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Description: vovada公司根据sip协议开发的关于voip通话的协议栈的源代码sip-1.3.0版-vovada companies under the development agreement sip voip calls on the protocol stack source code sip-version 1.3.0
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Size: 11367904 |
Author: 术化 |
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Description: Vovida公司根据sip协议编写的开放源代码。版本sip-1.4.0。-Vovida companies under sip agreement prepared by the open-source. Version sip - 1.4.0.
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Size: 9890536 |
Author: 术化 |
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Description: IETF RFC3261(SIP协议)的中文翻译版；觉得看英文比较慢的朋友可以先看看这个，有少量翻译错误请对照英文版校对-IETF RFC3261 (SIP) version of the Chinese translation; Think of English slower you can take a look at this, a small amount of translation errors in the English version control Please proofread
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Size: 180438 |
Author: 一家 |
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Description: 一个基于exosip,ortp标准协议库开发的sip软电话-One based on exosip, ortp standard protocol library development sip soft phone
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Size: 1529856 |
Author: lee |
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Description: 添加使用PHP对接sip协议的封装接口搜索(i don't speak English. you now? sometime i can say i don't used the port)
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Size: 7168 |
Author: charles22
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Description: java实现sip协议聊天通讯，采用Jain技术。(Java to achieve SIP chat communication)
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Size: 977920 |
Author: 老牛a
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Description: java中sip的应用，可以熟悉sip在java中是使用。。。。。。。(Application of SIP in Java)
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Size: 12179456 |
Author: Kobe blet
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Description: 学习sip的基础知识以及如何抓取sip包的方法(Learn the basics of SIP and how to grab SIP packages)
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Size: 2928640 |
Author: Kobe blet
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Description: chrome 浏览器 网页SIP WEBRTC 开发 呼叫中心需要的工具 开发网页软电话 必备(chrome web sip webrtc asterisk callcenter)
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Size: 435200 |
Author: jf_linux
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Description: SIP协议文档简体中文版。会话初始协议（Session Initiation Protocol）是一种信令协议，用于初始、管理和终止网络中的语音和视频会话，具体地说就是用来生成、修改和终结一个或多个参与者之间的会话。(Simplified Chinese version of the SIP protocol document. Session Initiation Protocol is a signaling protocol used to initiate, manage, and terminate voice and video sessions in a network. Specifically, it is used to generate, modify, and terminate a conversation between one or more participants Conversation.)
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Size: 516096 |
Author: wcchen
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Description: Sip Cracker used to guess passwords and users
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Size: 6827008 |
Author: Matrex.max
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Description: jain-sip-1.2.344 源代码(jain-sip-1.2.344 source code)
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Size: 6510592 |
Author: panther0525
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Description: Linux下搭建一个Sip服务器之前也在windows下找到了一个不错的Sip服务器:MiniSipServer(MSS)，可见这篇文章，但是可惜的是MSS要求本地的所有PHONE都必须发送Authentication也就是授权消息，我还没做这个，所以只能留在以后使用了。因此就找到了这个在linux平台下的OpenSips，开放了源代码，配置自由的工具(But unfortunately, MSS requires all the local PHONE must send Authentication, that is, the authorized message)
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Size: 92160 |
Author: breastist |
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Description: SIP学习资料与SIP真实环境下抓的数据包，供参考(SIP learning materials and data packets captured in the SIP real environment for reference)
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Size: 8154112 |
Author: 许胡 |
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Description: jshtml5通用的支持SIP协议的互联网电话软件(Jshtml5 universal Internet phone software supporting SIP protoco)
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Size: 175104 |
Author: zswlife |
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Description: SIP软电话,官方例子，经过测试，可用,可为IP电话开发作为为能考。(SIP soft phone, an official example, has been tested and available. It can be used as a test for IP phone development.)
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Size: 1605632 |
Author: netseas |
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Description: JAIN SIP的测试代码，单机测试多机测试均可成功运行，初学JAIN SIP十分实用(JAIN SIP test code, single machine test multi machine test can be successfully run, beginner JAIN SIP is very practical.)
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Size: 2659328 |
Author: zzhao1997 |
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Description: SIP UDP redirect and registrar server
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Size: 199680 |
Author: ph379411 |
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Description: 这是一款基于C＃的简单SIP（VOIP）呼出电话。此SIP应用程序已开发，目前正在使用“帮助 - >呼叫支持”。这个想法是创建一个零配置，非常简单的呼出电话，这就是它现在的样子（虽然支持基于IP的传入呼叫;例如：sip：test @ ip：7666,7666是端口SIP_Call out运行） 。(This is a simple SIP (VOIP) outgoing telephone based on C#. This SIP application has been developed and is currently using Help - > Call Support. The idea is to create a zero-configuration, very simple outgoing phone, which is what it is now (although it supports IP-based incoming calls; for example, sip: test @ ip: 7666, 7666 is port SIP_Callout running).)
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Size: 2122752 |
Author: 高软玩家01 |
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Description: 非常实用的SIP软电话API，可以直接在vs2008下编译，简单调用api的接口，就可以实现软电话的所有功能，作者使用开源的exosip、ortp、mediastream等接口，进行封装，用户也可以以这个为基础，做出自己想要的软电话接口，改API目前支持多种语音编码，在易用性，稳定性上超过著名的linphone，作者对linphone也比较了解，在开发前全面分析了解linphone的代码。(A very practical SIP soft phone API can be compiled directly under VS2008 and all functions of the soft phone can be realized by simply calling the interface of the api. The author uses open source interfaces such as exosip, ortp and mediastream to encapsulate it. Users can also use this as the basis to make the soft phone interface they want. The API can support a variety of voice coding at present. It is easy to use and stable. More than the well-known linphone, the author also has a better understanding of the linphone, before the development of a comprehensive analysis of the linphone code.)
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Size: 20572160 |
Author: 北京大牛 |
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