A novel radio resource management (RRM) scheme, which jointly considers the system characteristics from the physical, link and network layers, is proposed for cellular code division multiple access (CDMA) systems. Specifically, the power distribution at the physical layer distributes only the necessary amount of power to each connection in order to achieve its required signal-to-interference-plus-noise ratio (SINR). The rate allocation guarantees the required delay/jitter for real-time traffic and the minimum transmission rate requirement for non-real-time traffic. Efficient rate allocation is achieved by making use of the randomness and burstiness of the packet generation process. At the link layer, a packet scheduling scheme is developed based on the information of power distribution and rate allocation from the physical layer to achieve guaranteed quality of service (QoS). It schedules the system resource on a time slot basis to efficiently utilize the system resource in every time slot and to improve the packet throughput for non-real-time traffic. A connection admission control (CAC) scheme based on the lower layer resource allocation information is proposed at the network layer. The CAC scheme also makes use of user mobility information to reduce handoff connection dropping probability (HCDP). Theoretical analysis of the grade of service (GOS) performance, in terms of new connection blocking probability (NCBP), HCDP, and resource utilization, is given. Numerical results show that the proposed RRM scheme can achieve both effective QoS guarantee and efficient resource utilization.

An adaptive resource reservation and handoff priority scheme, which jointly considers the characteristics from the physical, link and network layers, is proposed for a packet switching Multicode (MC)-CDMA cellular network supporting multimedia applications. A call admission region is derived for call admission control (CAC) and handoff management with the satisfaction of quality of service (QoS) requirements for all kinds of multimedia traffic, where the QoS parameters include the wireless transmission bit error rate (BER), the packet loss rate (PLR) and delay requirement. The BER requirement is guaranteed by properly arranging simultaneous packet transmissions, whereas the PLR and delay requirements are guaranteed by the proposed packet scheduling and partial packet integration scheme. To give service priority to handoff calls, a threshold-based adaptive resource reservation scheme is proposed on the basis of a practical user mobility model and a proper handoff request prediction scheme. The resource reservation scheme gives handoff calls a higher admission priority over new calls, and is designed to adjust the reservation-request time threshold adaptively according to the varying traffic load. The individual reservation requests form a common reservation pool, and handoff calls are served on a first-come-first-serve basis. By exploiting the transmission rate adaptability of video calls to the available radio resources, the resources freed from rate-adaptive high-quality video calls by service degradation can be further used to prioritize handoff calls. With the proposed resource reservation and handoff priority scheme, the dynamic properties of the system can be closely captured and a better grade of service (GoS) in terms of new call blocking and handoff call dropping probabilities(rates) can be achieved compared to other schemes in literature. Numerical results are presented to show the improvement of the GoS performance and the efficient utilization of the radio resources.

With the rapid growth of multimedia applications and the advances of wireless communication technologies, quality-of-service (QoS) provisioning for multimedia services in heterogeneous wireless networks has been an important issue and drawn much attention from both academia and industry. Due to the hostile transmission environment and limited radio resources, QoS provisioning in wireless networks is much more complex and difficult than in its wired counterpart. Moreover, due to the lack of central controller in the networks, distributed network control is required, adding complexity to QoS provisioning. In this thesis, medium access control (MAC) with QoS provisioning is investigated for both single- and multi-hop wireless networks including wireless local area networks (WLANs), wireless ad hoc networks, and wireless mesh networks. Originally designed for high-rate data traffic, a WLAN has limited capability to support delay-sensitive voice traffic, and the service for voice traffic may be impacted by data traffic load, resulting in delay violation or large delay variance. Aiming at addressing these limitations, we propose an efficient MAC scheme and a call admission control algorithm to provide guaranteed QoS for voice traffic and, at the same time, increase the voice capacity significantly compared with the current WLAN standard. In addition to supporting voice traffic, providing better services for data traffic in WLANs is another focus of our research. In the current WLANs, all the data traffic receives the same best-effort service, and it is difficult to provide further service differentiation for data traffic based on some specific requirements of customers or network service providers. In order to address this problem, we propose a novel token-based scheduling scheme, which provides great flexibility and facility to the network service provider for service class management. As a WLAN has small coverage and cannot meet the growing demand for wireless service requiring communications ``at anywhere and at anytime", a large scale multi-hop wireless network (e.g., wireless ad hoc networks and wireless mesh networks) becomes a necessity. Due to the location-dependent contentions, a number of problems (e.g., hidden/exposed terminal problem, unfairness, and priority reversal problem) appear in a multi-hop wireless environment, posing more challenges for QoS provisioning. To address these challenges, we propose a novel busy-tone based distributed MAC scheme for wireless ad hoc networks, and a collision-free MAC scheme for wireless mesh networks, respectively, taking the different network characteristics into consideration. The proposed schemes enhance the QoS provisioning capability to real-time traffic and, at the same time, significantly improve the system throughput and fairness performance for data traffic, as compared with the most popular IEEE 802.11 MAC scheme.

