呼叫转移在IP电话中的实现

介绍了IP电话网守的概念，功能和呼叫信令的两种方式，讨论了呼叫转移在IP电话中的一种实现方法，在局域网、单网守的条件下得到了实现。     

MGCP和H.323协议互通的研究

通过对MGCP和H.323协议的分析和比较,提出了MGCP和H.323协议互通的实现方法,重点探讨了协议互通模型和呼叫处理模块的设计方法,并分别以MGCP和H.323终端发起呼叫为例,详细描述了MGCP和H.323互通的实现。     

如何运用Openh323协议栈来开发VoIP软件

Openh323是为开发使用H.323协议在IP网上进行多媒体通信的应用程序而专门设计的全功能协议栈。该协议栈封装了大量H.323实体,功能完善,是VoIP领域一个相当重要的基础项目。首先介绍Openh323项目计划,然后描述了协议栈类库结构,着重说明类的划分和协议过程实现机制,并且结合网守技术实现的基本思路从进程模式、类的继承等方面对如何将Openbh323协议栈运用到商用VoIP软件的设计开发中进行探讨。     

腾博视通推出新型视频通信服务器

8月8日，致力于提供高质量、创新视频通信解决方案的全球领先视频通讯供应商——腾博（TANDBERG）视通公司宣布，推出腾博视频通信服务器。该服务器内置FindMe^TM新型呼叫转移应用功能，使商业沟通更加个性化及统一化。该产品还具有支持H.323和SIP的呼叫控制和防火墙穿越等功能。     

基于H.323协议的VoIP安全问题探讨

本文简要介绍了H.323协议网络结构、子协议功能及工作流程。分析了VoIP网络系统潜在的威胁及安全要求,最后结合H.323 VoIP的通信流程,分析了H.235建议所采用的安全措施。     

IP电话系统网守的研究与实现

IP电话系统是具有广阔发展前景的新型电信业务。研究了在H .32 3协议下 ,IP电话系统中网守所起的作用 ,探讨了具有地址映射、域管理、呼叫代理、计费功能的网守的实现方案 ,并对该网守的特点进行了分析。     

SIP协议及其应用

由IETF的MMUSIC工作组提出的会话初始协议(SIP)采用与H.323不同的设计理念来实现Internet多媒体通信的信令功能,它借鉴了其它Internet的标准和协议的设计思想,坚持简练、开放、兼容和可扩展等原则,在复杂性、可扩展性以及呼叫建立过程等方面都优于H.323。SIP主要采用三种呼叫方式建立连接:直接呼叫、重定向呼叫和代理呼叫。     

H.323视频会议网守的设计与实现

设计和实现了基于H．323协议的视频会议网守，该网守具有地址翻译、接入控制、带宽控制，区域管理等基本功能，能够与符合H．323协议的端点进行通信。     

基于H．323协议的视频会议网守的设计与实现

本文设计和实现了基于H．323协议的视频会议网守，该网守具有地址翻译、接入控制、带宽控制、区域管理等基本功能，能够与符合H．323协议的端点进行通信。     

SIP与H.323的呼叫控制互通及媒体协商

SIP和H.323是构造IP电话网络的两大信令体系,实现SIP和H.323协议互通是保证IP电话网络顺利运营的一个关键问题。本文在方便控制呼叫、利于混合组网的基础上提出了一个互通模型,与通常的IWF实现作了比较,详细阐述了通过该方案的呼叫流程,并且以3PCC的实际例子说明了媒体能力协商的过程。     

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.     

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     

一种企业网内VoIP的呼叫控制协议

对Intranet环境中的VoIP呼叫控制的特点及目前相关协议进行了比较分析,提出了一套轻量级企业网VoIP协议SSI,并应用该信令系统简单高效地实现了企业网VoIP系统VCSI中的呼叫控制功能。     

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     

软交换实现机制

软交换技术是VoIP的核心技术之一,但它和基于H．323、SIP的VoIP系统有着本质区别,本文详细介绍了软交换系统既和PSTN交换机一样实现时隙交换,又解决用远距离分组交换通路取代长话通信线路的工作机制,给出了基于软交换系统的VoIP系统的呼叫连接建立过程。     

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.     

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。     

Evaluating the perceived voice quality on VoIP network using interpolated FIR filter algorithm

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.     

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.     

Effective VoIP call routing in WLAN and cellular integration

In cellular-WLAN integration, a dual-mode mobile station (MS) typically disables the WLAN module for power saving. A major problem is that for an incoming VoIP call (or data session), the MS will not be able to receive this call from the WLAN. It turns out that the call is directed to the cellular network. This letter proposes a simple push solution where an MS can accurately detect a VoIP call from paging signaling of the cellular network. Then the WLAN module of the MS is turned on and the VoIP call is connected to the MS through the relatively inexpensive WLAN.