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。     

ITU-T standardization activities for interactive multimedia communications on packet-based networks: H.323 and related recommendations

The Telecommunication Sector of the International Telecommunication Union (ITU-T) has developed a series of recommendations together comprising the H.323 system that provides for multimedia communications in packet-based (inter)networks. This series of recommendations describe the types and functions of H.323 terminals and other H.323 devices as well as their interactions. The H.323 series of recommendations includes audio, video and data streams, but an H.323 system minimally requires only an audio stream to be supported. Motivated by straightforward interoperability with the ISDN and PSTN networks and a variety of other protocols, the recommendation H.323 has been accepted as being the standard for IP telephony, developed by the ITU-T and broadly backed by the industry—which is also adopted by both the Voice over IP (VoIP) forum and the European Telecommunication Standards Institute (ETSI). This paper presents an overview of the H.323 system architecture with all its functional components and protocols and points out all the related specifications.     

基于Internet的视频会议系统的设计与实现

桌面视频会议是近年来网络应用的重点,国际电信联盟(ITU)也为基于分组交换网络(PBN)的多媒体会议系统制定了H.323建议。论文介绍了一个依据ITU-TH.323建议开发的、适用于TCP/IP网络的多点视频会议系统。在该系统中,用软件实现了MCU,大大降低了系统成本。该系统使用C/S结构和集中式的控制方式,适合在Internet上使用。测试结果表明,该系统具有较高的实用价值。     

A Methodology for Deriving VoIP Equipment Impairment Factors for a Mixed NB/WB Context

This paper proposes a novel approach to quantifying the quality degradation of Voice over IP (VoIP) telephony in the presence of codec and network-related impairments. This approach differs from the baisc ITU-T E-Model for VoIP quality estimation in that it addresses mixed narrowband/wideband scenarios. It makes novel use of instrumental models and symbolic regression via Genetic Programming (GP) to enable the evolution of degradation models from a modest set of initial parameters. Here, a two-step approach has been used. First, values of impairment factors are derived using WB-PESQ as a reference model. Secondly, a GP based symbolic regression approach has been utilized to automatically evolve the functional form of equipment impairment factors from a set of variables. Very few a priori assumptions are made about the model structure. The effectiveness of the approach is demonstrated by a number of generated models which compare favorably with WB-PESQ and outperform the traditional E-Model in terms of prediction accuracy when compared using WB-PESQ. A significant advantage of the approach is that new models are easily generated to account for continuing evolution of the VoIP standards.      

Using cross-layer metrics to improve the performance of end-to-end handover mechanisms

Network centric handover solutions for all IP wireless networks usually require modifications to network infrastructure which can stifle any potential rollout. This has led researchers to begin looking at alternative approaches. Endpoint centric handover solutions do not require network infrastructure modification, thereby alleviating a large barrier to deployment. Current endpoint centric solutions capable of meeting the delay requirements of Voice over Internet Protocol (VoIP) fail to consider the Quality of Service (QoS) that will be achieved after handoff. The main contribution of this paper is to demonstrate that QoS aware handover mechanisms which do not require network support are possible. This work proposes a Stream Control Transmission Protocol (SCTP) based handover solution for VoIP called Endpoint Centric Handover (ECHO). ECHO incorporates cross-layer metrics and the ITU-T E-Model for voice quality assessment to accurately estimate the QoS of candidate handover networks, thus facilitating a more intelligent handoff decision. An experimental testbed was developed to analyse the performance of the ECHO scheme. Results are presented showing both the accuracy of ECHO at estimating the QoS and that the addition of the QoS capabilities significantly improves the handover decisions that are made.     

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.     

