Today's home network usually involves connecting multiple PCs and peripheral devices, such as printers and scanners, together in a network. This provides the benefit of allowing the PCs in the network to share Internet access and other resources. However, it is expected in the future, the home area network (HAN) will grow and extend to other home devices such as home entertainment systems (including digital TV, hi-fi stereo, etc.), appliances, webcam, security alarm system, etc. Connecting other home devices to a HAN provides users with many benefits not available in today's home networks. For example, home devices capable of connecting to the future HAN are able to share the content downloaded from broadband access anywhere in the home. Users can also have remote access and control of their home devices. To extend the home area network to all these different home devices, however, means that the traffic between the ISP and future HAN will be very different from the traffic generated by today’s home network. In today's home network, which consists mainly of multiple PCs, a best-effort approach is able to satisfy the need, since most of the traffic generated by PCs is not real-time in nature. However, in future HANs, it is anticipated that traffic generated from home devices requiring real-time applications such as multimedia entertainment systems, teleconferencing, etc. will occupy a large proportion of the traffic between the ISP and future HANs. In addition, given the variety of home devices that could potentially be added to future HANs, the amount and variety of traffic between the ISP and a future HAN will certainly be very different from today's home network that is dominated by Internet/data traffic. To allow HAN users of these real-time applications and various types of home devices to continue enjoying seamless experiences in using their home devices without noticing significant delays or unnecessary interruptions, it is important for the ISP to be able to effectively manage the channel to the home so that it can provide sufficient bandwidth to ensure high QoS for home applications. The aim of this thesis is to understand the types of traffic that will be expected and to develop an analytical model that will represent the traffic behaviour between the ISP and future HANs to understand how to manage the channel to provide high QoS. In this thesis, we use the continuous-time PH/M/n/m preemptive priority queue to model the traffic behaviour between the ISP and a future HAN. Three classes of traffic are defined in this model: real-time, interactive, and unclassified. Each of these three traffic classes receives a unique priority level. From the model one can approximate the amount of bandwidth required to be allocated for each traffic class for each household so that the total bandwidth required is minimized while the QoS requirements (delay and blocking probability) of the traffic generated by the home devices are met. Thus this model could potentially be used as a network planning tool for ISPs to estimate how much bandwidth they need to provide per household for homes that use home area network. Alternatively, it could also be used to estimate what quality of service (e.g. what is the mean delay and blocking probability expected) given a certain amount of bandwidth per household. / October 2006

Our study investigates the performance of the WUSB standards and compares it to the Wimedia Standard. To the best of our knowledge, no technical contributions exist in the open literature at present simulating WUSB and its performance. The study showed that WUSB can achieve better throughput when bursting is enabled at the maximum burst size and it provides more accurate timing control of device activity than using the standard facilities of the WiMedia MAC. Our study also addresses protocol extensions and improvement to the original WUSB standard to support better Quality of Service (QoS). First improvement enables a di erent reservation mechanism along with contention based access to support higher priority security and medical system monitoring applications. Second improvement enables the host device to use an adaptive packet loss technique to change the packet size dynamically during the data transmission to achieve packet loss less than 10%. Third improvement enables redundancy in the cluster by adding a backup host to prevent mobility failures and changes. This backup host is chosen by a prede ned cost weighting function.

Quality of Service is a set of mechanisms created for differentiating users and data flows. Provisioning the QoS in service provider networks is always interesting for their business departments for they can offer and sell diverse services and to make more money. This thesis QoS for DSL subscribers of Pardis Online - the private ISP in Iran. The size of the network, the certain network connectivity limitations the ISP has in addition to the complexity of the services it provides, necessitate a comprehensive study to elucidate the different service types and requirements and then choosing a specific implementation of QoS in order to meet the ISP’s need. The internet bandwidth in the region that ISP operates is very expensive and the users suffer from the lack of bandwidth so the limited internet links are usually congested in the network. This document concludes with proposing a layered QoS that guarantees that the total bandwidth of an area in congested times is distributed to the users in that area proportionate to the subscribed band width. The resulting proposals are tested in the Lab and their performance under the simulated traffic is monitored and analyzed.

