VoIP insecurity

As voice over Internet Protocol (VoIP) moves into the mainstream in the UK, and new IP-based services such as IIPTV become available, there is a serious need to secure these systems. With the public switched telephone network (PSTN), users at all levels have peace of mind that the lines are not only reliable and available, but highly secure. A proactive security strategy which addresses security at multiple levels and understands the unique nature of telecommunication networks is critical to enable organizations and service providers to maximize the true potential of VoIP technology, create cost savings and deliver highly available and reliable services to their customers and employees     

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.     

Talk about a change [VoIP replacing PSTN]

The rapid rise of Internet telephony services means that existing telecoms networks could soon be redundant. This paper examines the technological and financial issues concerning voice over Internet protocol (VoIP) and the probability of it substantially replacing existing public switched telephone networks (PSTN) within the next few years.     

A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

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。     

VoIP reliability: a service provider's perspective

Voice over IP services offer important revenue-generating opportunities, as well as many technical challenges in providing high-quality services. Users have come to expect highly available telecommunications services with high-quality voice. Service providers need reliable high-performance networks to meet user expectations, and must be able to guarantee performance and reliability to their customers. In converged voice and data networks, the network infrastructure must deliver very high quality and availability for some customer needs, while also providing low-cost high-capacity bandwidth for other needs. The use of quality of service mechanisms to provide prioritization for various traffic types is a key element needed for voice and data network convergence. However, it is not sufficient if the underlying networks are unreliable. The focus of this article is to address the reliability aspects of VoIP services, including the underlying IP networks.     

Low‐cost and high‐performance: VoIP monitoring and full‐data retention at multi‐Gb/s rates using commodity hardware

Voice over IP (VoIP) is increasingly replacing the old public switched telephone network (PSTN) technology. In this new scenario, there are several challenges for VoIP providers. First, VoIP requires a detailed monitoring of both users' quality of service (QoS) and experience (QoE) to a greater extent than in traditional PSTNs. Second, such a monitoring process must be able to track VoIP traffic in high‐speed networks, nowadays typically of multi‐Gb/s rates. Third, recent government directives require that providers retain information from their users' calls. Similarly, the convergence of data and voice services allows operators to provide new services such as full‐data retention, in which users' calls can be recorded for either quality assessment (call centers, QoE) or security purposes (lawful interception). This implies a significant investment in infrastructure, especially in large‐scale networks which require multiple points of measurement and redundancy. This paper proposes a novel methodology, architecture and system to fulfill such challenges, called VoIPCallMon, as well as the data structures and necessary hardware‐tuning knowledge for its development. As distinguishing features, VoIPCallMon provides very high performance, being able to process VoIP traffic on‐the‐fly at high bitrates, novel services and significant cost reduction by using commodity hardware with minimal interference with operational VoIP networks. The performance evaluation shows that the system copes with the VoIP load of real‐world operators. We further evaluated the system performance at a fully saturated 10 Gb/s link and no packet loss was reported, therefore demonstrating the potential of commodity hardware solutions. Copyright © 2014 John Wiley & Sons, Ltd.     

基于ATM AAL-2的话音业务

电信网正朝着网络融合和业务综合的方向发展,要实现宽带网与现有窄带电信网络的有效融合,就要求宽带网能够很好地支持现有窄带网络或PSTN中的话音业务,实现话音的包传输与交换。ATM论坛最新通过的基于AAL-2承载话音业务的标准,为实现ATM宽带网与窄带网的融合提供了一个新的途径。本文根据ATM论坛基于AAL-2的话音中继标准,介绍AAL-2话音业务的原理,并给出基于AAL-2话音业务的网络融合方案。     

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。     

Security Challenge and Defense in VoIP Infrastructures

Voice over Internet protocol (VoIP) has become a popular alternative to traditional public-switched telephone network (PSTN) networks that provides advantages of low cost and flexible advanced ldquodigitalrdquo features. The flexibility of the VoIP system and the convergence of voice and data networks brings with it additional security risks. These are in addition to the common security concerns faced by the underlying IP data network facilities that a VoIP system relies on. The result being that the VoIP network further complicates the security assurance mission faced by enterprises employing this technology. It is time to document various security issues that a VoIP infrastructure may face and analyze the challenges and solutions that may guide future research and development efforts. In this paper, we examine and investigate the concerns and requirements of VoIP security. After a thorough review of security issues and defense mechanisms, we focus on attacks and countermeasures unique to VoIP systems that are essential for current and future VoIP implantations. Then, we analyze two popular industry best practices for securing VoIP networks and conclude this paper with further discussion on future research directions. This paper aims to direct future research efforts and to offer helpful guidelines for practitioners.     

