Packet Spacing: An Enabling Mechanism for Delivering Multimedia Content in Computational Grids

Streaming multimedia with UDP has become increasingly popular over distributed systems like the Internet. Scientific applications that stream multimedia include remote computational steering of visualization data and video-on-demand teleconferencing over the Access Grid. However, UDP does not possess a self-regulating, congestion-control mechanism; and most best-effort traffic is served by congestion-controlled TCP. Consequently, UDP steals bandwidth from TCP such that TCP flows starve for network resources. With the volume of Internet traffic continuing to increase, the perpetuation of UDP-based streaming will cause the Internet to collapse as it did in the mid-1980's due to the use of non-congestion-controlled TCP. To address this problem, we introduce the counter-intuitive notion of inter-packet spacing with control feedback to enable UDP-based applications to perform well in the next-generation Internet and computational grids. When compared with traditional UDP-based streaming, we illustrate that our approach can reduce packet loss over 50% without adversely affecting delivered throughput.     

Packet Spacing: An Enabling Mechanism for Delivering Multimedia Content in Computational Grids

Streaming multimedia with UDP has become increasingly popular over distributed systems like the Internet. Scientific applications that stream multimedia include remote computational steering of visualization data and video-on-demand teleconferencing over the Access Grid. However, UDP does not possess a self-regulating, congestion-control mechanism; and most best-effort traffic is served by congestion-controlled TCP. Consequently, UDP steals bandwidth from TCP such that TCP flows starve for network resources. With the volume of Internet traffic continuing to increase, the perpetuation of UDP-based streaming will cause the Internet to collapse as it did in the mid-1980's due to the use of non-congestion-controlled TCP.To address this problem, we introduce the counter-intuitive notion of inter-packet spacing with control feedback to enable UDP-based applications to perform well in the next-generation Internet and computational grids. When compared with traditional UDP-based streaming, we illustrate that our approach can reduce packet loss over 50% without adversely affecting delivered throughput.     

具有最小平方误差积分的主动队列管理算法ISE-GPM-PID

随着互联网流媒体应用的增多,流媒体流量在互联网总流量中所占比例越来越大。流媒体流量具有时延敏感和容忍丢包等特点,通常采用UDP协议传输;UDP流量是对TCP/AQM模型的干扰流量,但现有针对TCP长流设计的主动队列管理算法缺乏抗击UDP流量干扰的能力,不能适应互联网视频和音频流量日益增多的局面。利用TCP/AQM模型,设计了一个具有最小平方误差积分,且相角裕度在30°至60°之间,幅值裕度在2至5之间的基于PID控制器的主动队列管理算法ISE-GPM-PID。该算法对UDP流量具有良好的抗干扰能力,能适应互联网流媒体和Web应用日益增多的局面;同时其响应速度快,计算开销小,能用于大时滞网络环境,且鲁棒性好。     

Empirical evaluation of the congestion responsiveness of RealPlayer video streams

Increasingly popular commercial streaming media applications over the Internet often use UDP as the underlying transmission protocol for performance reasons. Hand-in-hand with the increase in streaming media comes the impending threat of unresponsive UDP traffic, often cited as the major threat to the stability of the Internet. Unfortunately, there are few empirical studies that analyze the responsiveness, or lack of it, of commercial streaming media applications. In this work, we evaluate the responsiveness of RealNetworks’ RealVideo over UDP by measuring the performance of numerous streaming video clips selected from a variety of RealServers on the Internet, analyze the TCP-Friendliness of the UDP streams and correlate the results with network and application layer statistics. We find that most RealVideo UDP streams respond to Internet congestion by reducing the application layer encoding rate, and streams with a minimum encoding rate less than the fair share of the capacity often achieve a TCP-Friendly rate. In addition, our results suggest that a reason streaming applications choose not to use TCP is that the TCP API hides network information, such as loss rate and round-trip time, making it difficult to estimate the available capacity for effective media scaling.     

