H.323和SIP在IP电话网络中的互通

SIP和H.323是构造IP电话网络的两大信令体系,因此,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。本文在介绍H.323和SIP协议的基础上,详细阐述了H.323和SIP在IP电话网络中互通的一些基本概念,如地址解析和映射、消息映射、媒体能力协商等。此外,还将通过一个基本的互通呼叫流程来介绍IWF(互通功能体)实现这两种协议的互联互通的方式。     

H.323和SIP在IP电话网络中的互通

SIP和H.323是构造IP电话网络的两大信令体系。因此，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。本在介绍H.323和SIP协议的基础上，详细阐述了H.323和SIP在IP电话网络中互通的一些基本概念，如地址解析和映、消息映射、媒体能力协商等。此外，还将通过一个基本的互通呼叫流程来介绍IWF（互通功能体）实现这两种协议的互联互通的方式。     

基于SIP和H.323的IP电话网互通研究

会话发起协议(SIP)和H.323是VoIP体系结构的两大主流信令协议,它们占有极其重要的地位,但两者的互通尚未解决。两者互通对VoIP系统更方便地提供多媒体业务有重要的实际意义。文中首先简单描述基于H.323和SIP的IP电话系统体系结构,提出了H.323和SIP互通的信令网关概念,然后着重讨论两种IP电话网络互通涉及的关键问题,包括地址解析、消息映射和终端能力协商等,最后分别以SIP和H.323侧发起呼叫为例,模拟SIP和H.323互通的实现。     

会话发起协议SIP的分析和研究

SIP是IETF的MMUSIC工作组提出的支持多媒体会话的信令控制协议,用于创建、修改、终止多媒体会话或呼叫,在现代的多媒体通信中有着相当广泛的应用,成为H.323协议族的有利竞争者。本文对SIP协议功能特点及SIP与H.323协议的异同点作了详细的阐述。     

SIP与H.323可视通话的互通实现

SIP和H．323是基于IP网络的多媒体通信的两大主流技术。实现两者的互通是尚待解决的一个很具现实意义的问题。文章在比较H．323协议和SIP协议的基础上，探讨了两者互操作的一些基本概念和实现要求，给出了互通模块IWF的结构，进而通过模型实例，讨论了实现互联互通的地址解析、呼叫建立等典型操作流程。     

基于SIP的嵌入式UA语音终端设计

SIP协议是NGN中的重要协议,将逐步取代H.323协议成为VoIP的标准信令协议。设计了一款基于SIP的语音用户代理,即SIP终端。系统硬件采用ARM9为平台,软件以LINUX为平台,采用C语言编写了UA用户程序,在实现过程中采用了osip2、eXosip2协议栈来处理信令,采用了Mediastreamer2协议栈和oRTP协议栈以及PCM等编码算法来处理音频流的采集/播放、编码/解码、发送/接收等。局域网内测试表明,该终端可以实现拨打、接听、取消、挂断、退出等基本呼叫流程,并且通话质量良好。     

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。     

基于软交换的SIP与H.323协议互通

SIP和H.323是下一代IP分组网上多媒体业务技术发展的两个方向,软交换可以同时支持这两种协议并实现两者的融合互通.在软交换网络的基础上,结合这两种协议的特征,本文简要介绍了SIP和H.323互通的实现过程和呼叫流程,并介绍了互通功能(IWF)单元在互通中的功能.     

H.323与SIP在VoIP和多媒体应用中的比较

目前,国际上IP网络通信的主要标准有H.323和SIP,两者都对IP电话系统信令提出了完整的解决方案,但两者的设计风格各有千秋,H.323采用的是传统的电话信令模式,包括一系列协议;而SIP借鉴互联网协议,采用基于文本的协议.采用H.323协议,各个不同厂商的多媒体产品和应用可以进行互相操作,用户不必考虑兼容性问题;而SIP协议应用较为灵活,可扩展性强.     

