Multirate video multicast over the Internet: an overview

Multirate multicast is an effective method for video distribution to a set of heterogeneous receivers. We present a comprehensive survey on multirate video multicast over the best effort Internet. We first review the key techniques in video encoding and network transport, and describe the representative approaches. We then study various trade-offs based on some important design issues and performance criteria, such as bandwidth economy, adaptation granularity, and coding complexity. Finally, we present some ongoing work and discuss possible avenues for future research.     

Multicast-based inference of network-internal loss characteristics

Robust measurements of network dynamics are increasingly important to the design and operation of large internetworks like the Internet. However, administrative diversity makes it impractical to monitor every link on an end-to-end path. At the same time, it is difficult to determine the performance characteristics of individual links from end-to-end measurements of unicast traffic. In this paper, we introduce the use of end-to-end measurements of multicast traffic to infer network-internal characteristics. The bandwidth efficiency of multicast traffic makes it suitable for large-scale measurements of both end-to-end and internal network dynamics. We develop a maximum-likelihood estimator for loss rates on internal links based on losses observed by multicast receivers. It exploits the inherent correlation between such observations to infer the performance of paths between branch points in the tree spanning a multicast source and its receivers. We derive its rate of convergence as the number of measurements increases, and we establish robustness with respect to certain generalizations of the underlying model. We validate these techniques through simulation and discuss possible extensions and applications of this work     

Scalable video coding and transport over broadband wirelessnetworks

With the emergence of broadband wireless networks and increasing demand of multimedia information on the Internet, wireless multimedia services are foreseen to become widely deployed in the next decade. Real-time video transmission typically has requirements on quality of service (QoS). However, wireless channels are unreliable and the channel bandwidth varies with time, which may cause severe degradation in video quality. In addition, for video multicast, the heterogeneity of receivers makes it difficult to achieve efficiency and flexibility. To address these issues, three techniques, namely, scalable video coding, network-aware adaptation of end systems, and adaptive QoS support from networks, have been developed. This paper unifies the three techniques and presents an adaptive framework, which specifically addresses video transport over wireless networks. The adaptive framework consists of three basic components: (1) scalable video representations; (2) network-aware end systems; and (3) adaptive services. Under this framework, as wireless channel conditions change, mobile terminals and network elements can scale the video streams and transport the scaled video streams to receivers with a smooth change of perceptual quality. The key advantages of the adaptive framework are: (1) perceptual quality is changed gracefully during periods of QoS fluctuations and hand-offs; and (2) the resources are shared in a fair manner     

Sharing the “cost” of multicast trees: an axiomaticanalysis

Given the need to provide users with reasonable feedback about the “costs” their network usage incurs and the increasingly commercial nature of the Internet, we believe that the allocation of cost among users will play an important role in future networks. This paper discusses cost allocation in the context of multicast flows. The question we discuss is this. When a single data flow is shared among many receivers, how does one split the cost of that flow among the receivers? Multicast routing increases network efficiency by using a single shared delivery tree. We address the issue of how these savings are allocated among the various members of the multicast group. We first consider an axiomatic approach to the problem, analyzing the implications of different distributive notions on the resulting allocations. We then consider a “one-pass” mechanism to implement such allocation schemes and investigate the family of allocation schemes such mechanisms can support     

Smoothing and transporting video in QoS IP networks

Real‐time traffic such as voice and video, when transported over the Internet, demand stringent quality of service (QoS) requirements. The current Internet as of today is still used as a best effort environment with no quality guarantees. An IP‐based Internet that supports different QoS requirements for different applications has been evolving for the past few years. Video streams are bursty in nature due to the instant variability of the video content being encoded. To help mitigate the transport of bursty video streams with minimal loss of information, rate‐adaptive shapers (RASs) are usually being used to reduce the burstiness and therefore help preserve the desired quality. When transporting video over a QoS IP network, each stream is classified under a specific traffic profile to which it must conform, to avoid packet loss and picture quality degradation. In this paper we study, evaluate and propose RASs for the transport of video over a QoS IP network. We utilize the encoding video parameters for choosing the appropriate configuration needed to support the real‐time transport of Variable Bit Rate (VBR) encoded video streams. The performance evaluation of the different RASs is based on the transport of MPEG‐4 video streams encoded as VBR. The performance is studied based on looking at the effect of various parameters associated with the RASs on the effectiveness of smoothing out the burstiness of video and minimizing the probability of packet loss. Copyright © 2005 John Wiley & Sons, Ltd.     

