[IP电话/视频会议] 如何使用拨号计划功能在C＃中构建简单的SIP PBX
PBX是Private Branch eXchange的短期内容。它是一个将公司的电话分机连接到外部公共电话网络以及移动网络的系统。今天的基于软件的IP PBX可以取代传统的硬件商务电话系统。PBX与VoIP技术协同工作，这就是为什么IP PBX通过TCP / IP协议栈为其内部网络提供音频/视频呼叫和即时消息通信，并将其内部网络与PSTN（公共交换电话网络）互连以进行电话通信。
随着VoIP技术的日益发展，通过一些额外的功能来提升您的PBX非常容易，以及拨号计划在那些通常有大量同时呼叫的通信网络中非常有用。拨号计划是服务器中的自动系统，用于管理内部和外部呼叫，呼叫转移，呼叫保持和限制。
It is a system that connects a company's telephone extension to external public telephone networks and mobile networks. Today's software-based IP PBX can replace the traditional hardware business telephone system. PBX works with VoIP technology, which is why IP PBX provides audio/video calls and instant messaging for its internal network through TCP/IP protocol stack, and interconnects its internal network with PSTN (Public Switched Telephone Network) for telephone communication.
With the development of VoIP technology, it's very easy to upgrade your PBX through some additional functions, and dialing plans are very useful in communication networks where there are usually a lot of simultaneous calls. Dial-up plan is an automated system in servers that manages internal and external calls, call transfer, call retention and restriction. (2019-04-28, C#, 4018KB, 下载4次)
[IP电话/视频会议] SIP协议.RFC3621中文版
协议文档，Internet 的许多应用都需要建立和管理一个会话，会话在这里的含义是在参与者之间的数据的交换。由于考虑到参与者的实际情况，这些应用的实现往往是很复杂的：参与者可能是在代理间移动，他们可能可以有多个名字，他们中间的通讯可能是基于不同的媒介（比如文本，多媒体，视频，音频等）－有时候是多种媒介一起交互。人们创造了无数种通讯协议应用于实时的多媒体会话数据比如声音，影像，或者文本。本SIP （会话初始协议）和这些协议一样，同样允许使用Internet 端点（用户代理）来寻找参与者并且允许建立一个可共享的会话描述。为了能够定位精确的会话参与者，并且也为了其他的目的， SIP 允许创建基础的network hosts （叫做代理服务器） ，并且允许终端用户注册上去，发出会话邀请，或者发出其他请求。SIP 是一个轻形的，多用途的工具，可以用来创建， 修改和终止会话， 它独立运作于通讯协议之下， 并且不依赖建立的会话类型。
Like these protocols, this SIP also allows the use of Internet endpoints (user agents) to find participants and allows the creation of a shared session description. In order to locate precise session participants and for other purposes, SIP allows the creation of basic network hosts (called proxy servers) and allows end users to register, issue session invitations, or make other requests. SIP is a lightweight, multi-purpose tool that can be used to create, modify and terminate sessions. It operates independently of the communication protocol and does not depend on the type of session established. (2019-02-28, Others, 394KB, 下载5次)
[IP电话/视频会议] internet_voip_vcPP
本源码分成两个问分：A.voip服务器，主要用于将两个终端连接起来，将拨号、挂机、接听等信令转发、语音数相互所转发。B.voip客户端，采用voip常用语音压缩g.729，压缩率可达1/16，采用8K16bit的采样，可以低网速的情况下稳定实现双方通话，软件已提供拨号、持机、接听、忙信号等电话基本功能。
The source code is divided into two sub-Q: A.voip servers, mainly used to connect the two terminals, dial, hang up, answering and other signaling forwarding, voice number forwarded to each other. B.voip client, using voip common voice compression g.729, the compression ratio of up to 1/16, using 8K16bit sampling, the situation can be stabilized at a low speed to achieve double-talk, the software has been providing dial-up, holding machine, answering, busy signals and other basic functions of the phone. (2015-04-03, Visual C++, 3814KB, 下载12次)
[IP电话/视频会议] apache-openmeetings-2.1.0-src.tar
Openmeetings 是一个网络视频会议系统，提供视频，白板，doc，ppt，pdf，jpg等文档，图像文件的白板共享。其最大特点是视频会议客户端不需要下载安装，openmeetings系统将客户端做成swf形式，通过网页浏览的方式自动加载，免去用户下载安装的步骤，所有支持flash的ie浏览器都可以使用openmeetings进行视频会议。
