3G网络中流媒体业务的发展策略

首先介绍了3G网络中的一些流媒体业务，接着分析了影响移动流媒体业务开展的几个因素，最后对流媒体业务在3G网络中的发展策略进行了分析，并给出网络建设的几种方案。     

流媒体技术及其在3G移动通信中的应用

随着3G移动通信的发展,移动流媒体业务成为新的移动增值业务热点。本文首先介绍了流媒体的出现和发展,然后介绍了流媒体业务及其应用,最后介绍了流媒体业务发展制约因素及解决办法。     

3G移动流媒体业务平台部署方案探讨

从流媒体业务端到端时延、视频质量、用户的安全访问控制等方面对3G移动流媒体业务平台部署方案进行了探讨,这对于构建以用户为中心的新型3G移动流媒体平台有一定的参考价值。     

3G前夜——移动流媒体春光乍泻

随着全球移动通信网络从2．5G向3G过渡，无线互联网带宽的变化让移动通信业务发展趋势产生重大变化，由以话音业务为主的初始形式到增值业务全面发展的局面。业内分析人士预计．随着3G业务的正式展开，未来三年移动流媒体视频业务将会有更大发展．移动流媒体业务的用户数将会不断增加．预计到2007年，移动流媒体视频用户在流媒体总用户中将占20％以上。虽然国内3G牌照尚未发放．     

中兴通讯3G流媒体业务解决方案

首先对流媒体业务的概况进行了介绍,接着介绍了其关键技术和业务特征,包括业务模式和网络组成等,最后给出了中兴通讯3G流媒体业务解决方案。     

流媒体与移动流媒体技术的研究

流媒体在移动通信中的应用将是3G和3G后移动通信增值的主要方式。文章介绍了流媒体与移动流媒体的基本原理,阐述了移动流媒体的主要实现技术——高容错性、H．264视频压缩标准和传输协议,简要分析了移动流媒体业务发展的制约因素。     

简述流媒体在3G平台的技术及应用

简单介绍了3G网络及流媒体技术,阐述了流媒体在3G平台上的三种典型业务模式,对流媒体的具体应用及在实际应用中存在的问题做了说明。最后对流媒体的未来市场前景进行展望。     

3G流媒体业务的QoS及解决之道

移动流媒体技术被认为是移动增值业务的一个热点,它开创了无线通信与互联网、视频融合的新时代。随着3G网络基础设施的完善、移动终端功能的增强和媒体内容的丰富,流媒体业务必将会成为未来移动通信业务新的增长点。运营商在开展流媒体业务中,最为关心的问题是服务质量(QoS)。流媒体的QoS问题在有线网络中已经得到了广泛的研究并提出了很好的解决方案,然而无线网络中,用户的移动性、传输信道的恶劣等使得移动流媒体的QoS保证更加复杂。本文就3G流媒体业务的QoS问题进行了分析,并给出了解决方法。     

3G流媒体业务发展特点及市场规模评估

1 前言 不论从国外已推出3G服务运营商的业务发展,还是国 内对3G业务的展望,移动流媒体业务被一致推崇为3G时代 最具代表性的核心增值业务。3G流媒体业务虽然起源和演 化于2G、2．5G移动数据业务,但由于网络技术的进步、业 务表现方式的多彩多样、受众感受的升华,其发展中的特点     

基于3G网络的流媒体直播技术系统的设计与实现

分析当前三网融合背景下新闻业务的功能需求,提出基于3G网络的流媒体直播技术系统的系统架构,结合具体需求构建基于3G网络的流媒体新闻业务工作流程,实现新闻3G直播和SDI直播输出。     

基于细粒度调整的同步播放控制算法

针对网络时延和时延抖动引起缓存波动而造成的流媒体播放不连续,基于接收端播放速率调整,提出一种播放速率自适应调整的播放缓存控制算法。算法依据缓存区的占用水平,通过构造控制序列,动态、细粒度地调整媒体的播放速率。在校园网环境下的模拟结果表明,该算法能够有效地减少缓存上溢或下溢引起的播放跳跃或停顿,从而实现流媒体的平滑播放。     