Achieving inter-session fairness for layered video multicast

The Internet is increasingly used to deliver multimedia services. Since there are heterogeneous receivers and changing network conditions, it has been proposed to use adaptive rate control techniques such as layered video multicast to adjust the video traffic according to the available Internet resources. A problem of layered video multicast is that it is unable to provide fair bandwidth sharing between competing video sessions. We propose two schemes, layered video multicast with congestion sensitivity and adaptive join-timer (LVMCA) and layered video multicast with priority dropping (LVMPD), to achieve inter-session fairness for layered video multicast. Receiver-driven layered multicast (RLM), layer-based congestion sensitivity, LVMCA, and LVMPD are simulated and compared. Results show both proposed schemes, especially LVMPD, are fairer and have shorter convergence time than the other two schemes.     

Network traffic demand patterns for video over relative differentiated services networks

Relative service differentiation mechanisms such as DiffServ Assured Forwarding are promising techniques for future delivery of video services due to their desirable scalability properties and the ability to trade-off cost and loss rate. This trade-off is important since the loss of certain parts of a video bitstream (e.g. motion information) has a catastrophic effect on the quality of the decoded sequence whereas the loss of other information (e.g. high-frequency transform coefficients) does not. We investigate the performance of these systems under realistic conditions where per-packet tariffs must remain fixed for relatively long periods of time but the network conditions may change over short timescales. We also consider how the presence of many users on a network, each with their own unique notions of utility interact in the context of a single shared network. We show that so long as the network behaves in a certain way as the overall load fluctuates the distribution of traffic among service classes remains largely unchanged. An essential consideration is not simply the mapping of video packets to service classes but also the rate of generation of video packets and thus the amount of protection that can be given to each packet. This is an important result as it demonstrates that such a network “makes sense” and will not simply regress to behave like a best-effort network.     

An extended discrete autoregressive model for variable bit rate video traffic encoded by scalable video codec

Designing an effective and high performance network requires accurate characterization and modeling of network traffic. This paper provides a study of the transmission, modeling, and analysis of variable bit rate (VBR) video, which is fundamental for designing protocols and for effective utilization of networks in video transmission. To meet the specified requirements of future networks, the scalable video codec (SVC) was chosen as the video compression standard. The main objective of this work was to propose a statistical model that will allow better coordination between generated SVC video traffic and original video traffic. The performance of the proposed model was evaluated by comparing the model with three existing low‐complex statistical models using graphical and statistical measurements as well as cross‐validation. For all evaluation techniques, the accuracy of the proposed model was evaluated, and the frame size distribution of the modeled traffic was found to match that of the original video traffic better than other existing models. Copyright © 2015 John Wiley & Sons, Ltd.     

Self-similar traffic and network dynamics

One of the most significant findings of traffic measurement studies over the last decade has been the observed self-similarity in packet network traffic. Subsequent research has focused on the origins of this self-similarity, and the network engineering significance of this phenomenon. This paper reviews what is currently known about network traffic self-similarity and its significance. We then consider a matter of current research, namely, the manner in which network dynamics (specifically, the dynamics of transmission control protocol (TCP), the predominant transport protocol used in today's Internet) can affect the observed self-similarity. To this end, we first discuss some of the pitfalls associated with applying traditional performance evaluation techniques to highly-interacting, large-scale networks such as the Internet. We then present one promising approach based on chaotic maps to capture and model the dynamics of TCP-type feedback control in such networks. Not only can appropriately chosen chaotic map models capture a range of realistic source characteristics, but by coupling these to network state equations, one can study the effects of network dynamics on the observed scaling behavior We consider several aspects of TCP feedback, and illustrate by examples that while TCP-type feedback can modify the self-similar scaling behavior of network traffic, it neither generates it nor eliminates it     