基于Cable Modem技术的VoIP话音解决方案

目前,一些大、中城市的有线电视公司已经拥有庞大的有线电视宽带用户群,而基于HFC网的VoIP系统正是目前在有线电视宽带网条件下综合接入语音、数据和多媒体业务的最佳解决方案之一,其中语音业务可利用中继媒体网关设备完成媒体流转换,并通过标准的E1协议接入PSTN网络,以达到充分利用HFC及PSTN网络资源迅速开展话音业务的目的.同时,作为NGN(下一代网络)中的标准部件,VoIP是面向未来、可持续发展的解决方案之一,在有线电视宽带网所及之处,可以为商业和家庭用户提供质优价廉的IP语音服务.     

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.     

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.     

Mobile IP-DECT internetworking architecture supporting IMT-2000applications

Realistic realization and mass acceptance of mobile data services require networking architectures offering acceptable quality of service and attractive tariffs. A novel strategy for this goal is maximum integration of popular data networking standards and their infrastructure into wireless networks. This article discusses a Mobile IP-based network architecture to provide IP services in DECT to support IMT-2000 applications. DECT offers micromobility within multicell subnets, while Mobile IP supports macromobility between multicell subnets. Incorporating Mobile IP into the DECT handoff mechanism in this way extends DECT micromobility with IP macromobility. Also, utilizing fast, seamless DECT handoff management reduces Mobile IP handoff delay to circumvent TCP throughput degradation during handoff and reduce frequency of Mobile IP signaling over the ether to conserve spectral efficiency. This feature seamlessly unifies DECT with the global Internet. Seamless integration of DECT with the Internet is crucial due to the continuing phenomenal popularity of the Internet and wireless communications, and ubiquity of DECT systems. To achieve the above DECT/IP interworking efficiently, the architecture introduces a network entity called a DECT service switching point, which is an extended DECT central control fixed part. DECT network-level services are mapped onto those of the IETF integrated services architecture to maintain QoS provided by DECT in the backbone Internet. Mobile Resource Reservation Protocol, an extended RSVP tailored to mobile networking, is adopted to provide the needed signaling in IntServ. The proposed architecture preserves traditional non-IP based services such as PSTN voice     

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.     

话音业务的多元化发展

随着电信网络和业务的发展,传统话音业务逐渐从电话网上分离,数据网上传话音进而提上日程。ＩＰ电话标准已然成熟,但相关产品仍缺乏兼容性,话音质量也不够理想;帧中继具有传送话音的巨大潜能,有较强的竞争优势;ＡＴＭ虽然已发展多年,但在传话音方面仍存在许多问题,不能完全尽如人意。作者对话音业务的多元化发展进行了分析和比较。     

Vulnerability analysis and best practices for adopting IP telephony in critical infrastructure sectors

IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.     

SIP-based enterprise converged networks for voice/video-over-IP: implementation and evaluation of components

The next generation of enterprise networks is undergoing major changes as a plethora of new architectures, applications, and services begin to roll out within businesses. In general, the world of voice/telephony, video, and data are "converging" into a global communications network. The purpose of this paper is twofold. First, the design, analysis, and performance of a session initiation protocol (SIP)-based videoconferencing desktop client, which has been developed and deployed over Internet2, is presented. Second, a guideline for managing SIP-based services to be deployed within enterprises, which addresses several challenges in each layer, such as network address translator (NAT)/FW issues, directory service integration issues, and interoperability issues, is proposed. Several detailed experimental results related to interoperability and conformance that were carried out are presented. Findings of extensive SIP/NAT traversal analysis through network traffic measurements are reported. The lessons learned from both the design of a new SIP-based voice/video client, as well as management challenges with enterprise deployment are highlighted.     

VoIP over cdma2000 1xEV-DO revision A

In this article we analyze performance of VoIP services over 1xEVDO-Revision A (DO-Rev A) networks and show that high-quality VoIP with unconstrained mobility and high capacity can be achieved. Together with quality of service (QoS) requirements, we emphasize practical issues such as mobility, degradation of feedback-channel quality, and packet overheads. Novel techniques are presented for voice processing such as smart blanking and adaptive dejitter playback buffer with time warping. These techniques help to meet QoS constraints to achieve a circuit-like voice quality while improving overall capacity. Detailed end-to-end simulations are presented and system capacity is analyzed under the QoS and system stability constraints. We claim that DO-Rev A can provide VoIP capacity comparable to circuit-switched cellular CDMA technologies (e.g., IS-2000) and simultaneously carry significant amount of other types of traffic such as non-delay sensitive applications and downlink multicast.     

Perceptual QoS assessment technologies for VoIP

Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy.