39 research outputs found
Scalable on-demand streaming of stored complex multimedia
Previous research has developed a number of efficient protocols for streaming popular multimedia files on-demand to potentially large
numbers of concurrent clients. These protocols can achieve server bandwidth usage that grows much slower than linearly with the file request rate, and with the inverse of client start-up delay.
This hesis makes the following three main contributions to the design and performance evaluation of such protocols.
The first contribution is an investigation of the network bandwidth requirements for scalable on-demand streaming. The results suggest that the minimum required network bandwidth for scalable on-demand streaming typically scales as K/ln(K) as the number of client sites K increases for fixed request rate per client site, and as ln(N/(ND+1)) as the total file request rate N increases or client start-up delay D decreases, for a fixed number of sites. Multicast delivery trees configured to minimize network bandwidth usage rather than latency are found to only modestly reduce the minimum required network bandwidth. Furthermore, it is possible to achieve close to the minimum possible network and server bandwidth usage simultaneously with practical scalable delivery protocols.
Second, the thesis addresses the problem of scalable on-demand streaming of a more complex type of media than is typically considered, namely variable bit rate (VBR) media. A lower bound on
the minimum required server bandwidth for scalable on-demand streaming
of VBR media is derived. The lower bound analysis motivates the design of a new immediate service protocol termed VBR bandwidth skimming (VBRBS) that uses constant bit rate streaming, when sufficient client storage space is available, yet fruitfully exploits the knowledge of a VBR profile.
Finally, the thesis proposes non-linear media containing parallel sequences of data frames, among which clients can dynamically select at designated branch points, and investigates the design and performance issues in scalable on-demand streaming of such media. Lower bounds on the minimum required server bandwidth for various non-linear media scalable on-demand streaming approaches are derived, practical non-linear media scalable delivery protocols are developed, and, as a proof-of-concept, a simple scalable delivery
protocol is implemented in a non-linear media streaming prototype system
ATOM : a distributed system for video retrieval via ATM networks
The convergence of high speed networks, powerful personal computer processors and improved storage technology has led to the development of video-on-demand services to the desktop that provide interactive controls and deliver Client-selected video information on a Client-specified schedule. This dissertation presents the design of a video-on-demand system for Asynchronous Transfer Mode (ATM) networks, incorporating an optimised topology for the nodes in the system and an architecture for Quality of Service (QoS). The system is called ATOM which stands for Asynchronous Transfer Mode Objects. Real-time video playback over a network consumes large bandwidth and requires strict bounds on delay and error in order to satisfy the visual and auditory needs of the user. Streamed video is a fundamentally different type of traffic to conventional IP (Internet Protocol) data since files are viewed in real-time, not downloaded and then viewed. This streaming data must arrive at the Client decoder when needed or it loses its interactive value. Characteristics of multimedia data are investigated including the use of compression to reduce the excessive bit rates and storage requirements of digital video. The suitability of MPEG-1 for video-on-demand is presented. Having considered the bandwidth, delay and error requirements of real-time video, the next step in designing the system is to evaluate current models of video-on-demand. The distributed nature of four such models is considered, focusing on how Clients discover Servers and locate videos. This evaluation eliminates a centralized approach in which Servers have no logical or physical connection to any other Servers in the network and also introduces the concept of a selection strategy to find alternative Servers when Servers are fully loaded. During this investigation, it becomes clear that another entity (called a Broker) could provide a central repository for Server information. Clients have logical access to all videos on every Server simply by connecting to a Broker. The ATOM Model for distributed video-on-demand is then presented by way of a diagram of the topology showing the interconnection of Servers, Brokers and Clients; a description of each node in the system; a list of the connectivity rules; a description of the protocol; a description of the Server selection strategy and the protocol if a Broker fails. A sample network is provided with an example of video selection and design issues are raised and solved including how nodes discover each other, a justification for using a mesh topology for the Broker connections, how Connection Admission Control (CAC) is achieved, how customer billing is achieved and how information security is maintained. A calculation of the number of Servers and Brokers required to service a particular number of Clients is presented. The advantages of ATOM are described. The underlying distributed connectivity is abstracted away from the Client. Redundant Server/Broker connections are eliminated and the total number of connections in the system are minimized by the rule stating that Clients and Servers may only connect to one Broker at a time. This reduces the total number of Switched Virtual Circuits (SVCs) which are a performance hindrance in ATM. ATOM can be easily scaled by adding more Servers which increases the total system capacity in terms of storage and bandwidth. In order to transport video satisfactorily, a guaranteed end-to-end Quality of Service architecture must be in place. The design methodology for such an architecture is investigated starting with a review of current QoS architectures in the literature which highlights important definitions including a flow, a service contract and flow management. A flow is a single media source which traverses resource modules between Server and Client. The concept of a flow is important because it enables the identification of the areas requiring consideration when designing a QoS architecture. It is shown that ATOM adheres to the principles motivating the design of a QoS architecture, namely the Integration, Separation and Transparency principles. The issue