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.

The imminent inception of third-generation (3G) mobile communication networks offers an unprecedented opportunity for the development of video streaming applications through wireless Internet access. Different design challenges exist in implementing video streaming connections spanning both wired and wireless domains. A split-domain TCP-friendly streaming video transmission protocol is presented based on adaptive rate encoding in the MPEG-4 video format. Network simulations are conducted to demonstrate the benefits and viability of such a video streaming scheme over existing options. Further feature enhancements and refinements are necessary for the proposed protocol to achieve its full potential.

In this thesis, a novel Scalable Multiple Description Coding (SMDC) framework is proposed. To address the bandwidth fluctuation, packet loss and heterogeneity problems in the wireless networks and further enhance the error resilience tools in Moving Pictures Experts Group 4 (MPEG-4), the joint design of layered coding (LC) and multiple description coding (MDC) is explored. It leverages a proposed distributed multimedia delivery mobile network (D-MDMN) to provide path diversity to combat streaming video outage due to handoff in Universal Mobile Telecommunications System (UMTS). The corresponding intra-RAN (Radio Access Network) handoff and inter-RAN handoff procedures in D-MDMN are studied in details, which employ the principle of video stream re-establishing to replace the principle of data forwarding in UMTS. Furthermore, a new IP (Internet Protocol) Differentiated Services (DiffServ) video marking algorithm is proposed to support the unequal error protection (UEP) of LC components of SMDC. Performance evaluation is carried through simulation using OPNET Modeler 9. 0. Simulation results show that the proposed handoff procedures in D-MDMN have better performance in terms of handoff latency, end-to-end delay and handoff scalability than that in UMTS. Performance evaluation of our proposed IP DiffServ video marking algorithm is also undertaken, which shows that it is more suitable for video streaming in IP mobile networks compared with the previously proposed DiffServ video marking algorithm (DVMA).

By equipping vehicles with the on-board wireless facility, the newly emerged vehicular networking targets to provision the broadband serves to vehicles. As such, a variety of novel and exciting applications can be provided to vehicular users to enhance their road safety and travel comfort, and finally raise a complete change to their on-road life. As the content distribution and media/video streaming, such as Youtube, Netflix, nowadays have become the most popular Internet applications, to enable the efficient content distribution and audio/video streaming services is thus of the paramount importance to the success of the vehicular networking. This, however, is fraught with fundamental challenges due to the distinguished natures of vehicular networking. On one hand, the vehicular communication is challenged by the spotty and volatile wireless connections caused by the high mobility of vehicles. This makes the download performance of connections very unstable and dramatically change over time, which directly threats to the on-top media applications. On the other hand, a vehicular network typically involves an extremely large-scale node population (e.g., hundreds or thousandths of vehicles in a region) with intense spatial and temporal variations across the network geometry at different times. This dictates any designs to be scalable and fully distributed which should not only be resilient to the network dynamics, but also provide the guaranteed quality-of-service (QoS) to users. The purpose of this dissertation is to address the challenges of the vehicular networking imposed by its intrinsic dynamic and large-scale natures, and build the efficient, scalable and, more importantly, practical systems to enable the cost-effective and QoS guaranteed content distribution and media streaming services to vehicular users. Note that to effective- ly deliver the content from the remote Internet to in-motion vehicles, it typically involves three parts as: 1.) an infrastructure grid of gateways which behave as the data depots or injection points of Internet contents and services to vehicles, 2.) protocol at gateways which schedules the bandwidth resource at gateways and coordinates the parallel transmissions to different vehicles, and 3.) the end-system control mechanism at receivers which adapts the receiver’s content download/playback strategy based on the available network throughput to provide users with the desired service experience. With above three parts in mind, the entire research work in this dissertation casts a systematic view to address each part in one topic with: 1.) design of large-scale cost-effective content distribution infrastructure, 2.) MAC (media access control) performance evaluation and channel time scheduling, and 3.) receiver adaptation and adaptive playout in dynamic download environment. In specific, in the first topic, we propose a practical solution to form a large-scale and cost-effective content distribution infrastructure in the city. We argue that a large-scale infrastructure with the dedicated resources, including storage, computing and communication capacity, is necessary for the vehicular network to become an alternative of 3G/4G cellular network as the dominating approach of ubiquitous content distribution and data services to vehicles. On addressing this issue, we propose a fully distributed scheme to form a large-scale infrastructure by the contributions of individual entities in the city, such as grocery stores, movie theaters, etc. That is to say, the installation and maintenance costs are shared by many individuals. In this topic, we explain the design rationale on how to motivate individuals to contribute, and specify the detailed design of the system, which is embodied with distributed protocols and performance evaluation. The second topic investigates on the MAC throughput performance of the vehicle-to- infrastructure (V2I) communications when vehicles drive through RSUs, namely drive-thru Internet. Note that with a large-scale population of fast-motion nodes contending the chan- nel for transmissions, the MAC performance determines the achievable nodal throughput and is crucial to the on-top applications. In this topic, using a simple yet accurate Marko- vian model, we first show the impacts of mobility (characterized by node velocity and moving directions) on the nodal and system throughput performance, respectively. Based on this analysis, we then propose three enhancement schemes to timely adjust the MAC parameters in tune with the vehicle mobility to achieve the maximal the system throughput. The last topic investigates on the end-system design to deliver the user desired media streaming services in the vehicular environment. In specific, the vehicular communications are notoriously known for the intermittent connectivity and dramatically varying throughput. Video streaming on top of vehicular networks therefore inevitably suffers from the severe network dynamics, resulting in the frequent jerkiness or even freezing video playback. To address this issue, an analytical model is first developed to unveil the impacts of network dynamics on the resultant video performance to users in terms of video start-up delay and smoothness of playback. Based on the analysis, the adaptive playout buffer mechanism is developed to adapt the video playback strategy at receivers towards the user-defined video quality. The proposals developed in the three topics are validated with the extensive and high fidelity simulations. We believe that our analysis developed in the dissertation can provide insightful lights on understanding the fundamental performance of the vehicular content distribution networks from the aspects of session-level download performance in urban vehicular networks (topic 1), MAC throughput performance (topic 2), and user perceived media quality (topic 3). The protocols developed in the three topics, respectively, offer practical and efficient solutions to build and optimize the vehicular content distribution networks.