Wireless systems of today face the dual challenge of both supporting large traffic flows and providing reliable quality of service to different delay-sensitive applications. For such applications, it is essential to derive meaningful performance measures such as queue-length distribution and packet loss probability, while providing service guarantees. The concepts of effective bandwidth and effective capacity offer a powerful cross-layer approach that provides suitable performance metrics for the bandwidth and capacity of wireless channels supporting delay-sensitive traffic. Many wireless systems rely on multihop forwarding to reach destinations outside the direct range of the source. This work extends part of the methodology available for the design of wireless systems to the multihop paradigm. It describes the analysis of a communication system with two hops using this cross-layer approach. A framework is developed to study the interplay between the allocation of physical resources across the wireless hops and overall service quality as defined by a queueing criterion based on large deviations. Decoupling techniques introduce simple ways of analyzing the queues independently. Numerical analysis helps identify fundamental performance limits for Rayleigh block fading wireless channel models with independent and identically distributed blocks. Simulation studies present comparable results akin to that obtained using the analytical framework. These results suggest that it is imperative to account for queueing aspects while analyzing delay-sensitive wireless communication systems.

With the maturity of wireless network technologies, more and more real-time data and applications, such as video on demand, video teleconferencing, are being transmitted on the wireless environment. In the next generation high-speed wireless networks, the important issue is how to provide quality-of-service (QoS) guarantees as they are expected to support multimedia applications. Although the QoS provisioning problem arises in wireline network as well, scarcity of bandwidth, mobility of hosts, high bit error rate, etc., provisioning a challenging task in wireless network. It is important to reduce the dropping rate caused by lack of available bandwidth in the handoff target cell. Call admission control is a key component for real-time, continuous media connections to guarantee QoS for these applications. In this thesis, we proposed a new call admission control scheme in wireless network. The scheme is integration bandwidth reservation and reallocation strategy and more effective to reach that decrease the handoff call dropping probability (CDP), new call blocking probability (CBP) and increase the bandwidth utilization (BU). Furthermore, our scheme also considered aspects of multi-class and fairness when the system allocated resource to calls. Extensive simulation results show that our scheme outperforms the best previously known schemes to provide QoS guarantees for multimedia traffic.

This paper proposed BRQ (Bandwidth-Reservation QoS scheme) for mobile Ad-Hoc wireless network. Unlike many schemes about IFS or backoff time interval of DCF in IEEE 802.11, BRQ uses the bandwidth reservation mechanism to support QoS. Because of the compatibility of BRQ with other schemes, we believe that BRQ scheme can achieve higher performance by integrating with other schemes. We evaluate the performance of BRQ, DCF in IEEE 802.11, EDCF in 802.11e, and BRQ+ (the combined scheme of BRQ and EDCF) in terms of their differentiation capability, network throughput, and average end-to-end delay. The simulation results demonstrate the effectiveness of the BRQ scheme and show that the BRQ+ scheme can achieve higher performance by integrating BRQ and EDCF.

GPRS is one of popular topics for mobile communication. To satisfy the Quality of Service (QoS) requirement for multimedia transmission, the QoS is divided into four classes, which are conversational, interactive, streaming and background class. When mobile communication network transmits the multimedia data, it needs a proper scheduling algorithm to assign the radio resource and make every data meet the requirement of the QoS in GPRS system. In this thesis, we discuss the properties of every traffic class. For each traffic class, we propose a transmission method. The proposed methods are integrated with the link adaptation to develop a scheduling algorithm to suit the QoS requirement in the GPRS system. In addition, we introduce the FIFO scheduling algorithm and the scheduling algorithm that has priority. Our methods are then compared with those existing algorithms. We use OPNET simulation system to study the FIFO scheduling algorithm, the priority-scheduling algorithm and our new scheduling algorithms. The effects on the delay time, packet loss ratio and throughput for every scheduling algorithm are analyzed and compared. The simulations show that the new scheduling algorithm can satisfy all QoS requirements, and the performance of the new scheduling algorithm is better than that of the other scheduling algorithms in the interactive class.

With the rapid proliferation of wireless networks and mobile computing applications, Quality of Service (QoS) for mobile ad hoc networks (MANETs) has received increased attention. Security is a critical aspect of QoS provisioning in the MANET environment. Without protection from a security mechanism, attacks on QoS signaling system could result in QoS routing malfunction, interference of resource reservation, or even failure of QoS provision. Due to the characteristics of the MANETs, such as rapid topology change and limited communication and computation capacity, the conventional security measures cannot be applied and new security techniques are necessary. However, little research has been done on this topic. In this dissertation, the security issues will be addressed for MANET QoS systems. The major contributions of this research are: (a) design of an authentication mechanism for ad hoc networks; (b) design of a security mechanism to prevent and detect attacks on the QoS signaling system; (c) design of an intrusion detection mechanism for bandwidth reservation to detect QoS attacks and Denial of Service (DoS) attacks. These three mechanisms are evaluated through simulation.

