A framework for network quality monitoring in the VoIP environment

Monitoring speech quality in Voice over IP (VoIP) networks is important to ensure a minimal acceptable level of speech quality for IP calls running through a managed network. Information such as packet loss, codec type, jitter, end‐to‐end delay and overall speech quality enables the network manager to verify and accurately tune parameters in order to adjust network problems. The present article proposes the deployment of a monitoring architecture that collects, stores and displays speech quality information about concluded voice calls. This architecture is based on our proposed MIB (Management Information Base) VOIPQOS, deployed for speech quality monitoring purposes. Currently, the architecture is totally implemented, but under adjustment and validation tests. In the future, the VOIPQOS MIB can be expanded to automatically analyze collected data and control VoIP clients and network parameters for tuning the overall speech quality of ongoing calls. Copyright © 2006 John Wiley & Sons, Ltd.     

Adaptive Speech Streaming Based on Packet Loss Prediction Using Support Vector Machine for Software‐Based Multipoint Control Unit over IP Networks

An adaptive speech streaming method to improve the perceived speech quality of a software‐based multipoint control unit (SW‐based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate‐narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW‐based MCU under various packet loss conditions in an IP network.     

A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

Jitter buffer plays an important role in Voice over IP (VoIP) applications because it provides a key mechanism for achieving good speech quality to meet technical and commercial requirements. The main objective of this paper is to propose a new, simple-to-use jitter buffer algorithm as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance, in terms of enhanced user-perceived speech quality and reduced end-to-end delay. Supported by signal processing features, the new algorithm, the so-called Play Late Algorithm, alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. The results show that the new algorithm achieves the best performance under different network conditions when compared to conventional static and adaptive jitter buffer algorithms. The results reported here are based on live tests and emulated network conditions on real mobile phone prototypes. The mobile phone prototypes use AMR codec and support full IP/UDP/RTP stack with IPSec function in some of the tests. The method for perceived speech quality measurement is based on the ITU-T standard for speech quality evaluation (PESQ).

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.     

基于AMR的抗分组丢失主次型语音编解码器

在基于分组传输的实时网络语音通信中,分组丢失不可避免,这也造成了语音质量的恶化.本文提出了一种基于AMR的主次型语音编解码方法,可以有效减少分组丢失对语音质量的影响.仿真实验证明,这种方法明显改善了IP网络语音通信的质量.     

基于PXA255的VoIP语音传输系统研究

文章研究IP语音传输系统的总体架构,实现了一个基于PXA255处理器的嵌入式IP电话终端硬件平台,为该平台建立了一个优化的嵌入式Linux环境,并研究基于GSM 06.10语音编解码实现,设计了一个IP语音实时传输系统,实现了IP语音的网络实时传输功能.     

Transport architecture evolution in UMTS/IMT‐2000 cellular networks

Mobile evolution from the second generation (2G) to the third generation (3G) raises several important questions for operators and manufacturers. How to ensure that the old and current investments can still be utilized in the future? What is the optimum architecture? ATM or IP? Voice or data? There is no single correct answer to these questions, as it all depends on individual cases. In this paper, we discuss the transport architecture evolution for the universal mobile telecommunications system (UMTS)/international mobile telecommunications—year 2000 (IMT‐2000), or 3G cellular networks and interworking aspects between 2G and 3G cellular networks. The interfaces between access nodes in a cellular network and the changes incorporated to support packet data services are described. Emerging services such as mobile data, virtual private networks (VPN) and location aware networking are described. Role of ATM and IP in this new transport architecture is presented. Control and data plane interworking issues between different transport technologies are described. The new ATM standard, ATM adaptation layer type 2 (AAL2) and its applicability for transporting compressed speech in an ATM based cellular network is described. A similar approach in IP, multiplexing in real‐time transport protocol (RTP) payload to transport compressed speech on selective interfaces of 3G network, is introduced. Transport network architecture evolution within four different scenarios is evaluated. Special interest is focused on the protocol stacks and flexible layered solutions that allow smooth migration from one transport technology to another. Copyright © 2000 John Wiley & Sons, Ltd.     

A New Method of Reducing Assembly Delay of Speech Packets

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声码器通用硬件平台的实现

基于低速率语音编解码算法实时实现的需求和IP电话终端以及小型IP电话网关的需求,设计开发了通用语音处理平台,实时实现了320b／s高质量极低速率语音编解码系统,通过测试得到了和定点优化后的语音算法完全相同的结果。     

Rate adaptive speech coding for universal multimedia access

This article reviews state-of-the-art in transport adaptation techniques for mobile networks. It discusses the mechanisms for rate adaptation to combat quality degradations of speech caused by the radio links. It begins with a review of dynamic schemes for adaptation of speech encoders in cellular networks where we observe two distinct approaches to rate adaptation: network controlled and source controlled. The issues associated with adaptive voice over IP (VoIP) mechanisms are considered next. Here, the encoder detects some form of network congestion to judge how to behave itself for the good of the network. It is noted that this altruistic behavior will only benefit coordinated IP networks such as private intranets and its application to the public Internet is improbable.     

