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Description: SIP称为会话发起协议（Session Initiation Protocol)。SIP是Internet工程任务组（IETF）提出的建议中的一项协议，其目的是为了替代H.323的部分协议，主要是考虑到在下一代网络（NGN）中，由于IP产品和IP网关将在网络中得到大规模使用和集成，使得端到端都可以采用IP以实现纯IP的业务应用；而基于纯IP的SIP借鉴了HTTP和SMTP，结构简单并具有可扩充性和可扩展性。此外，SIP还提供良好的QoS支持。对于NGN在IP网络上实现VOIP和多媒体通信来讲，SIP在全面满足NGN特性要求的应用上具有独特的优势，必将成为下一代网络VOIP的重要解决方案。
-called SIP Session Initiation Protocol (Session Initiation Protocol l). SIP is the Internet Engineering Task Force (IETF), the recommendations of an agreement, its purpose is to replace the H.323 part of the agreement is mainly on account of the next generation network (NGN), As IP and IP gateways in the network will be large-scale use and integration, making end-to-end IP can be applied to achieve a pure IP business applications; For pure IP SIP from the HTTP and SMTP, simple structure and is designed to be scalable and scalability. In addition, the SIP also provides good QoS support. IP NGN for the realization of the VoIP network and multimedia communications speaking, SIP to meet the full requirements of the characteristics of NGN application has unique advantages, the next generation w
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Author: Jee |
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Description: KPhone is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet. KPhone requires KDE (K Desktop Environment) or at least various libraries from the KDE package. KPhone also requires the OpenSSL libraries to be present in the system. KPhone has been somewhat tested on Red Hat 7.2 and 7.3, Slackware 8.0, SuSE 8.1 and Debian. Requirements on each distribution vary according to the installation layout.
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Size: 219405 |
Author: 张强 |
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Description: this sip voip sample on g711encode/decode.-this sip on 1/1/2006 sample g711encode/deco de.
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Size: 54219 |
Author: yokkie |
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Description: 文中详细介绍了SIP各个层次模块的功能和设计，描述了重要的协议流程
和设计方法，重点阐述了关键技术的使用和实现中需要解决的问题，最后设计了网
络拓扑对协议栈进行了功能测试，测试结果表明该 SIP 协议栈的实现完全满足 VoIP
网关对信令建链功能的需求。
-paper describes in detail the various levels of SIP module and functional design, Description of the important agreements and design process, focuses on the key technology use and realization of the need to address the problem, final design of the network topology of the protocol stack for functional testing, Test results show that the SIP protocol stack to achieve fully meet the VoIP Signaling Gateway to build the demand chain functions.
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Size: 583651 |
Author: 黄妮丽 |
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Description: 开放的Voip源代码，实现SIP协议，可用于Voip的开发。-open source code, achieving SIP can be used for Student Development.
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Size: 649393 |
Author: shiwei |
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Description: 著名的SIP协议栈，为GNU开源项目。通过其提供的API接口，可以轻易开发出基于SIP协议栈的VOIP应用程序。-famous SIP protocol stack for the GNU open source projects. Through its interface with the API, can easily develop SIP-based VoIP protocol stack applications.
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Size: 664050 |
Author: wxs |
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Description: voip压力测试工具源代码，sipp功能非常强大，可以对任何类型的sip服务器进行压力测试。该程序支持linux和win32。-1/1/2006 pressure testing tools source code, sipp is very powerful, can sip any type of server stress testing. The procedure support Linux and win32.
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Size: 379625 |
Author: 吴宗静 |
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Description: SIP Express Router, Linux下的SIP代理服务器,小巧实用,开发测试VoIP设备和应用的必备.-SIP Express Router, Linux under the SIP proxy server, Small practical, the development of VoIP test equipment and the necessary application.
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Size: 1846781 |
Author: nico zhu |
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Description: 实用于SIP、H323等协议软电话、单口网关、网络电话计费，支持市话单独计费。-applied to SIP, H323, and other agreements soft phone, the single gateway, VoIP billing, to support the then separate billing.