A new fast friendly window-based congestion control for real-time streaming media transmission

1 Introduction In the current Internet, not all applications use TCP and they do not follow the same concept of fairly sharing the available bandwidth. The rapid growing of real-time streaming media applications will bring much UDP traffic without integrating TCP compatible congestion control mechanism into Internet. It threats the quality of service (QoS) of real-time applications and the stability of the current Internet. For this reason, it is desirable to define appropriate rate rule…     

Evalvid-RA: trace driven simulation of rate adaptive MPEG-4 VBR video

Due to the increasing deployment of conversational real-time applications like VoIP and videoconferencing, the Internet is today facing new challenges. Low end-to-end delay is a vital QoS requirement for these applications, and the best effort Internet architecture does not support this natively. The delay and packet loss statistics are directly coupled to the aggregated traffic characteristics when link utilization is close to saturation. In order to investigate the behavior and quality of such applications under heavy network load, it is therefore necessary to create genuine traffic patterns. Trace files of real compressed video and audio are text files containing the number of bytes per video and audio frame. These can serve as material to construct mathematical traffic models. They can also serve as traffic generators in network simulators since they determine the packet sizes and their time schedule. However, to inspect perceived quality, the compressed binary content is needed to ensure decoding of received media. The EvalVid streaming video tool-set enables this using a sophisticated reassembly engine. Nevertheless, there has been a lack of research solutions for rate adaptive media content. The Internet community fears a congestion collapse if the usage of non-adaptive media content continues to grow. This paper presents a solution named Evalvid-RA for the simulation of true rate adaptive video. The solution generates real rate adaptive MPEG-4 streaming traffic, using the quantizer scale for adjusting the sending rate. A feedback based VBR rate controller is used at simulation time, supporting TFRC and a proprietary congestion control system named P-AQM. Example ns-2 simulations of TFRC and P-AQM demonstrate Evalvid-RA’s capabilities in performing close-to-true rate adaptive codec operation with low complexity to enable the simulation of large networks with many adaptive media sources on a single computer.     

可交互流媒体服务中的应用层组播技术研究

在开放、动态的网络环境中,网络构件致力于如何有效地整合和共享多样化资源.近年来,流媒体应用在In-ternet上日趋流行,由此带来了资源共享和节约带宽消耗等一系列挑战性的问题.应用层组播被认为是解决大规模流媒体应用网络拥塞的一种有效技术.然而,流媒体交互操作会引起组播树的频繁重构,从而降低系统性能.提出了一种支持可交互操作的应用层组播树构建协议ISMT(Interactive Streaming Multicast Tree),可以降低用户响应延时和改善系统的扩展性.通过仿真实验验证了ISMT协议的有效性.     

SProxy: A Caching Infrastructure to Support Internet Streaming

Many algorithmic efforts have been made to address technical issues in designing a streaming media caching proxy. Typical of those are segment-based caching approaches that efficiently cache large media objects in segments which reduces the startup latency while ensuring continuous streaming. However, few systems have been practically implemented and deployed. The implementation and deployment efforts are hindered by several factors: 1) streaming of media content in complicated data formats is difficult; 2) typical streaming protocols such as RTP often run on UDP; in practice, UDP traffic is likely to be blocked by firewalls at the client side due to security considerations; and 3) coordination between caching discrete object segments and streaming continuous media data is challenging. To address these problems, we have designed and implemented a segment-based streaming media proxy, called SProxy. This proxy system has the following merits. First, SProxy leverages existing Internet infrastructure to address the flash crowd. The content server is now free of the streaming duty while hosting streaming content through a regular Web server. Thus, UDP based streaming traffic from SProxy suffers less dropping and no blocking. Second, SProxy streams and caches media objects in small segments determined by the object popularity, causing very low startup latency, and significantly reducing network traffic. Finally, prefetching techniques are used to pro-actively preload uncached segments that are likely to be used soon, thus providing continuous streaming. SProxy has been extensively tested and we show that it provides high quality streaming delivery in both local area networks and wide area networks (e.g., between Japan and the U.S.).     

流媒体自适应传输控制的研究

流媒体是指多媒体数据流在网络上一边传输一边播放的一种多媒体通信服务.提供尽力而为服务的Internet不能为流媒体保证网络带宽、传输延迟、分组丢失以及分组错误等,而自适应传输控制机制能够提高流媒体服务的传输服务质量和传输服务的公平性.本文探讨流媒体自适应传输控制技术所涉及的各个方面,包括拥塞控制、质量自适应和错误控制.     

