Subjective MOS model and simplified E-model enhancement for Skype associated with packet loss effects: <Emphasis Type="Italic">a case using conversation-like tests with Thai users</Emphasis>

This paper proposes two mathematical models that can be used to estimate VoIP quality from Skype, which is one of the most popular VoIP applications. The first model is simple, it has been developed using data from the informal interview tests called Conversation-like tests, referring to packet loss of 0 %, 5 %, 10 %, …, and 30 %. The tests have been conducted with Skype using a non ITU-T’s codec called SILK via the Internet with over 180 native Thai participants, while packet loss effects were generated using a network emulation tool. The second model is called the Enhanced Simplified E-model, this has been developed by adding the Thai Bias factor into a generic Simplified E-model, which calculates by subtracting the subjective results from the computed results using the Simplified E-model formula. After obtaining the models, they were evaluated with the Test set from 36 native Thai participants (different from the other group of participants) using Mean Absolute Percentage Error technique (MAPE). It has been found that VoIP quality measurement performance of both models are classified as excellent and provide higher reliability and accuracy than the Simplified E-model. Subjective MOS model and Enhanced Simplified E-model error reduction compared to the simplified one was at about 21.9 % and 21.2 % respectively, which is the major contribution of this work.     

MOS estimation model development using ACR listening-opinion tests with Thai users referring to loss effects: a case of G.726 and G.729

This paper proposes two models of Mean Opinion Score (MOS) estimation based on Thai users and the Thai language, referring to packet loss effects, for G.726 and G.729 codecs. Based on Thai users and Thai speech referring to packet loss effects in this work, the Absolute Category Rate (ACR) listening tests were conducted with 89 participants and 107 participants for the MOS estimation model development of G.726 and G.729 respectively, while the same tests were conducted with totally 60 participants for the model evaluation of both codecs. Packet loss rates were 0–15% for G.726 with 5 test conditions and G.729 with 6 test conditions; each condition was conducted with at least 16 participants. After gathering the data, the MOS estimation models for both codecs were simply created and then evaluated with the test sets, comparing Perceptual Evaluation of Speech Quality (PESQ), a popular measurement method. For one of the contributions of this study, after the models were evaluated using Mean Absolute Percentage Error (MAPE), it was found that the proposed models for G.726 and G.729 provided better performance than PESQ, particularly by reducing the MAPE by about 30% and 17% respectively, compared to PESQ.     

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。     

Voice quality prediction models and their application in VoIP networks

The primary aim of this paper is to present new models for objective, nonintrusive, prediction of voice quality for IP networks and to illustrate their application to voice quality monitoring and playout buffer control in VoIP networks. The contributions of the paper are threefold. First, we present a new methodology for developing perceptually accurate models for nonintrusive prediction of voice quality which avoids time-consuming subjective tests. The methodology is generic and as such it has wide applicability in multimedia applications. Second, based on the new methodology, we present efficient regression models for predicting conversational voice quality nonintrusively for four modern codecs (G.729, G.723.1, AMR and iLBC). Third, we illustrate the usefulness of the models in two main applications - voice quality prediction for real Internet VoIP traces and perceived quality-driven playout buffer optimization. For voice quality prediction, the results show that the models have accuracy close to the combined ITU PESQ/E-model method using real Internet traces (correlation coefficient over 0.98). For playout buffer optimization, the proposed buffer algorithm provides an optimum voice quality when compared to five other buffer algorithms for all the traces considered.     

Vo IP 的语音质量分析与控制

分析了VoIP语音质量的影响因素，通过E模型定量地描述了语音质量与端到端延迟和丢包率的关系。为了控制VoIP的语音质量，计算出VolP系统在各种情况下的语音质量极限，提出一种自适应编码和分组封装的控制策略。将该方法应用于自行开发的IP电话网关，实际测试证明能在很大程度上提高VoIP的语音质量。     

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。     

骨干网中 VoIP语音质量的快速评估方法

提出一种基于被动模式流量分析的 VoIP语音质量评估方法 ,以帮助 VoIP运营商对主干网中大量并发 VoIP会话的通话质量进行实时监测。在流量采集的基础上 ,通过流跟踪算法获取 VoIP会话的丢包、时延等基本性能指标 ;通过快速评估模型对 VoIP会话的 R和 MOS值进行计算。实验证明 ,该方法性能较高 ,评估结果准确。     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

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.     