IP电话信令协议的比较

目前各标准化组织对在IP网络上承载实时业务采用的方案均相同,都是采用IETF的RTP协议,但是在呼叫建立和控制方面则有着不同的方案,有IETF的SIP协议、ITU－T的H323协议和IETF的MGCP协议。它们是关系到IP电话质量好坏的核心协议。本文分别介绍这三种协议,并进行比较。SIP协议（RFC2543）SIP（SerasionInitiationProtocol）协议是由IETF推出的会话初始化协议,它是一种用于建立、修改和终止多媒体会话或呼叫的应用层控制协议。SIP是一种基于文本的协议,它的语法和信令信息非常类似于HTI：7,P协议,不同2处在于,S…     

Developing applications for Internet telephony: a case study on the use of parlay call control APIs in SIP networks

Applications offered to end users as value-added services, or more simple, services, are crucial for the survival and success of service providers. Two main sets of standards have emerged for Internet telephony: H.323 from the ITU-T and SIP from the IETF. Unfortunately, the related application development frameworks are rather weak. Parlay, a set of standard object-oriented and signaling protocol-neural APIs, is an alternative. It allows applications to access network functionality, including call control, in a controller manner. Call control makes it possible to establish, modify, and tear down calls. It is the main functionality offered by Internet telephony networks. We have built a call control application in a SIP environment, using the call control APIs offered by Parlay. The application is a multiparty game. This article describes the case study. The mapping of the APIs onto SIP is presented, and its implementation is described. Related work reviewed, and the lessons learned are discussed. Parlay call control APIs are suitable for application development in Internet telephony. However, well isolated extensions are needed to realize their full potential.     

Internet telephony signaling

The term “multimedia session” refers to the integration of data coming from various sources, such as sound, video and text, within a computer application. Telephony over the Internet is among the more exciting current developments. The signaling of a telephone call consists of the set of messages and procedures used to establish a connection, to request changes in communication bandwidth, to obtain the message status for the end points participating in the conversation, and to close the link. At present there exist two competing signaling protocols for Internet telephony, viz., the H.323 protocol sponsored by the ITU and the Session Invitation Protocol (SIP) sponsored by the IETF. Each of them supplies its own signaling mechanisms.

In this paper, these two protocols in terms of their main functionalities are compared. Based on the results of this comparison, a Client/Server architecture for the development of an application that supports a basic SIP implementation, as well as the formulation of requests allowing the establishment and the disconnection of communications between a number of users in a multimedia session are then defined.     

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.     

软交换中SIP与H.323电话网的互通实现

首先简要介绍了SIP与H.323协议,接着根据互通的要求,给出了实现互通的四个典型流程,最后讨论了SIP和H.323消息的映射、参数的转换和状态机的思想。     

软交换网络中媒体网关控制器的研究与设计

300万像素对于专业摄影来说可能不值一提，但是对于家庭用户而言却已经十分完美了。下面三款数码相机是300万家庭中令人兴奋的新品，它们的价格都在3000元到4000元人民币之间，想想一年前3000万像素的相机要花多少钱？而在测试中它们的表现也的确让人觉得物有所值。     

The Distributed Gateway Interworking with SIP/H.323

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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     

Analysis and Applications of Session Initiation Protocol (SIP) in Internet Telephony

1　Introduction　Internet telephony, also known as Voice over IP (VoIP)or IP telephony (IPtel), is the real time delivery of voice(and possibly other multimedia data types) between two ormore parties, across networks using the Internet protocols,and the exchange of information required to control this de livery. Internet telephony offers the opportunity to design aglobal multimedia communications system that may eventual ly replace the existing telephony infrastructure.　Internet Engine…     

基于CS6220的VoIP终端的研究

以往的IP电话设备主要采用H．323协议实现，但是H．323协议实现过程的复杂，给实际使用造成很多的问题。提出一种基于CS6220的VoIP终端的实现方法，VoIP协议采用实现简单的SIP协议，同时也能够确保电话业务的服务质量，电话业务接口的技术性能指标满足相关规范要求；重点阐述了基于CS6220的VoIP终端的软硬件设计方法；电话接口的性能测试结果表明，所提出的方案在因特网运行良好。