一种基于机器学习的P2P网络流量识别方法

对等(P2P)覆盖网络作为一种典型的分布式系统日益受到人们的重视.P2P应用遍及文件共享、流媒体、即时通信等多个领域,P2P应用所产生的流量占据了互联网流量的60%以上.为了更好地管理和控制P2P流量,有必要对P2P流量识别模型进行深入的研究.提出一种基于小波支持向量机的机器学习模型(ML-WSVM)来识别已知和未知的P2P流量,ML-WSVM是通过满足小波框架和Mercer定理的小波基函数替换支持向量机核函数的方法,实现小波与支持向量机的结合.该模型充分利用了小波的多尺度特性与支持向量机在分类方面的优势.然后,提出基于损失函数的串行最小化算法来优化求解ML-WSVM的最优分类面.最后,理论分析和实验结果表明该方法大大提高了对P2P网络流量的识别精度和识别效率,尤其是对加密报文的识别.     

A transport-level proxy for secure multimedia streams

To provide a secure traversal service, firewalls need more than static packet filtering and application-level proxies. SOCKS (Secure sOCKets) is an application-independent transport-level proxy that offers user-level authentification and data encryption. An extended SOCKS UDP (user datagram protocol) binding model with appropriate socket calls is proposed to provide complete support for UDP-based multimedia streaming applications     

一种基于Sever和Proxy流媒体流行性的Caching策略

1 概述经过了2000、2001两年的社区宽带网建设的高速发展后,摆在中国ISP们面前的任务是如何在已建成的宽带网上开展增值服务,许多ISP尝试在宽带网上开展流媒体(Streaming Media)服务,如视频点播VOD(Video On-Demand)系统。然而,流媒体对网络带宽和实时性的要求使得流服务器必须能够进行端对端(End-to-End)的拥塞控制和质量调整,由于     

种基于Sever和Proxy流媒体流行性的Caching策略

It is expected that by 2003, continuous media will account for more than 50% of the data available on origin servers. This will provoke a significant change in Internet workload, due to the high bandwidth requirements and the long-lived nature of digital video, streaming server loads and network bandwidths are proving to be major limiting factors. Aiming at the characteristic of broadband network in a residential area, we propose a popularitybased on server-proxy caching strategy for streaming media. According to a streaming media popularity on streaming server and proxy, this strategy caches the content of this streaming media partially or completely, and plays an important role in decreasing server load, reducing the traffic from streaming server to proxy, and improving the startup latency of the client.     

Lightweight and Informative Traffic Metrics for Data Center Monitoring

In recent years, thousands of commodity servers have been deployed in Internet data centers to run large scale Internet applications or cloud computing services. Given the sheer volume of data communications between servers and millions of end users, it becomes a daunting task to continuously monitor the availability, performance and security of data centers in real-time operational environments. In this paper, we propose and evaluate a lightweight and informative traffic metric, streaming frequency, for network monitoring in Internet data centers. The power-series based metric that is extracted from the aggregated IP traffic streams, not only carries temporal characteristics of data center servers, but also helps uncover traffic patterns of these servers. We show the convergence and reconstructability properties of this metric through theoretical proof and algorithm analysis. Using real data-sets collected from multiple data centers of a large Internet content provider, we demonstrate its applications in detecting unwanted traffic towards data center servers. To the best of our knowledge, this paper is the first to introduce a streaming metric with a unique reconstruction capability that could aid data center operators in network management and security monitoring.     

Distributed WiFi detection and integration in dense urban mobile Peer-to-Peer networks

Running Peer-to-Peer applications??such as multimedia streaming or file sharing??on mobile devices significantly increases the congestion in 3G access networks. Offloading traffic from 3G to WiFi domains is promising in such scenarios, since communication is possible without generating any load in the WiFi??s uplink or in the Internet, given that peers are located in the same WiFi domain. However, in today??s urban areas devices are commonly in range of dozens of infrastructure-based WiFi domains, a fact that calls for an efficient rendezvous mechanism. In this article, we propose a rendezvous mechanism that efficiently enables physically close mobile devices running an arbitrary P2P application to peer with each other in a common WiFi domain. The mechanism builds upon tree-based collection, aggregation, and distribution of WiFi information. Using a stochastic model, we estimate the overhead of the mechanism based on WiFi density statistics from real world urban areas. We further show how to reduce this overhead on the expense of a reduced rendezvous success probability by applying Bloom Filters. Simulations of a tree-based Peer-to-Peer media streaming application demonstrate that the mechanism can in fact support effective offloading of P2P traffic to WiFi domains.     