The Openmeetings is a network of video conferencing systems, video, whiteboard, doc, ppt, pdf, jpg, and other documents, image files whiteboard sharing. Its greatest feature is video conferencing client does not need to download and install OpenMeetings client made swf form, automatically loaded through a web browser, replacing the user to download the installation steps, ie browser support flash can be used openmeetings for video conferencing. (2013-05-10, Java, 14765KB, 下载40次)
[IP电话/视频会议] chat-video
通过AnyChat音视频互动开发平台（SDK），可以开发具有企业特色的即时通讯系统、视频游戏系统、视频会议系统、网络教学系统以及在线客服系统等，系统的功能、界面完全由企业定制，底层通信协议加密传输，多重安全防护（参考： AnyChat的安全保障措施有哪些？），保密性强。
AnyChat SDK分为客户端SDK和服务器SDK两大部分
联系见下载包“联系方式”文本。
Can develop through AnyChat audio and video interactive development platform (SDK), the instant messaging system with enterprise features, video game systems, video conferencing systems, network teaching system, and online customer service system, the function of the system, the interface is completely customized by the enterprise, the underlying communication the protocol encrypted transmission, multiple security (Reference: What AnyChat security measures?), strong confidentiality. AnyChat SDK divided into the download package client SDK and server SDK two major contact, see " Contact" text. (2012-09-05, Visual C++, 7976KB, 下载16次)
[IP电话/视频会议] Asterisk_cn
Asterisk[1] 是一个开放源代码的软件VoIP PBX系统，它是一个运行在Linux环境下的纯软件实施方案。Asterisk是一种功能非常齐全的应用程序，提供了许多电信功能，能够把你的x86机 器变成你自己的交换机，还能够当作一台企业级的商用交换机。Asterisk让人激动的事情是它在小企业预算可承受的范围内提供了商业交换机的功能和可伸 缩性。你可以使用一台老式的奔腾3计算机，让你的机构看起来就同世界上的大企业一样。
Asterisk is an open source, converged telephony platform, which is designed primarily
to run on Linux. Asterisk combines more than 100 years of telephony knowledge into
a robust suite of tightly integrated telecommunications applications. The power of Asterisk
lies in its customizable nature, complemented by unmatched standards compliance.
No other PBX can be deployed in so many creative ways. (2012-03-30, DOS, 2210KB, 下载10次)
[IP电话/视频会议] Python
用pjsip的库编写web网络电话，说明：
1.先安装pjsip的库。
2.你需要两个sip账号，分别填入第114行和116行。
第一个参数是SIP服务器，第二是用户名，第三是密码。
3.被叫号在第49行，你可以修改成从文件读或者其他。
4.第131行，从E盘根目录获取“电话号码.call”文件。获取之后就把该文件删掉。并呼叫该号码。
5.第44行，等待音文件，必须是标准的wav文件，可以用windows自带的录音机自己录制，转码出来的有可能不被支持，会报异常，此处未作处理
6.本程序是拿来做网页免费电话（web800）用的，当用户在web上填入一个号码，就可以自动回拨他，播放等待音，并接通第49行设定的客服号码。
7.web部分考虑到各人服务器不一样，未给出，只需按格式写入一个空文件到131行设定的路径就可以
Written in the pjsip the library web network telephone, stating:
1 to install the pjsip the library.
You need two sip accounts, respectively, to fill 114 lines and 116 lines.
The first parameter is the SIP server, the second is the user name and password.
3 called in line 49, you can modify the file read or other.
Four. Line 131, obtained from the root directory of E telephone number call "file. Fetch and then put the file is deleted. And call the number.
5 line 44, waiting for the sound file must be a standard wav file, you can use windows built-in recorder to record your own out of transcoding may not be supported, exception will be reported here is not dealt with
This procedure is used as the web pages Toll-free (web800) used when the user fill in a number on the web, you can automatically call back him, the player waiting tone, and connected to the line 49 to set the customer service number.
