基于自适应变速率编码的VoIP网关的研究与应用

针对当前VoIP语音质量易受网络带宽影响的问题，提出了一种基于自适应变速率编码的VoIP网关。通过实时监测RTP语音分组的丢包率，并分析服务质量，自适应地选择最适合当前网络状况的编码速率，以提供一种语音质量和网络状况的最佳组合，从而降低语音分组的丢包率、有效地保障语音服务质量。结合开源项目Asterisk，实现了基于自适应变速率编码的VoIP网关。     

基于E-model的VoIP语音质量评估的研究

为准确评估VoIP）语音质量，对E—model算法进行了深入研究，剖析了E—model算法的组成部分Id，Is，Ie，A，探讨了丢包、延迟和抖动对VoIP质量的影响，并应用该算法对多种语音编码进行了评估。实验证明该客观评估算法主观与客观相关度高，有较强的适应性，可靠性，实用性，完全可用于VoIP语音质量评估。     

在区分服务上的VoIP仿真评价研究

分析了区分服务的工作原理和影响VoIP语音质量的主要因素,介绍了一种语音质量的客观评价方法——E模型,运用ns-2仿真器构建网络仿真模型,比较VoIP在区分服务和传统网络中的性能表现,利用E模型对VoIP的性能进行了定量的客观评价,并为区分服务对VoIP的支持能力提供了用户级语音质量的分析。     

基于语音质量预测的Speex自适应码率控制算法

为实现IP语音(VoIP)质量的动态管理与控制,提出了一种基于语音质量预测的自适应码率控制算法。通过实时预测VoIP通话的瞬时语音质量和总体语音质量,自适应地调整Speex编码参数,从而根据需要选择最佳编码速率。实验仿真结果表明,提出的算法能够有效减少网络拥塞,提高VoIP系统的语音质量。     

提高VoIP语音通信质量方法的研究

探讨了VoIP的关键技术,分析了影响VoIP语音通信质量的因素,提出了提高VoIP语言通信质量的方法,即:优化网络环境、选择合适的编解码、服务质量保障(QoS)、使用颤音缓存。该方法对提高VoIP语音通信质量,推动VoIP业务的普及具有实际意义。     

基于H．323协议的企业级VoIP语音网关的设计和实现

讨论了基于H．323的企业级VoIP语音网关的设计原则，并根据该原则设计和实现了VoIP语音网关的基本模块。文章还简要介绍了H．323协议，同时对VoIP语音网关和VoIP技术作了小结。     

基于网络性能的VoIP语音质量评价模型

在VoIP应用中,为了实现服务质量的监测和路径切换,通常需要测量路径的网络性能,并将网络性能映射到语音质量评价.本文提出一种基于网络性能的VoIP语音质量评价模型,该模型在E-Model的基础上进行了改进,只考虑网络性能的动态变化对语音质量的影响.新的模型考虑更少的影响因素,比E-Model更容易计算,因此更适用于VoIP系统的语音质量评价.通过实验比较了新的模型和简单的网络参数评价模型,结果显示该模型具有更好的语音质量描述能力.     

基于NS-2的VoIP模拟与仿真

提出了一种基于自适应多速率算法的新VoIP系统方案,该方案可以根据网络信道质量的好坏来自适应地选择一种最佳的语音传输模式,从而在一定程度上保障了重构后的语音质量.在NS-2仿真平台下,通过与传统VoIP体系比较发现,所提出的新VoIP系统方案在语音QoS保障方面明要显优于传统VoIP系统.     

基于E-Model的VoIP语音质量研究

针对目前网络电话语音质量难以准确评价及预测的情况,基于E-Model对VoIP的语音质量进行预测。分析几个主要影响因素,如延时、丢包等对话音质量的影响,构建VoIP语音质量预测模型,将E-Model中未考虑到的抖动因素引入模型公式,着重考虑抖动缓冲区的大小对语音质量的影响。通过设计相关验证实验,证明该模型对VoIP语音质量的预测具有较高的准确度。     

基于语音质量预测的VoIP自适应抖动缓冲算法

抖动缓冲是解决VoIP系统延时抖动问题的有效方法。为实现抖动缓冲的动态调整,获得更好的VoIP通话质量,提出了一种基于语音质量预测的自适应抖动缓冲算法。算法采用Pareto分布为延时建模,通过E-Model方法预测突发丢包模式下的瞬时语音质量,以最大化语音质量为目标,自适应选择出最优的抖动缓冲区大小。实验仿真结果表明,所提算法明显优于已有算法,能够有效提高VoIP系统的语音质量。     

