IPTV业务中的应用层差错控制技术分析

在IP网络上提供网络电视(IPTV)业务要面对传输带宽、传输延时和抖动、丢包等困难.以终端为中心的实现方案通过在接收终端以及信源编码中加入一定的控制策略如拥塞控制策略、差错控制策略,可以较好地改善图像质量.在IP网络上采用端到端的应用层可靠性解决方案,能很好的提供IPTV业务.应用层前向纠错(AL-FEC)技术解决网络丢包问题,从而保证端到端的可靠性.     

流化云平台建设思路及方案

流化云平台是基于云计算技术的理念,采用视频作为“云端”向“终端”呈现处理结果的一种云计算方案.应用在云端服务器上运行,将运行的显示输出、声音输出编码后经过网络实时传输给终端,终端进行实时解码后显示输出.终端同时可以进行操作,经过网络将操作控制信息实时传送给云端应用运行平台进行应用控制,终端“精简”为仅提供网络能力、视频解码能力和人机交互能力.     

针对移动视频终端的基于感兴趣区域的快速转换编码

提出一种针对移动终端,基于感兴趣区域(ROI)的快速转换编码方案.首先,根据移动终端的显示尺寸,在视频服务器端利用视觉关注度模型从H.264视频流自动地检测出ROI.然后,在代理服务器端根据ROI转换编码生成适合于移动终端的视频流.此外,针对此转码体系提出了一种快速模式选择算法.仿真实验结果表明,本方案可在降低网络占用带宽的情况下,获得较好的主观视觉效果,并且计算量小.     

基于负载自适应的无线自组网视频传输队列调度算法

移动终端多媒体业务的发展目前已经成为终端发展的必然趋势.如何在移动终端构成Ad hoc网络时有效地进行视频传输是无线网络研究的热点之一.提出了一种基于负载状况的跨层优化方案,其基本思想是结合应用层视频编码的特点和接入层的网络负载和资源的情况联合进行优化.在特定的资源下通过概率接入的方法对重要性更高的数据包进行更优先的接入.仿真结果显示,在网络处于高负载、多跳传输等场景下,提出的方案视频传输质量PSNR值提升3 dB以上.相对于传统调度算法,系统时延也可以大幅度降低.     

自适应传输速率MAC层协议的探讨

随着无线通信技术的发展和器件性能的提高,无线网络有能力支持更高的数据传输速率.随之而来的问题是如何提高网络的性能,即根据无线信道的状况而自适应地改变传输参数.文中讨论了媒体访问控制(MAC)层协议上多跳无线网络自适应传输速率的技术方案.该方案运用自适应调制和编码技术,可以最大限度地利用信道的容量,根据不同终端报告的信道情况提供个性的调制与编码选择;对位置较好的用户提供高速率的数据服务,增加系统的吞吐率;并且由于信道的自适应是通过改变调制和编码的方式,而不是像功率控制那样改变发射功率,因此系统中干扰变化很小.     

D2D网络中基于立即可解网络编码的时延最小化重传方案

该文针对D2D无线网络中多终端并发协作重传冲突避免问题,提出一种基于立即可解网络编码的时延最小化重传方案。在重传阶段,充分利用D2D无线网络终端协作传输数据的优势,结合各终端数据包接收状态,综合考虑时延的影响因素,选取单次重传时延增量较小的数据包生成编码包,最小化重传时延。同时,构建终端冲突图,在图中搜索极大独立集,根据各终端的编码包权重值,选择最大加权独立集中的终端作为并发协作重传终端,从而降低重传次数。仿真结果表明,所提方案能够进一步改善D2D无线网络的重传效率。     

基于混合人工鱼群的ABC支持型切换决策机制

面向下一代互联网(NGI, next generation Internet)的实际需要,提出了一种总最佳连接(ABC, always best connected)支持型切换决策机制.引入模糊数学和微观经济学等相关知识,刻画应用类型、服务质量(QoS, quality of service)需求、接入网络和移动终端,综合考虑接入网络状况、应用需求、用户对接入网络编码制式偏好、用户对接入网络供应商偏好、终端当前运动速度和终端当前剩余电量等因素.使用博弈分析,基于变异与模拟退火相结合的混合人工鱼群算法,寻找把N个终端分配到M个接入网络的最佳切换决策方案,使各方效用达到或接近Nash均衡下的Pareto最优.仿真结果表明,该机制是有效的,性能较好.     

提高卫星通信系统吞吐量的复数域网络编码算法

针对传统卫星通信系统吞吐量不高的问题,本文提出了一种基于复数域网络编码（CFNC）的卫星通信系统方案。该方案在发送前对信号进行符号级的CFNC编码,在不增加终端发射功率的情况下提供比传统卫星通信系统更高的吞吐量,并适用于多用户网络。该方案同时适用于基于处理转发器和透明转发器的卫星平台。特别是在双向卫星通信系统中,与基于伽罗瓦域网络编码（GFNC）的系统相比,本文提出的方案不仅具有更高的吞吐量而且更具有普适性。      

基于终端行为的可信网络连接控制方案

在可信网络连接(TNC)框架下,结合完整性度量方式,通过对终端活动进程的行为属性实时分析并计算终端的"健康度",进而提出实施网络连接控制的方案.与已有的基于终端静态特征的控制方法相比,该方案在识别和隔离潜在安全威胁方面更有效.实验结果表明利用该方案能够实时地将感染恶意代码的终端阻断在网络之外.     

