基于IIS平滑流式处理的移动流媒体系统

针对当前流媒体传输协议在无线网络中的不足,采用IIS平滑流式处理技术设计了基于服务器/客户端模式的移动流媒体系统。详细讨地论了微软的IIS平滑流式处理协议,采用该协议传输音/视频数据,搭建了基于IIS7 Web服务器的流媒体服务,设计了一款基于Windows Phone 7智能手机的流媒体播放器客户端。通过在WiFi网络环境下测试该系统,在直播和点播两种播放模式下,客户端播放的视频画面清晰流畅。通过仿真无线网络带宽的变化,验证了IIS平滑流式处理根据客户端的可用带宽实时调整传送到客户端视频流的质量的特性。     

An Efficient Transmission Scheme of MPEG2‐TS over RTP for a Hybrid DMB System

Hybrid digital multimedia broadcasting (DMB) is a next‐generation mobile TV system that combines broadcasting and wireless communication networks and can provide various high‐quality multimedia services. However, if a system adheres to the current standard of transmitting the DMB content in the form of MPEG2‐TS through wireless networks, it results in a burden on the network due to low transmission efficiency. The reasons for the low transmission efficiency are as follows. First, due to its constant bitrate characteristic, DMB MPEG2‐TS includes a considerable amount of needless information, such as NULL packets and stuffing bytes. Second, due to the inflexibility of the Real‐time Transport Protocol (RTP) standard, one cannot fully utilize the maximum transmission unit of the network when converting MPEG2‐TS to RTP stream for transmission. This paper proposes a new transmission scheme that resolves these problems. Experiment results show that the proposed scheme improves data bitrate transmission efficiency by 8% to 36%, compared to the standard scheme, in the streaming of various real‐DMB contents.     

Cloud‐Based Gaming Service Platform Supporting Multiple Devices

To implement a cloud game service platform supporting multiple users and devices based on real‐time streaming, there are many technical needs, including game screen and sound capturing, audio/video encoding in real time created by a high‐performance server‐generated game screen, and real‐time streaming to client devices, such as low‐cost PCs, smart devices, and set‐top boxes. We therefore present a game service platform for the running and management of the game screen, as well as running the sound on the server, in which the captured and encoded game screen and sound separately provide client devices through real‐time streaming. The proposed platform offers Web‐based services that allow game play on smaller end devices without requiring the games to be installed locally.     

基于ZigBee技术的智能家居系统

通常把智能家居定义为利用计算机、网络和自动化控制技术,通过一个固定的平台将与家居生活有关的设备集成到一个系统中。系统通过无线通信技术进行各个节点和平台之间的通信,类似于一个家庭的局域网。文中设计了一种基于ZigBee技术的智能家居系统,除了能够利用ZigBee无线通信技术对室内的温湿度信息的实时感知,开关控制外,本系统还设计了视频流服务器,能够将室内的视频信息通过网络传输到远程客户端,如手机或者PC Web客户端。另外,此视频流服务器具有较强的扩展功能,能够进行各种常用协议的传输以及支持各种常见视频设备的数据采集。     

Network Bandwidth Requirements for Scalable On-Demand Streaming

Previously proposed streaming protocols using broadcast or multicast are able to deliver multimedia files on-demand with required server bandwidth that grows much slower than linearly with request rate, or with the inverse of client start-up delay. The same efficiencies can be achieved for network bandwidth if delivery is over a true broadcast channel. This paper considers the required network bandwidth for on-demand streaming over multicast delivery trees. We consider both simple canonical delivery trees, and more complex cases in which delivery trees are constructed using both existing and new algorithms for randomly generated network topologies and client site locations. Results in this paper quantify the potential savings from use of multicast trees that are configured to minimize network bandwidth rather than the latency to the content server. Further, we determine the network bandwidth usage of particular immediate service and periodic broadcast on-demand streaming protocols. The periodic broadcast protocol is able to simultaneously achieve close to the minimum possible network and server bandwidth usage.     