7.web part of each one taking into account the server is not the same, is not given (2012-03-09, Python, 529KB, 下载76次)
[IP电话/视频会议] 201171312345296
这是一套完整的软实现VOD点播系统，系统主要应用在KTV、宾馆、自选电影室….场所，由前台开房收银、后台影片维护、客户端（房间）播放器几块组成，系统比较庞大，07年5.1开始开发，8月份结束，历时3个月，可供对VOD有兴趣的朋友研究，研究这套系统要有一定的耐心，相信里面的功能会对你有所帮助。
This is a complete software implementation VOD system, the system mainly used in KTV, hotel, cinema room choice .... Place, open house by the front cash register, background film to maintain the client (the room) player a few components, the system more large, 5.1 in 2007 began developing in August lasted three months, for friends who are interested on VOD study to the system to have some patience, I believe the function which will help you.
(2011-12-19, Visual Basic, 6423KB, 下载5次)
[IP电话/视频会议] Video-conferencing-system
视频会议系统使用户可以利用普通的PC机、标准的视频采集设备（USB摄像头或者视频头+视频采集卡）、耳机和麦克风，实现基于Internet、广域网、局域网的虚拟会议。与传统的基于硬件的H.323解决方案相比，本系统是纯软件解决方案。用户无需投入高昂的成本，就能够实现高质量、高可靠性的音视频通讯的会议功能，有效地节约时间和经费，提高企业的工作效率。
本视频会议系统由服务器软件和客户端软件组成。
Video conferencing system allows users to make use of common PC, a standard video capture device (USB cameras or video head+ video capture card), headphones and microphone, based on Internet, WAN, LAN virtual meetings. With traditional hardware-based H.323 solution, this system is a pure software solution. Users do not need the high cost of inputs to be able to achieve high-quality, high reliability, audio and video conferencing communication, effectively saving time and money, and improving the work efficiency. The video conference system consists of server software and client software. (2011-04-08, Visual C++, 6770KB, 下载116次)
[IP电话/视频会议] kamailio-3.0.2_linux_i386.tar
Kamailio是一个开源的SIP服务器，原名OpenSER
该版本主要修复了代码中的一些小问题，完善了文档，建议使用 3.0 和 3.0.1 版本的用户升级。
SIP Router (sip-router) is an industrial-strength, free VoIP server based
on the Session Initiation Protocol (SIP RFC3261).
It is engineered to power IP telephony and presence infrastructures up to large
scale. The server keeps track of users, sets up VoIP sessions, relays instant
messages and creates space for new plug-in applications. Its proven interoperability
guarantees seamless integration with components from other vendors, eliminating the
risk of a single-vendor trap. It has successfully participated in various
interoperability tests in which it worked with the products
of other leading SIP vendors. (2010-08-28, Unix_Linux, 6223KB, 下载8次)
[IP电话/视频会议] GlobalIPSound
Subject to Doc-Voip-2013
这是瑞典Global IP Sound公司的GIPS语音编解码引擎开发包（无实现代码，只提供库文件与.h文件）。该公司是全球顶尖的语音处理公司，Skype、Logitech、WebEx、TI、QQ的产品中都使用了该公司的语音开发包。
本下载包中包含GIPS的2部分产品：
VoiceEngine
能在最恶劣的网络下为VOIP提供最小延迟、最佳语音质量的软件开发包；
ConferenceEngine
为VOIP服务器软件（会议混音、会议中声音处理）提供支持的开发包；
开发包中还包括：AEC(自动回音消除)、AGC、VAD、DTMF、以及编码转换等各种处理接
Subject to Doc-Voip-2013 This is a Swedish company Global IP Sound GIPS voice codec engine development kit (no implementation code, only library files. H files). The company is the world' s leading voice processing company, Skype, Logitech, WebEx, TI, QQ' s products are used in the development of the company' s voice packets. The download package contains 2 parts of GIPS products: VoiceEngine in the worst under the VOIP network to provide the minimum delay, the best voice quality of the software development kit ConferenceEngine for the VOIP server software (conference mixing, sound processing session ) to provide support for the development package development kit also includes: AEC (Auto Echo Cancellation), AGC, VAD, DTMF, and then encoding conversion and other processing (2010-05-27, C/C++, 514KB, 下载158次)
[IP电话/视频会议] komssys-snap-19feb2007
出版《多媒体技术: 计算、通讯及应用(英文影印版)》牛人使用的RTP通信服务程序，可以用来做多媒体通信快速实验，只支持MPEG-1编码。服务器功能相当于IBM s VideoCharger.