Slicing Wi-Fi links based on QoE video streaming fairness

Dynamic Adaptive Streaming over HyperText Transfer Protocol (DASH) video streaming is one of the dominant sources of traffic on the Internet, and this traffic is often delivered to users via Wi-Fi access points. This makes the quality of experience (QoE) of the service susceptible to degradation when the Wi-Fi link is underperforming, and it can also inflict QoE unfairness, creating situations where users connected to the same Wi-Fi network feel different levels of QoE. This article presents a system that improves the QoE fairness of DASH video transmission network slicing in the context of Wi-Fi networks. The proposed system slices the wireless resources unevenly among flows, taking into account the characteristics of the video and the QoE demands for each user. Experimental results show that fairness improves by 56% when compared to an unmanaged network, while the mean QoE is reduced by only 5%.     

Distributed Medium Access Control Next-Generation CDMA Wireless Networks

The next-generation wireless networks are expected to have a simple infrastructure with distributed control. In this article, we consider a generic distributed network model for future wireless multimedia communications with a code-division multiple access (CDMA) air interface. For the medium access control (MAC) of the network model, we provide an overview of recent research efforts on distributed code assignment and interference control and identify their limitations when applied in next-generation wireless networks supporting multimedia traffic. We also propose a novel distributed MAC scheme to address these limitations, where active receivers determine whether a candidate transmitter should transmit its traffic or defer its transmission to a later time. Simulation results are given to demonstrate the effectiveness of the proposed distributed MAC scheme.     

Error spreading: a perception-driven approach to handling error incontinuous media streaming

With the growing popularity of the Internet, there is increasing interest in using it for audio and video transmission. Perceptual studies of audio and video viewing have shown that viewers find bursty losses, mostly caused by congestion, to be the most annoying disturbance, and hence these are critical issues to be addressed for continuous media streaming applications. Classical error handling techniques have mostly been geared toward ensuring that the transmission is correct, with no attention to timeliness. For isochronous traffic like audio and video, timeliness is a key criterion, and given the high degree of content redundancy, some loss of content is quite acceptable. We introduce the concept of error spreading, which is a transformation technique that permutes the input sequence of packets (from a continuous stream of data) before transmission. The packets are unscrambled at the receiving end. The transformation is designed to ensure that bursty losses in the transformed domain get spread all over the sequence in the original domain, thus improving the perceptual quality of the stream. Our error spreading idea deals with both cases where the stream has or does not have inter-frame dependencies. We next describe a continuous media transmission protocol and experimentally validate its performance based on this idea. We also show that our protocol can be used complementary to other error handling protocols     

Optimal quality adaptation for scalable encoded video

The dynamic behavior of the Internet's transmission resources makes it difficult to provide perceptually good quality streaming video. Scalable video encoding techniques have been proposed to deal with this problem. However, an encoded video generally exhibits significant data rate variability to provide consistent visual quality. We are, therefore, faced with the problem of accommodating the mismatch between the available bandwidth variability and the encoded video variability. We investigate quality adaptation algorithms for scalable encoded variable bit-rate video over the Internet. Our goal is to develop a quality adaptation scheme that maximizes perceptual video quality by minimizing quality variation, while at the same time increasing the usage of available bandwidth. We propose an optimal adaptation algorithm and a real-time adaptation algorithm based on whether the network conditions are known a priori. Experimental results show that the real-time adaptation as well as the optimal adaptation algorithm provide consistent video quality when used over both TCP-friendly rate control (TFRC) and transmission control protocol (TCP).     