Transmission algorithm for video streaming over cellular networks

2.5G and 3G cellular networks are becoming more widespread and the need for value added services increases rapidly. One of the key services that operators seek to provide is streaming of rich multimedia content. However, network characteristics make the use of streaming applications very difficult with an unacceptable quality of service (QoS). The 3GPP standardization body has standardized streaming services that will benefit operators and users. There is a need for a mechanism that will enable a good quality multimedia streaming that uses the 3GPP standard. This paper describes an adaptive streaming algorithm that uses the 3GPP standard. It improves significantly the QoS in varying network conditions while monitoring its performance using queueing methodologies. The algorithm utilizes the available buffers on the route of the streaming data in a unique way that guarantees high QoS. The system is analytically modeled: the streaming server, the cellular network and the cellular client are modeled as cascaded buffers and the data is sequentially streamed between them. The proposed Adaptive streaming algorithm (ASA) controls these buffers’ occupancy levels by controlling the transmission and the encoding rates of the streaming server to achieve high QoS for the streaming. It overcomes the inherent fluctuations of the network bandwidth. The algorithm was tested on General Packet Radio Service (GPRS), Enhanced Data rates for GSM Evolution (EDGE) and Universal Mobile Telecommunication System (UMTS) networks. The results showed substantial improvements over other standard streaming methods used today.     

Deployment of IP multimedia streaming services in third-generation mobile networks

In this article, an end-to-end quality of service framework for streaming services in 3G mobile networks is considered. Under this scenario, the interaction between UMTS and IETF's protocols and mechanisms for a streaming session is analyzed. By signaling flowcharts, it is shown that both groups of protocols and mechanisms can co-operate to provide seamless end-to-end real-time services. Specifically, the article proposes to make the IP multimedia subsystem aware of the real time streaming protocol, in order to extend its control from SIP to RTSP-based services, such as multimedia streaming services. Supported by this proposed framework, provisioning of audio streaming services over 3G mobile networks is also outlined.     

Real-Time Streaming Relay Mechanism for P2P Conferences on Hierarchical Overlay Networks

A peer-to-peer (P2P) multimedia conferencing service is operating that users share their resources to each other on the Internet. It can solve the problem in the centralized conferencing architecture, such as the centralized loading, single point error, and expensive infrastructure. However, P2P networks have the problem that a peer has a difference between the physical location and logical location in the overlay network. In the viewpoint of P2P networks, the nearest conference resource may be far away geographically. The P2P-session initiation protocol (P2P-SIP) multimedia conference is to construct an application-based logical multicast network efficiently according to physical network information. Thus, this paper proposes a real-time streaming relay mechanism for P2P conferences on hierarchical overlay networks. The real-time streaming relay mechanism can improve the transportation efficiency of conferencing stream exchange well based on the application-layer multicast (ALM) structure and the hierarchical overlay networks.     

A Framework for MPEG‐4 Contents Delivery over DMB

Digital Multimedia Broadcasting (DMB) is an upcoming standard in Korea used to provide mobile multimedia broadcasting service based on the Eureka‐147 Digital Audio Broadcasting (DAB) system. The current dominant multimedia coding standard, MPEG‐4, is foreseen to play an important role in forthcoming DMB services. However, the current approaches for transporting MPEG‐4 content over DMB networks are not optimized. To address this issue we propose a novel MPEG‐4 stream multiplexer, called M4SMux, which provides better stream multiplexing and delivery over DMB networks. M4SMux features an MPEG‐4 elementary‐stream interleaving mechanism that reduces the multiplexing overhead and a multiplex configuration mechanism that utilizes M4SLinkTable for easy content access. In addition, we propose an error correction method which enhances transport efficiency.     

基于流媒体技术的移动视频服务器的设计

针对3G无线互联网技术的迅猛发展和人们生活、工作中对手机实时视频的需求,应用MPEG-4视频编码标准和RTP/RTCP流媒体传输协议为基础,设计了一个流媒体传输的解决方案的移动实时视频服务器,服务器完成了对实时视频数据的采集和压缩,对实时源Filter传输和处理过程和服务器与客户端RTP/RTCP通信过程,实现服务器视频实时传输的编码设计和对视频传输的测试.     

