一种新的基于网络丢包率的主动队列管理策略

提出了一种新的基于网络丢包率的动态自适应的主动队列管理的改进算法。该算法首先用早期网络的丢包率标记到达的数据包,并作为数据包的丢弃概率。这样使得到达数据包的丢弃概率逼近当前实际网络的丢包率,然后经过自适应调整丢弃概率使得缓冲队列长度保持在一定范围内。通过NS仿真实验表明了该算法可以很好的控制队列长度,降低延迟抖动。     

无线环境下具有实时约束的主动队列管理机制

实时流业务的超期丢包率是衡量无线网络传输过程服务质量的常用参数,但是该参数并不能反映出具有不同重要度的数据包对视频质量的影响.该文将实时传输流的数据包重要度与超期丢包率结合,提出加权超期丢包率的概念,更加准确地描述丢包与视频质量的关系,在无线环境下提出基于数据包不同重要度的适用于实时约束流的主动队列管理算法,通过主动丢弃数据包以最小化服务过程中实时流的加权超期丢包率.仿真结果表明,相比于传统的队列管理算法和实时视频流滤波器算法,该超期数据包主动丢弃算法能够有效降低加权超期丢包率,提供更好的服务质量,提高大约0.5~1.5 dB的峰值信噪比(PSNR)增益.     

基于前向纠错的自适应低延时传输机制

针对无线网络中视频传输的延时以及用户数据报协议（User Datagram Protocol,UDP）因丢包造成的视频质量大幅下降问题,提出一种基于前向纠错的自适应低延时（Adaptive Low-Latency Forward Error Correction,AL-FEC）传输机制。该机制通过前向纠错方案避免在使用UDP协议传输过程中因丢包而造成的视频失真卡顿现象,通过发包策略及自适应冗余度达到降低端到端时延并提高视频传输质量的目的。根据当前的网络状况以及视频帧特性确定需要发送的源包个数,利用接收包接收到的数据包信息实时估计网络状况,从而根据网络波动情况动态更新冗余包比例。大量实验结果表明,相较于静态的前向纠错机制,AL-FEC能够使得平均端到端时延降低40～50 ms,在高丢包率情况下,视频的峰值信噪比（Peak Signal to Noise Ratio,PSNR）提高5～8 dB。     

基于机会式网络编码的低时延广播传输算法

为了提高无线网络中数据包广播传输的效率,本文提出了一种基于机会式网络编码的广播传输算法.该算法在发送端按一定顺序选择不同终端的丢包,并采用异或运算编码重传包,在终端采用从重传包中解码数据包的方法恢复丢包.该算法优先恢复时间重要性较高的丢包,并使多个终端同时从单个重传包恢复其丢包,因此有效地提高了广播传输效率并降低了传输...     

无线网络中视频流的协作传输研究分析

提出一种两跳的协作传输方案,该方案利用无线信道广播特性,并结合应用层视频包的信息,通过选择成功监听到数据包的节点来承担协作传输任务,以此来降低重传次数,降低包的接入和服务延时,提高传输效率,以解决在无线环境下流媒体传输存在丢包率高、吞吐量受误码率影响、端到端的传输延时等QoS问题.     

基于合作分集的新型自动重传协议

研究了一种基于合作分集的新型链路层自动重传协议——合作ARQ协议,传输错误的数据包将由一个中继节点进行重传。根据接收端处理重传数据包的不同方式,探讨了2种协议:1)合作简单ARQ协议,目的节点在收到重传的数据包之后丢弃原始的数据包;2)合作混合ARQ协议,目的节点将重传的数据包和原始数据包进行分集合并。分析了独立慢衰落信道下合作ARQ协议性能,仿真成果证明,合作ARQ协议相比传统ARQ协议能够降低链路层的丢包率,提高了系统的传输性能。     

有线-无线混合IP网络多媒体传输算法研究

提出了一种端到端的在异构的有线．无线混合IP网络上多媒体传输的改进算法WMTA（wireless multimedia transmission algorithm）。通过研究Gilbert无线差错模型仿真环境中包的长度与丢包率的关系,发现包长与丢包率呈线性增长的关系。基于上述观察,算法通过交替发送大小数据包探测随机和拥塞丢包数,并根据两种丢包的程度进行速率控制。针对不同网络状态转换时算法更新慢的缺点,添加了更新因子K,使算法在网络状态转换时能迅速适应当前网络的状态。仿真结果表明,与现有算法相比,WMTA无论是在运行单个流还是存在竞争流情况下,都能够达到更高的吞吐量和带宽利用率,有效地提高了网络中多媒体传输的服务质量（QoS）。     

