SIP协议在开放的VOIP模型中的实现

开放的VOIP模型为基于IP网络的语音通信提供了一个标准的、分布式的开发平台.SIP协议是支持开放的VOIP体系结构的标准.通过对SIP协议的研究,提出用基于web浏览器/服务器方式实现SIP协议的方法.     

基于SIP协议的IP电话技术

现在的IP电话系统大部分还是基于H.323协议族,而H.323沿袭了传统电信网的运行和管理模式,本身较复杂,在应用于IP电话时存在着不可避免的缺陷。会议初始化协议(SIP)是由IETF制订的用于在两个或多个用户之间建立、修改和终止多媒体会话的信令协议。文中主要介绍了基于SIP的IP电话系统协议栈结构、SIP协议的基本内容以及其在IP电话中的具体应用。最后在与H.323协议进行了比较之后,提出了一些现存的问题并且做出了相应的前景展望。     

基于SIP协议的IP电话服务器的设计与实现

SIP协议作为VoIP的应用层控制协议已经变得十分重要。该文提出了一种以功能插件形式实现基于SIP协议的IP电话服务器方法，介绍了SIP协议及SIP服务器组件，设计并实现了IP电话服务器的系统结构，说明了实现插件功能的细节。     

基于SIP协议的IP电话技术

现在的IP电话系统大部分还是基于H．323协议族，而H．323沿袭了传统电信网的运行和管理模式，本身较复杂，在应用于IP电话时存在着不可避免的缺陷。会议初始化协议（SIP）是由IETF制订的用于在两个或多个用户之间建立、修改和终止多媒体会话的信令协议。文中主要介绍了基于SIP的IP电话系统协议栈结构、SIP协议的基本内容以及其在IP电话中的具体应用。最后在与H．323协议进行了比较之后，提出了一些现存的问题并且做出了相应的前景展望。     

基于SIP协议的VoIP的设计与实现

会话初始化协议(SIP)是IETF提出的IP电话呼叫信令协议,是一个基于文本的协议,用来创建、修改和终止多媒体呼叫与会话。该文介绍了SIP协议及其分层实现方法,采用UML的类图、状态图和序列图等设计了SIP消息包的结构和SIP协议的事务层,并用C++加以具体的实现。     

SIP协议的认证机制及其性能分析

SIP协议是IETF提出的IP电话标准,有着很好的应用前景。讨论了SIP协议的安全认证机制,并通过基于开放源码的Java程序在一个SIP代理服务器上的执行,对该应用中SIP协议安全机制及有关性能进行了分析评价。     

基于SIP的企业指挥调度系统的设计与实现

SIP是下一代网络的核心协议，基于SIP协议的企业指挥调度系统是一个新的重要的研究课题。该文通过对SIP协议和指挥调度系统相关技术的研究，提出了一种以SIP协议作为控制信令的、集中式拓扑结构的IP调度系统，分析了系统的功能需求，介绍了主要功能模块的具体实现。     

基于SIP的无线局域网V2IP电话的设计与实现

对SIP (session initial protocol)协议和WAPI协议进行了研究与分析,在此基础上提出了一种新的基于SIP的V2IP电话模型并实现.与传统IP电话相比,V2IP电话不仅支持WiFi无线接入,而且支持WAPI无线安全接入并可与PC机进行音/视频通话.测试结果表明,基于新模型的V2IP电话具有无线安全接入、音视频传输质量高、可移动性强等优点,并且在稳定性、便携性和可扩展性方面表现良好.     

基于SIP的矿井通信终端协议栈设计

为实现矿用IP通信终端间的融合通信,基于SIP协议设计了一种矿井通信终端协议栈,介绍了该协议栈的架构及具体设计方案。该协议栈内部包括SIP语法解析模块、SIP语义解析模块、媒体处理模块等,可实现基于IP网络的矿井SIP通信终端会话建立和管理功能。测试结果表明,该协议栈能够与多种型号的SIP终端进行稳定的交互对接,且支持所有常用SIP会话管理功能,可满足矿用VOIP通信要求。     

SIP代理服务器的设计与实现

SIP(Session Initiation Protocol,会话初始化协议)是一个应用层信令控制协议,用来创建、修改和终结一个或多个参与者参加的会话,这些会话包括IP电话、分布式多媒体、多媒体会议等。本文基于SIP协议,提出了一个SIP 代理服务器的设计与实现方案。     

基于.NET的SIP实现

SIP(会话初始化协议)是由IETF制定的一套网络多媒体信令协议。该文首先简单介绍了SIP协议，然后在开发实践的基础上分析了基于．NET实现SIP的关键技术。     

在下一代网络中实现多媒体会议的研究

下一代网络中包含有各种网络,例如因特网、PSTN电话网和无线通信网。为了在不同的网络设备间提供统一的多媒体会议业务,该文结合SIP、VoiceXML和SIP会议技术提出了一种新架构,并明确了相关网络实体间的信令流程。     