LWIP在嵌入式视频服务器中的移植与性能测试

主要介绍一种开源的轻型TCP/IP协议栈LWIP(轻型Internet协议)及其在基于ADI-BF533的嵌入式视频服务器中的移植与性能测试.在分析LWIP特点的基础上选择直接调用LWIP专门的内部回调函数来实现网络控制和传输,而不是把LWIP移植到嵌入式操作系统上.试验结果表明,在10 Mbit/s局域网环境下服务器端至少可以同时接受8个客户端的请求,并向其实时传输MPEG4码流.     

Media Streaming With Network Diversity

Today's packet networks including the Internet offer an intrinsic diversity for media distribution in terms of available network paths and servers or information sources. Novel communication infrastructures such as ad hoc or wireless mesh networks use network diversity to extend their reach at low cost. Diversity can bring interesting benefits in supporting resource greedy applications such as media streaming services, by aggregation of bandwidth and computing resources. Typically, overlay network architectures compensate for lack of quality-of-service guarantees in the network by introducing redundancy in the media delivery system through network diversity. They can support efficient multimedia services when routing, coding, and scheduling algorithms are able to adapt to both the media information and the dynamic network status. This paper presents an overview of the distributed streaming solutions that profit from network diversity in order to improve the quality of multimedia applications. We discuss the coding techniques used for adaptive and flexible media streaming with network diversity. We describe the problem of media streaming with path diversity and focus on routing, path computation, and packet scheduling problems in multipath networks. Then, the advantages of server or source peer diversity in collaborative streaming solutions are discussed. Lastly, we present an overview of wireless mesh networks and focus on the typical constraints imposed by these novel communication models on media streaming with network diversity.     

VIRAL: coupling congestion control with fair video quality metric

Video streaming is often carried out by congestion controlled transport protocols to preserve network sustainability. However, the success of the growth of such non-live video flows is linked to the user quality of experience. Thus, one possible solution is to deploy complex quality of service systems inside the core network. Another possibility would be to keep the end-to-end principle while making aware transport protocols of video quality rather than throughput. The objective of this article is to investigate the latter by proposing a novel transport mechanism which targets video quality fairness among video flows. Our proposal, called VIRAL for virtual rate-quality curve, allows congestion controlled transport protocols to provide fairness in terms of both throughput and video quality. VIRAL is compliant with any rate-based congestion control mechanisms that enable a smooth sending rate for multimedia applications. Implemented inside TFRC a TCP-friendly protocol, we show that VIRAL enables both intra-fairness between video flows in terms of video quality and inter-fairness in terms of throughput between TCP and video flows.     

TelecomI+D06: Automatic recovering of RTSP sessions in mobile telephones using JADE-LEAP

Streaming is a technique used to transmit information over the network so that its issuing from the server, its communication along the network and its downloading and processing on the client overlapp, and moreover it must not be saved in the client memory. This technique is very effective for transmitting multimedia information (video) and is very suitable for mobile phones because there is no need to store the video in its memory that it is limited (data storage is still quite limited in the last generation mobile phones indeed). While the transmission speed of wireless networks has increased significantly, this technique is still very effective because it hides the latency of the network. On the other hand, the various wireless communication technologies used by mobile phones today (mainly Wireless Fidelity (WiFi), Bluetooth and 3G) are very likely to suffer physical intermittent disconnections making them miss the video streaming session every time a mobile phone loses network coverage for a certain time and forces him to reconnect manually, with the consequent loss of effectiveness of the streaming technique (that is lost due to the effectiveness of concealment of network latency). This work shows an automatic recovery solution for lost video stream and missed session by creating a new mechanism based on proxies implemented with software agents of the Java platform software DEvelopment framework-Lightweight Extensible Agent Platform (JADE-LEAP). As far as our knowledge, this is the first time the efficient use of this platform for streaming video on mobile phones is tested. In addition, the experimental results obtained are very promising for its effectiveness and relevance to the practical case studies we've tested.     

