一种自适应的3G网络流媒体速率控制算法

在3G移动无线网络中,流媒体的服务质量不仅受限于无线链路的不稳定特性,如时延、抖动、丢包率等,客户端的缓存与处理能力也是重要的制约要素,为自适应速率控制算法的研究带来了很大的挑战.本文首先介绍了在3G网络中流媒体自适应传输的机制和研究现状,然后分析了点对点自适应传输的策略问题,并综合考虑通信链路与客户端能力问题,提出了基于客户端缓存的AIMD自适应速率控制算法(ABB-AIMD),最后通过仿真实验对算法进行验证.     

试论一种自适应3G网络流媒体速率控制算法

随着我国3G移动网络的发展,流媒体业务已经成为了我国移动通信系统中的一个重要组成部分.对3G移动无线网络而言,流媒体的服务质量常常受到无线链路的不稳定性影响,同时还受到客户端缓存和处理能力的制约.本文从自适应AIMD速率控制算法设计入手,并通过仿真实验验证自适应AIMD速率控制算法的性能.     

自适应流媒体传输技术分析与研究

在介绍流媒体自适应传输技术发展现状的基础上,根据自适应策略实现实体的不同,将当前主流技术分为服务器端实现、客户端实现两种类型。重点分析了客户端实现类型的系统架构、工作流程,并对主流超文本传输协议(HTTP)自适应流媒体系统的特点进行了对比分析。由于实现简单,且在穿越防火墙/NAT方面更具优势,基于HTTP的自适应流媒体技术已成为流媒体技术的发展趋势。     

基于P2P的3G流媒体代理缓存策略

3G时代已经到来,伴随着3G的快速发展,移动流媒体服务的需求迅速增长。由于移动网络带宽远达不到互联网的带宽,流媒体业务的增长将增加网络延迟,影响流媒体的观看质量,因此提出了一种新的基于P2P的3G流媒体缓存代理结构PSPA,配合PSPA设计了代理缓存置换策略SCOP,该设计能有效降低对移动核心网带宽的占用。最后,仿真实验和实验结果分析表明,PSPA体系与代理缓存置换策略SCOP的结合,能有效提高移动流媒体的服务质量,对3G流媒体的应用具有一定的参考价值。     

有线无线网络TCP友好流媒体拥塞控制机制

在TCP友好拥塞控制方法比较基础上,为了实现流媒体平滑传输以及保证TCP流友好性,提出了TCP友好速率控制算法WTFCC。该算法能够在接收端区分网络拥塞丢包和链路错误随机丢包,准确判断网络拥塞状况;结合接收端缓存区占用程度,自适应实施多级速率调节。仿真实验结果表明,该机制对TCP流是友好的,并且保障了媒体播放质量,在有线无线混和网络中具有很好的性能。     

基于批处理补丁的流媒体后缀动态缓存算法

在IP网络上高效传输流式存储型A／V数据是实现诸如VOD等应用的基础。当前一些典型的传输方案考虑了服务器调度策略以降低骨干网带宽消耗和服务器负载，但并未考虑媒体后缀的缓存策略。本文在带前缀的OBP算法基础上提出了流媒体对象后缀的增量式缓存及快速释放算法ICBR，并推导出了采用IC算法所需的骨干网带宽的理论结果。通过针对上述两种算法的仿真实验，本文的结果表明：即使在有限的缓存容量的前提下，采用IC算法和ICBR算法对媒体对象的后缀进行动态缓存可以显著降低骨干网链路上传输的补丁数据量，其骨干网带宽消耗显著优于OBP，从而在保证客户端较小的播放启动延迟的情况下有效降低了流媒体传输中骨干网带宽的消耗和服务器的负载。     

3G网络中流媒体缓存系统的设计与实现

针对移动流媒体系统中无线接入网带宽窄、终端缓存空间小的特点，提出了一种适用于3G WCDMA网络的流媒体缓存体系结构WSCA，设计了基于缓存的移动流媒体调度策略WSCS，探讨了WSCA系统在实现中的技术细节。分析表明，WSCA体系与WSCS调度策略结合，能有效提高用户服务质量，降低核心网络带宽消耗，对3G流媒体应用具有一定的参考价值。     