安全的物理层网络编码的研究

无线信道的广播特性使得其中的碰撞现象普遍存在,充分利用这一特性的物理层网络编码可较大幅度地提高系统吞吐量,但也存在其特有的安全隐患.本文的主要贡献:从物理层网络编码的原理出发,挖掘了其中存在的必要前提破坏、信号强度攻击、信号能量攻击等安全隐患,并提出了应对的基本思路:综述了物理层网络编码的研究现状,并展望了未来的研究方向;提出了结合利用云计算和网络编码构建从终端到交换节点、从底层到高层、无处不在的计算网络的概念.     

家庭网关承载语音业务的研究与实现

根据当前网络发展趋势和语音业务的承载需求,提出了采用家庭网关承载语音业务的实现方案,包括家庭网关的工作模式、IP地址规划、PVC规划、VLAN规划、QoS实现策略及软交换改造建议等,指出了家庭网关与原有承载方式相比,仅需在网络中进行统一配置,应用QoS技术对业务进行保障,就可以实现线路中的多业务承载,并且给出了采用家庭网关的方式承载多种业务适用的应用场景。     

Improving the R-score of an adaptive VoIP codec in IEEE 802.16 networks

Worldwide Interoperability for Microwave Access (WiMAX) technology, which is based on the IEEE 802.16 standard, supports different quality of service (QoS) for different services. WiMAX is expected to support QoS in real-time applications such as Voice over Internet Protocol (VoIP). When network congestion occurs, the VoIP bit rate needs to be adjusted to achieve the best speech quality. In this study, we propose a new scheme called Adaptive VoIP Level Coding (AVLC). This scheme takes into consideration network conditions (packet delay and packet loss) and a connection’s modulation scheme. The amount of data that can be transmitted increases with the speed of the modulation scheme. When network congestion occurs, AVLC scheme prioritizes reducing the bit rate of a connection that has a slower modulation scheme to mitigate congestion. Depending on network conditions, such as modulation scheme, packet delay, packet loss, and residual time slot, we use the G.722.2 codec to adjust each connection’s bit rate. Simulations are conducted to test the performance (network delay, packet loss, number of modulation symbols, and R-score) of the proposed scheme. The simulation results indicate that speech quality is improved by the use of AVLC.     

Type-2 fuzzy logic control of PQoS driven adaptive VoIP scheme

Adaptive VoIP schemes have potentially suboptimal performance owing to imprecision in the metrics used to infer network state. An interval Type-2 fuzzy logic controlled scheme for VoIP services is presented. It infers network state from average delivered perceived quality of service and its degradation due to network congestion and updates an AMR codec mode to match voice quality to available network bandwidth. Tests showed that the scheme maximised delivered voice quality and outperformed an existing adaptive scheme. The scheme achieves robust performance in the presence of input imprecision and can be implemented in VoIP terminals, and the fuzzy rule base is easy to understand and change by non-experts because of its similarity to the human decision-making process.     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

Perceptual QoS assessment technologies for VoIP

Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy.     

Supporting multimedia applications through network redesign

Recent evolutions in high‐performance computing and high speed broadband Internet access have paved a way to enterprise‐wide multimedia applications, which require stern QoS from the underlying networks. In this paper, we have explored threefold studies on existing enterprise network, whereby we proposed an analytical approach to evaluate the performance of the existing network; we have examined the feasibility of existing enterprise networks to accommodate voice over Internet protocol (VoIP) services with acceptable QoS, and we have redesigned the enterprise network to accommodate VoIP services to comply with the user defined QoS. The network performance is evaluated by number of VoIP calls sustained by the network, bandwidth utilization, loss rate and latency through Network Simulation (NS‐2) tool. We have derived a cost model to show the cost‐effectiveness of VoIP services over telephonic network. For a medium‐size enterprise network of 200 clients and 9 servers, our simulation results show that the redesign improves the network performance by increasing the number of VoIP calls by 57% and decreasing bandwidth utilization and packet loss rate by 20% and 7%, respectively. Moreover, the proposed network redesign demonstrates that the network can be scalable and it can handle up to 4% increased voice calls in the future maintaining QoS standards. Copyright © 2012 John Wiley & Sons, Ltd.     

一种动态时分窄带多业务接入新方案

如何在窄带低比特率链路上进行高效的语音数据等多业务综合接入,并保证语音等实时业务的质量是目前多业务接入的一个重点问题.本文提出的动态时分多业务接入方案解决了Digital Data Network (DDN)专线上窄带压缩语音,ADPCM语音,传真,以及数据的同时接入问题,有效地解决了DDN专线上多业务接入的质量保证问题,提高了链路利用率.本文对于该方案的性能进行了分析,并与目前基于IP的多业务接入方案进行了比较.本方案能够提供目前的VoIP方案下所无法提供的语音,传真业务的质量保证,在多业务的支持上比VoIP更加简单,更有吸引力.     

A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

VoIP语音时延的分析和研究

文章介绍了VoIP(IP网络上传送语音)语音质量的测试方法,分析了影响VoIP语音质量的主要因素:延迟、抖动、丢包率和时延.利用E模型定量地分析了语音质量与端到端时延的关系,通过建立数学模型,指出了VoIP 系统中主要的时延分量,并研究了这些时延分量产生的机理和影响它们的参数.在设计实际的VoIP系统时,可以通过优化影响时延分量的主要参数,改善VoIP系统的时延.