speed services. This dissertation analyses the performance of a random access protocol that uses Hybrid Network-assisted Diversity Multiple Access (H-NDMA), for a LEO satellite system network, named by Satellite Random NDMA (SR-NDMA). The protocol also considers a Single Carrier-Frequency Domain Equalization (SC-FDE) scheme for the uplink transmission and a Multipacket Reception (MPR) receiver. In this scenario, the transmission of data packets between MTs and the Base Station (BS) is made through random access and schedule access slots, organized into super-frames with the duration of a Round Trip Time (RTT). A SR-NDMA simulator is implemented to measure the system performance in matters of throughput, energy consumption, system delay and also the protocol capacity to meet Quality of Service (QoS) requirements. A set of simulations tests were made with a random Poisson process tra c generation to validate the analytical model. The capacity to ful l the QoS requirements of a real-time tra c class was also tested.FCT/MEC: MPSat - PTDC/EEA-TEL/099074/2008, OPPORTUNISTIC CR - PTDC/EEA-TEL/115981/2009, Femtocells - PTDC/EEA-TEL/120666/2010 e ADIN - PTDC/EEI-TEL/2990/201

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

Traffic modelling in WLANs and cellular networks.

Over the past several years there has been a considerable amount of research in the field of traffic modelling for WLANs and Cellular Networks as well as the integration of these networks. To date, the focus of published work has been largely on the operation of delay sensitive calls. Because the voice calls are no longer the only service in wireless and cellular systems, multi-service traffic networks now consist of integrated services with distinctive Quality of Service (QoS) requirements. Therefore, a number of different schemes have been proposed to deal with this problem. Most of these schemes only consider mobility and multi-service traffic characteristics. However, few studies have considered the impact of buffering of voice calls in integrated voice and data services. Therefore, we aim to make a critical investigation of existing traffic models and offer generic traffic schemes for WLAN and Cellular networks in order to analyse the impact of buffering of voice calls in hybrid networks. For this purpose, an analytical model for performance evaluation of a single server network with voice and data traffic is considered. In this system, voice is given priority but can be buffered in a limited way. The analysis shows that this approach can be used in fast mobile systems

Application-aware scheduling for VoIP in Wireless Mesh Networks

Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit switched networks is a major challenge. This paper analyses different scheduling mechanisms for TDMA based access control in mesh networks as specified in the IEEE 802.16 2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns 2 simulation and mathematical analysis. The paper concludes that on demand scheduling of VoIP traffic typically deployed in 802.11 based WMNs is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator’s point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP aware scheduler significantly increases the number of supported calls