一种基于SoPC技术的iLBC语音编解码器设计实现

基于VoIP技术的语音通信发展迅速,单芯片VoIP处理器的设计方法成为当前的研究热点.iLBC作为专为窄带通信而设计的VoIP语音编解码器,可以在丢包率和延迟较高的网络环境中保持良好的语音通话质量,具有广泛的应用前景.传统的基于DSP处理器实现方法具有芯片面积大、功耗高等缺点,难以满足VoIP系统集成度高、低功耗和易于升级等需求.本文提出了一种基于SoPC技术的iLBC语音编解码器实现方案,并对自相关计算算法进行了并行计算硬件IP核设计,提高了系统的集成度、计算性能和可扩展性.理论分析和实验结果表明并行自相关计算结构有效减少了访存次数,可以获得接近30的加速比.     

Hybrid Signal-and-Link-Parametric Speech Quality Measurement for VoIP Communications

A hybrid signal-and-link-parametric approach to speech quality measurement for voice-over-Internet protocol (VoIP) communications is described. Connection parameters are used to determine a base quality representative of the transmission link. Degradation factors, computed from perceptual features extracted from the decoded speech signal, are used to quantify distortions not captured by the connection parameters. The algorithm is tested on speech degraded by acoustic noise, temporal clippings, and noise suppression artifacts, thus simulating degradations present in wireless-VoIP tandem connections. Hybrid measurement is shown to overcome the limitations of pure link parametric and pure signal-based measurement methods, resulting in better measurement accuracy for modern VoIP communications. In addition, the proposed algorithm incurs modest computational overhead relative to pure link parametric measurement and attains up to 88% reduction in processing time relative to the ITU-T standard P.563 signal-based algorithm.      

Evolutionary speech quality estimation in VoIP

Estimating the quality of Voice over Internet Protocol (VoIP) as perceived by humans is considered a formidable task. This is partly due to the relatively large number of variables that are involved as determinants of quality. Moreover, discerning the significance of one variable over the other is difficult. In this paper a novel approach based on genetic programming (GP) is presented. It maps the effect of network traffic parameters on listeners’ perception of speech quality. The ITU-T Recommendation P.862 (PESQ) algorithm is used as a reference model in this research. The GP discovered models that provide effective VoIP quality estimation are highly correlated to ITU-T Recommendation P.862 (PESQ). They also outperform the ITU-T Recommendation P.563 in estimating the effect that packet loss has on speech quality. The GP discovered models prove suited to real-time and in vivo evaluation of VoIP calls. Additionally, they are deployable on a wide variety of hardware platforms.     

马尔可夫模型在VoIP通话质量实时评估中的应用

针对运用国际电联G.107 E模型评估VoIP通话质量时如何准确计算有效设备损伤系数的问题，提出一种基于马尔可夫模型的实时评估算法，通过分别为随机信息包丢失概率和突发比建立三态和二态马尔可夫模型，推导出估算有效设备损伤系数的运算公式和相应统计算法。商用测试结果表明，该评估算法能够在实时环境中较准确地评估VoIP通话质量。     

Packet based echo cancellation for VoIP networks

In today’s modern telephony network, VoIP is fast emerging as one of the main communication techniques. However, the performance and the quality of VoIP are affected by echo. Packet Based Echo Canceller (PBEC) is introduced, as a solution to cancel echo in the VoIP network. PBEC can replace the current echo cancellers, which are located in the Public Switched Telephony Network (PSTN) central switches. The operating principle of the PBEC is explained and its advantages are highlighted. The performance of the PBEC using different speech codecs is also studied. Using the PBEC, a maximum Echo Return Loss Enhancement (ERLE) of 37.39 dB has been achieved when used with the Pulse Code Modulation (PCM) based speech codec. From the simulation results, it can be seen that the performance of the Adaptive Differential Pulse Code Modulation (ADPCM) clearly matches the performance of the PCM based speech codec. The other major problem affecting the VoIP network is the issue of packet loss. This issue of packet loss has been successfully addressed in this paper by the insertion of random values. With the insertion of random values, the ERLE increases by 4.81 dB compared to when there is no insertion of random value. The PBEC with the utilization of random values would make the VoIP a better communication tool.