Cost-based cache replacement and server selection for multimedia proxy across wireless Internet

Multimedia proxy plays an important role in multimedia streaming over wireless Internet. Since wireless network exhibits different characteristics from the Internet, multimedia proxy caching over wireless Internet faces additional challenges. In this paper, we present a study of cache replacement for a single server and server selection for multiple servers across wireless Internet. By considering multiple objectives of multimedia proxy, we design a unified cost metric to measure proxy performance in wireless Internet. Based on the defined unified cost metric, we propose a novel replacement algorithm for single-server and a new server-selection policy for multiple servers to improve the end-to-end performance such as throughput, media quality, and start-up latency. To effectively handle errors occurred on wireless link, channel-adaptive unequal error protection is deployed according to distinct quality of service requirements of layered or scalable media. Simulation results demonstrate that our approaches achieve significantly better performance than the known cache-replacement algorithms and sever selection schemes, respectively.     

An end-to-end robust approach for scalable video over the internet

With the growth of the Internet and abundant network resources, video streaming is becoming one of the increasingly important Internet applications. However, the current IP-based network provides only a single class best effort service. Video packet can be regarded as a packet loss by the video decoder either due to network congestion or due to exceeding the maximum delay threshold. It remains an open challenging task as to how to cope with the packet loss in the video streaming over the Int…     

Quality adaptive end-to-end packet scheduling to avoid playout interruptions in Internet video streaming systems

In Internet multimedia streaming, the quality of the delivered media can be adapted to the Quality of Service provided by the underlying network, thanks to encoding algorithms. These allow a fine grained enhancement of a low quality base layer at streaming time. The main objective that should be satisfied in such systems is to avoid the starvation of the decoding process and consequent playout interruptions. In this work, we tackle the problem using a control theoretic approach. In particular, we design and implement the novel end-to-end Quality Adaptive Scheduler for properly distributing the network available bandwidth among base and enhancement layers. The developed solution can be adopted in many contexts given that it has been designed without assumptions on the delivered media nor on the protocol stack. Anyway, to test its effectiveness, we have casted it in a H.264/AVC SVC based video streaming architecture for unicast Internet applications. The performance of the scheduler has been experimentally evaluated in both a controlled testbed and several “wild” Internet scenarios, including also UMTS and satellite radio links. Results have clearly demonstrated that our Quality Adaptive Scheduler is able to significantly improve the performance of the video streaming system in all operative conditions.     

Failure-Aware, Open-Loop, Adaptive Video Streaming With Packet-Level Optimized Redundancy

A plethora of coding and streaming mechanisms have been proposed for real-time multimedia transmission over the Internet. However, most proposed mechanisms rely only on global (e.g. based on end-to-end measurements), delayed (at least by the round-trip-time), or statistical (often based on simplistic network models) information available about the network state. Based on recently-proposed state-of-the-art open-loop video coding schemes, we propose a new integrated streaming and routing framework for robust and efficient video transmission over networks exhibiting path failures. Our approach explicitly takes into account the network dynamics, path diversity, and the modeled video distortion at the receiver side to optimize the packet redundancy and scheduling. In the derived framework, multimedia streams can be adapted dynamically at the video server based on instantaneous routing-layer information or failure-modeling statistics. The performance of our integrated application and network-layer method is simulated against equivalent approaches that are not optimized based on routing-layer feedback and distortion modeling, and the obtained gains in video quality are quantified     

端到端的流媒体传输控制技术研究综述

随着多媒体应用的发展,因特网上的流媒体传输技术已成为研究热点。在因特网上传输音频或视频流需要有带宽、延迟、丢包率等诸多的QoS要求,但当前的因特网并不提供任何QoS保证,这对流媒体在因特网上的传输提出巨大挑战。该文从端到端的传输控制技术角度入手,给出了一个因特网上流媒体传输的总体框架,然后依此框架为线索,对流媒体传输所必需的协议栈、拥塞控制、自适应速率编码、速率整形、差错控制等技术的研究进展进行了概括总结并进行了对比,同时提出了进一步的研究建议。     

