适用于Pbit/s光分组交换网络的丢包率下限模型

文章针对通过优化路由来改善光分组交换网络性能这一方法进行了深入地分析。文章认为在输出光缓存结构下可以计算出网络丢包率下限的理论模型,通过将该模型得到的丢包率下限与在已有路由算法下由仿真得到的丢包率进行比较,能够预测出最佳路由算法下的丢包率,从而进一步预测通过优化路由所能获得的最大性能提升。     

SPN波长转换竞争解决的研究

对光分组交换中节点共享(SPN)的结构进行了理论分析,推导出SPN系统的丢包率公式,并对系统丢包率与负载、输入光纤端口数和每纤波长数的关系进行了数值计算。结果表明:当共享波长转换器的数目从0开始增加时,首先改善的是低负载的丢包率,当低负载的丢包率接近极限后,开始明显改善中负载的丢包率,最后改善高负载的丢包率。     

SPL波长转换竞争解决的研究

对光分组交换中每条链路共享波长转换器(SPL,share-per-link)的结构进行了理论分析,推导出SPL系统的丢包率公式,并对系统丢包率与负载、输入光纤端口数和每纤波长数的关系进行了数值计算.结果表明:SPL的丢包率极限是随着波长数的增加而减小,在SPL结构中当波长转换器数目较小且一定时,波长数大的系统的丢包率比波长数小的系统要大.     

SPC、SPL和SPN波长转换竞争解决的研究

对光分组交换中波长共享(SPC)、链路共享(SPL)和节点共享(SPN)的结构进行了理论分析,推导出三种结构的丢包率公式,并对其系统丢包率与负载、输入光纤端口数和每纤波长数的关系进行了数值计算。结果表明:在相同的波长转换器数目(没有达到极限值)时,SPN结构的丢包率低于SPL的丢包率,并且在接近极限值时,SPN结构所需要的波长转换器数目也小于SPL结构所需要的波长转换器数目。     

上行信噪比对VoLTE分组丢失影响及分场景优化

VOLTE是LTE网络语音解决终极方案,而丢包率又是VOLTE的关键指标之一。高丢包会导致语音卡顿、吞字、断续、单通等,严重影响用户的感知体验。本文从上行信噪比与VOLTE丢包率之间关系进行分析研究,分场景提出优化方案并展示优化效果。     

基于NS的自适应多速率网络语音通信的性能分析

目前,几乎所有的语音电话系统（VoIP）都采用固定速率传输,这使得网络丢包,特别是连续丢包无法避免,因此导致了严重的语音质量下降.针对这一问题,给出了一种新的抗分组丢失的网络语音通信系统,并用网络仿真软件NS（network simulator）对该系统进行了性能分析,仿真实验证明,所提出的网络语音通信系统在网络丢包、平均延迟和主观听觉方面明显优于传统的IP语音电话系统.     

拥塞控制卫星PRMA-AC协议及性能分析

卫星通信中PRMA协议在负荷较重时由于终端竞争加剧会引起信道拥塞,而较长的传播时延更进一步加剧了拥塞引起的丢包。该文提出一种利用话音终端在通话的不同阶段对信道资源的不同需求对终端进行区分的方案PRMA-AC,据此引入一种接入控制机制,以减少信道竞争,提高系统服务质量。文中给出了系统模型,对协议性能进行了理论分析,获得了新协议下的接入阻塞率、丢包率等性能指标,最后通过仿真与几种卫星通信中常用的PRMA协议进行了对比,证明了协议的性能。     

全光网络中一种基于完全图的分组调度算法

在无缓存的全光OBS网络中由于相互竞争而被丢弃的数据突发是造成其高丢包率的主要原因,而分组调度策略可以从整体上改善OBS网络的这种缺点。本文建立了一种改进的分组调度映射模型,并提出了一种基于完全图的分组调度算法,该算法相对简单且容易实现,能够较好地解决OBS网络中由于数据突发的无序竞争所致的丢包问题,从而进一步提高网络的服务质量及其信道利用率。     

AOPS节点中基于业务分级的均衡调度算法

异步光分组交换(AOPS)节点配置流量均衡功能后,可极大降低丢包率与平均时延。因此,提出了一种基于业务分级的动态均衡算法,仿真结果表明此算法可根据业务等级实现异步变长数据的动态均衡输出,端口丢包率最大降低2.05%,系统整体丢包率降低1.67%,小于0.017%;端口时延最大降低0.128ns,系统整体时延降低0.11ns,小于0.56ns。     

