基于区分服务的流媒体传输模型

流媒体在网络上的应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流（实时流媒体数据流）占用更多的带宽。仿真结果表明：该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

基于自适应FEC的区分服务流媒体传输模型

实时流媒体在目前的因特网中传输经常会因为网络带宽的不足或数据包丢失严重,使得接收方播放质量受到严重影响.基于跨层设计的思想,在应用层使用自适应前向纠错算法,即使流媒体数据有一定的丢包率,接收方仍然能完整地恢复出原视频序列.在数据链路层采用有利于流媒体传输的区分服务模型,用以增大高优先级数据流(如实时流媒体数据流)传榆过程的吞吐量.数学分析和仿真结果均表明,该策略能使接收方获得最大的可播放帧率,从而有效提高流媒体传输的可靠性和实时性.     

边传送 边播放——谈运营商流媒体应用策略

流媒体技术是在数据网络上以流的方式传输多媒体信息技术。近年来,在宽带网络的高速发展和用户需求的驱动下,流媒体应用已经和3G、NGN 联系在一起,受到越来越多的关注。与传统播放方式不同,流媒体在播放前并不下载整个文件,只将部分内容缓存,使流媒体数据流边传送边播放,节省下载等待时间和存储空间。流媒体数据流具有三个特点:连续性、实时性、时序性,即其数据流具有严格的前后时序关系。除了宽带网络外,流媒体技术还可以广泛地应用于其他网络,例如无线流媒体传输是3G 网络的主要应用之一。在 NGN网络中,流媒体也扮演重要的角色。目前,应用提供商、内容开发商、业务支撑系统运营商、网络提供商、用户构成了完整的产业链。其中,用户最终获得宽带流媒体     

针对流媒体应用的扩展UDP协议

流媒体(Streaming Media)是一种以音视频数据流的方式在网络上传递多媒体信息的技术。但是,一直以来由于视频数据量大和网络带宽有限的矛盾,流媒体技术的应用受到很大制约,因此必须根据流媒体的特点采用合适的网络协议和传输算法,才能达到较好的传输效果。     

流媒体技术在证券股评系统中的应用研究

流媒体技术是在数据网络上以数据流的方式传输多媒体信息的技术,被认为是未来高速网络的主流应用之一。该文研究流媒体关键技术中的音视频数据采集和编码技术,解决在多媒体数据应用平台中双数据流的同步和传输问题。综合应用到某证券公司视讯管理平台和证券股评系统,有效地实现了在数据网络上的实时多点视频数据流同步传输和交互。     

流媒体数据传输主动式滤波整流模型

针对采用VBR编码的流媒体数据，由于其码率变化大，常出现数据猝发传输，而传统的流控技术又需要大量的客户端反馈。致使带宽调整具有较大的滞后性而不适合基于UDP的VBR流媒体服务的问题。提出了一种基于流媒体数据传输速率本身特征的主动滤波整流模型。由于此模型无需客户反馈信息，可利用服务器的数据缓冲对数据流进行滤波整流处理，从而不仅减少了由VBR编码带来的数据猝发传输。而且可直接在服务器方提供稳定的数据流传输服务。     

基于累积时延的流媒体传输模糊拥塞控制

为保证网络流媒体传输质量,在流媒体的传输中需要采用有效的拥塞控制策略.结合流媒体数据对时延敏感的特点,提出了一种基于累积时延的模糊拥塞控制算法,该算法在流媒体数据流传输过程中检测和跟踪其时延,在转发分组数据前,根据容忍时延阈值,丢弃超时数据包,减少不必要的带宽浪费,并且对所到达的数据流按照累积时延进行优先级分类,把全局性缓冲区和各队列的局部性缓冲区按照正常、拥塞避免和拥塞的规则划分为3个具有交叉过渡域的阶段,然后采用整体和局部相结合的拥塞控制方法,实现队列调度过程中的模糊处理,从而对网络拥塞进行有效的控制.理论分析和实验结果表明,使用基于累积时延的模糊拥塞控制算法,能有效改善流媒体的传输性能,是解决流媒体传输拥塞控制的有效途径,并能对提高网络性能起到重要作用.     

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.     

