Dynamic bandwidth allocation in ATM networks

Includes bibliographical references.This thesis investigates bandwidth allocation methodologies to transport new emerging bursty traffic types in ATM networks. However, existing ATM traffic management solutions are not readily able to handle the inevitable problem of congestion as result of the bursty traffic from the new emerging services. This research basically addresses bandwidth allocation issues for bursty traffic by proposing and exploring the concept of dynamic bandwidth allocation and comparing it to the traditional static bandwidth allocation schemes

Buffer control algorithm for low bit-rate video compression

In this paper, a new buffer control algorithm for motion-compensated hybrid DPCM/DCT coding (like H.261 and MPEG-1 I pictures) is presented. The algorithm uses the bit allocation algorithm to determine the quantization scale factor of each macroblock under a given target bit rate. An important advantage of the algorithm is that it has precise control of the buffer and avoids buffer overflow events which is a severe problem in low bit rate video coder. Furthermore, the coder is able to allocate bits to the picture as a whole, resulting in better rate-distortion trade-off. Simulation results show that the H.261 coder, using the proposed algorithm, can achieve a higher PSNR and better visual quality than codec using conventional buffer control algorithm.published_or_final_versio

A modified H.263 algorithm using bit allocation buffer control algorithm

Buffer control is an important problem in very low bitrate video coding. In a recent work [ 111, the authors had proposed a new buffer control algorithm for motion-compensated hybrid DPCMiDCT coding. The algorithm is based on the use of bit allocation algorithm to determine the quantization scale factors in such coder to meet a given target bit rate. Simulation results showed that, using the proposed algorithm, the H.261 coder can achieve a higher PSNR and better visual quality than the coder using traditional buffer control algorithm. In this paper, we apply this buffer control algorithm to a modified version of the H.263 algorithm for very low bit-rate video coding. Comparing the performance of the modified H.263 codec with the TMN5 model also shows that better visual quality can be obtained at comparable PSNR values.published_or_final_versio

Improving the Performance of Wide Area Networks

Research in to the performance of wide area data networks is described in this thesis. A model of wide area network packet delays is developed and used to direct the research in to methods of improving performance. Wide area networks are slow and expensive compared to the computer systems that rely on them for communication. Typically data networks are packet switched in order to make efficient use of resources. This can lead to contention, and the mechanisms for resolving contention can bring about further delays when demand for resources is high. In this thesis, network users are viewed as interacting decision makers with conflicting interests, and Game Theory is used to analyse the effects users have on each other’s performance. It is asserted in this thesis that wide area network performance is an ethical issue as well as a technical one. Compression is examined as a technique for reducing network traffic load. While load reductions can reduce the time packets spend waiting in buffer queues experimental results show the compression process itself can present a bottleneck if CPU resources are limited. The other inhibiting factor with regard to wide area network performance is the time it takes for a signal to propagate through a transmission medium. Propagation delays are bounded by the speed of light and becomes significant as the distance between computer systems increases. Mirrors and Caches are methods of bringing data closer to the user, thereby reducing propagation delays and capping traffic loads on long haul communication facilities. The performance benefits of replicating data within a wide area network environment are studied in this thesis

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

An admission control algorithm for providing quality-of-service guarantee for individual connection in a video-on-demand system.

by Xiaoqing Wang.Thesis (M.Phil.)--Chinese University of Hong Kong, 2000.Includes bibliographical references (leaves 43-45).Abstracts in English and Chinese.Acknowledgments --- p.iiAbstract --- p.iiiChapter 1 --- Introduction --- p.1Chapter 2 --- The General Architecture of the VoD System and the Related Issues --- p.4Chapter 2.1 --- A Brief Description of VoD System --- p.4Chapter 2.2 --- Why Video Streams in VoD Service are VBR in Nature? --- p.6Chapter 2.3 --- The Video Storage Media in the VoD Systems --- p.8Chapter 2.4 --- The Data Placement Scheme in the VoD System --- p.9Chapter 2.5 --- An Overview of Disk Scheduling in VoD System --- p.10Chapter 2.6 --- The Admission Control in VoD System --- p.12Chapter 3 --- Our Admission Control Algorithm for VoD System --- p.14Chapter 3.1 --- QoS Requirements We Choose --- p.14Chapter 3.2 --- System Model --- p.15Chapter 3.3 --- The Admission Control for the Storage Sub-system --- p.19Chapter 3.4 --- The Admission Control for Network Sub-system --- p.21Chapter 3.4.1 --- Preliminaries --- p.22Chapter 3.4.2 --- The Admission Control Algorithm for Network Sub-system --- p.27Chapter 4 --- Experiment --- p.33Chapter 5 --- Conclusion and Future Work --- p.41Chapter 5.1 --- Conclusion --- p.41Chapter 5.2 --- Future Work --- p.4