Cross-Layer Error Control for Multimedia Streaming in Wireless/Wireline Packet Networks

In this paper, we propose a cross-layer error control framework for robust and low delay multimedia streaming in tandem-connected IEEE 802.11 wireless LANs and the Internet. For this network configuration, we model the end-to-end delay and packet loss rate as a function of the automatic repeat request (ARQ) and forward error correction (FEC) error control mechanisms that are employed at the application and wireless link layers. The analytical model is used as the basis of a delay-constrained error control algorithm that adapts the protection level at the application and link layers so that the end-to-end packet loss rate is minimized. With extensive simulations, we validate the efficiency of the proposed cross-layer error control methodology for delay-sensitive pre-compressed video streaming.      

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.     

Modeling and generating realistic streaming media server workloads

Currently, Internet hosting centers and content distribution networks leverage statistical multiplexing to meet the performance requirements of a number of competing hosted network services. Developing efficient resource allocation mechanisms for such services requires an understanding of both the short-term and long-term behavior of client access patterns to these competing services. At the same time, streaming media services are becoming increasingly popular, presenting new challenges for designers of shared hosting services. These new challenges result from fundamentally new characteristics of streaming media relative to traditional web objects, principally different client access patterns and significantly larger computational and bandwidth overhead associated with a streaming request. To understand the characteristics of these new workloads we use two long-term traces of streaming media services to develop MediSyn, a publicly available streaming media workload generator. In summary, this paper makes the following contributions: (i) we propose a framework for modeling long-term behavior of network services by capturing the process of file introduction, non-stationary popularity of media accesses, file duration, encoding bit rate, and session duration. (ii) We propose a variety of practical models based on the study of the two workloads. (iii) We develop an open-source synthetic streaming service workload generator to demonstrate the capability of our framework to capture the models.     

rStream: Resilient and Optimal Peer-to-Peer Streaming with Rateless Codes

Due to the lack of stability and reliability in peer-to-peer networks, multimedia streaming over peer-to-peer networks represents several fundamental engineering challenges. First, multimedia streaming sessions need to be resilient to volatile network dynamics and node departures that are characteristic in peer-to-peer networks. Second, they need to take full advantage of the existing bandwidth capacities by minimizing the delivery of redundant content and the need for content reconciliation among peers during streaming. Finally, streaming peers need to be optimally selected to construct high-quality streaming topologies so that end-to-end latencies are taken into consideration. The original contributions of this paper are twofold. First, we propose to use a recent coding technique, referred to as rateless codes, to code the multimedia bitstreams before they are transmitted over peer-to-peer links. The use of rateless codes eliminates the requirements of content reconciliation, as well as the risks of delivering redundant content over the network. Rateless codes also help the streaming sessions to adapt to volatile network dynamics. Second, we minimize end-to-end latencies in streaming sessions by optimizing toward a latency-related objective in a linear optimization problem, the solution to which can be efficiently derived in a decentralized and iterative fashion. The validity and effectiveness of our new contributions are demonstrated in extensive experiments in emulated realistic peer-to-peer environments with our rStream implementation.     

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.     

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.     

流媒体对象的缓存管理策略

基于流媒体服务的代理技术是流媒体研究领域中的重要课题.随着流媒体技术在Internet和无线网络环境中的高速发展,对流媒体代理服务器的研究也正在逐步深入.本文主要讨论通过代理技术改善媒体的服务质量,降低媒体的传输延迟以及减轻网络负载.在Internet环境下,对流媒体代理服务器的研究集中于流媒体的访问特性、缓存替换算法,构建和实现一个流媒体代理服务器是对流媒体代理技术研究的基础.     

