SIP在视讯会议中的应用和发展——当今最热门的通信协议

SIP(Session Initiation Protocol，会话发起协议)是由IETF(互联网工程任务组)提出的IP电话信令协议。它的主要目的是解决IP网中的信令控制，以及同Soft Switch(软交换)的通信，从而构成下一代增值业务平台，为电信、银行、金融等行业提供更好的增值业务。     

SIP会话启动协议

下一代网络是一个能够提供话音、数据、多媒体等多种业务的，集通信、信息、电子商务、娱乐于一体，满足自由通信的分组融合网络。SIP协议主要目的是为了解决IP网中的信令控制以及软交换设备间的通信，从而构成了下一代有增值业务平台作为电话信令协议。SIP协议凭借其简单、易于扩展、便于实现等诸多优点成为未来网络语音通信采用的主流协议，越来越受到业界的青睐。     

基于SIP协议的IP电话实现技术

随着IP电话已经在我国得到普遍认可,如何降低设备运维成本,促进运营商新的市场的开拓。因此,文章对基于SIP协议的IP电话进行研究,主要从SIP电话的相关协议进行介绍,最后对基于SIP协议的IP电话实现进行分析。     

软交换中的分组协议

软交换是下一代网络(NGN)的核心,是电路交换网与IP网的协调中心,它通过对媒体网关的控制,实现不同网络之间的业务层融合。在通信系统中,控制通过协商实现,因此必须有相应的协议。在NGN体系结构中,软交换是控制中心,它支持H.248/MeGaCo、SIP、MGCP、H.323等多种协议。1MGCP协议1.1MGCP基本概念MGCP协议与H.323和SIP不同,H.323和SIP提出两套IP电话体系结构,二者完全独立,不能互相兼容,只能互通。MGCP不涉及IP电话的体系结构,只涉及网关分解问题,因而不仅可用于H.323IP电话系统,也可用于SIP IP电话系统。网关可分解成媒…     

SIP协议在3G多媒体子系统中的应用

SIP协议是由IETF提出的IP电话信令协议，它用于建立、修改和终结多个用户终端之间的多媒体会话。第三代无线系统(3G)的R5结构中，IP多媒体子系统域(IMS)选择SIP作为终端和IMS以及IMS内部各元素之间的信令协议。本文针对SIP协议在IMS中的应用展开介绍，重点介绍了IMS中由SIP协议实现的业务注册和会话建立流程。     

一种SIP软交换系统智能业务服务器的设计与实现

SIP（Session Initiation Protocol,会话初始协议）是一种典型的软交换协议,SIP相关的技术研发是当前软交换技术的研发热点。SipX是一种基于SIP协议的IP电话交换系统。本文在介绍了SIP协议相关问题和SipX系统的体系结构后,重点论述了＂智能业务服务器＂相关的实现原理、协议分析、设计和实现机制。     

IP电话多协议栈支持的解决方案

介绍了IP电话网关、SIP协议和H．248协议,并在H．323协议栈基础上提出支持H．248和SIP协议的IP电话网关解决方案。     

基于SIP协议的IP智能网应用及测试分析

基于SIP协议的IP智能网已成为VOIP网络发展的重要方向,但是新型智能业务的实现仍需调用部分传统智能网功能.以新型智能业务UC业务为例,分析了新型智能业务在基于SIP的IP智能网上的实现方式.通过测试对比UC业务下H.248协议和SIP协议的性能,证实基于SIP协议的IP智能网性能相较基于H.248协议的IP智能网更优越且更符合未来发展趋势的结论.     

会话的初始化协议

会话初始化协议(SIP),它的主要目的是为了解决IP网中的信令控制,以及同软交换机(SoftSwitch)的通信,从而构成下一代的增值业务平台,对电信、银行、金融等行业提供更好的增值业务.本文简单介绍了SIP的系统结构、主要功能、消息机制、优越性和发展现状,最后通过介绍了它的应用实例,模拟了SIP的呼叫建立及终止过程.     

