会话发起协议

会话发起协议（SessionInitiationProtocol，SIP）通常被视为与电信领域建立联系的事实协议。

The Session Initiation Protocol ( SIP ) is generally seen as the de-facto protocol to tap into the world of telecom .

IP上语音和视频聚合的一个主要基础部分是会话发起协议（SIP）。

One main element in the foundation of voice and video convergence on IP is the Session Initiation Protocol ( SIP ) .

本课程将对会话发起协议（SIP）的特征及功能进行深入探讨。

This session will delve deep into the Session Initiation Protocol ( SIP ) features and functionality .

WebSphereApplicationServer为在一个应用程序中无缝利用会话发起协议（SIP）和超文本传输协议（HTTP）提供了简单的编程模型。

WebSphere Application Server provides a simple programming model for utilizing the Session Initiation Protocol ( SIP ) and Hypertext Transfer Protocol ( HTTP ) seamlessly within one application .

举例说明了该模型在利用组合单播协议研究多播协议以及描述和分析会话发起协议（SIP）方面的应用。

Examples illustrate its applications in research of multicast protocol by composing unicast protocols and in describing and analyzing Session Initiate Protocol .

自从1999年变成了IETF标准以后，会话发起协议（SIP）有了很大的发展。

Session Initiation Protocol ( SIP ) has grown a lot since it first became an IETF standard in1999 .

会话发起协议（SIP）将在3G网络中发挥重要的作用，而JAINSIP是用Java语言实现的SIP类库。

SIP ( Session Initiation Protocol ) plays an important role in 3G environment , and JAIN SIP implemented SIP by Java language .

它为企业级应用程序提供了运行时环境，包括J2EE应用程序、Portlet应用程序和会话发起协议（SessionInitiationProtocol，SIP）应用程序。

It provides a run time environment for enterprise class applications , including J2EE applications , portlet applications , and Session Initiation Protocol ( SIP ) applications .

利用会话发起协议开发网络电话，满足IP网络上的多媒体会话需求，并且添加服务质量保证网络电话的语音质量。

Using Session Initiation Protocol to develop telephone network , which meets the needs of multi-media session on IP network , and add quality of service to guarantee the voice quality of Internet phone .

随着会话发起协议（SIP）应用领域的不断扩大，SIP实体间通讯的安全问题成为了亟待解决的热点问题。

The Community of Session Initiation Protocol ( SIP ) entity is becoming an urgent problem to be solved with the continued enlargement of SIP application area .

在此基础上，对简单的SIP网络模型进行了分析，并给出了新的SIP网络模型：基于事件通告的会话发起协议网络模型。

On the basis of exploring , analyze the simple SIP network model and present the way of improving this system : SIP Model based on Specific Event Notification .

针对会话发起协议（SIP）穿越网络地址转换（NAT）的问题，提出一种基于端口探测的解决方案。

To solve the problem of Session Initiation Protocol ( SIP ) traversing Network Address Translator ( NAT ), this paper describes a new method based on port detection .

主要研究和探讨了会话发起协议的结构及其实现机制，具体分析了SIP协议的请求消息和应答消息，功能实体和消息体。

The purpose of this thesis is to explore the designing idea , content , and realization mechanism of SIP , analyzes the request and response message , entity and message body of SIP in detail .

文章首先描述了FT/HA技术的基本原理，然后提出了一种基于温备份的FT/HA解决方案，最后，重点探讨了该解决方案在会话发起协议（SIP）开发中的应用。

We firstly describe fundament of FT / HA technology , and then propose a sort of FT / HA solving scheme based on warm-backup . Finally , we mostly discuss this solving scheme which is applied to SIP protocol development .

为此，IETF于1999年提出会话发起协议SIP（sessioninitiationprotocol），用来解决IP网上的信令控制，是一个基于应用层的控制协议，用来建立、修正以及终止一个多媒体会话。

So in 1999 , IETF issued a signaling protocol & SIP ( Session Initiation Protocol ) . It is used to control signal based on IP-Network and create , modify and terminate a multimedia session as a controlling protocol based on application layer .

随着传统电信网络向IP数据网络的转换，会话发起协议SIP作为一种多媒体控制协议，以其简单、通用、扩展性强的优势已成为当今研究热点，基于SIP协议的各种应用也开始层出不穷。

As the conversion from traditional telecommunications to IP data networks , Session Initiation Protocol as a multimedia control protocol has become the hottest research for its advantages of simple , common and strong expansion , and variety of applications based on SIP appear .

基于会话发起协议的存在服务器及其应用研究

Study on a SIP-Based Presence Server and Its Application

在下一代网络中应用会话发起协议

Applied the Session Initiation Protocol in Next Generation Network

分析会话发起协议在移动通信应用中存在的问题，指出对其信令消息进行压缩的重要性。

By analyzing the problems while implementing SIP in mobile communication system , necessity of Signaling Compression ( SigComp ) is indicated .