No description available.

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.

With the emergence of computer networks as one of the primary modes of communication, and with their adoption for an increasingly wide range of applications, there is a growing need to understand and characterize the traffic they carry. The rise of large scale network attacks adds urgency to this need. However, the large size, high speed and increasing complexity of these networks imply that tracking and characterizing the traffic they carry is an increasingly difficult problem. Dealing with higher level aggregates, such as flows instead of packets, does not solve the problem because these aggregates tend to be quite numerous and exhibit dynamics of their own. In this thesis, we investigate a novel approach to deal with the immense amounts of data associated with problems in network measurement and monitoring. Building upon the paradigm of Data Streaming, which processes a large stream of data using a small working memory to answer a class of queries, we develop an architecture for Network Data Streaming that can accommodate additional constraints imposed in the context of network monitoring. Using this architecture, we design algorithms for monitoring properties of network traffic that have traditionally been considered too difficult to monitor at high speed network links and routers. Our first algorithm provides the ability to accurately estimate the size of individual flows. A second algorithm to estimate the distribution of flow sizes enables network operators to monitor anomalies in the traffic. Incorporating the use of packet sampling, we can extend the latter algorithm to estimate the flow size distribution of arbitrary subpopulations. Finally, we apply the tools of Network Data Streaming to the operation of packet sampling itself. Using the ability to efficiently estimate flow-statistics such as approximate per-flow size, we design a family of mechanisms where the sampling decision is guided by this knowledge. The individual solutions developed in this thesis share a common architectural theme, supporting the monitoring of highly dynamic populations. Integrating this with the traditional sampling based framework for network monitoring will enable a broad range of applications for accurate and comprehensive monitoring of network traffic.

As multimedia processing and networking technologies, products and services evolve, the number of users communicating, collaborating and entertaining over the IP networks is growing rapidly. With the emergence of pervasive and ubiquitous multimedia services, this proliferation creates an abundant increase in the amount of the Internet backbone traffic. This brings the problem of efficient transmission of real-time and time-sensitive media content to the fore. Effective multimedia services demand appropriate application-specific and media-aware solutions, without which the full benefits of such services will not be realized. Poor approaches often lead to system performance degradations such as unacceptable presentation quality perceived by the users, possible network collapses due to the high-bandwidth nature of the multimedia applications, and poor performance observed by other data-oriented applications due to the unresponsiveness of multimedia flows. From a networking perspective, traditional approaches consider the application data as "sacred" and do not differentiate any part of it from the rest. While this keeps the data-delivery mechanisms, namely, the transport-layer protocols, as plain as possible, it also precludes these mechanisms from interpreting the media content and tailoring their actions according to the importance of the content. Given that this naive approach cannot satisfy the specific needs of each and every one of the today's emerging applications ranging from videotelephony to video-on-demand, from distance education to telemedicine, from remote surveillance to online video gaming, the study of Multimedia Transport Protocols (MMTP) is overdue. An MMTP solution basically integrates the multimedia content information into the responsible data-delivery mechanisms along with the requirements of the invoking application and network characteristics to deliver the highest level of service quality. In other words, an MMTP solution offers a unified environment where all cooperating protocol components interact with each other and make the best use of this collaboration to fulfill their respective duties. The focus of this thesis is on the design and evaluation of a set of end-to-end and system-level MMTP solutions for scalable, reliable, and high quality multimedia services in ever-changing, complex and heterogeneous computing and communication environments.