SCTP故障恢复机制改进

流传输控制协议（SCTP协议）是一种新的Internet传输层协议,Internet工作组设计SCTP的最初目的是在IP网络上传输PSTN信令消息,而且还能够充当通用传输协议。与传统的传输协议相比,SCTP协议允许在一个单一的连接中传输多个数据子流,这种功能可以大大改善高损耗的环境中多媒体流延迟问题,同时SCTP协议支持多宿功能,能够为网络提供冗余备份功能。对SCTP故障恢复机制进行了改进,充分利用SCTP多宿特性为移动Ad hoc网络提供可靠性保障,使其能够适应移动Ad hoc网络的特点,仿真结果表明：该功能极大地减少了故障恢复时间,提高了其在移动Ad hoc网络中的性能。     

HTTP自适应流媒体直播系统中的用户体验质量优化

分析HTTP自适应流媒体直播系统中对终端用户体验质量(QoE)产生影响的各类因素及其相互之间的作用关系,对基于服务器端、网络传输以及客户端的QoE优化策略进行总结。认为HTTP自适应流媒体直播系统的QoE优化重点在于降低延时,提出结合网络层和应用层影响因素来降低时延并提升用户QoE的建议。     

Segment‐aware dynamic routing for DASH flows over software‐defined networks

Most of the video streaming applications running over the Internet send video data over HTTP and provide an architecture for video clients to adapt video quality during streaming. In HTTP adaptive streaming, a raw video is encoded at various qualities, each encoded video file is divided into small segments, and the clients may change the segment quality by sending requests for segments having different qualities over time. MPEG has standardized dynamic adaptive streaming over HTTP (MPEG‐DASH) due to this tendency. In this work, we focus on DASH over software‐defined networks (SDN), and we dynamically reroute DASH flows by considering the current network capacity, available bandwidth of the paths, and bitrate of the segments in order to provide high quality of experience (QoE) and fairness among DASH clients. Simulations performed under various network conditions show that the proposed study provides higher QoE and fairness compared with the max‐flow routing approach.     

实现视音频信息服务器传输的IP网络

运用用户数据报协议 (UDP) ,实时传输控制协议 (RTP/RTCP)和H .32 3协议 ,设计实现解决视音频信息的实时传输服务器。设计中对视音频信息的打包格式采用MPEG 1的格式将信息打包后作为RTP报文的应用数据加上一个RTP头 ,给信息包再加上一个UDP头和IP头 ,组成一个完整的信息包后 ,采用非连接的UDP用户数据报的方式 ,将它发送出去 ,实现组播和多组会议 ,采用MPEG 1的数据压缩技术 ,可以较高的传输速率在宽带IP网络中的传输 ,实时性比较好     

SyncML数据同步协议研究

SyncML(Synchronization Markup Language)数据同步规范是数据同步的国际标准,它支持多种传输协议,能在多种网络和设备上进行数据同步。文中主要介绍了SyncML规范的结构,SyncML同步协议,分析了同步过程中客户端和服务器的会话,并提出了协议的改进意见。     

Adaptive and reliable video transmission over UMTS for enhanced performance

This paper proposes a mechanism for the congestion control for video transmission over universal mobile telecommunications system (UMTS). Our scheme is applied when the mobile user experiences real‐time multimedia content and adopts the theory of a widely accepted rate control method in wired networks, namely equation‐based rate control. In this approach, the transmission rate of the multimedia data is determined as a function of the packet loss rate, the round trip time and the packet size and the server explicitly adjusts its sending rate as a function of these parameters. Furthermore, we examine the performance of the UMTS for real‐time video transmission using real‐time protocols. Through a number of experiments, we measure performance parameters such as end‐to‐end delay, delay in radio access network, delay jitter and throughput in the wireless link. Copyright © 2006 John Wiley & Sons, Ltd.