网络视频服务的代理缓存策略

流媒体传输已经成为Internet通信中的重要组成部分之一，虽然它可以从代理缓存中受益，但传统的代理缓存策略不能满足媒体对象所特有的特征，必须提出新颖的缓存方法。该文讨论了代理对视频流的缓存，包括前缀、分块和部分视频对象的全部缓存的问题和挑战。同时研究了缓存视频流的代理网络结构，有分布式、层次式和覆盖式，对它们进行了描述和评论，还把缓存和代理网络同组播相结合进行了讨论。     

A real-time protocol for the Internet based on the least mean square algorithm

Generally, real-time applications based on the User Datagram Protocol (UDP) generate large volumes of data and are not sensitive to network congestion. In contrast, Transmission Control Protocol (TCP) traffic is considered "well-behaved" because it prevents the network becoming congested by means of closed-loop control of packet-loss and round-trip-time. The integration of both sorts of traffic is a complex problem, and depends on solutions such as admission control that have not yet been deployed on the Internet. Moreover, the problem of quality-of-service (QoS) and resource allocation is extremely relevant from the point of view of convergence of streaming media and data transmission on the Internet. In this paper an adaptive real-time protocol based on the least mean square (LMS) algorithm is proposed to estimate the application UDP bandwidth in order to reduce the quadratic error between the packet loss and a target. Moreover, the LMS algorithm is also applied to make sure that the reduction in the average bandwidth allocated to each TCP process will not be higher than a given percentage of the average bandwidth allocated before the beginning of the UDP application.     

Media streaming via TFRC: An analytical study of the impact of TFRC on user-perceived media quality

TCP-Friendly Rate Control (TFRC) is being adopted in Internet standards for congestion control of streaming media applications. In this paper, we consider the transmission of prerecorded media from a server to a client by using TFRC, and analytically study the impact of TFRC on user-perceived media quality, which is roughly measured by calculating the rebuffering probability. A rebuffering probability is defined to be the probability that the total duration of all rebuffering events experienced by a user is longer than a certain threshold. Several approaches are presented to help an application determine an appropriate initial buffering delay and media playback rate in order to achieve a certain rebuffering probability under a given network condition. First, we derive a closed-form expression to approximate the average TFRC sending rate, which could be used as the maximum allowed playback rate of a media stream. Second, we develop a queueing model for a TFRC client buffer with the traffic described by a Markov-Renewal-Modulated Deterministic Process (MRMDP), which captures the fundamental behavior of TFRC that predicts the immediate future TCP sending rate based on the history of past loss intervals. We present a closed-form solution and a more accurate iterative method to solve the queueing model and calculate the rebuffering probability.     

On the source switching problem of Peer-to-Peer streaming

Peer-to-Peer(P2P) streaming has been proved a popular and efficient paradigm of Internet media streaming. In some applications, such as an Internet video distance education system, there are multiple media sources which work alternately. A fundamental problem in designing such kind of P2P streaming system is how to achieve fast source switching so that the startup delay of the new source can be minimized. In this paper, we propose an efficient solution to this problem. We model the source switch process, formulate it into an optimization problem and derive its theoretical optimal solution. Then we propose a practical greedy algorithm, named fast source switch algorithm, which approximates the optimal solution by properly interleaving the data delivery of different media sources. The algorithm can adapt to the dynamics and heterogeneity of real Internet environments. We have carried out extensive simulations on various real-trace P2P overlay topologies to demonstrate the effectiveness of our model and algorithm. The simulation results show that our proposed algorithm outperforms the normal source switch algorithm by reducing the source switch time by 20%–30% without bringing extra communication overhead. The reduction in source switching time is more obvious as the network scale increases.     

视频细粒度可分级编码研究进展

Internet的迅速发展使越来越多的应用使用流媒体技术。作为流媒体的核心技术之一,视频的可分级编码技术已经成为一个重要的研究领域。本文首先对MPEG-4修订版中FGS的编码机制、可扩展特性和所存在的问题进行了讨论,然后对细粒度可分级视频编码的研究进展进行了分析,最后对视频细粒度可分级编码的未来发展趋势进行了展望。