流媒体技术详解

流媒体(Streaming Media)是一种新兴的网络传输技术，在互联网上实时顺序地传输和播放视／音频等多媒体内容的连续时基数据流，流媒体技术包括流媒体数据采集、视／音频编解码、存储、传输、播放等领域。     

FSMS无线流媒体传输模型

无线流媒体传输技术可以满足用户利用无线网络实现在线视频播放,但较恶劣的无线网络环境使在线流媒体,特别对于需实况转播的流媒体系统,视频画面可能出现延迟、抖动和失真等问题,这在很大程度上影响了用户观看视频的直观感受。为了降低无线流媒体传输技术对无线网络环境的要求,提高无线视频传输的QoS(Quality of Service,服务质量),在国外视频分帧传输的思想上,提出一种FSMS(Frame Splitting Multichannel Streaming,分帧多信道传输)无线流媒体传输模型,并且基于该模型在Android移动平台上设计并实现了一整套无线流媒体传输系统。系统测试表明,即使在大量帧丢失的较恶劣网络环境下,运用此模型可以显著提高无线流媒体视频播放的流畅性。     

Rate-distortion optimized distributed packet scheduling of multiple video streams over shared communication resources

We consider the problem of distributed packet selection and scheduling for multiple video streams sharing a communication channel. An optimization framework is proposed, which enables the multiple senders to coordinate their packet transmission schedules, such that the average quality over all video clients is maximized. The framework relies on rate-distortion information that is used to characterize a video packet. This information consists of two quantities: the size of the packet in bits, and its importance for the reconstruction quality of the corresponding stream. A distributed streaming strategy then allows for trading off rate and distortion, not only within a single video stream, but also across different streams. Each of the senders allocates to its own video packets a share of the available bandwidth on the channel in proportion to their importance. We evaluate the performance of the distributed packet scheduling algorithm for two canonical problems in streaming media, namely adaptation to available bandwidth and adaptation to packet loss through prioritized packet retransmissions. Simulation results demonstrate that, for the difficult case of scheduling nonscalably encoded video streams, our framework is very efficient in terms of video quality, both over all streams jointly and also over the individual videos. Compared to a conventional streaming system that does not consider the relative importance of the video packets, the gains in performance range up to 6 dB for the scenario of bandwidth adaptation, and even up to 10 dB for the scenario of random packet loss adaptation.     

基于GM(1,1)模型的自适应链路层ARQ控制策略

提出了一种用于无线实时流媒体传输的自适应链路层ARQ控制策略,用以提高接收方的播放质量。该策略采用跨层设计的方法,基于GM（1,1）模型预测当前的网络状态,考虑GOP可解码帧数的特性,自适应地调整ARQ参数Nmax;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。     

Wireless network interface energy consumption

With the proliferation of mobile streaming multimedia, available battery capacity constrains the end-user experience. Since streaming applications are expected to be long running, wireless network interface card's (WNIC) energy consumption is particularly an acute problem. In this work, we explore various mechanisms to conserve client WNIC energy consumption for popular streaming formats such as Microsoft Windows media, Real and Apple Quicktime. First, we investigate the WNIC energy consumption characteristics for these popular multimedia streaming formats under varying stream bandwidth and network loss rates. We show that even for a high bandwidth 2000 kbps stream, the WNIC unnecessarily spent over 56% of the time in idle state; illustrating the potential for significant energy savings.Based on these observations, we explore two mechanisms to conserve the client WNIC energy consumption. First we show the limitations of IEEE 802.11 power saving mode for multimedia streams. Without an understanding of the stream requirements, these scheduled rendezvous mechanisms do not offer any energy savings for multimedia streams over 56 kbps. We also develop history-based client-side strategies to reduce the energy consumed by transitioning the WNICs to a lower power consuming sleep state. We show that streams optimized for 28.8 kbps can save over 80% in energy consumption with 2% data loss. A high bandwidth stream (768 kbps) can still save 57% in energy consumption with less than 0.3% data loss. We also show that Real and Quicktime packets are harder to predict at the network level without understanding the packet semantics. As the amount of cross traffic generated by other clients that share the same wireless segment increases, the potential energy savings from our client side policies deteriorate further. Our work enables multimedia proxy and server developers to suitably customize the stream to lower client energy consumption.     