Wireless mesh networking has emerged as a promising technology for future broadband wireless access, providing a viable and economical solution for both peer-to-peer applications and Internet access. The success of wireless mesh networks (WMNs) is highly contingent on effective radio resource management. In conventional wireless networks, system throughput is usually a common performance metric. However, next-generation broadband wireless access networks including WMNs are anticipated to support multimedia traffic (e.g., voice, video, and data traffic). With heterogeneous traffic, quality-of-service (QoS) provisioning and fairness support are also imperative. Recently, wireless mesh networking for suburban/rural residential areas has been attracting a plethora of attentions from industry and academia. With austere suburban and rural networking environments, multi-hop communications with decentralized resource allocation are preferred. In WMNs without powerful centralized control, simple yet effective resource allocation approaches are desired for the sake of system performance melioration. In this dissertation, we conduct a comprehensive research study on the topic of radio resource management for WMNs supporting multimedia traffic. In specific, this dissertation is intended to shed light on how to effectively and efficiently manage a WMN for suburban/rural residential areas, provide users with high-speed wireless access, support the QoS of multimedia applications, and improve spectrum utilization by means of novel radio resource allocation. As such, five important resource allocation problems for WMNs are addressed, and our research accomplishments are briefly outlined as follows: Firstly, we propose a novel node clustering algorithm with effective subcarrier allocation for WMNs. The proposed node clustering algorithm is QoS-aware, and the subcarrier allocation is optimality-driven and can be performed in a decentralized manner. Simulation results show that, compared to a conventional conflict-graph approach, our proposed approach effectively fosters frequency reuse, thereby improving system performance; Secondly, we propose three approaches for joint power-frequency-time resource allocation. Simulation results show that all of the proposed approaches are effective in provisioning packet-level QoS over their conventional resource allocation counterparts. Our proposed approaches are of low complexity, leading to preferred candidates for practical implementation; Thirdly, to further enhance system performance, we propose two low-complexity node cooperative resource allocation approaches for WMNs with partner selection/allocation. Simulation results show that, with beneficial node cooperation, both proposed approaches are promising in supporting QoS and elevating system throughput over their non-cooperative counterparts; Fourthly, to further utilize the temporarily available radio spectrum, we propose a simple channel sensing order for unlicensed secondary users. By sensing the channels according to the descending order of their achievable rates, we prove that a secondary user should stop at the first sensed free channel for the sake of optimality; and Lastly, we derive a unified optimization framework to effectively attain different degrees of performance tradeoff between throughput and fairness with QoS support. By introducing a bargaining floor, the optimal tradeoff curve between system throughput and fairness can be obtained by solving the proposed optimization problem iteratively.

Internet protocol TV (IPTV) is predicted to be the key technology winner in the future. Efforts to accelerate the deployment of IPTV centralized model which is combined of VHO, encoders, controller, access network and Home network. Regardless of whether the network is delivering live TV, VOD, or Time-shift TV, all content and network traffic resulting from subscriber requests must traverse the entire network from the super-headend all the way to each subscriber's Set-Top Box (STB).IPTV services require very stringent QoS guarantees When IPTV traffic shares the network resources with other traffic like data and voice, how to ensure their QoS and efficiently utilize the network resources is a key and challenging issue. For QoS measured in the network-centric terms of delay jitter, packet losses and bounds on delay. The main focus of this thesis is on the optimized bandwidth allocation and smooth datatransmission. The proposed traffic model for smooth delivering video service IPTV network with its QoS performance evaluation. According to Maglaris et al [5] First, analyze the coding bit rate of a single video source. Various statistical quantities are derived from bit rate data collected with a conditional replenishment inter frame coding scheme. Two correlated Markov process models (one in discrete time and one incontinuous time) are shown to fit the experimental data and are used to model the input rates of several independent sources into a statistical multiplexer. Preventive control mechanism which is to be include CAC, traffic policing used for traffic control.QoS has been evaluated of common bandwidth scheduler( FIFO) by use fluid models with Markovian queuing method and analysis the result by using simulator andanalytically, Which is measured the performance of the packet loss, overflow and mean waiting time among the network users.

Internet protocol TV (IPTV) is predicted to be the key technology winner in the future. Efforts to accelerate the deployment of IPTV centralized model which is combined of VHO, encoders, controller, access network and Home network. Regardless of whether the network is delivering live TV, VOD, or Time-shift TV, all content and network traffic resulting from subscriber requests must traverse the entire network from the super-headend all the way to each subscriber's Set-Top Box (STB). IPTV services require very stringent QoS guarantees When IPTV traffic shares the network resources with other traffic like data and voice, how to ensure their QoS and efficiently utilize the network resources is a key and challenging issue. For QoS measured in the network-centric terms of delay jitter, packet losses and bounds on delay. The main focus of this thesis is on the optimized bandwidth allocation and smooth data transmission. The proposed traffic model for smooth delivering video service IPTV network with its QoS performance evaluation. According to Maglaris et al [5] first, analyze the coding bit rate of a single video source. Various statistical quantities are derived from bit rate data collected with a conditional replenishment inter frame coding scheme. Two correlated Markov process models (one in discrete time and one in continuous time) are shown to fit the experimental data and are used to model the input rates of several independent sources into a statistical multiplexer. Preventive control mechanism which is to be including CAC, traffic policing used for traffic control. QoS has been evaluated of common bandwidth scheduler( FIFO) by use fluid models with Markovian queuing method and analysis the result by using simulator and analytically, Which is measured the performance of the packet loss, overflow and mean waiting time among the network users.