MPEG-2传送流传输方法的研究与改进

高品质视频流媒体的传输在流媒体业务中占有很重要的地位，同时带来许多难题．以传统的恒定速率传输变比特率压缩的视频时，播放终端往往需要很大的缓存．PCR协助的恒定速率传输是一种新的恒定速率传输机制．这个方法的详细研究表明PCBR方法利用嵌入MPEG-2传送流的程序参考时钟来定期校正传送速率，是以较高传输速率为代价减少了缓存需求．因此对该方法进行了改进，通过缩小传输时间尺度来降低传送速率但没有增加额外的缓存需求，在NS通信系统仿真软件和实验平台的测试结果都表明，改进的方法能够更好服务于流媒体传输与终端回放．     

基于状态机的视频码率自适应算法

由于链路带宽存在随机性,已有的基于超文本传输协议的动态自适应流媒体传输技术（DASH）的码率自适应算法不能很好解决播放流畅性和视频质量之间的矛盾。为解决该问题,提出一种基于状态机的DASH（SDASH）算法,将码率切换过程用状态机进行分析与控制。首先充分考虑客户端观看体验质量（QoE）的影响因素,对影响因素进行数值分析,并设定6个码率等级状态;然后将视频码率与影响因素的数值变化之间的联系作为状态转移条件;最后在保证播放缓存和码率偏移率处于一定阈值的条件下将视频码率切换至视频质量和播放流畅性整体性能相对最佳的码率等级上。实验结果表明,该算法与基于模糊逻辑控制的码率自适应算法相比能够提高客户端请求视频的平均码率,且尽量避免出现码率骤降等情况,从而较好地平衡播放流畅性和视频质量之间的关系,提升了视频观看过程的体验质量。     

SMTVS:一种基于H.264的手机电视系统

手机电视系统必须能够适应网络和客户端的异构性、无线信道的多变性等特点.提出一种手机电视系统SMTVS,流媒体服务器在不同码率的H.264视频流之间进行切换以适应网络状况的变化;代理服务器之间构成一个内容分送网络,视频数据通过应用层组播的方式传输到各个代理服务器,降低了服务器的负载,提高了网络资源利用率;代理服务器接收到视频数据之后,针对无线链路的状况对数据进行处理后再发送到客户端,以适应最终用户的不同需求;客户端根据网络状况来动态调整播放速率,避免显示缓冲区下溢和上溢.通过流媒体服务器、代理服务器以及客户端的协作,实现了视频数据在无线信道上的自适应传输.     

Network-adaptive autonomic transcoding algorithm for seamless streaming media service of mobile clients

As a result of improvements in wireless communication technologies, a multimedia data streaming service can now be provided for mobile clients. Since mobile devices have low computing power and work on a low network bandwidth, a transcoding technology is needed to adapt the original streaming media for mobile environments. However, wireless networks have variable bandwidths depending on the movement of clients and the communication distance from Access Point (AP). These characteristics make it hard to support stable Quality of Service (QoS) streams for mobile clients. In this paper, a target transcoding bit-rate decision algorithm is proposed to provide stable QoS streams for mobile clients. In our experiments, the proposed algorithm provides seamless streaming media services based on the network adaptive bit rate control and reduces transmission failure.     

Resource allocation for multimedia streaming over the Internet

This paper addresses the resource allocation problem for multiple media streaming over the Internet. First, we present an end-to-end transport architecture for multimedia streaming over the Internet. Second, we propose a new multimedia streaming TCP-friendly protocol (MSTFP), which combines forward estimation of network conditions with information feedback control to optimally track the network conditions. Third, we propose a novel resource allocation scheme to adapt media rate to the estimated network bandwidth using each media's rate-distortion function under various network conditions. By dynamically allocating resources according to network status and media characteristics, we improve the end-to-end quality of services (QoS). Simulation results demonstrate the effectiveness of our proposed schemes     

ARTP: A buffer-aware rate control protocol for media streaming

We present adaptive real-time transport protocol (ARTP), a media streaming transport protocol that implements a congestion control mechanism. With this mechanism, the sender adapts its sending rate to network conditions and to the buffering capacity of the receiver. This adaptiveness takes into account the real-time constraints of media streaming. It aims at ensuring media playback continuity, and at achieving a low packet loss rate during media streaming sessions. ARTP ensures the continuity of media playback by buffering media packets during congestion-free periods and reduces the loss rate by reducing the transmission rate during congestion periods. This protocol considers the size of the buffer that the receiver dedicates to rate control in order to avoid overflow or underflow of the buffer. This approach allows limited memory devices such as cellular phones and PDAs to take advantage of rate control.ARTP is based on the feedback that the real-time control protocol (RTCP) reports give with the addition of two new parameters that we define in this paper: the steady state loss event rate and the duration to the next feedback report. It also requires that the real-time streaming protocol (RTSP) provide the server with the size of the buffer that the client dedicates to rate control.Our NS-2 simulations show that, besides buffer protection, ARTP significantly reduces the loss rate. Compared to Additive Increase Multiplicative Decrease (AIMD) rate control techniques, ARTP provides a better media quality by ensuring the continuity of media playback and, compared to equation-based rate control techniques, it achieves a better loss rate and reduces the bandwidth used for feedback.     

