The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.

The third generation, 3G, mobile telephone systems are designed for multimedia communication and will offer us similar services as in our stationary computers. This will involve large traffic loads, especially in the downlink direction, i.e. from base station to terminal. To improve the downlink capacity for packet data services a new concept is included in evolved 3G networks. The concept is called High Speed Data Packet Access, HSDPA, and provides peak bit rates of 14 Mbps. HSDPA uses a so-called best effort channel, i.e. it is developed for services that do not require guaranteed bit rates. The channel is divided in time between the users and a scheduling algorithm is used to allocate the channel among them. Streaming is a common technology for video transmission over the Internet and with 3G it is supposed to become popular also in our mobiles. Streaming generates lots of data traffic in the downlink direction and it would thus be satisfying to make use of the high bit rates HSDPA provides. The problem is that streaming requires reasonable stable bit rates, which is not guaranteed using HSDPA. The aim of this study is to modify the scheduling algorithms to prioritise streaming over web users and provide streaming Quality of Service, QoS. QoS is the ability to guarantee certain transmission characteristics. The results of the study show that it is hard to improve the streaming capacity by modifications of the scheduling. Of course, a consequence is that the web user throughput is decreased and to avoid this, new users have to be rejected by the admission control.The solution is to prioritise the streaming users both in the scheduling algorithm and in the admission control, i.e. when the system is nearly full new web users are rejected. By doing so the results are significantly improved.

Wireless Mesh Networking is envisioned as an economically viable paradigm and a promising technology in providing wireless broadband services. The wireless mesh backbone consists of fixed mesh routers that interconnect different mesh clients to themselves and to the wireline backbone network. In order to approach the wireline servicing level and provide same or near QoS guarantees to different traffic flows, the wireless mesh backbone should be quality-of-service (QoS) aware. A key factor in designing protocols for a wireless mesh network (WMN) is to exploit its distinct characteristics, mainly immobility of mesh routers and less-constrained power consumption. In this work, we study the effect of varying the transmission power to achieve the required signal-to-interference noise ratio for each link and, at the same time, to maximize the number of simultaneously active links. We propose a QoS-aware routing framework by using transmission power control. The framework addresses both the link scheduling and QoS routing problems with a cross-layer design taking into consideration the spatial reuse of the network bandwidth. We formulate an optimization problem to find the optimal link schedule and use it as a fitness function in a genetic algorithm to find candidate routes. Using computer simulations, we show that by optimal power allocation the QoS constraints for the different traffic flows are met with more efficient bandwidth utilization than the minimum power allocations.

With the exploding increase of mobile users and the release of new wireless applications, the high bandwidth requirement has been taking as a main concern for the design and development of the wireless techniques. There is no doubt that broadband wireless access with the support of heterogeneous kinds of applications is the trend in the next generation wireless networks. As a promising broadband wireless access standard, IEEE 802.16 has attracted extensive attentions from both industry and academia due to its high data rate and the inherent media access control (MAC) mechanism, which takes the service differentiation and quality of service (QoS) provisioning into account. To achieve service differentiation and QoS satisfaction for heterogenous applications is a very complicated issue. It refers to many fields, such as connection admission control (CAC), congestion control, routing algorithm, MAC protocol, and scheduling scheme. Among these fields, packet scheduling plays one of the most important roles in fulfilling service differentiation and QoS provisioning. It decides the order of packet transmissions, and provides mechanisms for the resource allocation and multiplexing at the packet level to ensure that different types of applications meet their service requirements and the network maintains a high resource utilization. In this thesis, we focus on the packet scheduling for difficult types of services in IEEE 802.16 networks, where unicast and mulitcast scheduling are investigated. For unicast scheduling, two types of services are considered: non-real-time polling service (nrtPS) and best effort (BE) service. We propose a flexible and efficient resource allocation and scheduling framework for nrtPS applications to achieve a tradeoff between the delivery delay and resource utilization, where automatic repeat request (ARQ) mechanisms and the adaptive modulation and coding (AMC) technique are jointly considered. For BE service, considering the heterogeneity of subscriber stations (SSs) in IEEE 802.16 networks, we propose the weighted proportional fairness scheduling scheme to achieve the flexible scheduling and resource allocation among SSs based on their traffic demands/patterns. For multicast scheduling, a cooperative multicast scheduling is proposed to achieve high throughput and reliable transmission. By using the two-phase transmission model to exploit the spatial diversity gain in the multicast scenario, the proposed scheduling scheme can significantly improve the throughput not only for all multicast groups, but also for each group member. Analytical models are developed to investigate the performance of the proposed schemes in terms of some important performance measurements, such as throughput, resource utilization, and service probability. Extensive simulations are conducted to illustrate the efficient of the proposed schemes and the accuracy of the analytical models. The research work should provide meaningful guidelines for the system design and the selection of operational parameters, such as the number of TV channels supported by the network, the achieved video quality of each SS in the network, and the setting of weights for SSs under different BE traffic demands.