基于P2P-DHT节点搜索的SIP网络电话设计

结合P2P-DHT(CHORD)节点搜索算法,对SIP协议进行扩展,把基于DHT的全分布式结构化P2P网络架构引入基于SIP协议的网络电话系统,设计了一种新型的P2P-SIP网络电话系统。该方案克服了原有系统需要集中式目录服务器的缺点,在中小规模的网络电话领域具有很好的应用前景。     

基于VoiceXML的语音浏览器的研究与实现

语音浏览器系统能够提供更易为人们所接受的网络浏览模式,拓展了Internet的发展空间。VoiceXML语言是XML语言在语音浏览器方面的应用,文章设计并实现了一个基于VoiceXML技术的语音浏览器系统。     

Internet Telephony: architecture and protocols – an IETF perspective

Internet telephony offers the opportunity to design a global multimedia communications system that may eventually replace the existing telephony infrastructure. We describe the upper-layer protocol components that are specific to Internet telephony services: the Real-Time Transport Protocol (RTP) to carry voice and video data, and the Session Initiation Protocol (SIP) for signaling. We also mention some complementary protocols, including the Real Time Streaming Protocol (RTSP) for control of streaming media, and the Wide Area Service Discovery Protocol (WASRV) for location of telephony gateways.     

Integrating Internet telephony services

Cost savings and the ease of developing and adding new services have motivated great interest in Internet telephony, which integrates services provided by the Internet with the public switched telephone network (PSTN). Internet telephony relies on several protocols, including the real-time transport protocol (RTP) for multimedia data transport and the session initiation protocol (SIP) or H.323 for establishing and controlling sessions. SIP can integrate with other Internet services, such as email, the Web, voice mail, instant messaging, conference calling, and multimedia collaboration. We have implemented a SIP-based software suite called the Columbia Internet extensible multimedia architecture (Cinema), which we installed and integrated with the existing private branch exchange (PBX) infrastructure in the computer science department at Columbia University. The Cinema environment provides interoperability with the PSTN, programmable Internet telephony services, and IP-based voice mail. It also integrates Web access and e-mail for unified messaging and supports multiparty multimedia conferencing. The setup lets us extend our PBX capacity and will eventually let us replace it while keeping our existing phone numbers. It also provides an environment in which we can easily add new services and features, including interoperation with existing multimedia tools, e-mail access from standard. telephones, network appliance control, and instant messaging support     

Emerging alternatives to today's advanced service architectures for Internet telephony: IN and beyond

Two principal sets of standards are emerging for Internet telephony: H.323 from the ITU-T, and session initiation protocol (SIP) from the IETF. The advanced service architecture that comes with H.323 is rather archaic, and reminiscent of the old days of circuit–switched telephony. The one that comes with SIP is more modern, but has its share of deficiencies. While improvements are being made, alternatives are being sought. Two trends are noticeable in the search for alternatives: the return to the old and well-known intelligent network (IN) architectural framework, and the exploration of more recent approaches, such as mobile agent technology, that go beyond IN. This paper scrutinizes the trends in the search for alternatives to today's ITU-T and IETF advanced service architectures for Internet telephony. We successively review the IN-based architectures that are emerging, and the mobile agent-based architectures that are being explored. Many circuit–switched networks that adhere to the IN architectural framework have been deployed worldwide. The emerging IN-based advanced service architectures may facilitate the reuse of this installed base. They will however fall short when it comes to supporting the wide range of advanced services expected from Internet telephony. Mobile-agent-based advanced services architectures are much more promising, although the technology is not yet mature. Their inherent flexibility can easily help in exploiting to the fullest extent the host of opportunities Internet telephony brings. The return to the IN architectural framework may be economically viable in the short and medium terms. There is however no doubt that it will be imperative to go further in the long term. Mobile agent technology is certainly among the venues worth being explored for going beyond IN.     

The promise of a voice-enabled Web

Many computing professionals have heard of XML, and some use it to describe text, images and other data with rich structure. The author discusses an innovative use of XML, called VoiceXML, to support human-computer dialogs via spoken input and audio output. VoiceXML defines dialogs between humans and machines in terms of audio files to be played, text-to-speech synthesis and speech recognition capabilities, and touch-tone input. The author reviews the existing architectures for World Wide Web and telephone services, describes how VoiceXML enables consolidation of service logic for Web and phone, and summarizes the features of the VoiceXML 1.0 specification. Implementation of VoiceXML clients and VoiceXML services has begun in many of the VoiceXML Forum's member companies and will soon be available in the marketplace. The World Wide Web Voice Browser working group has adopted VoiceXML 1.0 as the basis for the dialog markup language that is part of its speech user interface framework     

基于P2P的SIP研究与实现

使用SIP协议的P2P互联网技术构槊支持基本的用户注册和呼叫建立．Chord算法保证了DHT节点能被有效查找和更新，目的是构建一个基于SIP信令的纯P2P架构的IP电话系统。