基于RTP协议的IP电话QoS监测及提高策略

介绍了多媒体通信的发展趋势和当前存在的问题,对基于RTP协议的网络电话中音频数据传输技术进行了研究,对影响实时传输质量QoS的典型因素进行了分析。在局域网的环境下进行了语音包分析实验,探讨了基于RTP协议的QoS动态监测方法,并提出可行的改进方案。     

基于C/S模型的视频信息传输系统研究

讨论了IP网络与C／S模型，并对多媒体网络的传输协议作了分析，借助RTP／RTCP协议来确保视频信息传输的质量，采用C／S模型，在Windows2000环境下，设计了一个支持IP组播的视频信息传输系统，用它能够实现IP网络上的实时视频信息传输。     

基于NGN网络语音质量测量的探讨

基于IP承载网络和软交换技术的下一代网络（NGN）有着很大的发展趋势,NGN网络的基本业务是语音业务,影响语音质量的因素是由多个方面决定的,主要包括时延、丢包、抖动等。语音质量的好坏直接影响用户对运营商的选择,因此对NGN网络语音服务质量进行有效的分析和测量是十分重要的。     

基于TMS320C5402的IP语音通信系统的设计

介绍了基于DSP技术的IP语音通信系统的设计。详细论述了系统结构以及音频接口模块、网络接口模块的设计方法。针对软件结构,给出了G.729语音编码以及UDP传输协议的选择与在TMS320C5402上的实现。     

移动软交换核心网IP化实现

IP技术的发展给移动软交换网的改进提供了契机,运营商为提高竞争实力,均进行了移动软交换的IP化改革。文中通过规划IP地址、组建话路网络以及革新网络架构3种方法,探究了对新兴软交换核心网进行IP化的可行性及改造方法。     

Sprachübertragung über alternative Netze

The importance of multimedia can not be foreseen. However the combined transport of speech and broadband data will rapidly develop. Voice/data integration is the technology of the future and will more and more substitute the traditional way of information transport. The paper introduces UTA as the alternative to the incumbant operator und describes UTAs network infrastructure, SDH-backbone and broadband multiservice network. Three methods of voice/data integration are presented:

1. voice over ATM
2. voice over Frame Relay
3. voice over IP.

     

A Wavelet Based Hybrid Threshold Transform Method for Speech Intelligibility and Quality in Noisy Speech Patterns of English Language

The paper proposes a method to improve the performance of speech communication system in a highly noisy industrial environment. For the improvement, different speech signals are considered which includes signals from different environments such as car noise, railway station, babble noise, street noise which are corrupted with additional noise as input data set for processing. This database is processed using suitable filters which will remove the effect of noise to some extent. Different algorithms have been proposed to minimize the effect of noise to a certain limit. The denoising algorithms are generally the different wavelet thresholding method which removes the noise from the speech signal. Many researchers have worked on soft and hard thresholding for image processing. The proposed method of hybrid thresholding comprises of both soft and hard thresholding process which is comparatively better method than the previous methods. The method can be implemented for the non-stationary noise and it also removes the problems of edges. Unlike the traditional way of using single value, different values are used for the adaptive filtering to remove the edges. During the course of experiments, the dataset of IIIT-H with a set of noisy files from Noizeus and AURORA database having sampling rate of 16 kHz has been used. Results are calculated with subjective and objective measures for fine and broad level quality assessment. SNR, SSNR, PSNR, NRMSE, and PESQ parameters are used as performance parameters and outperform with other combinations as compared to conventional methods. The hybrid threshold method yields better results with significant improvement in speech quality and intelligibility.

     

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.     

Privileged Treatment of UMTS Subscribers in GSM Networks

Different user segments have various requirements and expectations towards the performance of mobile networks. Subscribers having experienced the high quality of UMTS networks desire to maintain high speech quality and excellent data throughput also in areas of missing UMTS but existing GSM coverage. In GSM networks a privileged treatment of UMTS subscribers by means of proper resource allocation provides a substantial quality improvement with respect to standard GSM subscribers. This strategy allows network operators to reduce the performance gap between both network areas experienced by UMTS subscribers. A detailed study on the performance of circuit switched speech and packet data services has been performed based on system level simulations. The results show significant speech quality advantages for users with dual-RAT terminals compared to standard GSM users as well as notably higher data throughput rates.     

基于TCP/IP的说话人确认系统

随着网络与语音信号处理技术的快速发展,把说话人识别系统应用于Internet,使其作为身份识别的一种方法是势在必行。文中介绍了一个基于TCP/IP的实时说话人确认系统,它基于C/S(客户/服务器)模型,采用TCP/IP,以期能够实现Internet上的语音登录系统。介绍了该系统的框架及具体算法,给出了实验结果及其分析。