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     

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.     

VTOA/VolP/N-ISDN Telephony Gateway

This article presents the architecture and implementation of a telephony gateway for interworking between N- ISDN, ATM and IP telephony. In this way, interworking is achieved both within private networks and with the PSTN, address translation being performed according to both the vtoa (atm interface) and H .323 (ip interface) specifications. The gateway implementation is based on a PC, presenting a cost- effective alternative to the equipment currently available on the market. Moreover, its highly modular software architecture allows new telephony interfaces to be easily added.     

A practical architecture for implementing end-to-end QoS in an IPnetwork

All critical elements now exist for implementing a QoS-enabled IP network. It can be built on commercially available platforms and then evolve by adopting emerging standards and technologies. This article describes a practical architecture for end-to-end QoS in an IP environment including incorporation of established, as well as developing, IP and QoS technologies. The article combines the IETF QoS mechanisms with the LAN aspects of QoS and QoS for VoIP-areas usually considered separately. Proposed solutions span across different technologies, e.g., preservation of IP-based classification in MPLS headers, identification of flows encrypted within IPSec during WAN handling, traffic shaping in the access to enable grooming diverse applications and VPNs in the WAN, and so on. VoIP receives special emphasis because of its unique features, such as call setup signaling and call admission control, rarely addressed in traditional IP QoS discussions. An attractive scenario for the IP QoS implementation is to provide a multiservice environment between large enterprise premises over a service provider's core network. A successful end-to-end realization of this service presumes well-defined interworking between the SP's and customers' networks. It will take place on several levels including IP signaling, VoIP setup and CAC, policy interworking, and exchange of billing information. The article recommends to establish SP's presence at the enterprise premises and to implement interworking entities such as the proposed QoS customer server and QoS network server     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future.     

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。     

Resource management for QoS support in cellular/WLAN interworking

To provide mobile users with seamless Internet access anywhere and anytime/ there is a strong demand for interworking mechanisms between cellular networks and wireless local area networks in the next-generation all-IP wireless networks. In this article we focus on resource management and call admission control for QoS support in cellular/WLAN interworking. In specific, a DiffServ interworking architecture with loose coupling is presented. Resource allocation in the interworking environment is investigated/ taking into account the network characteristics, vertical handoff, user mobility, and service types. An effective call admission control strategy with service differentiation is proposed for QoS provisioning and efficient resource utilization. Numerical results demonstrate the effectiveness of the proposed call admission control scheme.     

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.     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls     

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.     

An architecture for residential Internet telephony service

A new architecture that can be used for offering an Internet telephony service to residential customers is introduced. The architecture addresses scalability and availability requirements of mass-market deployment of carrier-grade services and supports interconnection with SS7 for Internet telephony calls to the public switched telephone network. The architecture is based on the concept of a gateway decomposition that separates the media transformation function of today's H.323 gateways from the gateway control function of the gateways and centralizes the intelligence in a call agent. The media gateway control protocol is introduced as the protocol between the call agent that assumes the gateway control function and the gateway that provides just the media transformation function. Interworking between the architecture and the public switched telephone network, the session initiation protocol, and H.323 are also discussed     

IP multimedia services: analysis of mobile IP and SIP interactions in 3G networks

Third-generation cellular networks have been designed to provide a variety of IP data services. Both IPv4 and IPv6 are supported in order to provide future-proof solutions. Mobility is supported through both cellular-specific and IP mechanisms. Mobile IP is becoming a key technology for managing mobility wireless networks. At the same time, the session initiation protocol is the key to realizing and provisioning services in IP-based cellular networks. The need for mobility of future real-time service independent of terminal mobility requires SIP to seamlessly interwork with mobile IP operations. In this article, we investigate the issues related to interworking between SIP and mobile IP, with a focus on IPv6 and the applicability to 3G networks being standardized in 3GPP and 3GPP2.     

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.     

Tool assisted H.323 dial‐plan design following formally defined strategies

This paper considers H.323 v4 VoIP networks consisting of gateways and gatekeepers, addressing the task of terminating calls to multiple ISDN networks. Our goal is the maximal utilization of the gatekeeper's centralized routing mechanism for call termination to ISDN networks based on load balancing and resource availability indication (RAI), thus minimizing the need for usage of unwanted (second attempt) overflow mechanisms. We elaborate on the drawbacks of overflow mechanisms and give an overview on different solutions offered in practice by vendors. The discussion justifies the emphasis on load balancing, thus we present an algorithm and tool assisted approach for the systematic assignment of PRIs from ISDN networks to gateways. We follow strictly defined design criteria that lead to an optimized network dial‐plan. We thus offer a stand alone, off‐line solution without the need of any extra add‐ons or upgrades. This is intended to replace the purely intuitive approach followed by today's planners. Our scope is the design rather than the runtime environment; the latter continues to feature the simplicity of a single global RAI per gateway. We consider desirable and non‐desirable link distribution topologies between gateways and ISDN networks based on operational and economic evaluation criteria. These measure the number of ‘wasted’ gateway ports or ‘inappropriately’ allocated network interfaces. Our approach is illustrated through a worked out example showing key features of the proposed algorithm and revealing characteristic cases met in the ISDN interface allocation. Finally, we show how the presented methodology, which presently addresses a H.323 design issue, can provide performance benefits in the VoIP technologies of the immediate future. Copyright © 2005 John Wiley & Sons, Ltd.     

IP电话组网技术

文章首先概述了IP电话的基础——H.323协议,然后介绍了中国电信IP电话网的组网技术——VocalTec综合体系结构(VEA)以及深圳市IP电话网的状况和组网,最后对IP电话存在的问题进行了分析并提出了相应的对策。     

The IETF Internet telephony architecture and protocols

Internet telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with telephony. Internet telephony and traditional circuit-switched telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem     

H.323和SIP在IP多媒体网络中互通的实现

随着IP电话和视频通信的发展,H.323和SIP作为IP多媒体通信领域中被广泛采纳的两种信令控制协议,受到业界的普遍重视。如何有效地实现这两种协议之间的互通,成为近年来国内外研究的热点。本文在简要分析H.323和SIP互通要求的基础上,提出了两者互通的实现方案,并对互通需要解决的关键问题进行了讨论。     

Modeling VoIP Call Holding Times for Telecommunications

Voice over IP is one of the most popular applications in broadband access networks. It is anticipated that the characteristics of call holding times (CHTs) for VoIP calls will be quite different from traditional phone calls. This article analyzes the CHTs for mobile VoIP calls based on measured data collected from commercial operation. Previous approaches directly used the Kolmogorov-Smirnov (K-S) test to derive the CHT distributions, which may cause inaccuracy. In this article we propose a new approach to derive the CHT distributions for mobile VoIP calls and other call types. Specifically, our approach uses hazard rate to select an appropriate distribution, and then utilizes the K-S test to validate our selection. We show that the mobile VoIP CHT distribution can be accurately approximated by a mix of two log-normal distributions. Based on the derived distributions, we compare the mobile VoIP CHTs with those for non-VoIP calls and fixed-network VoIP calls. Our study indicates that the characteristics for mobile VoIP calls are quite different from those of the non-VoIP mobile phone calls and are more close to those of fixed-network phone calls.     

IP电话发展新动向

介绍了NTT公司为参与到IP电话领域而进行的可行性论证、公司内部IP电话实验以及面向商业用户而推出的三种VoIP服务。对新的IP电话网从呼叫控制、通话建立、QoS和编译码、计费方面进行了介绍，并对IP国际电话、呼叫中心及新推出的服务进行了说明。     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented