The design and simulated performance of a mobile video telephonyapplication for satellite third-generation wireless systems

The design and performance of a low bit-rate video telephony service for mobile third-generation (3G) systems is presented. The ITU-T G.723.1 speech coding and the ITU-T H.263 video coding recommendations are used, as proposed by the ITU-T H.324 low bit-rate multimedia communications recommendation. The target bit-rate for the H.324 service is 64 kb/s. The design is performed in conjunction with that of a wideband-code division multiple access (W-CDMA) radio transmission technology (RTT) system, proposed by the European Space Agency (ESA) for the satellite component of the ITU IMT-2000 standard. Most of the results could also be applied to the 3G terrestrial systems. The use of concatenated channel coding with convolutional inner coding and Reed-Solomon (RS) outer coding is investigated. Service designs based on equal error protection (EEP) and unequal error protection (UEP) schemes for the audio and video sources are compared. The simulation of the proposed video telephony services shows that significantly more graceful video and audio degradation is obtained with the proposed UEP scheme than with a more straightforward EEP method. The UEP scheme reduces significantly the occurrence of highly annoying audio and video artefacts, allowing satellite-based video telephony services that are compatible with the current Internet-based applications     

PROQOS—Dynamic SLA Management in DiffServ Space Links

Several base elements for the provision of quality of service guarantees have been developed in the recent past. Of these, the Differentiated Services (DiffServ) architecture stands out as the most promising. In spite of this, various issues remain, especially when multidomain DiffServ services are concerned. In this case, some forms of distributed management of Service Level Agreements that allow the specification, exchange, enforcement and monitoring of quality of service data must be in place. Although, again, some isolated solutions exist for each of these problems, considerable effort is necessary to make them work together. The project presented in this paper tried to assess the feasibility of providing differentiated quality of service in satellite IP networks, by developing a dynamic Service Level Agreement management solution for an IP over Digital Video Broadcast Satellite system. The functionality of the implemented system comprises system configuration, dynamic SLA negotiation, QoS monitoring and metering, SLA conformance checking, and QoS reporting to customers.     

Internet桌面视频会议系统的设计与实现

本文研究在ＰＣ机Ｗｉｎｄｏｗｓ操作系统的平台上，如何建立一个基于ＴＣＰ／ＩＰ网络协议的支持视频、音频流的实时、双向传输和通用数据传输（如支持白板功能）的桌面视频会议系统。目前实现了ＴＣＰ／ＩＰ协议下的通信模块和ＶｉｄｅｏｆｏｒＷｉｎｄｏｗｓ标准的视频捕获与播放功能。对视频等实时数据在Ｉｎｔｅｒｎｅｔ上的传输策略与同步算法进行了研究，力图在带宽有限、数据包可能丢失和数据包顺序可能错误的情况下保持较高的视频、音频的同步和回放质量。     

分布式多媒体交互系统音频／视频流传输的设计与实现

本文简要介绍了RTP／RTCP协议和IP Multicast技术,结合分布式多媒体交互系统中音频、视频在网络上的多点实时传输的实现,详细地介绍了RTP／RTCP协议在音频和视频数据流传输及传输控制中的实现过程,以及利用IP Multicast技术实现多媒体多点传输过程中多用户之间的信息管理。     

Packet loss probability for DiffServ over IP and MPLS reliable homogeneous multicast networks

Multicasting has become increasingly important with the emergence of Internet-based applications such as IP telephony, audio/video conferencing, distributed databases and software upgrading. IP multicasting is an efficient way to distribute information from a single source to multiple destinations at different locations. In practice IP is considered as a layer 3 protocol. Multiprotocol Label Switching (MPLS) replaces the IP forwarding by a simple label lookup. MPLS combines the flexibility of layer 3 routing and layer 2 switching.In order to provide QoS in group communications for real time applications such as video conferencing, reliable multicasting is used. Miscellaneous efforts have been undertaken to provide reliability on top of IP multicast. Two error control strategies have been popular in practice. These are the FEC (Forward Error Correction) strategy, which uses error correction alone, and the ARQ (Automatic Repeat Request) strategy, which uses error detection, combined with retransmission of data.In this paper, we present a new fair share policy (FSP) that utilizes Differentiated Services to solve the problems of QoS and congestion control when reliable ARQ multicast is used. The results should provide insight into the comparisons of the residual packet loss probability between IP multicast in MPLS networks using FSP and plain IP multicasting using the same policy when DiffServ are adopted and when reliable ARQ multicast is considered.     

