一种TCP友好的Internet DTV组播发送速率控制机制

当前的Internet没有对流媒体应用提供足够的QoS的保证,同时Internet DTV的组播发送速率必须满足以下两点要求:1)能够自适应网络拥塞的变化;2)能够适应节目码率的要求,因此需要在服务器端进行组播发送速率控制.分析了Internet DTV组播的难点,提出了一种基于缓冲区管理的具有网络自适应特性的组播发送速率控制方法.通过合理地控制发送方的发送速率,既能自适应网络状况的变化,又能满足流媒体实时播放的需求.仿真实验结果表明,该算法在保证服务器端发送缓冲区不溢出前提下,与单纯采用TFMCC方法比较,减少了接收端的平均丢包率,提高了网络电视节目流的传输质量.     

基于多媒体网络的请求式测量系统

介绍了通过Internet的请求式测量 (MoD)系统 :允许访问、流式传输已存档的过程数据。客户可以通过访问服务器来接收以媒体流的形式存储的过程数据。MoD服务器和MoD客户之间的交互作用基于适于测量的实时流协议。在接收端 ,利用基于多媒体的分布式虚拟仪器捕获媒体流     

分布式多媒体系统中的媒体同步

本文介绍分布式多媒体系统中的媒体同步问题，着重介绍了基于MPEG－I的分布式多媒体系统中，视频和音频的同步以及媒体服务器和客户之间的流同步。     

开放型网络英语教学系统中流媒体QoS的研究

采用设置本地端缓冲服务器的方法提高流传榆质量．在开放型网络英语教学系统中应用流媒体提供QoS的管理功能,解决音视频流缓冲问题,并提供相应机制支持网络环境下的流媒体QoS。实验结果表明,流体系结构较好实现网络教学环境下的流媒体播放,保证音视频流的QoS。采用此流体系结构能较好地实现对流的管理和控制。从而保证多媒体课件的传输质量。     

开放型网络英语教学系统中流媒体QoS的研究

采用设置本地端缓冲服务器的方法提高流传输质量,在开放型网络英语教学系统中应用流媒体提供QoS的管理功能,解决音视频流缓冲问题,并提供相应机制支持网络环境下的流媒体QoS。实验结果表明,流体系结构较好实现网络教学环境下的流媒体播放,保证音视频流的QoS。采用此流体系结构能较好地实现对流的管理和控制,从而保证多媒体课件的传输质量。     

嵌入式系统中流媒体服务质量研究

围绕网络流媒体技术以及嵌入式环境的特点,为保障流媒体的服务质量,研究了基于客户端的缓冲区控制策略,以及具有流量控制机制的流媒体网络通信技术.在探讨嵌入式视频点播系统架构基础上,提出了利用平均传输速率、时延抖动、解码速率和辅助系数这四个参数进行网络缓冲区长度估算的简化算法;并提出在客户端按照数据的重要性或优先级的不同来实施主动抛弃策略;该系统也针对服务器端引入了流量控制算法.结果表明,应用上述策略明显节约了缓存,并且实现了流量控制功能,改善了系统的服务质量(Quality of Service,QoS).     

基于实时传输协议的QOS研究

随着互联网技术的发展与普及,多媒体等实时业务的应用变得越来越广泛,同时实时业务对网络传输的服务质量（QoS）要求也越来越高。在分析了影响网络实时传输QoS因素的基础上,利用丢包率作为判断当前网络拥塞与否的参考量,设计了一种自适应的流量控制机制,该机制使用变常数增长和乘减少的方式对发送速率进行自适应调整。仿真结果表明,该机制有效地提高了实时传输的QoS。     

A measurement-based approach for dynamic QoS adaptation in DiffServ networks

This paper presents a practical approach to managing multimedia traffic in DiffServ network, using network monitoring feedback and control. We exploit the flexibility of multimedia traffic and process network level parameters to adapt the traffic according to the current state of the network. The latter is determined based on reports sent by bandwidth monitors installed on each node of a DiffServ Domain. The bandwidth monitors interact with a policy server which, depending on the network state, decides the policy(ies) that should be enforced by the DiffServ network. The implementation of the selected policies typically leads to accepting, remarking, or dropping the multimedia traffic entering the network. Multimedia streams may be assigned different levels of QoS, as interpreted by the marker at the DiffServ edge router and marked according to network state. To achieve such dynamic QoS adaptation for multimedia applications, we have implemented and evaluated a policy-based management system. Performance evaluation shows that multimedia applications adapt better to network conditions using our approach.     