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Size: 4918804 |
Author: 孙昊 |
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Description: VoIP using SIP and Java， VoIP using SIP and Java-VoIP using SIP and Java. VoIP using SIP and Java, VoIP using SIP and Java
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Size: 403318 |
Author: xiaowang |
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Description:
KphoneSI (kpsi) is a SIP (Session Initiation Protocol) user agent
for Linux, with which you can initiate VoIP (Voice over IP) connections
over the Internet, send Instant Messages, subscribe your friends
presence information and start other applications of your choice
-KphoneSI (kpsi) is a SIP (Session Initiati on Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet, send Instant Messages. subscribe your friends a presence information nd start other applications of your choice
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Author: 刘晓军 |
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Description: 关于voip以及sip的一些课件,里面把这些内容阐述的很清楚.-on 1/1/2006 and sip some courseware, which have these elements set out in very clear.
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Size: 137116 |
Author: 张东 |
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Description: linux下的sip voip程序，这个是日本人开发的，对开发sip voip程序十分有参考意义-under linux sip 1/1/2006 procedures, the development of the Japanese, development of procedures sip 1/1/2006 very reference
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Size: 39578 |
Author: 蒋飓风 |
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Description: IAX client库, 一个VOIP的库. 支持H.323和SIP, PBX就是采用的它-the IAX client, a VoIP libraries. Support H.323 and SIP, PBX it is the
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Size: 765644 |
Author: 读卡 |
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Description: Session Initiation Protocol (SIP) [1] is fast becoming the protocol of choice for IP based communication
– specifically telephony (Voice over Internet Protocol - VoIP), video, and instant
messaging.-Session Initiation Protocol (SIP) [1] is f ast becoming the protocol of choice for IP based communication-specifically telephony (Voic e over Internet Protocol-VoIP), video, and instant messaging.
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Size: 381147 |
Author: 刘志合 |
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Description: 本文从 VoIP 技术的发展现状开始，分析了 VoIP 的发展方向，引入了软
交换和下一代网络的架构，讨论了软交换和下一代网络的功能、主要协议、
组网用途、和演进策略，介绍了软交换架构在下一代移动通讯网 UMTS 中的
应用。然后本文讨论了 H.323 协议的体系结构和它在软交换中应用的必要性
和可行性。之后，介绍了 SIP 网络与 H.323 网络地址解析体系的不同与互通
问题，研究分析了他们之间的互通寻址定位方案。 -paper from the VoIP technology development, the analysis of the VoIP development direction, introduction of the next generation soft switch and the network structure, discussed the next generation soft switch and network functions, the main agreement, Network use, and evolution strategy, introduced the soft structure of the exchange in the next generation UMTS mobile communications network applications. It then discussed the H.323 architecture and its application of soft switching the necessity and feasibility. , Introduced a SIP network with H.323 network address system of different analytical and communication problems Analysis of the exchange between them Addressable positioning program.
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Size: 693012 |
Author: 周东 |
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Description: 通过RTP协议进行通讯的完整例子，本人已经在VC6上编译通过。
可以向支持SIP协议的服务器发起VOIP呼叫,并可以进行文字、视频、文件的传输。-through RTP for a complete communications example, I have compiled the VC6 through. To support SIP Server launched VoIP call, and can be text, video, document transmission.
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Size: 126524 |
Author: zwd |
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Description: 一个用vc开发的小程序，用于voip通讯中，解析sip协议中sdp协议的包文，采用面向对象的设计思想，还是不错的。-vc development with a small procedure for 1/1/2006 communication, Analysis sip agreement sdp agreement package text, object-oriented design ideas, or good.
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Size: 68720 |
Author: lvzhuqiang |
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Description: IP电话的设计与实现.rar
本文从VoIP相关知识入手，介绍了IP电话的软硬件结构设计及实现方案，并描述了IP电话基于SIP的呼叫流程，并通过设计杂项的方式详细描述了系统中的几个关键点。内容仅供大家参考，更详细内容可参见源代码，源代码是最好的老师。
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Size: 162040 |
Author: benter |
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Description: voip实现的原代码的一个部分 sip协议的实现
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Size: 158776 |
Author: zhl |
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