OFDM系统原理及仿真实现

多媒体系统必须能够限制时延抖动的变化,通过在播放端设置缓冲区以抵消时延抖动,从而保证媒体内及媒体间的同步。根据多源流媒体的分发特点,文章设计了一种分组的媒体缓冲区大小的设置方案,该方案在保证流媒体同步的基础上,巧妙地解决了流媒体不同码率的缓冲区大小设置问题。     

常见3G流媒体业务投诉的分析与排查

3G流媒体业务,是移动通信发展的方向.随着3G多媒体手机的大规模上市,以手机为载体的相关产业也将随之得到空前发展,通过音频、视频工具提供的服务将占主导地位.但运营商要发展流媒体业务还需解决诸多技术问题,如文件制作技术、文件定位调度管理技术、节点选择技术、同步技术、容错技术、终端能力以及安全机制等.结合日常流媒体投诉处理的一些经验,总结出几种常见流媒体业务投诉的处理和分析方法.     

OV7670及其在3G流媒体服务中的应用

在3G网络视频监控迅速崛起的时代背景下,介绍一种CMOS图像传感器OV7670。从其基本结构、编程功能及数字图像的输出方面做了具体研究,最后给出一种OV7670在3G流媒体（视频流）服务中的应用方案,对3G无线视频传输系统设计具有指导意义。     

Concurrent streaming over heterogeneous wireless networks: technology and realization

Some emerging services such as augmented/virtual reality need high data rates. Whereas, the existing 2nd Generation (2G), the 3rd Generation (3G) and the 4th Generation (4G) networks cannot provide such high transmission rates. To meet these requirements, in this paper, the technology and the realization of a “virtual super terminal” that can perform the concurrent streaming of video via the cooperation of multiple heterogeneous terminals are introduced. This paper uses concurrent streaming scheduling algorithm, buffer overhead and packet delivery method, and packet loss prevention and recovery method to improve the performances of the system. Testbed measurements show that compared with the scheme without cooperative transmission this new design can significantly increase system throughput and decrease packet delay.     

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.     

Radio Link Buffer Management and Scheduling for Wireless Video Streaming

In this paper we compare strategies for joint radio link buffer management and scheduling for wireless video streaming. Based on previous work in this area [8], we search for an optimal combination of scheduler and drop strategy for different end-to-end streaming options including timestamp-based streaming and ahead-of-time streaming, both with variable initial playout delay. We will show that a performance gain versus the two best drop strategies in Liebl et al. [8], i.e. drop the HOL packet or drop the packet with the lowest priority starting from HOL, is possible: Provided that some basic side-information on the structure of the incoming video stream is available, a more sophisticated drop strategy removes packets from an HOL group of packets in such a way that the temporal dependencies usually present in video streams are not violated. This advanced buffer management scheme yields significant improvements for almost all investigated scheduling algorithms and streaming options. In addition, we will demonstrate the importance of fairness among users when selecting a suitable scheduler, especially if ahead-of-time streaming is to be applied: Given a reasonable initial playout delay at the streaming media client, both the overall achievable quality averaged over all users, as well as the individual quality of users with bad channel conditions can be increased significantly by trading off fairness with maximum throughput of the system.     

无线网络中TCP友好流媒体传输改进机制

为保持无线网络中多媒体业务对TCP的友好性，提出了一种适用于无线网络的动态自适应的流媒体传输速率调节机制。该机制通过在接收端区分网络拥塞丢包和链路错误随机丢包，准确判断网络的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，实现了TCP流友好性和流媒体服务质量（QoS）的折中。由于准确区分出无线链路误码丢包和动态调整流媒体QoS要求，该机制能维持较高的网络利用率。仿真实验结果显示在连接数为2和32，链路误码率从0到0．1变化时TCP,TFRC和吞吐量幅度下降幅度较大，WTFCC幅度下降相对较慢，最大相差达2M；在网络负载重时，尽管链路误码率较低，WTFCC区分链路错误与拥塞丢包，因此，端到端丢包率高于TCP和TFRC，但整体传输吞吐量也高于两者。归一化吞吐量显示WTFCC对TCP流友好。     

AES算法在流媒体加密中的应用

文中在分析高级加密标准AES算法原理的基础上，探讨了采用AES加密算法针对实时流媒体进行加密的具体实现，最后文中提出了一种流媒体安全传输系统中保持共享缓冲区的读写同步机制．     

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.