基于H.264的具有抗分组丢失能力的视频通信系统

针对IP网络环境下实时视频通信的需要，提出了具有抗分组丢失能力的视频通信系统（PRVCS）。系统以H．264编解码算法为核心，在满足实时视频通信要求的同时，将前向分组保护，误码掩盖和交互式防误码扩散三种算法有机结合，在丢包环境下保证恢复视频的主客观质量。其中重点研究了PRVCS所使用的算法、体系结构以及系统的具体实现。     

Design and performance evaluation of a packet voice system

A packet voice network has been designed using adaptive delta modulators. Tests were performed to examine the effects of packet length and packet loss rate on digital voice intelligibility. The packet voice network was simulated using a SLAM network model to find queue size requirements as a function of packet size, delay distribution, and expiring time out. The queuing strategy is investigated in order to minimize total information loss during channel acquisition delays.     

关于VOLTE语音质量分组丢失率高小区整治的研究

阐述4G语音分组丢失对用户感知的影响,通过大数据对语音分组丢失进行四维五域分析定界,聚焦在“单通”、“断续”、“音质“三个影响用户感知的现象,最后根据VoLTE语音质量定界法,对无线侧分组丢失问题这一影响用户感知主要原因进行分析,并列出影响分组丢失最常见因素及优化解决方法。     

Performance Evaluation of a Packetized Voice System--Simulation Study

Introduction of the packet switching technique into digitized voice communication may afford great advantages in efficient use of the channel, compared to both circuit-switched and DSI systems. Detailed characteristics, however, have not been obtained because of difficulty in the exact analysis. Hence, simalation models are developed in this paper for the packetized voice transmission system, and various characteristics such as tranmission delays and loss probability of voice packets are obtained. We further evaluate three types of voice packet reassembly strategy at the receiving terminal, and obtain the optimal packet length, which keeps both overall packet transmission delay and packet loss probabilty less than a certain permissible value. Comparison among three strategies is also stated.     

Study of information loss in packet voice systems

Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay     

AAL2分组话音复接器的研究与实现

该文研究在ATM虚通路带宽利用率一定的条件下,AAL2分组话音复接器性能随ATM虚通路输出速率的增加而变化的情况。得出结论:当ATM虚通路带宽利用率一定时,ATM虚通路输出速率越高,AAL2分组话音复接器的分组丢弃概率和平均分组排队时延越小。并提出了一种AAL2分组话音复接器的实现方案。该方案可以随着ATM虚通路输出速率的增加,方便地复接多个E1话音电路上的话音数据。     

Delta Modulators in Packet Voice Networks

In this paper, the performances of the Song Voice Adaptive Delta Modulator (SVADM) and the Continuously Variable Slope Delta Modulator (CVSD) in terms of dynamic range, sampling rate and the channel errors are compared. The use of the SVADM and the CVSD in a packet voice system, the algorithms for digital detection of silent periods and the performance of a packet voice system using the SVADM and the CVSD as source encoders are presented. The parameters employed for subjective evaluation of the packet voice system are packet size, silence detection algorithm, bit rate and packet loss rate.     

Anti-packet-loss joint encoding for voice-over-IP steganography

VoIP (voice over IP) is a kind of voice communication technology based on UDP/IP protocols. Packet loss will inevitably happen when the channel environment deteriorates, which can pose challenges to the reliable transmission of VoIP steganography. A steganographic model based on joint encoding was proposed. In this model, packet erasure coding was introduced to preprocess the secret data. And the encoded data were embedded into voice packets with minimum dis-tortion using matrix embedding. Furthermore, the influences of key parameters on the performance of joint coding were studied. The selection algorithm for optimal parameters was also given. Experimental results show that the proposed joint coding can effectively improve steganographic resistance to packet loss, and decrease the number of modifications to the voice stream.     

VoLTE高丢包问题分层优化方法研究

VoLTE是LTE网络语音演进的最终目标方案。VoLTE语音业务中,丢包问题是影响语音质量的关键因素之一。高丢包会导致语音吞字、断续、单通等问题,严重影响用户感知。本文从VoLTE协议栈入手,分析各协议层主要功能,提出高丢包优化方法,展示参数试点效果。     

AAL2分组话音复接器缓冲器队列门限值的确定方法

该文研究了带比特丢弃的AAL2分组话音复接器缓冲器队列门限值的确定方法,提出用话音分组作为缓冲器队列门限值的单位,给出了确定门限值的计算公式,并对输出链路容量为384kb/s的情况进行了计算机仿真。仿真结果表明,作者提出的门限值的确定方法可获得较小的平均分组时延和较低的平均分组丢失率,计算简便,易于实现,是一种很好的确定缓冲器队列门限值的方法。