Quality of experience (QoE) driven adaptation scheme for?voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.     

VoIP语音时延的分析和研究

文章介绍了VoIP(IP网络上传送语音)语音质量的测试方法,分析了影响VoIP语音质量的主要因素:延迟、抖动、丢包率和时延.利用E模型定量地分析了语音质量与端到端时延的关系,通过建立数学模型,指出了VoIP 系统中主要的时延分量,并研究了这些时延分量产生的机理和影响它们的参数.在设计实际的VoIP系统时,可以通过优化影响时延分量的主要参数,改善VoIP系统的时延.     

An analytical simulator for deploying IP telephony

Deploying IP telephony or voice over IP (VoIP) is a major and challenging task. This paper describes an analytical design and planning simulator to assess the readiness of existing IP networks for the deployment of VoIP. The analytical simulator utilizes techniques used for network flows and queuing network analysis to compute two key performance bounds for VoIP: delay and bandwidth. The simulator is GUI‐based and has an interface with drag‐and‐drop features to easily construct any generic network topology. The simulator has an engine that automates and implements the analytical techniques. The engine determines the number of VoIP calls that can be sustained by the constructed network while satisfying VoIP QoS requirements and leaving adequate capacity for future growth. As a case study, the paper illustrates how the simulator can be utilized to assess the readiness to deploy VoIP for a typical network of a small enterprise. We have made the analytical simulator publicly available in order to improve and ease the process of VoIP deployment. Copyright © 2008 John Wiley & Sons, Ltd.     

基于VoIP的车内话音通信系统的设计

文章对VoIP技术进行了研究,分析了VoIP的技术原理及与电路交换相比具有的优势,比较了VoIP两种体制ITU-U的H．323和IETF的SIP的优劣。在此基础上根据车内通信系统发展的现状提出基于VoIP的设计方案。给出了系统体系架构,以及话音综合接入设备的参考设计,为未来多业务终端接人的车内话音通信系统的应用提供了新思路。     

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。     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

An enhanced SIP proxy server for wireless VoIP in wireless mesh networks

The wireless mesh network (WMN) has emerged recently as a promising technology for next-generation wireless networking. In WMNs, it is important to provide high quality multimedia service in a flexible and intelligent manner. To address this issue in this article, we study the Session Initiation Protocol (SIP) for wireless voice over IP (VoIP) applications. Especially, we investigate the technical challenges in WMN VoIP systems and propose a design of an enhanced SIP proxy server to overcome them. An analysis of the signaling process and a study of simulation results have shown the advantages of our proposed approach.     

Mobility management for VoIP in 3G systems: evaluation of low-latency handoff schemes

The introduction of IP-based real-time services in next-generation mobile systems requires coupling mobility with quality of service. The mobility of the node can disrupt or even intermittently disconnect an ongoing real-time session. The duration of such an interruption is called disruption time or handover latency, and can heavily affect user satisfaction. Therefore, this delay needs to be minimized to provide good quality of VoIP services. In this article, we focus on network-layer mobility and mobile IP since it is a natural candidate for providing such mobility. We evaluate different low-latency schemes based on mobile IP and compare their performances in terms of disruption time for VoIP services. Low-latency handoffs are performed by anticipating and/or postponing the mobile IP registration process. With these methods, disruption time is reduced to 200 ms in most considered cases.     

Speex中语音增强算法的仿真与评估

随着VoIP技术在近年来的快速发展,互联网语音通信得到了越来越多的应用,IP电话的语音通信质量成为制约其发展的重要因素,因此应用了许多新技术使其语音质量更符合用户的要求。在VoIP系统中使用语音增强算法,可以消除发送端噪声干扰,提高通话质量。研究的Speex算法中的STSA-MMSE语音增强算法是在计算先验和后验信噪比的基础上估计噪声,再通过计算得出一个增益值,作用于带噪语音,对噪声段进行抑制,对叠加了噪声的语音尽可能分离噪声,最终达到增强语音清晰度,提高语音舒适度的目的。     

VoIP over DVB-RCS with QoS and bandwidth on demand

Motivated by the need for compliance/interoperability above the satellite-specific layers, this article proposes a consolidated approach for voice over IP over satellite networks based on the ETSI DVB-RCS standard. Voice communication is a real-time service that needs priority over other services in IP environments with limited bandwidth, such as IP satellite networks. Bandwidth utilization in such networks needs to be optimized in order to reduce service costs, and this requires the use of dynamic bandwidth allocation schemes. This article therefore addresses the role of bandwidth on demand in the optimization of bandwidth allocation for VoIP and assesses the impact of BoD mechanisms on voice quality. The trade-off between voice quality and bandwidth efficiency is investigated under different DVB-RCS-specific capacity request/allocation strategies, and it is demonstrated that DVB-RCS provides an efficient platform for integrated support for a variety of VoIP applications over satellite. The main contribution of this article consists of the identification of the mechanisms capable of responding to the key challenges raised by the VoIP application in the satellite environment.     

VoIP Performance over HSPA with Different VoIP Clients

High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network).     