一种基于SIP的交互式高清晰视频会议系统

文中提出一种专门用于远程教育中实时授课的视频会议系统(HDVShow),它采用SIP作为会话控制协议,采用可靠混合组播协议(RCMP)作为多媒体传输协议,可传输1080iHD的教室全场景高清晰视频,完全再现教室中发生的教学活动.RCMP结合了IP组播和应用层组播(ESM)的优点,解决了传统C/S结构带来的服务器负载过重问题.系统采用重传和FEC编码相结合的机制恢复网络传输过程中丢失的视频帧,从而有效改善了网络状况欠佳时的视频质量.实验表明,和传统的视频会议相比, HDVShow在高带宽下能充分利用可用带宽,从而有更高质量的视音频;在低带宽下也可有更加稳定的视音频质量.     

基于SIP的无线局域网V2IP电话的设计与实现

对SIP (session initial protocol)协议和WAPI协议进行了研究与分析,在此基础上提出了一种新的基于SIP的V2IP电话模型并实现.与传统IP电话相比,V2IP电话不仅支持WiFi无线接入,而且支持WAPI无线安全接入并可与PC机进行音/视频通话.测试结果表明,基于新模型的V2IP电话具有无线安全接入、音视频传输质量高、可移动性强等优点,并且在稳定性、便携性和可扩展性方面表现良好.     

基于Internet的远程实时教学系统的研究与实践

讨论如何构建一种基于Internet的实时教学系统。系统采用TCP／IP协议和Client／Server框架结构，实现教师和学生的异地实时相连。学生端可以看到教师端的图象，听到教师端的声音，同时，可以用语音或文字形式向教师端提问，或同其他学生端进行讨论。     

Meet In the Middle Cross-Layer Adaptation for Audiovisual Content Delivery

This paper describes a new architecture and implementation of an adaptive streaming system (e.g., television over IP, video on demand) based on cross-layer interactions. At the center of the proposed architecture is the meet in the middle concept involving both bottom-up and top-down cross layer interactions. Each streaming session is entirely controlled at the RTP layer where we maintain a rich context that centralizes the collection of (i) instantaneous network conditions measured at the underlying layers (i.e.: link, network, and transport layers) and (ii) user- and terminal-triggered events that impose new real-time QoS adaptation strategies. Thus, each active multimedia session is tied to a broad range of parameters, which enable it to coordinate the QoS adaptation throughout the protocol layers and thus eliminating the overhead and preventing counter-productiveness among separate mechanisms implemented at different layers. The MPEG-21 framework is used to provide a common support for implementing and managing the end-to-end QoS of audio/video streams. Performance evaluations using peak signal to noise ratio (PSNR) and structural similarity index (SSIM) objective video quality metrics show the benefits of using the proposed Meet In the Middle cross-layer design compared to traditional media delivery approaches.     

A client-driven media synchronization mechanism for RTP packet-based video streaming

Media synchronization is used to correctly playback a video stream with its associated audio. To support synchronization between video and audio streams transported over IP networks, an RTP/RTCP protocol suite is usually employed. In conventional server-driven media synchronization, the server needs to periodically transmit an RTCP sender report (SR) packet to provide the client with a UTC time in NTP format corresponding to the RTP timestamp carried by each RTP packet. In this paper, we propose a precise client-driven media synchronization mechanism for an RTP packet-based multimedia streaming service. In the proposed method, the server does not need to send any RTCP SR packets for synchronization. Instead, the client device derives the precise normal play time (NPT) for each video and audio stream from the received RTP packets containing an RTP timestamp. Simulations show that the proposed client-driven synchronization method can provide accurate media synchronization without employing an RTCP SR packet and accordingly reduce the initial synchronization delay, the processing complexity at the client device, the number of required user datagram protocol ports, and the amount of control traffic injected into the network.     

基于主动服务结点的分布式主动服务框架

主动服务结点既是服务的提供者,又是服务的使用者,具有自主移动计算和提供服务能力。主动服务结点采用私有UDDI和构件资源库,通过私有UDDI及WEB服务描述语言与UCDL之间的转换提供主动服务。文中结合基于私有UDDI的WEB服务模型和本地构件资源库方式,提出主动服务结点模型和基于主动服务结点的分布式主动服务框架,克服了基于UDDI框架的服务集中管理的问题。     

天地互连视频会议系统视音频传输模块

介绍了基于天地网的视频会议系统中视音频传输模块的设计和实现。它利用RTP协议来传输视音频数据和相关的统计信息，实现了视音频数据在天网(卫星)和地网(CERNFT)的互连互通，在卫星回传带宽小、双向信道不对称的条件下较好地解决了诸如延时和乱序、带宽自适应以及唇音同步等难题。     