An adaptive multimedia streaming dissemination system for vehicular networks

An adaptive seamless streaming dissemination system for vehicular networks is presented in this work. An adaptive streaming system is established at each local server to prefetch and buffer stream data. The adaptive streaming system computes the parts of prefetched stream data for each user and stores them temporarily at the local server, based on current situation of the users and the environments where they are located. Thus, users can download the prefetched stream data from the local servers instead of from the Internet directly, meaning that the video playing problem caused by network congestion can be avoided. Several techniques such as stream data prefetching, stream data forwarding, and adaptive dynamic decoding were utilized for enhancing the adaptability of different users and environments and achieving the best transmission efficiency. Fuzzy logic inference systems are utilized to determine if a roadside base station or a vehicle can be chosen to transfer stream data for users. Considering the uneven deployment of BSs and vehicles, a bandwidth reservation mechanism for premium users was proposed to ensure the QoS of the stream data premium users received. A series of simulations were conducted, with the experimental results verifying the effectiveness and feasibility of the proposed work.     

Effective client-driven three-level rate adaptation (TLRA) approach for adaptive HTTP streaming

Multimedia streaming allows consumers to view multimedia content anywhere. However, quality of service is a major concern amid heightened levels of network traffic caused by increasing user demand. Accordingly, media streaming technology is adopting a new paradigm: adaptive HTTP streaming (AHS). AHS is widely used for real-time streaming content delivery in the Internet environment. In streaming, selection of appropriate bitrate is crucial for adapting media rate to network variations and client processing capabilities while ensuring optimal service for the consumer. We evaluate a proposed client-driven three-level optimized rate adaptation algorithm for adaptive HTTP media streaming. In the first stage, the algorithm chooses a suitable starting bitrate close to the available channel capacity. Next, the algorithm monitors the client parameters in real time, precisely detecting network variations and choosing a likely available bit representation for the next download segment. The algorithm controls and minimizes the effects of buffer stalls and overflow resulting from the brief network variations occurring between consecutive segments. The proposed algorithm is implemented in Dynamic Adaptive Streaming over HTTP (DASH) player and its performance compared to that of commercially available Gstreamer-based HTTP Live Streaming (HLS) and DASH players which use conventional segment fetch time–based adaptation and throughput-based adaptation algorithms respectively. This evaluation uses a real-time cloud server client and test bed streaming setup. The resulting analysis shows that the client-driven three-level rate adaptation (TLRA) approach allows adaptive streaming clients to maximize use of end-to-end network capacity, delivering an ideal user experience by precisely predicting network variations and rapidly adapting to available bandwidth, minimizing rebuffering events and bitrate level changes.     

具有优先级特征的多媒体流的资源管理

本文研究了具有优先级特征的分布式多媒体流的资源管理，提出了一种基于节优先级的资源管理的设计方法，包括资源管理机制，资源管理策略，服务质量（ＱｏＳ）协商调整算法和高优先级节枪占算法，它能够保证稳定的具估优先级特征的多媒体流，能够极大地调度并发的多媒体流，特别是在系统资源不足时，能够最大限度地调度高优先级的多媒体流，并能保证各多媒体流量了的ＱｏＳ。     

An integrated admission control scheme for the delivery of streaming media

Quality of service (QoS) assurance is a major concern in media-on-demand (MoD) systems. Admission control is one of the most important issues that need to be addressed for QoS assurance. Also, smoothing is a basic technique for the media server to improve its bandwidth and buffer utilization. However, existing approaches cannot achieve the best resource utilization because (1) they cannot fully utilize the time-varying buffer spaces available at both server and client sides due to the separation of admission control and smoothing processes, and (2) the computing time is unacceptable for media of long duration. In this paper, we formulate the admission control as an integer programming problem and propose several heuristic methods for solving the problem. Specially, we introduce an efficient scheme, called batched admission (BA) scheme, which integrates admission control, transmission rate smoothing, and batching, to achieve best resource utilization with guaranteed QoS. Experimental studies show that the BA scheme outperforms existing approaches.     