Objective Assessment Methodology for Estimating Conversational Quality in VoIP

Voice over IP (VoIP) is becoming one of the key technologies for telecommunications. Since IP networks generally do not guarantee transmission quality, it is extremely important to design and manage the quality of service (QoS) properly. To do this, it is desirable to develop an objective quality assessment method that estimates subjective quality based on the physical characteristics of the VoIP system. This paper first proposes a framework of objective models that can be applied not only to quality planning, which is an intended application of the existing standard methodology known as International Telecommunication Union—Telecommunication Standardization Sector (ITU-T) Recommendation G.107, “the E-model,” but also to quality benchmarking and management. Then, it proposes a model that complies with the proposed framework. Experimental results show that the proposed model has sufficient accuracy in the evaluation of practical VoIP systems. In addition, we attempt to integrate the opinion model with other objective quality measures, such as perceptual evaluation of speech quality (PESQ), standardized in Recommendation P.862 in ITU-T. Finally, we examine the task dependence of the performance of the proposed model.     

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。     

IEEE 802.11b无线局域网的VoIP通信容量计算及分析

针对用IEEE 802.11b网络建立低成本的无线VoIP网络进行分析研究,本文评估在不同的时延限制、信道质量参数和语音音质指标下运载语音通话的能力,并比较G.711和G.729两种语音编码方案下不同的语音数据分组长度时的效果,为在无线环境下以IP方式构造移动语音通信网络奠定了良好的基础.     

Adaptive VoIP playout scheduling: assessing user satisfaction

Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics.     

Time synchronization for VoIP quality of service

Various approaches seek to optimize the quality of service of VoIP applications. We propose a system that uses synchronized time to combine the useful characteristics of both fixed and adaptive buffer strategies, thereby improving VoIP quality of service. Using a combination of global positioning system (GPS) technologies and the network time protocol (NTP), hosts can learn the precise end-to-end delay for each packet. This information can benefit both domestic and business Internet telephony users. We outline our proposed system and discuss issues arising from the use of synchronized time     

无线VoIP系统语音加密算法的研究

随着无线网络的飞速发展和VoIP技术的日渐成熟,无线VoIP技术应运而生。由于无线网络安全的脆弱性和VoIP系统本身的安全问题使得无线VoIP系统在安全方面存在着各种隐患。为提高无线VoIP的安全性,采用高压缩率的语音编解码技术G.729提高无线VoIP的通话质量;采用高级加密标准(AES算法)加解密采用G.729压缩过的语音信息;选用椭圆曲线密码(ECC算法)传输AES算法中用到的会话密钥;利用混沌系统可以提供可重复的随机数序列且其序列仅与系统参数和初值有关的性质确保了会话密钥的保密性。     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

Proposed modifications in ITU-T G.729 8 kbps CS-ACELP speech codec and its overall performance analysis

This paper proposes modification in the transmission of excitation codevector and its non-zero pulse sign magnitude using “codebook partition and label assignment” approach, which in turn reduces the number of bits required to transmit it through the communication channel in legacy CS-ACELP 8 kbps speech codec. The proposed approach uses the excitation codebook structure of forward mode standard G.729E 11.8 kbps with two non-zero pulses per track which avoids the use of two algebraic codebook structure for forward mode as well as for backward mode of G.729E with least significant pulse replacement approach for finding optimized excitation codevector. Proposed modification in legacy 8 kbps CS-ACELP (80 bits/10 ms) speech codec actuates the bit rate of 10.6 kbps (106 bits/10 ms) with a better objective and subjective analysis in stark contrast with legacy 8 kbps CS-ACELP speech coder and also avoids the switching of codebook modes of standard 11.8 kbps (G.729E) CS-ACELP speech coder. This paper also aims to propose the reduction in the number of searches in the final codevector of excitation structure by considering initial codevector as a final codevector which improves the quality of the speech compared to the output speech quality of legacy G.729 CS-ACELP working at 8 kbps. Both legacy CS-ACELP 8 kbps speech codec and proposed CS-ACELP 10.6 kbps are implemented in MATLAB. Subjective and objective analysis are carried out on a proposed CS-ACELP 10.6 kbps speech codec in order to evaluate its performance and the results obtained are then cross- compared with the results of legacy CS-ACELP (8 kbps) using set of tables and graphs. It is evident from obtained results that both PESQ and MOS scores are quite comparable for each set of wave files even though bitrates are reduced. Consistency and efficiency of proposed algorithm is assured by calculating the population mean of 95% confidence interval based on obtained objective and subjective parameter results.     

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.     

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。