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。     

Non-intrusive speech quality assessment using several combinations of auditory features

Quality estimation of speech is essential for monitoring and maintenance of the quality of service at different nodes of modern telecommunication networks. It is also required in the selection of codecs in speech communication systems. There is no requirement of the original clean speech signal as a reference in non-intrusive speech quality evaluation, and thus it is of importance in evaluating the quality of speech at any node of the communication network. In this paper, non-intrusive speech quality assessment of narrowband speech is done by Gaussian Mixture Model (GMM) training using several combinations of auditory perception and speech production features, which include principal components of Lyon’s auditory model features, MFCC, LSF and their first and second differences. Results are obtained and compared for several combinations of auditory features for three sets of databases. The results are also compared with ITU-T Recommendation P.563 for non-intrusive speech quality assessment. It is found that many combinations of these feature sets outperform the ITU-T P.563 Recommendation under the test conditions.     

Improving the transcoding capability of speech coders

With the trend of merging various communication networks, a need arises to provide transcoding between different speech coding formats. Presently this means a cross tandem between the two coders in each case. This results in both quality loss and extra delay. A possible alternative is using a bitstream mapping approach that directly converts parameter values. For several standard coders having a similar coding structure, it should be possible to generate comparable or better quality without adding much delay or complexity. This paper proposes a bitstream mapping method between ITU-T Recommendation G.729 and TIA IS-641. Informal listening tests and the perceptual subjective quality measure (PSQM) scores show that the proposed method has better quality than the cross tandem method, while it has at least 5 ms less delay and six times less computation.     

VoIP quality measurement: subjective VoIP quality estimation model for G.711 and G.729 based on native Thai users

This paper presents a mathematical model that has been created from the subjective MOS, instead of modifying or improving the existing objective measurement methods (e.g., E-model) for VoIP quality measurement. The proposed model of VoIP quality measurement method is based on native Thai users who communicate to each other using Thai language, which is a tonal language, unlike English and most western languages. The data have been gathered using conversation-opinion tests with 400 and 354 native Thai subjects for two popular codecs, G.711 and G.729, respectively, referring to effects from two major network factors, packet loss and packet delay. This model is called the Thai subjective VoIP quality evaluation model (ThaiVQE). It has been evaluated using two test sets of subjective MOS, from 50 native Thai subjects for G.711 and 64 native Thai subjects for G.729, then the results have been compared with the E-model results. Based on native Thai users, the evaluation result surprisingly shows that ThaiVQE can contribute better accuracy and reliability than the standard E-model with error reduction of over 13 % for G.711 and 28 % for G.729. Therefore, this is an example study for other countries that have their own languages and cultures to create their subjective MOS model.     

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。     

提高VoIP通信服务质量的策略研究

基于IP技术的语音信号分组传输((voiceoverIP,VoIP)电话目前被广泛使用。但是IP电话在Internet传输信号过程中,通话质量还存在回声、抖动、断续、掉线等许多问题。本文分析了影响通话效果的主要因素,并提出引入基于路由模块的多目标处理策略,仿真结果表明提高了IP语音通信服务质量(Quality-of-Service,QOS)。     

VoIP中为提高语音质量所采用的关键技术

VoIP(Voice over IP)即IP电话,是将话音编码、压缩转换成数据包,在IP网络中进行传输的技术。为应对传统电话公司的竞争,IP电话的语音质量成为决定其未来命运的关键因素。该文首先介绍了几个界定QoS的参数和目前评价IP电话业务语音质量的三种模型：MOS模型、PSQM模型、E模型,然后重点介绍了在终端和网络上提高VoIP语音质量所采用的一些关键技术,应用于终端的技术中比较重要的是语音的编码与压缩、差错控制等,而应用于网络的技术则是解决IP QoS的两种基本模型：综合业务模型和区分业务模型。     

Adaptive partial-matching steganography for voice over IP using triple M sequences