of mapping human requirements to network QoS parameters is investigated and the action of a QoS framework is introduced, including several possible causes of QoS degradation. The design of the ATOM Quality of Service Architecture (AQOSA) is then presented. AQOSA consists of 11 modules which interact to provide end-to-end QoS guarantees for each stream. Several important results arise from the design. It is shown that intelligent choice of stored videos in respect of peak bandwidth can improve overall system capacity. The concept of disk striping over a disk array is introduced and a Data Placement Strategy is designed which eliminates disk hot spots (i.e. Overuse of some disks whilst others lie idle.) A novel parameter (the B-P Ratio) is presented which can be used by the Server to predict future bursts from each video stream. The use of Traffic Shaping to decrease the load on the network from each stream is presented. Having investigated four algorithms for rewind and fast-forward in the literature, a rewind and fast-forward algorithm is presented. The method produces a significant decrease in bandwidth, and the resultant stream is very constant, reducing the chance that the stream will add to network congestion. The C++ classes of the Server, Broker and Client are described emphasizing the interaction between classes. The use of ATOM in the Virtual Private Network and the multimedia teaching laboratory is considered. Conclusions and recommendations for future work are presented. It is concluded that digital video applications require high bandwidth, low error, low delay networks; a video-on-demand system to support large Client volumes must be distributed, not centralized; control and operation (transport) must be separated; the number of ATM Switched Virtual Circuits (SVCs) must be minimized; the increased connections caused by the Broker mesh is justified by the distributed information gain; a Quality of Service solution must address end-to-end issues. It is recommended that a web front-end for Brokers be developed; the system be tested in a wide area A TM network; the Broker protocol be tested by forcing failure of a Broker and that a proprietary file format for disk striping be implemented
Theories and Models for Internet Quality of Service
We survey recent advances in theories and models for Internet Quality of Service (QoS). We start with the theory of network calculus, which lays the foundation for support of deterministic performance guarantees in networks, and illustrate its applications to integrated services, differentiated services, and streaming media playback delays. We also present mechanisms and architecture for scalable support of guaranteed services in the Internet, based on the concept of a stateless core. Methods for scalable control operations are also briefly discussed. We then turn our attention to statistical performance guarantees, and describe several new probabilistic results that can be used for a statistical dimensioning of differentiated services. Lastly, we review recent proposals and results in supporting performance guarantees in a best effort context. These include models for elastic throughput guarantees based on TCP performance modeling, techniques for some quality of service differentiation without access control, and methods that allow an application to control the performance it receives, in the absence of network support
Joint Playback Delay and Buffer Optimization in Scalable Video Streaming
This paper addresses the problem of the transmission of scalable video streams to a set of heterogeneous clients through a common bottleneck channel. The packet scheduling policy is typically crucial in such systems that target smooth media playback at all the receivers. In particular, the playback delays and the transmission strategy for the packets of the different layers have to be chosen carefully. When the same video is sent simultaneously to multiple clients that subscribe to different parts of the stream, the playback delay cannot be jointly minimized for all the clients. We therefore propose delay optimization strategies along with low complexity solutions for a fair distribution of the delay penalty among the different receivers. Once the delays are selected, we show that there exists a unique scheduling solution that minimizes the buffer occupancy at all the receivers. We derive an algorithm for computing the optimal sending trace, and we show that optimal scheduling has to respect the order of the packets in each media layer. Interestingly enough, solving both delay and buffer optimization problems sequentially leads to a jointly optimal solution when the channel is known. We finally propose a simple rate adaptation mechanism that copes with unexpected channel bandwidth variations by controlling the sending rate and dropping layers when the bandwidth becomes insufficient. Experimental results shows that it permits to reach close to optimal performances even if the channel knowledge is reduced. Rate adaptation provides an interesting alternative to conservative scheduling strategies, providing minor and controllable quality variations, but with a higher resulting average quality
Advances in Internet Quality of Service
We describe recent advances in theories and architecture that support performance guarantees needed for quality of service networks. We start with deterministic computations and give applications to integrated services, differentiated services, and playback delays. We review the methods used for obtaining a scalable integrated services support, based on the concept of a stateless core. New probabilistic results that can be used for a statistical dimensioning of differentiated services are explained; some are based on classical queuing theory, while others capitalize on the deterministic results. Then we discuss performance guarantees in a best effort context; we review: methods to provide some quality of service in a pure best effort environment; methods to provide some quality of service differentiation without access control, and methods that allow an application to control the performance it receives, in the absence of network support
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Multimedia streaming using multiple TCP connections
Packet loss, delay and time-varying bandwidth are three main problems facing multimedia streaming applications over the Internet. Existing techniques such as Media-aware network protocol, network adaptive source and channel coding, etc. have been proposed to either overcome or alleviate these drawbacks of the Internet. But these techniques either need specialized codecs or
