VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems     

vGPRS: A Mechanism for Voice over GPRS

This paper proposes vGPRS, a voice over IP (VoIP) mechanism for general packet radio service (GPRS) network. In this approach, a new network element called VoIP mobile switching center (VMSC) is introduced to replace standard GSM MSC. Both standard GSM and GPRS mobile stations can be used to receive real-time VoIP service, which need not be equipped with the VoIP (i.e., H.323) terminal capabilities. The vGPRS approach is implemented using standard H.323, GPRS, and GSM protocols. Thus, existing GPRS and H.323 network elements are not modified. Furthermore, the message flows for vGPRS registration, call origination, call release and call termination procedures are described to show the feasibility of our vGPRS system.     

海事卫星通信系统VoIP技术的应用分析

随着VoIP技术的发展,VoIP技术结合卫星通信网络的应用越来越广泛。Inmarsat卫星系统是地球同步轨道系统,网络传播时延大,卫星VoIP电话的语音通信是否可行值得研究。结合VoIP关键技术和海事卫星通信语音通信应用场景,探讨了基于Inmarsat卫星网络实现VoIP技术的方案,并分析出此方案下VoIP系统通话过程的单向时延为350 ms,低于ITU G.114的400 ms的要求。在实际使用环境中进行了测试和验证,结果表明,基于Inmarsat网络下实现VoIP的方案是可行的。该方案实现复杂度低,可以方便地实现Inmarsat网络与地面电话网之间的互联互通,也可以为我国自主研制的宽带卫星通信系统实现VoIP技术提供参考。     

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).     

Internet telephony and modem delay

The vast majority of consumer Internet access is via dialup modems. These modems are a primary source of the delay experienced on voice over IP cells. The author focuses on understanding the causes of delay within analog modems, with the objective of developing recommendations to minimize delay for VoIP applications. First, the relative importance of modem delay is assessed versus other causes of VolP delay (e.g., PC client, IP network), and delay from other access types (e.g., ISDN, cable modems). Second, the characteristics of VoIP data streams are examined as a key determinant of modem delay. Third, the internal operation of modems is examined with respect to delay when transmitting VoIP data streams. Finally, some recommendations and conclusions are presented     

A multiplexing scheme for H.323 voice-over-IP applications

Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.     

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.     

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。     

Mobile Internet telephony: mobility extension to H.323

This paper studies mobility extensions to ITU-T Rec. H.323 for the support of mobile Internet telephony. Internet telephony, also known as voice-over Internet protocol (IP) (VoIP), requires the transmission of two-way and real-time traffic over IP-based networks. The current version of H.323 allows IP telephony and the interoperability of the Internet with switched circuit networks (SCN). However, VoIP mobility has not been previously widely considered, where VoIP mobility refers to the mobility within the scope of IP telephony. We focus on terminal mobility for VoIP. We investigate the influence of mobility on the H.323 layer and propose an H.323 mobility solution to be implemented over the IP layer. Two approaches to mobility extensions to H.323 are described: using ad hoc multipoint conference expansion and using IP multicasting to emulate mobility. Besides, we have also shown that the proposed ad hoc expansion approach shares many properties with the alternative of using IP multicasting for mobility. Hence, the call signaling procedure for the ad hoc expansion approach is also applicable to the multicasting approach. Since ad hoc multipoint expansion has been defined in H.323, our solution introduces no additional entities to H.323 and requires minimal modifications to the existing H.323 protocol. Such mobility extensions can serve as a value-added feature for the Internet telephony systems compliant to the H.323 standard     

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.     

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.     

民航甚高频VoIP新技术的应用研究与展望

VoIP(Voice over Internet Protocol)即网络话音通信,其工作原理是将模拟的声音数字化,经过压缩与封包之后,以数据包形式在IP网络实时传输。VoIP也叫互联网语音通信或IP电话。甚高频VoIP语音通信使用IP技术,在IP网络上布置支持数字音频的甚高频电台和内话系统,传输话音,区别于基于PCM(Pulse-code modulation,即脉冲编码调制)技术的传统数字基带信号传输。VoIP兼具操作功能性及灵活性,这是基于TDM的传统系统所不具备的。     

基于以太网的数字话音传输系统设计

针对在工程应用中如何通过以太网进行话音传输提出了一种设计方案,分析了话音在网络上传输的特点,介绍了一种基于以太网的数字话音传输系统方案。系统以自带网络协议的嵌入式ARM微控制器LM3S9B96为核心平台,采用IP上传送语音（Voice over IP,VoIP）技术实现话音的以太网传输。对系统的话音实际传输效果进行了仿真测试分析,结果表明,话音清晰、失真度和时延小,整体性能满足实际话音通信的要求。     

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.     

Toward an all-IP-based UMTS system architecture

Looking into the future, two main drivers for the mobile telecommunications market can be identified: third-generation mobile systems (e.g., UMTS) and the Internet (e.g., the introduction of IP technologies like voice/multimedia over IP in mobile networks). UMTS is seen as the enabler of wireless multimedia applications and portability of a personalized service set across network/terminal boundaries, as defined within the virtual home environment (VHE) system concept. In light of these evolutions, this article investigates the impact of the evolution toward an all-IP UMTS network architecture on the UMTS service architecture, which is based on the VHE concept. The article discusses two possible scenarios for supporting VoIP services in the UMTS service architecture and analyzes their applicability in an all-IP-based UMTS network. The first is based on the traditional centralized IN service architecture. The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces     

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。     

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.     

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.     

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.     

VoIP信令协议SIP和H.323的比较

VoIP系统凭借其低廉的话费和较好的语音质量,已经成为重要的电信业务,并有取代传统长途业务的趋势.许多组织研究并制定了IP网络上呼叫的协议标准,但有两种IP电话信令和控制标准最具有影响力.一种是ITU推荐的H.323协议,另一种是IETF的SIP.这两种协议代表了解决同一问题的两种不同的方法:H.323是信令基于ISDN Q.931和早期推荐的H系列协议的传统的电路交换的方法,而SIP是一种支持基于HTTP的IP网络的超轻量协议标准.本文,我们主要针对SIP和H.323的体系结构,可靠性,复杂性,可扩展性,可伸缩性以及支持业务类型方面进行比较.