Enabling adaptive live streaming in P2P multipath networks

Live Internet streaming can be regarded as a major current multimedia delivery mode. Efficient delivery under changing network conditions is a severe challenge in the design of live streaming systems. This study analyzes the key considerations and factors influencing live stream quality during system operations, and attempts to improve present P2P (peer-to-peer) live streaming systems by allowing users to enjoy high quality of service under the limitations of network resources. The proposed R-D (Rate-Distortion) optimized-dynamic-nodes-join algorithm is based on multipath streaming concept and receiver-driven approach. This distributed algorithm enables the system to evaluate the current network status in order to optimize the end-to-end distortion of P2P networks and stay in the optimal status. Experiment results of this study demonstrate the effectiveness of the proposed approach.     

流媒体技术分析及其在新闻网站的应用

随着互联网的普及和多媒体技术的发展,流媒体依靠其强大的技术优势迅速渗透到社会生活的各个领域。在国际上,对webcasting（或webcaster）的研究已经成为网络传播研究的一项重要课题。本文以流媒体的概念为切入点,深入分析了流媒体系统及其关键技术,研究了流媒体技术在新闻网站中的应用。     

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…     

Media Flow Rate Allocation in Multipath Networks

We address the problem of joint path selection and source rate allocation in order to optimize the media specific quality of service in streaming of stored video sequences on multipath networks. An optimization problem is proposed in order to minimize the end-to-end distortion, which depends on video sequence dependent parameters, and network properties. An in-depth analysis of the media distortion characteristics allows us to define a low complexity algorithm for an optimal flow rate allocation in multipath network scenarios. In particular, we show that a greedy allocation of rate along paths with increasing error probability leads to an optimal solution. We argue that a network path shall not be chosen for transmission, unless all other available paths with lower error probability have been chosen. Moreover, the chosen paths should be used at their maximum available end-to-end bandwidth. Simulation results show that the optimal flow rate allocation carefully adapts the total streaming rate and the number of chosen paths, to the end-to-end transmission error probability. In many scenarios, the optimal rate allocation provides more than 20% improvement in received video quality, compared to heuristic-based algorithms. This motivates its use in multipath networks, where it optimizes media specific quality of service, and simultaneously saves network resources at the price of a very low computational complexity.     

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。     

流媒体技术及其在网络教学中的应用

Internet的迅猛发展和普及为流媒体业务发展提供了强大的市场动力,流媒体业务正变得日益流行.流媒体技术广泛用于多媒体新闻发布、在线直播、网络广告、电子商务、视频点播(VOD)、远程教育、远程医疗、网络电台、实时视频会议等互联网信息服务的方方面面.特别是在教育领域,流媒体技术已经成为通过网络进行多媒体交互教学的最基本的技术之一.随着网络及流媒体技术的发展,越来越多的远程教育网站开始采用流媒体作为主要的网络教学方式.正因为这个原因,利用流媒体技术为网络教学服务,提高网络课堂的质量,增强网络课堂的交互性,必将成为今后远程教育发展的一个重要课题.本文主要从流媒体技术及其在网络教学中的应用方面进行简单的探讨.     

Dynamic QoS queuing control mechanism for multimedia differentiated services

An essential goal of communication networks is to provide multimedia services with QoS streaming. A properly designed multimedia QoS system must reserve requested resources according to user QoS requirements and the available network resources. However, the static resource allocation among priority queues in DiffServ networks leads to insufficient resource usage when a burst occurs in one priority queue while other queues starve. This study presents a User-Oriented QoS Streaming System to achieve perceptible satisfaction based on novel streaming and media differentiation policies in DiffServ networks. This study also proposes that the Dynamic QoS Queue Mapping (DQ²M) mechanism dynamically control queue scheduling by adaptively maximizing the utilization of queues and network resources according to the soft states of the DiffServ network. Evaluation results indicate that the proposed DQ²M algorithm can improve the fairness and efficiency of resource utilization for low-priority queues.     

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。