Cognitive Radio Networks (CRNs) provide a solution for the spectrum scarcity problem facing the wireless communications community. However, due to the infancy of CRNs, further research is needed before we can truly benefit from CRNs. The basic concept of CRNs relies on utilizing the unused spectrum of a primary network, without interfering with the activity of primary users (PUs). In order to successfully achieve that, users in a CRN has to perform spectrum sensing, spectrum management, spectrum mobility, and spectrum sharing. The latter, which is the focus of our research, deals with how secondary users (SUs) share the unused spectrum. Furthermore, to be able to utilize CRNs in practical applications, a certain level of quality-of-service (QoS) should be guaranteed to SUs in such networks. QoS requirements vary according to the application. Interested in voice communications, we propose a packet scheduling scheme that orders the SUs' transmissions according to the packet dropping rate and the number of packets queued waiting for transmission. Two medium access control (MAC) layer protocols, based on the mentioned scheduling scheme, are proposed for a centralized CRN. In addition, the scheduling scheme is adapted for a distributed CRN, by introducing a feature that allows SUs to organize access to the available spectrum without the need for a central unit. Finally, extensive simulation based experiments are carried out to evaluate the proposed protocols and compare their performance with that of other MAC protocols designed for CRNs. These results reflect the effectiveness of our proposed protocols to guarantee the required QoS for voice packet transmission, while maintaining fairness among SUs in a CRN.

Web service selection problem focuses on how to choose component Web services to satisfy user¡¦s non-functional (or QoS) need, and it has been extensively studied in the past. The QoS measures include reliability, response time, and execution cost. However, in some applications, execution result, as demonstrated on some output values, matters, and this is seldom addressed by previous researches. In our work, we proposed an approach to guide the WS selection with the goal to meet user¡¦s preferences on the composition execution outcome. In addition, we consider the partner relationship between Web services. Some partner Web services may produce more desired execution result, such as better quality or a discount, than others. In our approach, we use Bayesian Network to guide Web services selection. Specifically, we propose two Bayesian Network-based methods: Partner-first Bayesian Network and Probability-first Bayesian Network. Both methods rank Web services by considering user¡¦s preference, user¡¦s input variables, and the past execution results of Web services. The experiment result shows that the proposed Bayesian Network methods perform better than the other more straightforward methods.

Data intensive applications, requiring reliability and strict delay constraints, have emerged recently and they necessitate a different approach to analyzing system performance. In my work, I establish a framework that relates physical channel parameters to the queueing performance for a single-user wireless system. I then seek to assess the potential benefits of multirate techniques, such as hybrid-ARQ (Automatic Repeat reQuest), in the context of delay-sensitive communications. Present methods of analysis in an information theoretic paradigm define capacity assuming that long codewords can be used to take advantage of the ergodic properties of the fading wireless channel. This definition provides only a limited characterization of the channel in the light of delay constraints. The assumption of independent and identically distributed channel realizations tends to over-estimate the system performance by not considering the inherent time correlation. A finite-state continuous time Markov channel model that I formulate enables me to partition the instantaneous data-rate received at the destination into a finite number of states, representing layers in a hybrid-ARQ scheme. The correlation of channel has been incorporated through level crossing rates as transition rates in the Markov model. The large deviation principle governing the buffer overflow of the Markov model, is very sensitive to channel memory, is tractable, and gives a good estimate of the system performance. Metrics such as effective capacity and probability of buffer overflow, that are obtained through large deviations have been related to the wireless physical layer parameters through the model. Using the above metrics under QoS constraints, I establish the quantitative performance advantage of using hybrid-ARQ over traditional systems. I conduct this inquiry by restricting attention to the case where the expected transmit power is fixed at the transmitter. The results show that hybrid-ARQ helps us in obtaining higher effective capacity, but it is very difficult to support delay sensitive communication over wireless channel in the absence of channel knowledge and dynamic power allocation strategies.