Bluetooth audio and video streaming on the J2ME platform

With the increase in bandwidth, more widespread distribution of media, and increased capability of mobile devices, multimedia streaming has not only become feasible, but more economical in terms of space occupied by the media file and the costs involved in attaining it. Although much attention has been paid to peer to peer media streaming over the Internet using HTTP and RTSP, little research has focussed on the use of the Bluetooth protocol for streaming audio and video between mobile devices. This project investigates the feasibility of Bluetooth as a protocol for audio and video streaming between mobile phones using the J2ME platform, through the analysis of Bluetooth protocols, media formats, optimum packet sizes, and the effects of distance on transfer speed. A comparison was made between RFCOMM and L2CAP to determine which protocol could support the fastest transfer speed between two mobile devices. The L2CAP protocol proved to be the most suitable, providing average transfer rates of 136.17 KBps. Using this protocol a second experiment was undertaken to determine the most suitable media format for streaming in terms of: file size, bandwidth usage, quality, and ease of implementation. Out of the eight media formats investigated, the MP3 format provided the smallest file size, smallest bandwidth usage, best quality and highest ease of implementation. Another experiment was conducted to determine the optimum packet size for transfer between devices. A tradeoff was found between packet size and the quality of the sound file, with highest transfer rates being recorded with the MTU size of 668 bytes (136.58 KBps). The class of Bluetooth transmitter typically used in mobile devices (class 2) is considered a weak signal and is adversely affected by distance. As such, the final investigation that was undertaken was aimed at determining the effects of distance on audio streaming and playback. As can be expected, when devices were situated close to each other, the transfer speeds obtained were higher than when devices were far apart. Readings were taken at varying distances (1-15 metres), with erratic transfer speeds observed from 7 metres onwards. This research showed that audio streaming on the J2ME platform is feasible, however using the currently available class of Bluetooth transmitter, video streaming is not feasible. Video files were only playable once the entire media file had been transferred

Statistical characterisation and stochastic modelling of 1-layer variable bit rate H.261 video codec traffic

The Integrated Services Digital Network(ISDN) is under re-design to provide flexibility which will ensure efficient network utilisation in the provision of broadband services. The main broadband services envisaged for provision on the Broadband ISDN(B-ISDN) are : Videophone; Videoconferencing; Television and High Definition TV. The B-ISDN will be a packet switched network where the packets(cells) will be transferred by the Asynchronous Transfer Mode(ATM) concept. Unlike voice and data services, the impact video services will have on the BISDN is unknown and hence loss of information is difficult to predict. Present videophone terminals are based on the CCITT H.261 Video Coding standard hence the picture quality is variable because video codec traffic is transmitted at a constant rate. To maintain a constant quality picture the codec output data must be transmitted at a variable rate or alternatively, for constant rate video codecs extra information must be made available to achieve constant picture quality. This latter technique is 2- Layer video coding where the first layer transmits at a constant rate and the second layer at a variable rate. The ATM B-ISDN promises constant picture quality video services, therefore to achieve this aim the impact variable rate video sources will have on the network must be determined by network simulation, thus variable rate video source models must be derived. To statistically characterise and stochastically model 1-Layer VBR(Variable Bit Rate) H.261 Video Codec traffic, here a videophone sequence is analysed by two alternative strategies : Talk-Listen and Motion Level. This analysis also found that 2-Layer H.261 Video Codec traffic can be stochastically modelled via a 1-Layer VBR H.261 Video Codec traffic model. Numerous hierarchical stochastic models with the ability to capture the statistical characteristics of long video sequences, in particular the short-term and long-term autocorrelations, are presented. One such model was simulated and the resulting simulated traffic was analysed to confirm the advantage hierarchical stochastic models have over non-hierarchical stochastic models in modelling video source traffic