无线视频流媒体异构接入的二维自适应流控模型研究

TFRC（TCP-Friendly Rate Control）机制适用于视频流媒体UDP流传输的流控,它保证UDP流的吞吐量具备良好的TCP友好特征。异构用户接入也可以借助TFRC机制探测可用带宽,但其在无线信道中面临新的挑战。基于无线信道特征,本文提出一种无线流媒体接入二维自适应流控模型。该模型建立在丢包率（PLR,Packct Loss Ratio）、误码率（BER,Bit Error Ratio）统计的基础上,分别针对包长和帧速率进行二维调节。首先,基于BER统计来调整包长和发送间隔以提供稳定帧速率的流量;其次,根据TFRC方程,由RTT（RoundTripTime）、RTO（Retransmission timeout）、PLR等计算可用带宽,结合帧错误率（FER,Frame Error Ratio）作为帧速率调整指标。仿真结果验证了该模型的有效性。     

Packet Permutation: A Robust Transmission Technique for Continuous Media Streaming Over the Internet

With the growing popularity of the Internet, there is an increasing demand to deliver continuous media (CM) streams over the Internet. However, packets may be damaged or lost during transmission over the current Internet. In particular, periodic network overloads often result in bursty packet losses, degrading the perceptual quality of CM streaming. In this paper, we focus on reducing the impact of this bursty loss behavior. We propose a novel robust end-to-end transmission scheme, referred to as packet permutation (PP), to deliver pre-compressed continuous media streams over the Internet. At the server side, PP permutes, prior to transmission, the normal packet delivery sequence of CM streams in a specific way. The packets are then re-permuted at the receiver side before they are presented to the application. In this way, the probability of losing a large number of packets within each CM frame can be significantly reduced. To validate the effectiveness of PP, a series of trace-driven simulations are conducted. Our results show that for a given quality of service (QoS) requirement of CM streaming, PP greatly reduces the overhead required by traditional error control schemes, such as forward error correction (FEC) and feedback/retransmission-based schemes.     

Server side,play buffer based quality control for adaptive media streaming

Existing media streaming protocols provide bandwidth adaptation features in order to deliver seamless video streams in an abrupt bandwidth shortage on the networks. For instance, popular HTTP streaming protocols such as HTTP Live Streaming (HLS) and MPEG-DASH are designed to select the most appropriate streaming quality based on client side bandwidth estimation. Unfortunately, controlling the quality at the client side means the effectiveness of the adaptive streaming is not controlled by service providers, and it harms the consistency in quality-of-service. In addition, recent studies show that selecting media quality based on bandwidth estimation may exhibit unstable behavior in certain network conditions. In this paper, we demonstrate that the drawbacks of existing protocols can be overcome with a server side, buffer based quality control scheme. Server side quality control solves the service quality problem by eliminating client assistance. Buffer based control scheme eliminates the side effects of bandwidth based stream selection. We achieve this without client assistance by designing a play buffer estimation algorithm. We prototyped the proposed scheme in our streaming service testbed which supports pre-transcoding and live-transcoding of the source media file. Our evaluation results show that the proposed quality control performs very well both in simulated and real environments.     

面向端用户的流媒体业务性能监测与分析

目前,已有的流媒体业务性能测量工具主要用于对服务进行压力测试,不能反映终端用户的实际使用性能。本文从端用户角度设计了合理反映流媒体业务性能的指标,提出并实现了一种基于主动业务仿真的流媒体业务性能测量方法,能及时向流媒体服务器反馈信息,实现动态的质量控制,并为流媒体业务系统的优化提供依据。     

基于电力载波和流媒体的视频监控服务器的实现

文章首先介绍了电力载波通信、视频数据传输以及RTP/RTCP协议的特点,在此基础上提出了一种以电力载波为传输媒介和采用流媒体技术的视频监控服务器设计方案,该方案首先给出了系统的设计框架,然后从软件开发角度具体阐述流媒体服务器实现流程,最后给出实验结果和分析。实验结果表明,在基于电力载波的视频监控服务器中采用RTP/RTCP协议传输视频数据可有效利用网络带宽和降低丢包率。     

一种适用于流媒体传输的无线TCP友好速率控制机制

综合无线错误丢包以及拥塞丢包事件的统计特征,基于丢包区分,本文提出一种适用于流媒体传输的无线TCP友好速率控制(Wireless TCP-Friendly Rate Control,WTFRC)方程并建立实用的速率控制机制.该机制包括服务器或者代理发送端和用户接收端两部分,分别负责往返传输时间(Round-Trip Time,RTT)估计、发送速率估计调整和丢包区分、丢包率(Pack-et Loss Ratio,PLR)统计、接收速率估计.仿真结果验证了该机制能够获得较高的吞吐量和较平缓的发送速率,并具备较好的TCP友好特征.