平安城市系统GB/T 28181级联设计方案简析

本文主要针对平安城市建设中数字视频系统和系统联网的需求,提出一种数字高清和国标GB/T 28181-2011标准级联解决方案.该方案采用基于数字高清和GB/T 28181-2011技术,利用数字高清、GB/T 28181-2011系统互联标准接口开放的特性,为平安城市视频监控平台提供了安全可靠的系统联网技术.在这种联网模式下,不同厂家平台通过GB/T 28181-2011标准接入接口可以相互访问,从而保证上级平台通过GB/T 28181-2011访问下级平台视频数据的联通性.在平安惠州和平安佛山城市项目的建设中,该技术实现了与省公安厅视频监控平台标准域上联接入不同厂家的异构平台高清视频传输服务质量要求高的应用业务,证明了本方案的有效性,对于向其他智慧平安城市视频监控系统联网业务领域推广具有重要意义.     

MAC层实时业务满意度评价模型及基于满意度的包调度与资源分配方法

该文给出了无线通信系统中MAC层可实现的评价用户VoIP业务和实时视频业务满意度的方法,该方法根据MAC层统计的丢包率及丢包模式等信息计算当前业务的用户满意度。基于文中的业务满意度评价准则,该文进一步提出一种在WiMax系统中基于满意度的包调度和资源分配算法。仿真结果表明,该方法在保证系统获得较高吞吐率的同时,能够更好的保证实时业务的满意度。     

基于时频二维竞争解决机制的光数据包交换节点性能分析

针对光数据包交换的应用需求,提出了基于时频二维竞争解决机制的光数据包交换节点结构,建立了该节点基于排队论的丢包率分析模型,分析讨论了光交换节点竞争解决机制对丢包率的影响。输出端口归一化负载为0.8时,时频二维竞争解决机制在8波复用和8根延迟线结构下使丢包率降低至约10-7。数值分析结果表明,高端口负载下增加复用波长个数对减小端口丢包率比较有效,低端口负载下增加FDL延迟线数量对减小端口丢包率比较有效。     

基于实测流量的网际分组丢失率分析

分组丢失率是评价网络性能的一个重要指标。提出了一种利用完整TCP流计算实测数据中分组丢失率的方法,并基于实测数据对CERNET江苏省网与外部不同网络分区的分组丢失率进行分析。算法给出了基于实测数据的分组丢失率包括首次分组丢失率pfl、整体分组丢失率psl和重传率prp 3个测度。统计分析表明,在绝大多数情况下,pfl比psl小,而prp比pfl和psl明显要高,验证了分组丢失对网络性能的影响以及重传率明显比分组丢失率高。     

网络流量丢包率预测模型

无线信道的网络流量的及时预测,是目前国内外研究的前沿课题。通过对实时网络流量的测量分析,提出了一种基于最小二乘法的多项式拟合网络流量丢包率预测模型。该模型根据当前获得的网络流量信息,对网络丢包率趋势进行预测,为无线网络下信号丢包率的预测供了可行的手段。实测丢包率与预测丢包率的对比分析表明提出的预测模型能很好的预测丢包率趋势。     

Unequal packet loss protection for layered video transmission

A general model for the transmission of layered video is introduced. The effect of packet dropping in different layers on the reconstruction of pictures is analysed. The application of unequal packet loss protection (UPLP) for layered video is proposed and the performance is investigated. An algorithm for the design of the UPLP parameters is described. The numerical analysis shows that the unequal packet loss protection can significantly improve the communication quality, in the sense of packet loss rate on reconstructing pictures, without any additional complexity in the network infrastructure. It is also shown that application of UPLP can achieve a more efficient bandwidth utilization than the application of ARQ protocols. This technique is suitable for multimedia multicasting in heterogeneous networks, as well as for digital video broadcasting systems which support a variety of receivers with different resolutions and signal processing powers     

Video quality protection for real time video streams over wireless networks

Received video quality is dependent on the available link rate and the packet loss ratio, which are correlated in a busy network link. Even low packet loss ratios (PLRs) can significantly reduce the video quality. In this paper, a packet level parity Forward Error Correction (FEC) is applied to the video stream in order to reduce the video PLR. A constant gross data rate is assumed, such that adding a FEC leads to a decrease in effective video data rate. The FEC block is truncated at the end of each video VOP, such that there are no inter-VOP dependencies for FEC correction. An algorithm is proposed to optimize the FEC length, based on the Quality of Experience as modelled by the ITU-T R G.1070 standard. It is shown that the optimization algorithm can significantly increase the video quality, without increasing the gross data rate. The algorithm has been evaluated both analytically and through simulations, which confirm the very significant increases in subjective video quality.     