基于模糊控制的ATM网络VBR视频传输平滑策略

VBR视频传输的突发性是影响ATM网络服务质量的关键因素,文中通过模糊控制方法对视频传输流量的阈值进行监控,实现了接入的平滑并可动态调整传输速度。文中以传输流量及这发级作为评价指标进行分析,结果表明VBR视频传输自适应平滑策略的实效性。     

A multi-layer experimental study of multimedia and QoS communication in wireless mesh networks

A significant issue in Mesh networks is to support multimedia transmissions while providing Quality of Service (QoS) guarantees to mobile users. For real-time multimedia streaming, unstable throughput or insufficient bandwidth will incur unexpected delay or jitter, and it remains difficult to provide comprehensive service guarantees for a wireless mesh environment. In this paper, we target the problem of providing multimedia QoS in wireless mesh networks. We design and implement a campus test-bed for supporting multimedia traffic in mobile wireless mesh networks, and investigate in detail some possible improvements on a number of layers to enable multimedia transmission over wireless networks with QoS support. We first study a number of improvements of some existing routing protocols to support multimedia transmissions. Some new admission control and rate control mechanisms are studied and their performance gains are verified in our experiments. In our new cross-layer adaptive rate control (CLARC) mechanism, we adaptively change the video encoder’s output bit rate based on the available network bandwidth to improve the quality of the received video. We also design a mobile gateway protocol to connect the MANET to Internet and a wireless LAN management protocol to automatically manage WLAN to provide some QoS.     

Design and Implementation of a Streaming System for MPEG-1 Videos

A new streaming application has been developed for the Internet environment. The system has client-server structure together with multithreaded architecture and pipelining. RTP protocol is used to transmit packets belonging to MPEG videos. RTCP protocol collects transmission statistics. System is adaptive in the sense that it reacts to dynamic network conditions. A feedback mechanism controls both the frame interval and frame rate depending on the frame-loss statistics and buffered video level at the client. A flow control module at the client side controls buffer underflows and overflows. Performance results of the implementation are reported and discussed. The performance of the proposed buffering strategy is compared with other proposed methods from the literature. The comparisons showed that the proposed strategy is more robust than other methods.     

Efficient Selective Frame Discard Algorithms for Stored Video Delivery Across Resource Constrained Networks

Video delivery from a server to a client across a network is an important component of many multimedia applications. While delivering a video stream across a resource constrained network, loss of frames may be unavoidable. Under such circumstances, it is desirable to find a server transmission schedule that can efficiently utilize the network resources while maximizing the perceived quality-of-service (QoS) at the client. To address this issue, in this paper we introduce the notion ofselective frame discard at the server and formulate the optimal selective frame discard problem using a QoS-based cost function. Given network bandwidth and client buffer constraints, we develop an O (N log N) algorithm to find the minimum number of frames that must be discarded in order to meet these constraints. The correctness of the algorithm is also formally established. We present a dynamic programming based algorithm for solving the problem of optimal selective frame discard. Since the computational complexity of the optimal algorithm is prohibitively high in general, we also develop several efficient heuristic algorithms for selective frame discard. These algorithms are evaluated using JPEG and MPEG video traces.     

下一代网络多媒体流自适应QoS研究

下一代网络是一个支持话音、数据和视频的多媒体网络。随着网络的发展,如何保障下一代网络多媒体流QoS是一个重要的研究课题。为保证多媒体流QoS,系统必须分配足够的CPU、I/O、网络带宽资源,在多媒体传输过程中,这些资源是动态变化的。论文提出一种带优先级的自适应节调节方法,采用AIMD调节技术调节数据帧长度,自适应调节QoS,实验表明,该方法保证了NGN多媒体流的QoS,也使得网络资源得到充分利用。     

Protocol considerations for a prefix-caching proxy for multimedia streams

The increasing popularity of multimedia streaming applications introduces new challenges in content distribution. Web-initiated multimedia streams typically experience high start-up delay, due to large protocol overheads and the poor delay, throughput, and loss properties of the Internet. Internet service providers can improve performance by caching the initial segment (the prefix) of popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while simultaneously requesting the remainder of the stream from the server. This paper analyzes the challenges of realizing a prefix-caching service in the context of the IETF's Real-Time Streaming Protocol (RTSP), a multimedia streaming protocol that derives from HTTP. We describe how to exploit existing RTSP features, such as the Range header, and how to avoid several round-trip delays by caching protocol information at the proxy. Based on our experiences, we propose extensions to RTSP that would ease the development of new multimedia proxy services. In addition, we discuss how caching the partial contents of multimedia streams introduces new challenges in cache coherency and feedback control. Then, we briefly present our preliminary implementation of prefix caching on a Linux-based PC, and describe how the proxy interoperates with the RealNetworks server and client.     

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.