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。     

应用于PSTN的IP中继传输及利用度分析

本文分析了在PSTN中引入IP中继传输的基本需求,给出了承载信道IP中继传输的协议设计,并针对IP中继传输的效益性能做了简要的计算分析。结果表明,在典型话务量负载下,通过IP中继传输,带宽利用率能降至约13%。     

新兴电信运营商构建智能宽带本地电话网的策略分析

如何利用宽带IP城域网快速地向用户提供现有的业务已成为每一个电信运营商急需解决的问题。为此,本文提出了建设智能宽带IP城域网的思想,将智能网业务与因特网业务有机地融合在一起,并提供了在此基础上逐步建设宽带本地电话网的解决方案,为我国城域网的建设提供了有益的参考。     

An Approach to Integrating SIP in Converged Multimodal/Multimedia Communication Services

In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.     

SIP和H.323协议互通的研究

下一代网络是当前的研究热点,H.323和SIP是VoIP网络的两大主流技术,基于它们的网络间互联互通是一个亟需解决的问题。文章从协议角度提供一个从SIP网络到H.323网络的互通单元(IWF),重点介绍互通单元的结构和功能模型。通过对H.323与SIP协议的分析比较,指出互通过程中需要处理的主要问题,实现网络寻址、地址转换和消息映射的基本方法,最后给出实现互通最典型的通信流程,为实现基于软交换的下一代网络提供参考。     

基于IP承载的软交换核心网的建设

本文主要阐述2G软交换IP承载语音测试完成后,软交换网络结构发生的变化、建设过程中需要考虑的新问题以及IP软交换设备在2G网络中的建设方案.     

面向固定移动融合环境的P2PSIP系统设计

全业务运营是电信市场继语音和宽带接入服务之后的下一个增长点,而基于IP的融合有线网络和无线网络的语音服务则是全业务的重点之一。本文通过分析现有VoIP网络存在的问题以及固定移动融合网络环境下VoIP的特点,提出一种新型双层重叠网架构的P2PSIP架构,并阐述了新型架构的优点及双层重叠网之间的通信机制。新型架构能有效提高系统的安全性、健壮性和用户节点资源利用效率,更好的满足固定移动融合网络环境下VoIP对带宽、网络质量和安全性的要求。     

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。     

漫谈VoIP技术发展趋势——H.325:第三代多媒体系统

目前在Internet或IP网络上应用的VoIP技术主要是基于H．323或者SIP开发的。随着技术和需求的发展，VoIP要求能够同时提供话音、数据和视频等多种业务，向下一代网络NGN演进。为了更好地满足NGN的需求，弥补现有系统的不足，ITU提出了下一代多媒体系统H．325协议的概念，它的重点在于实现控制单元和服务单元的分离，更好地支持多种媒体编码协议的互通，提高系统的QoS以及安全性。H．325有望成为下一代VoIP技术的支撑协议。     

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.     

基于SIP协议的VoIP网关的设计和实现

在传统方式下,话音业务和数据业务是由两个互相独立的网络———公共电话交换网(PSTN)和数据通信网分别提供的。公共电话交换网基于电路交换的结构,这种技术保证语音有优良的品质,但同时也给PSTN带来线路资源分配方案的效率低和通话传真成本高的局限性。数据通信网则是基于包交换,由于采用了简单健壮的IP协议,使得其在这几年发展迅速。分析了目前在IP通信领域的两大主流协议H.323和SIP的特点和差异,给出了对支持SIP协议的VOIP网关的软硬件设计方案。     

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     

Integrated Mobility Management Methods for Mobile IP and SIP in IP based Wireless Data Networks

With developments in voice over IP (VoIP), IP-based wireless data networks and their application services have received increased attention. While multimedia applications of mobile nodes are served by Session Initiation Protocol (SIP) as a signaling protocol, the mobility of mobile nodes may be supported via Mobile IP protocol. For a mobile node that uses both Mobile IP and SIP, there is a severe redundant registration overhead because the mobile node has to make location registration separately to a home agent for Mobile IP and to a home registrar for SIP, respectively. Therefore, we propose two new schemes that integrate mobility management functionality in Mobile IP and SIP. We show performance comparisons among the previous method, which makes separate registration for Mobile IP and SIP without integration, and our two integrated methods. Numerical results show that the proposed methods efficiently reduce the amount of signaling messages and delay time related to the idle handoff and the active handoff.     

IP电信网的业务模型

文章在概要分析当前有关下一代网络的共识和存在的问题的基础上，讨论了IP电信网相关技术的概念与相互关系，阐述了VoIP技术，软交换技术在IP电信网中的作用，给出了IP电信网的业务模型和实现策略。     

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。     

VoIP网关中呼叫处理模块的实现

电信网络向下一代网络演进是发展的必然趋势,以软交换为核心的下一代网络的快速发展,必然导致电信技术的革新。简要分析了VoIP网关产生的必然性,给出了基于SIP协议的VoIP网关中呼叫处理模块的设计方案,并对呼叫处理任务的邮件结构,呼叫处理任务流程、接口和程序进行了详细的设计。     

IP组播在集群软交换中的应用

集群系统组呼要求交换机具备大的汇接交换容量,因此提出了组呼汇接交换的几种方案并进行了分析比较。其中,IP软交换和IP组播技术应用改善了大容量汇接交换的话音质量,降低了成本,提高了网络带宽利用率,是下一代集群系统交换技术发展的必然趋势。同时也论述了利用IP组播实现集群组呼交换需要的协议和对IP网络条件的要求。     

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.     

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     

电力程控交换中心软交换组网设计

近年来,我国的电力企业取得了卓有成效的发展成就,电力通信业务突破了传统程控语音通信等单一的业务模式,逐渐朝着网络融合方向发展.传统的程控交换技术仅能够提供语音通话服务,难以满足多样化综合业务需求,因此,必须探索新的能够实现对电话交换网管理的新型通信技术.此次研究将宽带IP技术作为基础,结合最新通信技术软交换系统,在信令控制及媒体处理技术支持下,构建了电力程控交换中心软交换组网系统平台,为企业生产及通信业务发展提供帮助.