The KOM(S) Streaming System provides a server, a client, and a proxy cache for audio/video streaming. The first and foremost encoding format that we use is MPEG-1 System. The primary platforms is Linux.
It does not handle data in pull mode (File, HTTP and FTP) but uses RTSP/RTP (and Files). It uses RTSP as control protocol to communicate with possible streaming servers. The video data is transported via RTP.
On all working platforms, a rudimentary but functional streaming server is available. On AIX 4.3, the server is able to act as a replacement RTSP server for IBM s VideoCharger. README for further information. Please read the file COPYRIGHT for copyright information.
(2009-10-25, Unix_Linux, 2515KB, 下载8次)
[IP电话/视频会议] vkControls
软件名： 语音聊天录音软件
想知道他/她在网上和别人语音聊过什么吗？
语音聊天录音软件是一款电脑自动录音软件,该监控软件能够自动记录所有QQ语音聊天、新浪UC语音聊天 、skype语音聊天及在线聊天室聊天等所有语音聊天内容。该软件也可以记录所有的环境声音，把软件装 在家里的电脑上后，就像在家里装上了一个监听器，所有电脑附近发出的声音都能够被完全地记录下来， 包括打电话的聊天内容等所有声音。
Software name: voice chat recording software would like to know his/her online voice chat and what others do? Voice chat software is a computer recording automatically recording software, the monitoring software can automatically record all voice chat QQ, Sina UC voice chat, skype voice chat and online chat rooms chat voice chat and all other content. The software can also record all the voices of the environment, the software installed on your computer at home after, just like at home fitted with a listener, all the sound near the computer can be fully recorded, including phone chat the contents of all the voices. (2009-07-12, Visual Basic, 250KB, 下载34次)
[IP电话/视频会议] VOIP7
详细描述了基于SIP的VoIP复杂的安全问题，并且阐述了身份认证、加载MIME消息体
来扩展SIP消息和加密包等解决方法。提出一套新的安全框架，在链路层和网络层(交换机)、传输和会话层(防火墙)及应用层(基f-jl~．务器的软件等)多个层实施安全措施来根本解决诸多安全隐患。
A detailed description of SIP-based VoIP complex security issues and expounded on authentication, loading MIME message body to extend the SIP messages and packages, such as encryption solution. Proposed a new security framework, in the link layer and network layer (switches), transport and session layer (firewall) and the application layer (base f-jl ~. Server software, etc.) more than the implementation of safety measures to the fundamental layer solve the many security risks. (2009-01-20, Others, 301KB, 下载2次)
[IP电话/视频会议] H323BeaconClient-v1.4_win_src
本视频会议系统使用户可以利用普通的PC机、标准的视频采集设备（USB摄像头或者视频头+视频采集卡）、耳机和麦克风，实现基于Internet、广域网、局域网的虚拟会议。与传统的基于硬件的H.323解决方案相比，本系统是纯软件解决方案。用户无需投入高昂的成本，就能够实现高质量、高可靠性的音视频通讯的会议功能，有效地节约时间和经费，提高企业的工作效率。
本视频会议系统由服务器软件和客户端软件组成。
The video conferencing system allows users to use an ordinary PC, the standard video capture device (USB cameras or video head+ Video capture card), headphones and microphone, based on Internet, WAN, LAN virtual meetings. With traditional hardware-based solutions compared to H.323, the system is a pure software solution. Users without the high cost of inputs, we can achieve high-quality, high reliability, communications, audio and video conferencing capabilities, effectively saving time and funds to enhance the work efficiency. The video conferencing system from server software and client software. (2008-09-11, Others, 4226KB, 下载150次)