The increasing usage of information technology in military affairs raises the need for robust high capacity radio networks. The network will be used to provide several different types of services, for example group calls and situation awareness services. All services have specific demands on packet delays and packet losses in order to be fully functional, and therefore there is a need for a Quality of Service (QoS) mechanism in the network. In this master thesis we examine the possibility to provide a QoS mechanism in Ad Hoc networks by using priority queues. The study includes two different queuing schemes, namely fixed priority queuing and weighted fair queuing. The performance of the two queuing schemes are evaluated and compared with respect to the ability to provide differentiation in network delay, i.e., provide high priority traffic with lower delays than low priority traffic. The study is mainly done by simulations, but for fixed priority queuing we also derive a analytical approximation of the network delay. Our simulations show that fixed priority queuing provides a sharp delay differentiation between service classes, while weighted fair queuing gives the ability to control the delay differentiation. One of those queuing schemes alone might not be the best solution for providing QoS, instead we suggest that a combination of them is used.

With the Internet expansion, real-time transmission over Internet is becoming a new promising application. Successful real-time communication over IP networks requires reasonably reliable, low delay, low loss date transport. Since Internet is a non-synchronous packet switching network, high load and lack of guarantees on data delivery make real-time communication such as Voice and Video over IP a challenging application to become realistic on the Internet. This thesis work is composed of two parts within real-time voice and video communication: network simulation and measurement on the real Internet. In the network simulation, I investigate the requirement for the network"overprovisioning"in order to reach certain quality-of-service. In the experiments on the real Internet, I simulate real-time transmission with UDP packets along two different traffic routes and analyze the quality-of- service I get in each case. The overall contribution of this work is: To create scenarios to understand the concept of overprovisioning and how it affects the quality-of-service. To develop a mechanism to measure the quality-of-service for real-time traffic provided by the current best-effort network.

A novel radio resource management (RRM) scheme, which jointly considers the system characteristics from the physical, link and network layers, is proposed for cellular code division multiple access (CDMA) systems. Specifically, the power distribution at the physical layer distributes only the necessary amount of power to each connection in order to achieve its required signal-to-interference-plus-noise ratio (SINR). The rate allocation guarantees the required delay/jitter for real-time traffic and the minimum transmission rate requirement for non-real-time traffic. Efficient rate allocation is achieved by making use of the randomness and burstiness of the packet generation process. At the link layer, a packet scheduling scheme is developed based on the information of power distribution and rate allocation from the physical layer to achieve guaranteed quality of service (QoS). It schedules the system resource on a time slot basis to efficiently utilize the system resource in every time slot and to improve the packet throughput for non-real-time traffic. A connection admission control (CAC) scheme based on the lower layer resource allocation information is proposed at the network layer. The CAC scheme also makes use of user mobility information to reduce handoff connection dropping probability (HCDP). Theoretical analysis of the grade of service (GOS) performance, in terms of new connection blocking probability (NCBP), HCDP, and resource utilization, is given. Numerical results show that the proposed RRM scheme can achieve both effective QoS guarantee and efficient resource utilization.

An adaptive resource reservation and handoff priority scheme, which jointly considers the characteristics from the physical, link and network layers, is proposed for a packet switching Multicode (MC)-CDMA cellular network supporting multimedia applications. A call admission region is derived for call admission control (CAC) and handoff management with the satisfaction of quality of service (QoS) requirements for all kinds of multimedia traffic, where the QoS parameters include the wireless transmission bit error rate (BER), the packet loss rate (PLR) and delay requirement. The BER requirement is guaranteed by properly arranging simultaneous packet transmissions, whereas the PLR and delay requirements are guaranteed by the proposed packet scheduling and partial packet integration scheme. To give service priority to handoff calls, a threshold-based adaptive resource reservation scheme is proposed on the basis of a practical user mobility model and a proper handoff request prediction scheme. The resource reservation scheme gives handoff calls a higher admission priority over new calls, and is designed to adjust the reservation-request time threshold adaptively according to the varying traffic load. The individual reservation requests form a common reservation pool, and handoff calls are served on a first-come-first-serve basis. By exploiting the transmission rate adaptability of video calls to the available radio resources, the resources freed from rate-adaptive high-quality video calls by service degradation can be further used to prioritize handoff calls. With the proposed resource reservation and handoff priority scheme, the dynamic properties of the system can be closely captured and a better grade of service (GoS) in terms of new call blocking and handoff call dropping probabilities(rates) can be achieved compared to other schemes in literature. Numerical results are presented to show the improvement of the GoS performance and the efficient utilization of the radio resources.