Although steganographic transparency and steganographic bandwidth are believed to be two conflicting objectives in the design of steganographic systems, it is possible and necessary to strike an optimal balance between them. This paper presents an adaptive partial-matching steganography for voice over IP (VoIP). We introduce the notion of partial similarity value (PSV) to evaluate the partial matching between covers and secret messages. By properly setting a low threshold of PSV and a high threshold of PSV, we can adaptively balance steganographic transparency and bandwidth. Moreover, we employ triple m sequences to eliminate the correlation among secret messages, guide the adaptive embedding process, and encrypt synchronization signaling patterns. In addition, we introduce an improved strategy that takes into account the similarity between not only covers and encrypted messages but also covers and original messages. We evaluate the proposed approach and its improved strategy with ITU-T G.729a as the codec of the cover speech in StegVoIP that is a prototypical covert communication system based on VoIP and compare them with some existing approaches. The experimental results demonstrate that the proposed approaches can provide a better balance between steganographic transparency and bandwidth. Furthermore, the results of delay tests show that they adequately meet the real-time requirement of VoIP.     

H.323协议安全特性的研究

为了在H.323系统中确保多媒体通信的安全性,需要在H.323框架内使用ITU T的H.235建议所提供的身份认证、完整性检查和保密性等服务。本文首先描述H.235建议的主要功能,并分析其第三版中提供的新特性,然后提出了一个分层的H.235实现方案,最后介绍H.235建议目前真正的实现状况。     

The ITU-T Software Tool Library

The advancement of speech coding technology has led to new and improved standards. The Telecommunication Standardization Sector of the International Telecommunication Union (ITU-T) has been a leader in developing speech coding standards for telecommunications applications. To increase the efficiency of the coding standard selection process, a group called User's Group on Software Tools (UGST) was established in 1990 to produce a framework of speech coding-related software tools that would allow a common ground for activities across different organizations. Over the years, this group produced a high-quality software tool library (STL), which has been used in the definition of ITU and non-ITU standards, such as G.723.1, G.729, half-rate and enhanced full-rate GSM, among others. However, despite its usefulness, the ITU-T STL is largely unknown outside the telecommunications standards community. This paper introduces the ITU-T STL concepts and functionalities to the wider communications community.     

IP环境下ATN网间互联模型的设计与实现

为了解决传统的航空电信网（ATN）无法通过IP网络进行互联的问题,提出了一种通过IP专网实现各ATN域之间互联的ATN网间互联模型。该模型通过修改ATN网络层协议,使用IP 子网相关汇聚功能（IP SNDCF）实现了封装后的ATN数据包（CLNP包）在各ATN域与IP专网间的传输。最后,通过在一个已建立的ATN 试验网上传输数据,验证了该模型的可行性和有效性。     

Voice-quality monitoring and control for VoIP

Voice over IP offers important opportunities for the telecommunications market to deploy more advanced services, but it must overcome many obstacles. Users expect toll-quality voice, which calls for end-to-end quality of service (QoS) - a challenge for IP service providers. To make VoIP attractive to end users, the only feasible and directly implementable alternative is to deploy an efficient mechanism within the endpoints. To that end, the authors propose the scalable, modular, call quality monitoring and control framework for maintaining voice quality at acceptable levels over networks that don't offer QoS guarantees.     

What are suspicious VoIP delays?

Voice over IP (VoIP) is unquestionably the most popular real-time service in IP networks today. Recent studies have shown that it is also a suitable carrier for information hiding. Hidden communication may pose security concerns as it can lead to confidential information leakage. In VoIP, RTP (Real-time Transport Protocol) in particular, which provides the means for the successful transport of voice packets through IP networks, is suitable for steganographic purposes. It is characterised by a high packet rate compared to other protocols used in IP telephony, resulting in a potentially high steganographic bandwidth. The modification of an RTP packet stream provides many opportunities for hidden communication as the packets may be delayed, reordered or intentionally lost. In this paper, to enable the detection of steganographic exchanges in VoIP, we examined real RTP traffic traces to answer the questions, what do the “normal” delays in RTP packet streams look like? and, is it possible to detect the use of known RTP steganographic methods based on this knowledge?