智能数据流传输在网络视频中的应用与实现

为了进一步提高视频传输质量,充分发挥网络视频在教学中的作用,提出了一种新的基于智能数据流的传输模式,摒弃了以往单一改善网络传输环境的思路,建立更加智能的视频源服务环境,并详细阐述了智能数据流传输的几个关键技术。最后,从网络接收延时方面对基于智能数据流传输的网络视频教学系统进行仿真试验。实验表明,该系统可以有效地减少网络接收延时,提高网络视频教学的服务质量和效率。     

Video Streaming with Network Coding

Recent years have witnessed an explosive growth in multimedia streaming applications over the Internet. Notably, Content Delivery Networks (CDN) and Peer-to-Peer (P2P) networks have emerged as two effective paradigms for delivering multimedia contents over the Internet. One salient feature shared between these two networks is the inherent support for path diversity streaming where a receiver receives multiple streams simultaneously on different network paths as a result of having multiple senders. In this paper, we propose a network coding framework for efficient video streaming in CDNs and P2P networks in which, multiple servers/peers are employed to simultaneously stream a video to a single receiver. We show that network coding techniques can (a) eliminate the need for tight synchronization between the senders, (b) be integrated easily with TCP, and (c) reduce server’s storage in CDN settings. Importantly, we propose the Hierarchical Network Coding (HNC) technique to be used with scalable video bit stream to combat bandwidth fluctuation on the Internet. Simulations demonstrate that under certain scenarios, our proposed network coding techniques can result in bandwidth saving up to 60% over the traditional schemes.     

Low-complexity video coding for receiver-driven layered multicast

The “Internet Multicast Backbone,” or MBone, has risen from a small, research curiosity to a large-scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. We describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a “real” application-the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone     

Progressive Streaming of Video Data for Traffic Surveillance

Traffic surveillance has been one of the essential attributes in smart city concept. Nowadays, in such applications rotating camera is preferred over static camera. Motivation behind this substitution is to reduce the cost of data transmission and Total of cost of ownership. To design an optimal and performant wireless ‘smart city area network’ for video surveillance systems, this paper focuses on some key areas, namely, transmission efficiency, lossless video data coding, data congestion, edge computing at transmission nodes. The end objective is to achieve high quality received video stream in spite of compressed data transmission. Some research initiatives in this domain are pertinent. For example, Structural Similarity Index (SSIM) based rate distortion optimization is an effective tool in enhancing the perceptual video quality in wireless environments. However, prevailing system does not consider the network congestion conditions, affecting quality of received video. Also, effect of distortion introduced by ‘channel noise’ is unattended. This motivated us to propose a new dual metric traffic control mechanism, wherein both metrics i.e. distortion and data congestion are considered. It is based on an ‘improvised SSIM’ method which incorporates the ‘Rate of allocation’ algorithm as a function. Experimental results unveil that the proposed traffic control using similarity index under noise diversity can achieve better video quality and more data throughput.     

有限长序列卷积和求解法

有很多简单而直观的方法计算有限长序列的卷积和,但国内一些教材对此缺乏系统介绍。个别教材仅介绍了一种方法,且含糊不清,不便于教师讲授和学生自学。本文简洁地介绍了四种方法,其中包括笔作者提出的序号和匹配法。     

基于分层多播的视频传输拥塞控制算法研究

分层多播以多速率方式解决了多播接收者异构性问题,对于提高网络服务质量具有重要意义。本文分析了分层多播传输特性,通过提取MPEG视频流中的I帧、P帧、B帧组成3个帧流,分别放到分层多播的基层、增强层1和增强层2上传输,并在中间节点采用优先级队列机制,提出了一种面向视频流传输的分层多播拥塞控制(VLMCC)算法。仿真实验表明,本文提出的VLMCC算法能够适应视频多播接收者的异构性,大大提高了视频多播传输质量。