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。     

基于SIP协议的IP电话语音系统设计与实现

本文基于SIP协议的IP电话作为主要研究内容,探讨了IP电话的相关议、相关标准和关键技术,对SIP电话协议进行了研究分析,设计提出一套结构合理的VoIP系统,并对系统组成、系统流程等作了详细的规划,安装配置OpenSER服务器,并能够使用MySql数据库系统来存储用户信息,基于分层设计的思想设计了客户端软件,最后利用该软件进行了系统测试,结果表明系统性能优良。     

SIP security issues: the SIP authentication procedure and its processing load

Session Initiation Protocol (SIP) is currently receiving much attention and seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services, also becoming a real competitor to the plain old telephone service. For the realization of such a scenario, there is an obvious need to provide a certain level of quality and security, comparable to that provided by the traditional telephone systems. While the problem of QoS mostly refers to the network layer, the problem of security is strictly related to the signaling mechanisms and the service provisioning model. For this reason, at present, a very hot topic in the SIP and IP telephony standardization track is security support. In this work, the security model used by SIP is described, and the different open issues are highlighted. We focus, in particular, on the problem of authentication providing a short tutorial on the solution under standardization. The architecture of a possible commercial IP telephony service including user authentication is also described. Finally, we focus on performance issues. By means of a real testbed implementation, we provide an experimental performance analysis of the SIP security mechanisms, based on our open source Java implementation of a SIP proxy server. The performance of the server has been compared with and without security support, under various scenarios.     

True number portability and advanced call screening in a SIP-basedIP telephony system

Custom local area signaling service features offered in the PSTN have certain limitations due to the closed nature of PSTN network signaling. The adoption of telephony over IP (IP telephony) will enable a new paradigm of services and features that are not possible to implement in today's PSTN. This is especially the case for services that make use of personal, trusted information, which can be provided by a user's personal digital assistant. We demonstrate how personal information can be coupled with an IP telephony service to provide user-customized call handling by the network. In particular, we describe a demonstration architecture that includes Ethernet-attached phones running SIP, with an interface to synchronize with PDAs that supply personal information. The proposed architecture is quite flexible; it can support enhanced versions of the current PSTN and private branch exchange services, in addition to many new features and services. We describe true number portability and advanced call screening as examples of new services in a hybrid PSTN/IP telephony environment     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future.     

Using PARLAY APIs Over a SIP System in a Distributed Service Platform for Carrier Grade Multimedia Services

The implementation of new mobile communication technologies developed in the third generation partnership project (3GPP) will allow to access the Internet not only from a PC but also via mobile phones, palmtops and other devices. New applications will emerge, combining several basic services like voice telephony, e-mail, voice over IP, mobility or web-browsing, and thus wiping out the borders between the fixed telephone network, mobile radio and the Internet. Offering those value-added services will be the key factor for success of network and service providers in an increasingly competitive market. In 3GPP's service framework the use of the Parlay APIs is proposed that allow application development by third parties in order to speed up service creation and deployment. 3GPP has also adopted SIP for session control of multimedia communications in an IP network. This article proposes a mapping of SIP functionality to Parlay services and describes a prototype implementation using the SIP Servlet API. Furthermore, an architecture of a Service Platform is presented that offers a framework for the creation, execution and management of carrier grade multimedia services in heterogeneous networks.     

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…     

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.     

移动通信与IP的融合

Ｉｎｔｅｒｎｅｔ产业的热点之一是ＩＰ电话。与传统电话相比,ＩＰ电话具有诸多优点,其中最吸引用户的是资费。我国电话业务因资费过高而导致网络利用率低,移动袖应抓住ＩＰ电话建设的契机,将ＩＰ引入到移动网中,既可节省长途中继,又可以降低花费以吸引用户。     

Vulnerability analysis and best practices for adopting IP telephony in critical infrastructure sectors

IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.     

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.