Double feedback streaming agent for real-time delivery of media over 3G wireless networks

A network agent located at the junction of wired and wireless networks can provide additional feedback information to streaming media servers to supplement feedbacks from clients. Specifically, it has been shown that feedbacks from the network agent have lower latency, and they can be used in conjunction with client feedbacks to effect proper congestion control. In this work, we propose the double-feedback streaming agent (DFSA) which further allows the detection of discrepancies in the transmission constraints of the wired and wireless networks. By working together with the streaming server and client, DFSA reduces overall packet losses by exploiting the excess capacity Of the path with more capacity. We show how DFSA can be used to support three modes of operation tailored for different delay requirements of streaming applications. Simulation results under high wireless latency show significant improvement of media quality using DFSA over non-agent-based and earlier agent-based streaming systems.     

Real-time video transport optimization using streaming agent over 3G wireless networks

Feedback adaptation has been the basis for many media streaming schemes, whereby the media being sent is adapted in real time according to feedback information about the observed network state and application state. Central to the success of such adaptive schemes, the feedback must: 1) arrive in a timely manner and 2) carry enough information to effect useful adaptation. In this paper, we examine the use of feedback adaptation for media streaming in 3G wireless networks, where the media servers are located in wired networks while the clients are wireless. We argue that end-to-end feedback adaptation using only information provided by 3G standards is neither timely nor contain enough information for media adaptation at the server. We first show how the introduction of a streaming agent (SA) at the junction of the wired and wireless network can be used to provide useful information in a timely manner for media adaptation. We then show how optimization algorithms can be designed to take advantage of SA feedbacks to improve performance. The improvement of SA feedbacks in peak signal-to-noise ratio is significant over nonagent-based systems.     

基于TCP的支持多格式实时编码的流媒体直播系统设计与实现

随着多媒体技术的发展和宽带网络的普及,IPTV系统在国内已经得到越来越广泛的应用,直播系统作为其中的一个子系统虽然满足用户的需求可以进行时移操作,然后面向的基本都是机顶盒用户群,对于大部分PC用户来说,能够支持时移的直播系统几乎没有。基于DirectShow和Socket技术,建立了一种流媒体音视频实时编码及网络传输系统,并且可以支持时移功能。     

基于P2P网络模式VOD系统的数据传输新方法

随着计算机和网络技术发展,流媒体应用越来越受到关注。本文提出并实现了一种基于P2P网络模式的流媒体视频点播系统数据传输结构,这种新型的流媒体视频点播系统能够利用客户机的系统空间和网络资源进行视频的存储和传输。实验证明,该系统有效地降低了服务器的负荷,使服务器能够满足更多客户的点播请求,而降低了系统成本。     

面向TCP的流媒体传输编码码率自适应算法

由于网络的时变性和异构性,以及在拥塞情况下的高丢包率,利用TCP传输流媒体数据是Internet流媒体分发系统提高流媒体分发质量的首选方案。由于TCP具有超时或错误重传机制,在网络拥塞情况下,难以保证高码率流媒体数据传输的实时性,因此提出一种面向TCP流媒体传输的编码码率自适应算法(TCP_RA)。该算法根据流媒体发送应用层缓冲区读写指针差值调整流媒体发送端的编码码率适应网络带宽的变化。仿真实验对比分析了该算法与基于UDP之上TFRC协议的流媒体传输码率自适应算法在流媒体传输质量上的差别。结果表明,该算法在网络环境较差的情况下有效地提高了流媒体传输质量。并且该算法容易实现,值得推广。     

一种基于反馈的流媒体拥塞控制算法的改进

目前流媒体的拥塞控制算法大都采用控制发送端速率的方法，但这种方法存在着反馈延迟和参数选择的问题。针对上述问题，在一种基于反馈的流媒体拥塞控制算法FCA基础之上，从反馈信息的即时性和参数调整两方面进行了改进。仿真实验结果表明，改进算法IFCA在延时抖动和吞吐量方面都有所改进，更适合流媒体的传输，并较好地保持了TCP友好特性。