Multimedia applications over the Internet are getting more and more popular. While non-real-time streaming services, such as YouTube and Megavideo, are attracting millions of visiting per day, real-time conferencing applications, of which some instances are Skype and Yahoo Voice Chat, provide an interesting experience of communication. Together, they make the fancy Internet world become more and more amusing. Undoubtedly, multimedia flows will eventually dominate the computer network in the future. As the population of multimedia flows increases gradually on the Internet, quality of their service (QoS) is more of a concern. At the moment, the Internet does not have any guarantee on the quality of multimedia services. To completely surpass this limitation, modifications to the network structure is a must. However, it will take years and billions of dollars in investment to achieve this goal. Meanwhile, it is essential to find alternative ways to improve the quality of multimedia services over the Internet. In the past few years, many endeavors have been carried on to solve the problem. One interesting approach focuses on the development of end-to-end congestion control strategies for UDP multimedia flows. Traditionally, packet losses and delays have been commonly used to develop many known control schemes. Each of them only characterizes some different aspects of network congestion; hence, they are not ideal as feedback signals alone. In this research, the flow accumulation is the signal used in feedback for flow control. It has the advantage of reflecting both packet losses and delays; therefore, it is a better choice. Using network simulations, the accumulations of real-time audio applications are collected to construct adaptive flow controllers. The reason for choosing these applications is that they introduce more control challenges than non-real-time services. One promising flow control strategy was proposed by Bhattacharya and it was based on Model Predictive Control (MPC). The controller was constructed from an ARX predictor. It was demonstrated that this control scheme delivers a good QoS while reducing bandwidth use in the controlled flows by 31 percent to 44 percent. However, the controller sometime shows erratic response and bandwidth usage jumps frequently between lowest and highest values. This is not desirable. For an ideal controller, the controlled bandwidth should vary near its mean. To eliminate the deficiency in the strategy proposed by Bhattacharya, it is proposed to introduce a feed forward term into the MPC formulation, in addition to the feedback terms. Simulations show that the modified MPC strategy maintains the benefits of the Bhattacharya strategy. Furthermore, it increases the probability of bandwidth savings from 58 percent for the case of Bhattacharya model to about 99 percent for this work.

We propose an object-based multimedia model for specifying the QoS (quality of service) requirements, such as the maximum data-dropping rate or the maximum data-delay rate. We also present a resource allocation model, called the net-profit model, in which the satisfaction of user¡¦s QoS requirements is measured by the benefit earned by the system. Based on the net-profit model, the system is rewarded if it can allocate enough resources to a multimedia delivery request and fulfill the QoS requirements specified by the user. At the same time, the system is penalized if it cannot allocate enough resources to a multimedia delivery request. In this dissertation, we present our research in developing optimal solutions for multimedia stream delivery in bandwidth limited networks. To fulfill the QoS requirements, the resource, such as bandwidth, should be reserved in advance. Hence, we first investigate how to allocate a resource such that the QoS satisfaction is maximized, assuming that the QoS requirements are given a priori. The proposed optimal solution has significant improvement over the based line algorithm, EDF (Earliest Deadline First). Among all the optimal solutions found from the above problem, the net-profit may be distributed unevenly among the multimedia delivery requests. Furthermore, we tackle the fairness problem -- how to allocate a resource efficiently so that the difference of the net-profit between two requests is minimized over all the possible optimal solutions of the maximum total net-profit. A dynamic programming based algorithm is proposed to find all the possible optimal solutions and, in addition, three filters are conducted to improve the efficiency of the proposed algorithm. The experimental results show that the filters prune out unnecessary searches and improve the performance significantly, especially when the number of tasks increases. For some multimedia objects, they might need to be delivered in whole, indivisible, so we extend the proposed multimedia object-based model to indivisible objects. A dynamic programming based algorithm is presented to find an optimal solution of the delivery problem, where the total net-profit is maximized.

IEEE 802.11 has become the standard in wireless LAN. Originally, 802.11 is designed for the best-effort services only. To support the increasing demand of delay-sensitive applications, IEEE 802.11 Task Group E is developing a QoS-aware MAC protocol, EDCF, for differentiated services. However, when the network becomes congested, there exists unexpected packet delay due to collisions and retransmissions. This thesis proposes a dynamic contention control (DCC) scheme to reduce packet delay and increase the percentage of packets arriving in time. DCC estimates per-hop delay, denoted as Mrtt, and end-to-end delay, denoted as Sigma_t, based on either the received MAC-layer ACK or the control packets of a reactive routing protocol. Then, Mrtt and Sigma_t are used to dynamically adjust the associated contention window for each priority. Besides, when a frame is retransmitted, the backoff time is determined according to the remaining end-to-end delay instead of a uniformly distributed random number. For the propose of evaluation, we perform simulations on the well-known network simulator, NS-2. DCC is compared with the EDCF and one previously proposed scheme, AEDCF. The simulation results demonstrate the effectiveness and superiority of DCC.