基于IP的小型船舶监控系统的设计和实现

运用IP控制技术,将单片机与网络芯片结合在一起,通过TCP/IP协议,构成一个基于网络的分布式数据采集系统,并将音频、视频、雷达、GPS等信号完全数字化后,通过TCP/IP协议在网络上传输,实现了一个高速、稳定、灵活、扩展性强的船舶监控系统。     

远程多媒体协同诊断系统(RMCD)

该系统支持位于不同地方的用户通过连入Ｉｎｔｅｒｎｅｔ的站点进行交互,利用ＷＹＳＩＷＩＳ的共享窗口以及实时的语音和视频完成用户之间的协同工作,并在ＴＣＰ／ＩＰ协议的基础上实现了支持ＣＳＣＷ的远程通信协议扩展。     

基于Linux的实时音频与视频传输

讨论了多媒体会议中音频与视频的传输特性、要求和协议标准。并给出了程序实现。     

视频会议系统及应用

视频会议是指利用通信网,以电视实况方式召开的会议。通过视频会议系统,位于各地的与会者在开会时即可听到对方的声音,又可看到对方的影像,会议的场景,以及会议中展示的图片表格等。其好处是可以大大节省会议差旅费,提高办事效率,节省时间。     

区分服务网络的一种基于聚集状态的动态资源管理方案

提出了一种适用于区分服务网络的确保服务的动态资源提供方案:基于聚集状态的分布式动态资源管理。该方案在网络节点基于聚集状态进行接纳控制和动态资源预留,不需要核心节点保存单个流的状态,因此,该方案是可扩展的。方案包括一种轻量级动态资源预留协议和网络节点动态资源管理的相关算法,如状态建立、管理和接纳控制决策算法。该方案中,网络节点聚集状态的建立使用了基于实时流量测量的方法,在提供服务质量(QualityofService,QoS)保证的同时,得到了统计复用的增益,提高了资源的利用率,简化了信令协议,也使该方案具有很好的健壮性。     

移动通讯媒体流仿真器的设计与实现

介绍了IMS(IP多媒体子系统)中所使用的音频和视频编码;针对IMS系统设计实现了一种应用于移动通讯系统的IP媒体流仿真器,此种仿真器实现了IMS系兢中音频的AMR编码和视频的H.263编码,使用RTP协议产生模拟的实时媒体流。它可对IMS系统的媒体承载节点进行测试,配合其他的会话控制协议仿真器,可对IMS系统进行综合测试。给出了此种仿真器的设计原理和实现方法。在实际的应用中,此媒体仿真器作为IMS协议仿真器的重要组成部分得到了良好的应用效果。     

一种基于域名错误的DNS重定向协议①

域名系统（DNS）重定向是为互联网服务提供商（ISP）和DNS应用服务提供商（ASP）的用户实现增值服务的有效功能。定义以不同于错误查询域名的按相关性排序的推荐域名列表作为错误描述。提出重定向服务器的两个转交过程,以利于正确及时地找到推荐域名。此外,提出使用延长资源记录的请求处理算法。定义重定向报文结构以在DNS协议内容纳重定向IP地址信息。     

Adaptive service scheduling for workflow applications in Service-Oriented Grid

When the workflow application is executed in Service-Oriented Grid (SOG), performance issues such as service scheduling should be considered, to achieve high and stable performance in execution. However, most of the prior works on workflow management neither study the performance issues nor provide evaluation methodologies on the performance of Grid Services. Therefore, it is infeasible to apply for the service scheduling problem in SOG. In this paper, we propose and model evaluation metrics for the Grid Service performance. The metrics are extracted based on common properties of Grid Services and are used to quantify and evaluate the performance of an individual Grid Service. With these metrics, we develop a service scheduling scheme with a list scheduling heuristic, to choose proper and optimal Grid Services for tasks in workflow applications. It ensures high performance in the execution of the workflow applications. In addition, we propose a low-overhead rescheduling method, referred to as Adaptive List Scheduling for Service (ALSS), to adapt to the dynamic nature of a grid environment. ALSS provides stable performance for workflow applications, even in abnormal circumstances. Finally, we design an experimental environment with actual traces and perform simulations to quantify the benefits of our approach. Throughout the experiments, we demonstrate that ALSS outperforms conventional scheduling methods. Our scheme produces a scheduling performance that is superior to AHEFT by 50.2%, SLACK by 50.8%, HEFT by 68.3%, MaxMin by 72.0%, MinMin by 71.0%, and Myopic by 69.8%.