require significant changes in the network infrastructure. In this thesis, we propose the MultiTCP system, a receiver-driven, TCP-based application-layer transmission protocol for multimedia streaming over the Internet. The proposed algorithm aims at providing resilience against SHORT TERM insufficient bandwidth by using MULTIPLE TCP connections for the same application. Our proposed system enables the application to achieve and control the desired sending
rate during congested periods, by using multiple TCP connections and dynamically changing the receiver's window size for each connection, which cannot be
achieved using traditional TCP. Finally, the proposed system is implemented at
the application layer. Thus no kernel modification is necessary, which ensures
easy deployment. To demonstrate the performance of the proposed system,
we present simulation and experimental results on the PlanetLab network to
establish its advantages over the traditional single TCP based approach
Virtualization of multicast services in WiMAX networks
Multicast service is one of the methods used to efficiently manage bandwidth when sending multimedia content. To improve bandwidth utilisation, virtualization is often invoked because of its additional features such as bandwidth sharing and support of services that require high volumes of transactional data. Currently, network providers are concerned with the bandwidth amount for efficient use of the limited wireless network capabilities and the provision of a better quality of service. The virtualization design of a multicast service framework should satisfy several objectives. For example, it should enable the interchange of service delivery between multiple networks with one shareable network infrastructure. Also, it should ensure efficient use of network resources and guarantee users' demands of Quality of Service (QoS). Thus, the design of virtualization of multicast service framework is a complex research study. Due to the bandwidth-related arguments, a strong focus has been put on technical issues that facilitate virtualization in wireless networks. A well-designed virtualized network guarantees users with the required quality service. Similarly, virtualization of multicast service is invoked to improve efficient utilisation of bandwidth in wireless networks. As wireless links prove to be unstable, packet loss is unavoidable when multicast service-oriented virtual artefacts are incorporated in wireless networks. In this thesis, a virtualized multicast framework was modelled by using Generalized Assignment Problem (GAP) methodology. Mixed Integer Linear Programing (MILP) was implemented in MATLAB to solve the GAP model. This was to optimise the allocation of multicast traffic to the appropriate virtual networks. Thus, the developed model allows users to have interchangeable services offered by multiple networks. Furthermore, Network Simulator version 3 (NS-3) was used to evaluate the performance of the virtualized multicast framework. Three applications, namely, voice over IP (VoIP), video streaming, and file download have been used to evaluate the performance of a multicast service virtualization framework in Worldwide Interoperability for Microwave Access (WiMAX) networks using NS-3. The performance evaluation was based on whether MILP is used or not used. The results of experimentation have revealed that there is good performance of virtual networks when multicast traffic is sent over one single virtual network instead of sending it over multiple virtual networks. Similarly, the results show that the bandwidth is efficiently used because the multicast traffic is not delivered through multiple virtual networks. Overall, the concepts, the investigations and the model presented in this thesis can enable mobile network providers to achieve efficient use of bandwidth and provide the necessary means to support services for QoS differentiations and guarantees. Also, the multicast service virtualization framework provides an excellent tool that can enable network providers to interchange services. The developed model can serve as a basis for further extension. Specifically, the extension of the model can boost load balancing in the flow allocation problem and activate a virtual network to deliver traffic. This may rely on the QoS policy between network providers. Therefore, the model should consider the number of users in order to guarantee improved QoS
Layer-based coding, smoothing, and scheduling of low-bit-rate video for teleconferencing over tactical ATM networks
This work investigates issues related to distribution of low bit rate video within the context of a teleconferencing application deployed over a tactical ATM network. The main objective is to develop mechanisms that support transmission of low bit rate video streams as a series of scalable layers that progressively improve quality. The hierarchical nature of the layered video stream is actively exploited along the transmission path from the sender to the recipients to facilitate transmission. A new layered coder design tailored to video teleconferencing in the tactical environment is proposed. Macroblocks selected due to scene motion are layered via subband decomposition using the fast Haar transform. A generalized layering scheme groups the subbands to form an arbitrary number of layers. As a layering scheme suitable for low motion video is unsuitable for static slides, the coder adapts the layering scheme to the video content. A suboptimal rate control mechanism that reduces the kappa dimensional rate distortion problem resulting from the use of multiple quantizers tailored to each layer to a 1 dimensional problem by creating a single rate distortion curve for the coder in terms of a suboptimal set of kappa dimensional quantizer vectors is investigated. Rate control is thus simplified into a table lookup of a codebook containing the suboptimal quantizer vectors. The rate controller is ideal for real time video and limits fluctuations in the bit stream with no corresponding visible fluctuations in perceptual quality. A traffic smoother prior to network entry is developed to increase queuing and scheduler efficiency. Three levels of smoothing are studied: frame, layer, and cell interarrival. Frame level smoothing occurs via rate control at the application. Interleaving and cell interarrival smoothing are accomplished using a leaky bucket mechanism inserted prior to the adaptation layer or within the adaptation layerhttp://www.archive.org/details/layerbasedcoding00parkLieutenant Commander, United States NavyApproved for public release; distribution is unlimited