基于网络丢包率的混合FSO/RF系统切换技术研究

利用C++ Builder编程软件提供的相关网络组件,提出了基于网络丢包率的混合FSO/RF系统切换判决机制,在网络中实现了对链路的监控功能,并采用无线路由器、光纤收发器等器件搭建了混合FSO/RF测试系统,对基于网络丢包率的混合系统切换机制进行了测试,结果表明采用网络切换判决方案,能够自动选择链路通信,保证数据不间断...     

On network bandwidth sharing for transporting rate‐adaptive packet video using feedback

While existing research shows that feedback‐based congestion control mechanisms are capable of providing better video quality and higher link utilization for rate‐adaptive packet video, there has been relatively little study on how to share network bandwidth among competing rate‐adaptive video connections, when feedback control is used in a fully distributed network. This paper addresses this issue by presenting a framework of network bandwidth sharing for transporting rate‐adaptive packet video using feedback. We show how a weight‐based bandwidth sharing policy can be used to allocate network bandwidth among competing video connections and design a feedback control algorithm using an Available Bit Rate (ABR)‐like flow control mechanism. A novel video source rate adaptation algorithm is also introduced to decouple a video source's actual transmission rate from the rate used for distributed protocol convergence. Our feedback control algorithm provides guaranteed convergence and smooth source rate adaptation to our weight‐based bandwidth sharing policy under any network configuration and any set of link distances. Finally, we show the on‐line minimum rate renegotiation and weight adjustment options in our feedback control algorithm, which offer further flexibility in network bandwidth sharing for video connections. Copyright © 2000 John Wiley & Sons, Ltd.     

有限带宽网络视频流传输速率控制策略研究

为促进公共视频监控领域新技术的开发和应用,立足解决工程实际问题,提出了一种有限带宽网络下视频流传输速率的控制策略。系统可以根据数据包丢失率、传输时间和时限等动态参数对网络带宽进行估测,并对实时视频的发送速率进行自适应调整,实现稳健控制,很好地解决了网络的传输带宽与视频流码率不匹配等问题。仿真实验表明,系统在降低视频时延和包丢失率、减少网络拥塞、提高带宽利用率和改善视频质量等方面效果明显。     

Rate control for streaming video over wireless

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is TCP-friendly rate control (TFRC) (Floyd, 2000). It is equation-based rate control in which the TCP-friendly rate is determined as a function of packet loss rate, round-trip time, and packet size. TFRC assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the main cause of packet loss is at the physical layer. In this article we review existing approaches to solve this problem. Then we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over a 1/spl times/RTT CDMA wireless data network, and video streaming simulations using traces from the actual experiments are carried out to characterize the performance and show the efficiency of our proposed approach.     

Estimation of accurate effective loss rate for FEC video transmission

We propose an algorithm for adjusting data transmission parameters, such as the packet size and the code rate of forward error correction (FEC), to obtain maximum video quality under dynamic channel conditions. When determining transmission parameters, it is essential to calculate an accurate effective loss rate that reflects FEC recovery failures and over-deadline packets. To this end, we analyze the delays caused by FEC coding and the potential packet size variations. In our analysis, we consider the effect of delayed transmission of video packets incurred by the parity packets as well as the encoder and decoder buffers. With the analysis reflecting the delay effect, we are able to accurately estimate the delay patterns of all video packets. Based on the analysis results, we establish an accurate model for estimating the effective loss rate. Simulations show that the proposed effective loss rate model accurately estimates the effective loss rate and significantly improves the reconstructed video quality at the receiver.     

利用SRUDP的3G无线网络视频监控系统设计

为促进无线视频监控领域新技术的开发和应用,立足解决工程实际问题,提出了一种利用简单可靠用户数据报协议(SRUDP)的3G无线网络视频监控系统的新型设计方案.系统可以对H.264视频流进行分层模式的高效编码、压缩纠错和实时传输控制,并重新封装用户数据报协议(UDP)、SRUDP和IP报头.系统采用滑动窗口机制和重传机制来进行消息的收发同步和流量控制,可以实现任意用户终端对任意监控点的有效监控.系统视频QoS的控制方式运用迭代算法,发送端可以根据反馈信息通过编码器进行自适应调整速率,并参考SRUDP缓存进行参数修改,实现鲁棒控制.仿真实验表明,在3G网络环境下,基于SRUDP的视频应用在降低视频时延和包丢失率、减少网络拥塞、提高传输速率和改善视频质量等方面具有明显的优越性.