Best effort networks fail to deliver the level of service emerging Internet applications demand. As a result many networks are being transformed to Quality of Service (QoS) networks, of which most are Differentiated Services (DiffServ) networks. While the deployment of such networks has been feasible, it is extremely difficult to overhaul the transport layer protocols such as Transmission Control Protocol (TCP) running on hundreds of millions of end nodes around the world. TCP, which has been designed to run on a best effort network, perform poorly in a DiffServ network. It fails to deliver the performance guarantees expected of DiffServ. In this thesis we investigate two aspects of TCP performance in a DiffServ network unaccounted for in previous studies. We develop a deterministic model of TCP that intrinsically captures flow aggregation, a key component of DiffServ. The other important aspect of TCP considered in this thesis is its' transient behavior. Using our deterministic model we derive a classical control system model of TCP applicable in a DiffServ network. Performance issues of TCP can potentially inhibit the adoption of DiffServ. A DiffServ network commonly use token buckets, that are placed at the edge of the network, to mark packets according to their conformance to Service Level Agreements (SLA). We propose two token bucket variants designed to mitigate TCP issues present in a DiffServ network. Our first proposal incorporates a packet queue alongside the token bucket. The other proposal introduces a feedback controller around the token bucket. We validate both analytically and experimentally the performance of the proposed token buckets. By confining our changes to the token bucket we avoid any changes at end-nodes. The proposed token buckets can also be incrementally deployed. Most part of the Internet still remains as a best effort network. However, most nodes run various QoS functions locally. We look at one such important QoS function, i.e. the ability to survive against flows that are non-responsive to congestion, the equivalent of a Denial of Service (DoS) attack. We analyze existing techniques and propose improvements.

Fixed Broadband Wireless Access (BWA) for the last mile is a promising technology which can offer high speed voice, video and data service and fill the technology gap between Wireless LANs and wide area networks. This is seen as a challenging competitor to conventional wired last mile access systems like DSL and cable, even in areas where those technologies are already available. More importantly the technology can provide a cost-effective broadband access solution in rural areas beyond the reach of DSL or cable and in developing countries with little or no wired last mile infrastructure. Earlier BWA systems were based on proprietary technologies which made them costly and impossible to interoperate. The IEEE 802.16 set of standards was developed to level the playing field. An industry group the WiMAX Forum, was established to promote interoperability and compliance to this standard. This thesis gives an overview of the IEEE 802.16 WirelessMAN OFDM standard which is the basis for Fixed WiMAX. An in depth description of the medium access control (MAC) layer is provided and functionality of its components explained. We have concentrated our effort on enhancing the performance of Fixed WiMAX for VoIP services, and best effort traffic which includes e-mail, web browsing, peer-to-peer traffic etc. The MAC layer defines four native service classes for differentiated QoS levels from the onset. The unsolicited grant service (UGS) class is designed to support real-time data streams consisting of fixed-size data packets issued at periodic intervals, such as T1/E1 and Voice over IP without silence suppression, while the non-real-time polling service (nrtPS) and best effort (BE) are meant for lower priority traffic. QoS and efficiency are at opposite ends of the scale in most cases, which makes it important to identify the trade-off between these two performance measures of a system. We have analyzed the effect the packetization interval of a UGS based VoIP stream has on system performance. The UGS service class has been modified so that the optimal packetization interval for VoIP can be dynamically selected based on PHY OFDM characteristics. This involves cross layer communication between the PHY, MAC and the Application Layer and selection of packetization intervals which keep the flow within packet loss and latency bounds while increasing efficiency. A low latency retransmission scheme and a new ARQ feedback scheme for UGS have also been introduced. The goal being to guarantee QoS while increasing system efficiency. BE traffic when serviced by contention based access is variable in speed and latency, and low in efficiency. A detailed analysis of the contention based access scheme is done using Markov chains. This leads to optimization of system parameters to increase utilization and reduce overheads, while taking into account TCP as the most common transport layer protocol. nrtPS is considered as a replacement for contention based access. Several enhancements have been proposed to increase efficiency and facilitate better connection management. The effects of proposed changes are validated using analytical models in Matlab and verified using simulations. A simulation model was specifically created for IEEE 802.16 WirelessMAN OFDM in the QualNet simulation package. In essence the aim of this work was, to develop means to support a maximum number of users, with the required level of service, using the limited wireless resource.