Mobility management for VoIP service: Mobile IP vs. SIP

Wireless Internet access has gained significant attention as wireless/mobile communications and networking become widespread. The voice over IP service is likely to play a key role in the convergence of IP-based Internet and mobile cellular networks. We explore different mobility management schemes from the perspective of VoIP services, with a focus on Mobile IP and session initiation protocol. After illustrating the signaling message flows in these two protocols for diverse cases of mobility management, we propose a shadow registration concept to reduce the interdomain handoff (macro-mobility) delay in the VoIP service in mobile environments. We also analytically compute and compare the delay and disruption time for exchanging signaling messages associated with the Mobile IP and SIP-based solutions.     

软交换中基于VoIP语音识别的业务部署方案

本文提出了一套基于VoIP语音捕获及其识别的业务部署方案,该方案可以在不改变现有通信网络结构和业务部署的情况下,利用语音识别技术,继承或扩展非IP网的通信业务,也可以在IP网中部署新的增值业务。实验环境下的验证结果表明,该方案具备实际部署的可行性。     

Integrated management architecture for IP-based networks

IP telephony will bring about a dramatic change in the way IP services are planned, provisioned, managed, and billed. In order to build and retain a strong customer base for these new services, service providers need to meet, if not exceed, the customer expectations set by today's traditional voice services. Acceptance of IP telephony will depend on the quality and efficiency with which service providers offer, deliver, and manage IP services. Installation, configuration, and activation must be rapid and error-free. Furthermore, customers will want direct control over the reconfiguration of services and real-time visibility into the impact change has on their operating costs. Once the service is activated, customers will want the provider to guarantee service quality as defined by industry standards. Corporate customers in particular will need to be assured that the provider is proactively monitoring performance to avoid problems and providing them visibility into the performance data collected. This article discusses an integrated management support system for IP-based networks illustrating the functions needed to support the unique challenges of managing VoIP services. An example of a service management system is also described     

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.     

基于H.323的VoIP系统QoS实现研究

文章提出了一种QoS增强的VoIP体系结构，增加了连接准入控制功能，提供了语音分组的IP优先服务，并改进了抖动缓冲的设计，使VoIP系统的QoS得到很大改善。     

WLAN中基于MIPv6的移动VoIP切换性能仿真分析

移动IP是一个在Internet上基于网络层提供移动性支持功能的要求较高的VoIP业务,切换延迟将直接影响到话音质量,严重时甚至会中断正在进行的会话.文章借助ns2网络模拟器仿真分析了WLAN中基于MIPv6的移动VoIP切换性能.结果表明,MIPv6及其扩展协议的切换性能优劣顺序依次为:F-HMIPv6、FMIPv6、HMIPv6、MIPv6.尤其是F-HMIPv6协议,无论端到端延迟还是切换延迟,都得到了最大的改善.所得结论能为网络切换性能的进一步优化提供重要依据.     

Voice over the internet: A tutorial discussing problems and solutions associated with alternative transport

This article provides a tutorial overview of voice over the Internet, examining the effects of moving voice traffic over the packet switched Internet and comparing this with the effects of moving voice over the more traditional circuit-switched telephone system. The emphasis of this document is on areas of concern to a backbone service provider implementing Voice over IP (VoIP). We begin by providing overviews of the Plain Old Telephone Service (POTS) and VoIP. We then discuss techniques service providers can use to help preserve service quality on their VoIP networks. Next, we briefly discuss Voice over ATM (VoATM) as an alternative to VoIP. Finally, we offer some conclusions.     

Implementing a secure VoIP communication over SIP-based networks

Recent years the Session Initiation Protocol (SIP) is commonly used in establishing Voice over IP (VoIP) calls and has become the centerpiece for most VoIP architecture. As wireless and mobile all-IP networks become prosperous, free VoIP applications are utilized in all places. Consequently, the security VoIP is a crucial requirements for its adoption. Many authentication and key agreement schemes are proposed to protect the SIP messages, however, lacking concrete implementations. The performance of VoIP is critical for users’ impressions. In view of this, this paper studies the performance impact of using key agreements, elliptic curve Diffie–Hellman and elliptic curve Menezes–Qu–Vanstone, for making a SIP-based VoIP call. We evaluate the key agreement cost using spongycastle.jce.provider package in Java running on android-based mobile phones, the effect of using different elliptic curves and analyze the security of both key agreements. Furthermore, we design a practical and efficient authentication mechanism to deploy our VoIP architecture and show that a VoIP call can be established in an acceptable interval. As a result, this paper provides a concrete and feasible architecture to secure a VoIP call.     

VoWLAN通信平台的设计与实现

VoWLAN也叫VoWiFi或者WiFi VoIP。它是基于无线网络技术和VoIP网络,是两者的有机结合。即是通过WLAN提供VoIP业务,使得终端用户通过WLAN拨打IP电话成为现实。本文提出了在基于Linux操作系统的SIP应用服务器及VoIP网关中,如何通过SIP信令和传统的PSTN数据通信线路与无线网络无缝连接方案,从而实现IP网络与传统电话间的实时语音通信、电话会议、语音信箱、视频通信、短消息、数据传输等业务。本设计已成功应用于某企业的实时语音通信平台,获得良好的效果。