Video traffic modeling and delivery

Video is becoming a major component of the network traffic, and thus there has been a great interest to model video traffic. It is known that video traffic possesses short range dependence (SRD) and long range dependence (LRD) properties, which can drastically affect network performance. By decomposing a video sequence into three parts, according to its motion activity, Markov-modulated self-similar process model is first proposed to capture autocorrelation function (ACF) characteristics of MPEG video traffic. Furthermore, generalized Beta distribution is proposed to model the probability density functions (PDFs) of MPEG video traffic. It is observed that the ACF of MPEG video traffic fluctuates around three envelopes, reflecting the fact that different coding methods reduce the data dependency by different amount. This observation has led to a more accurate model, structurally modulated self-similar process model, which captures the ACF of the traffic, both SRD and LRD, by exploiting the MPEG structure. This model is subsequently simplified by simply modulating three self-similar processes, resulting in a much simpler model having the same accuracy as the structurally modulated self-similar process model. To justify the validity of the proposed models for video transmission, the cell loss ratios (CLRs) of a server with a limited buffer size driven by the empirical trace are compared to those driven by the proposed models. The differences are within one order, which are hardly achievable by other models, even for the case of JPEG video traffic. In the second part of this dissertation, two dynamic bandwidth allocation algorithms are proposed for pre-recorded and real-time video delivery, respectively. One is based on scene change identification, and the other is based on frame differences. The proposed algorithms can increase the bandwidth utilization by a factor of two to five, as compared to the constant bit rate (CBR) service using peak rate assignment

A study of the transmission of VBR encoded video over ATM networks.

by Ngai Li.Thesis (M.Phil.)--Chinese University of Hong Kong, 1997.Includes bibliographical references (leaves 66-69).Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Video Compression and Transport --- p.2Chapter 1.2 --- Research Contributions --- p.6Chapter 1.2.1 --- Joint Rate Control of VBR Encoded Video --- p.6Chapter 1.2.2 --- Transporting VBR Video on LB Controlled Channel --- p.7Chapter 1.3 --- Organization of Thesis --- p.7Chapter 2 --- Preliminary --- p.9Chapter 2.1 --- Statistical Characteristics of MPEG-1 Encoded Video --- p.9Chapter 2.2 --- Temporal and Spatial Smoothing --- p.14Chapter 2.2.1 --- Temporal Smoothing --- p.14Chapter 2.2.2 --- Spatial Smoothing --- p.15Chapter 2.3 --- A Single Source Control-Theoretic Framework for VBR-to-CBR Video Adaptation --- p.16Chapter 3 --- Joint Rate Control of VBR Encoded Video --- p.19Chapter 3.1 --- Analytical Models --- p.21Chapter 3.2 --- Analysis --- p.27Chapter 3.2.1 --- Stable Region --- p.29Chapter 3.2.2 --- Final Value of the State Variables --- p.33Chapter 3.2.3 --- Peak Values of Buffer-occupancy Deviation and Image- quality Fluctuation --- p.35Chapter 3.2.4 --- SAE of Buffer-occupancy Deviation and Image-quality Fluc- tuation --- p.42Chapter 3.3 --- Experimental Results --- p.43Chapter 3.4 --- Concluding Remarks --- p.48Chapter 4 --- Transporting VBR Video on LB Controlled Channel --- p.50Chapter 4.1 --- Leaky Bucket Access Control --- p.51Chapter 4.2 --- Greedy Token-usage Strategy --- p.53Chapter 4.3 --- Non-greedy Token-usage Strategy --- p.57Chapter 4.4 --- Concluding Remarks --- p.60Chapter 5 --- Conclusions --- p.62Chapter 5.1 --- Joint Rate Control of Multiple VBR Videos --- p.62Chapter 5.2 --- LB Video Compression --- p.63Chapter 5.3 --- Further Study --- p.64Chapter 5.4 --- Publications --- p.65Bibliography --- p.6