For more than a decade, streaming media services, including on-line conferences, distance education and movie broadcasting, have gained much popularity on the Internet. Due to the high bandwidth requirements and long lived nature of video streaming, it requires huge transmission cost to support these streaming media services. In addition, how to adapt rich multimedia content to satisfy various resource-constrained devices presents a challenge. The limited and time-varying network bandwidth complicates the content adaptation tasks. Differentiated content delivery may be required to meet diverse client profiles and user preferences. Therefore, in order to reduce transmission cost to serve heterogeneous clients for efficient streaming, in this dissertation, several novel schemes including transcoding-enable proxy caching scheme, reactive transmission schemes, and network coding P2P content distribution scheme, are proposed to support efficient multiple-version and layered video delivery in the proxy-attached network environment as well as to provide efficient interactive IPTV service in a peer-to-peer network. Firstly, for multiple-version cache consideration in the transcoding-enable proxy, we focus on reducing the required server bandwidth and startup delay by caching the optimal versions of the video. A generalized video object profit function is derived from the extended weighted transcoding graph to calculate the individual cache profit of certain version of a video object, and the aggregate profit from caching multiple versions of the same video object. This proposed function takes into account the popularity of certain version of a video object, the transcoding delay among versions and the average access duration of each version. Based on the profit function, cache replacement algorithms are proposed to reduce the startup delay and network traffic by efficiently caching video objects with maximum profits. Next, a set of proxy-assisted transmission schemes are proposed to reduce the transmission cost for layered video streaming by integrating the proxy caching with reactive transmission schemes, peer-to-peer mesh networks and multicast capability. These proposed transmission schemes make multiple requests to be serviced by the single transmission and thus to significantly reduce the total required transmission cost. The optimal proxy prefix cache allocation is also calculated for each transmission scheme to identify the cache layers and cache length of each video to minimize the aggregate transmission cost. The process considers the fact that reduction in transmission cost by caching X layers of a video is not only from requests on X layers, but also from requests on less than X layers. Finally, we proposed a network coding equivalent content distribution (NCECD) scheme to decrease server stress, startup delay and jumping latency to support random access operations which are desirable for peer-to-peer on-demand video streaming. The random access operations are difficult to be efficiently supported, due to the asynchronous interactive behaviors of users and the dynamic nature of peers. In NCECD, videos are divided into segments which are then further divided into blocks. These blocks are then encoded into independent encoded blocks that are distributed to the local storage of different peers. With NCECD, a new client only needs to connect to a sufficient number of parent peers in order to view the whole video and rarely needs to find new parents when performing random access operations. Whereas most existing methods must search for parent peers containing interested segments, NCECD uses the properties of network coding to cache equivalent content on most peers, so that searches are rarely needed. The analysis of system parameters is given to achieve reasonable block loss rates for peer-to-peer interactive video-on-demand streaming. Experimental results demonstrate that these proposed schemes can lead to significant transmission cost saving, high delay saving ratio, high bandwidth saving ratio, low startup and jumping searching delays, connecting to a new parent peer delay and less server resources. Hence, these proposed schemes can further be integrated and utilized to build an efficient video streaming platform for providing high-performance and high-quality IPTV services to a diversity of clients.

In this paper, we propose two schemes for layer allocations to adjust the number of layers of SVC (Scalable Video Coding) segments according to the bandwidth variation in P2P video streaming networks. The first scheme is Periodical Layer Allocation (PLA) that can adjust the number of layers to fully satisfy the available bandwidth measured periodically. However, when the available bandwidth is changed abruptly, two major drawbacks may be occurred by PLA algorithm; first, the quality of video frames may become unsmooth so that users would feel uncomfortable about the picture quality, and second, the cost is increased due to the periodical measurement of the available bandwidth. Therefore, we propose Dynamic Layer Allocation (DLA), to dynamically change the time interval for adjusting SVC layers. When freeze-up occurred or when there was not enough buffer space to store the video segments during the interval, the interval would be reduced. When the interval for adjusting SVC layers was expired, available bandwidth can be determined by the number of video segments waiting in the buffer. Compared with PLA, DLA adjusts the SVC layers gracefully so that the quality of picture becomes smoother and users feel more comfortable while watching the film. We built a simulator written in C++ under two scenarios: the available bandwidth is changed abruptly and the one changed gradually. Simulation results show that the performance of PLA is quite similar to DLA when the available bandwidth is changed gradually. However, when the available bandwidth is changed abruptly, DLA can not only obtain the smoother video film but also decrease the freeze-up time significantly.