[IP电话/视频会议] libosip2-3.1.0.tar
SIP是IETF致力于将电话服务带入IP网络众多协议的一个组成部分（它与SDP、RTP、RTCP、RTSP、RSVP、TRIP等众多协议构成SIP系统协议栈）. 这是sip协议栈实现最新版, 可用于开发sip客户端/代理/服务器
SIP is the IETF working on telephone services into the IP network number of an integral part of the agreement (with the SDP, RTP, RTCP, RTSP, RSVP, TRIP, and many other agreements constitute a SIP system protocol stack). This is the sip protocol stack to achieve the latest version, can be used to develop sip client/agent/server (2008-03-26, Windows_Unix, 640KB, 下载75次)
[IP电话/视频会议] H263example
这个程序有PC的服务端和Pocket PC上的客户端两个部分。服务端运行时，按下开始就可以从PC机的摄像头捕获视频，并且使用H.263编码进行压缩。这时服务端将在TCP的8765端口进行监听；这时运行客户端，输入服务器IP地址并点击“连接”，在网络正常的情况下，将实时显示服务端摄像头捕捉到的画面，画面的实时性取决于网络的带宽。
艕芒 赂 枚艂臍膼艌脫膼PC木脛牛脦艅艣脣艧脥Pocket PC脡膸木脛偶脥钮 搂 艣脣脕 藵 赂 枚 藳 偶脰膭艁牛脦艅艣脣脭脣膼膼臉膮艁殴 掳 (2007-07-24, C/C++, 455KB, 下载100次)
[IP电话/视频会议] iphone2
本程序自带反向回传语音代理服务，可以使两个不同局域网内的客户机通过国际互连网进行语音通讯，本程序使用点对点方式，不通过任何中介服务器速度快，语音延迟极小，声音清晰逼真，其效果和真实IP电话不相上下，程序界面简洁，操作简单，是在互连网上打IP电话的极好工具，唯一的要求是通话双方都要有电脑，并连入互连网。
本程序能实现以下连网操作功能：
1.通话双方是同一局域网内的用户
2.通话双方是不同局域网内的用户，并通过各自的网关连入互连网
3.通话双方一个直接连入互连网，另一个处在局域网中，并通过网关连入互连网
4.通话双方直接连入互连网
一般的语音聊天软件只能实现1，4两种连网功能，而本程序能完全支持所有这四种连网情况，因而能使所有用户享受到语音聊天所带来的好处.
reverse the procedure own voice back-agency service, can make two different LAN client through the Internet for voice communications, point-to-point use of this procedure, not through any intermediary server speed, voice minimal delay, loud realistic, and the real effect of IP phone lines, procedural interface simple, simple operation, the interconnection of the IP phone online playing an excellent instrument, the only requirement is to call both sides have to have a computer, and connect to the Internet. This program can achieve the following networking feature : 1. Both the caller is within the same LAN users. Both the caller is different users within the LAN and through their gateways connect to the Internet 3. Both the caller and a direct link into the Internet, and the other at LAN, a (2005-10-05, Delphi, 18KB, 下载156次)
[IP电话/视频会议] 35925983
网络电话。采用GSM语音格式，2.2KB/s的数据流量，支持HTTP, Socks4/5单一代理或二者同时代理。本版本修复了以前版本的一些小问题，增加了动态调整延时，打开、关闭话筒和声音，保存和自动加载通话地址列表，代理服务器列表，使用全部或部分本机地址开机
telephone network. GSM voice format, 2.2KB/s data traffic, supports HTTP, Socks4/5 of a single agent or combination of both agents. This version fixes previous versions of some small problem, the dynamic adjustment of delay, opening, closing and microphone voice, preservation and automatically load calls address list, the proxy list, the use of all or part of the machine addresses boot (2005-02-12, Delphi, 263KB, 下载534次)
[IP电话/视频会议] jrtplib-2.8
一个外国牛人用C++写的实时传输协议RTP,源代码的编写风格很好，可读性很强。可以用于网络视频和音频的传输，enjoy it
A foreign Niu Ren real-time transmission agreement RTP which
writes with C, the source code compilation style very good, the
readability is very strong. May use in the network video frequency and
the audio frequency transmission, enjoy it (2004-06-30, LINUX, 139KB, 下载947次)