The Design of a single chip 8x8 ATM switch in 0.5 micrometers CMOS VLSI

This thesis illustrates the design of a single chip Asynchronous Transfer Mode (ATM) protocol switch using Very Large Scale Integration (VLSI). The ATM protocol is the data communications protocol used in the implementation of the Broadband Integrated Services Digital Network (B-ISDN), A number of switch architecture are first studied and a new architecture is developed based on optimizing performance and practicality of implementation in VLSI. A fully interconnected switch architecture is implemented by permanently connecting every input port to all the output ports. An output buffering scheme is used to handle cells that cannot be routed right away. This new architecture is caned the High Performance (HiPer) Switch Architecture. The performance of the architecture is simulated using a C++ model. Simulation results for a randomly distributed traffic pattern with a 90% probability of cells arriving in a time slot produces a Cell Loss Ratio of 1.Ox 10^-8 with output buffers that can hold 64 cells. The device is then modeled in VHDL to verify its functionality. Finally the layout of an 8x8 switch is produced using a 0.5 micrometer CMOS VLSI process and simulations of that circuit show that a peak throughput of 200 Mbps per output port can be achieve

Optimal Cell Loss Equalization for Video Multiplexers

Video traffic multiplexers in high speed ATM networks are prone to fairness problems with respect to the per-connection cell loss ratio experienced by multiplexed video sources. The problem is a result of the random, but fixed over time, relation of the frame transmission epochs that feed a multiplexer. This paper presents a solution to this fairness problem which is based on the enforcement of controlled per--connection delays. The amount of delay imposed on each source is calculated by an optimization process at connection admission and termination instants. Two different optimization objectives, one minimax and one minisum, are considered. Their performance and their relation to buffer space constraints is examined. The loss and delay performance of the scheme is also evaluated through simulations. In particular, very low per--connection delay variance is observed, indicating reduced jitter. Finally, two implementation alternatives of the scheme on an ATM network are presented: (a) as a protocol between multiplexer and sources and (b) as a non--work conserving service discipline for multiplexers. The engineering aspects and, in particular, the buffer demands of the two alternative implementations are discussed in detail

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

Variable bit rate video time-series and scene modeling using discrete-time statistically self-similar systems

This thesis investigates the application of discrete-time statistically self-similar (DTSS) systems to modeling of variable bit rate (VBR) video traffic data. The work is motivated by the fact that while VBR video has been characterized as self-similar by various researchers, models based on self-similarity considerations have not been previously studied. Given the relationship between self-similarity and long-range dependence the potential for using DTSS model in applications involving modeling of VBR MPEG video traffic data is presented. This thesis initially explores the characteristic properties of the model and then establishes relationships between the discrete-time self-similar model and fractional order transfer function systems. Using white noise as the input, the modeling approach is presented using least-square fitting technique of the output autocorrelations to the correlations of various VBR video trace sequences. This measure is used to compare the model performance with the performance of other existing models such as Markovian, long-range dependent and M/G/(infinity) . The study shows that using heavy-tailed inputs the output of these models can be used to match both the scene time-series correlations as well as scene density functions. Furthermore, the discrete-time self-similar model is applied to scene classification in VBR MPEG video to provide a demonstration of potential application of discrete-time self-similar models in modeling self-similar and long-range dependent data. Simulation results have shown that the proposed modeling technique is indeed a better approach than several earlier approaches and finds application is areas such as automatic scene classification, estimation of motion intensity and metadata generation for MPEG-7 applications

On the Queue Length Distribution in BMAP Systems

Batch Markovian Arrival Process – BMAP – is a teletraffic model which combines high ability to imitate complex statistical behaviour of network traces with relative simplicity in analysis and simulation. It is also a generalization of a wide class of Markovian processes, a class which in particular include the Poisson process, the compound Poisson process, the Markovmodulated Poisson process, the phase-type renewal process and others. In this paper we study the main queueing performance characteristic of a finite-buffer queue fed by the BMAP, namely the queue length distribution. In particular, we show a formula for the Laplace transform of the queue length distribution. The main benefit of this formula is that it may be used to obtain both transient and stationary characteristics. To demonstrate this, several numerical results are presented

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today