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)
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)
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)
本书是详细的sip协议入门指导书。SIP是类似于HTTP的基于文本的协议。SIP可以减少应用特别是高级应用的开发时间。由于基于IP协议的SIP利用了IP网络，固定网运营商也会逐渐认识到SIP技术对于他们的深远意义。
This book is a detailed guide book entry sip protocol. SIP is HTTP-like text-based protocol. SIP applications can be reduced, especially senior application development time. As the use of IP-based SIP IP network, fixed network operators will also come to realize the SIP technology for their far-reaching significance. (2011-12-05, Java, 5064KB, 下载11次)
协同视频会议系统解决方案
视频会议系统由服务器端软件、客户端软件组成。
视频会议系统是纯软件的解决方案，可运行于互联网、专网、VPN 等多种类型的网
络中，支持B/S 和C/S 架构，方便与企业网站、OA、ERP 等系统无缝整合；支持多台服
务器级联，均衡系统带宽的使用；支持更多用户并发，实现运营级视频会议的应用；实
现远程协同办公、多方视频会议等应用，方便用户部署和使用；除支持桌面终端和大型
会场终端外，还支持VOIP 和标准协议的硬件终端接入，可满足大型会场视频会议、个
人商务视频会议等多种形式会议需求。
Collaborative video conferencing solutions (2011-11-30, PDF, 4123KB, 下载15次)
你是不是在上网是很怕别人盗了你的密码呢，会不会真的中病毒了，连QQ都不敢开呢。不用怕不会误你事了，有了这个用mfc制作的聊天程序，你不会现要什么密码，更无密码可盗。因为你就是另一个QQ公司的老总
Are you afraid of someone else in the Internet is stealing your password it will not really of the virus, and even the QQ did not dare to open it. Do not be afraid you will not be misled things, with this chat program to use mfc produced, you do not now what to passwords, but no password can be stolen. Because you are another QQ the company' s CEOs (2010-02-03, MultiPlatform, 211KB, 下载2次)
yate是一个软交换的sip电话。也是一个voip服务器或客户端。
主要支持功能：
VoIP 服务器 VoIP 客户端
VoIP to PSTN 网关 PC2Phone and Phone2PC 网关
H.323 网守
H.323 多端点服务器
H.323<->SIP 转换代理
SIP session border controller
SIP 路由 S
IP 注册服务
Jingle 即时聊天 I
SDN passive and active recorder
IAX2服务器客户端
电话服务器和客户端
呼叫中心服务器 (会议，队列)
IVR语音交互应答
预付费，后付费电话卡系统
兼容Asteirsk的zaptel中继卡
支持linux /windows
Yate is a next-generation telephony engine while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate s flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. (2009-08-13, Windows_Unix, 1962KB, 下载249次)
Cisco的网络产品已经使世界发生了革命性的变化。准确地讲，Cisco网关、路由器和交换机构筑了Internet的主干网。每天都会有数亿美元的商业利润通过它们滚滚涌来。而近几年来发生的这些惊人变化还只是这场革命的开端。在网络短暂的历史进程中，最具活力的事件也许即将发生：把语音和数据技术结合成一种统一的传输模式。随着Cisco新的语音解决方案系列的发布，从传统的PBX语音解决方案到基于IP的语音解决方案的转移将会以指数级加速进行。
Cisco' s networking products has made the world a revolutionary change took place. Accurately speaking, Cisco gateways, routers and switches to build the Internet backbone. Every day there are hundreds of millions of dollars of business profits come rolling through them. And alarming in recent years these changes have occurred is only the beginning of the revolution. In the network of short-term historical process, perhaps the most dynamic upcoming events: the voice and data technologies into a single transfer mode. With the new Cisco voice solutions series of releases, from the traditional PBX voice solutions to IP-based voice solutions to the transfer will be accelerated exponentially. (2009-08-11, PDF, 5962KB, 下载2次)
】会话初始协议(srP)是IETF制订的多媒体通信系统框架协议之一，也是3GPP的rP多媒体子系统(IMs)的重要组成
部分。面对复杂、开放的因特网环境，SIP协议自身缺乏有力的安全机制，使其在安全性方面显得较为薄弱。该文从分析SIP
的安全威胁入手，针对SIP协议报文明文传送、缺乏有力的身份认证这两大脆弱性，从数据加密和身份鉴定两方面研究了相应
的安全解决方案，讨论了如何利用现有技术和手段改善SIP的安全性，并提出了进一步改善SIP安全性的一些思路。
】 Session Initiation Protocol (srP) is the IETF to develop a multimedia communication system, one of the framework agreement, rP is the 3GPP s Multimedia Subsystem (IMs) an important part. The face of complex, open Internet environment, SIP protocol lack strong security mechanisms, so that security seems weak. In this paper, from the analysis of security threats SIP start with SIP protocol for text transmission of civilization reported the lack of strong authentication vulnerability of these two, from data encryption and the identification of two studies the corresponding security solutions, discussed how to make use of the existing technology and means to improve the security of SIP and SIP to further improve the safety of some ideas. (2009-01-20, Others, 222KB, 下载16次)
Openh323是为开发使用 H．323协议在 IP网上进行多媒体通信的应用程序 而专门设计 的全功能协议栈 。
该协议栈封装 了大量 H．323实体，功能完善 ，是 v0IP领域一个相 当重要的基础项 目。首先介绍 Openh323项 目计
划．然后描述 了协议栈类库结构，着重说明类的划分和协议过程实现机制 ，并且结合 网守技 术实现的基本思路从进
程模式 、类的继承等方面对如何将 Openbh323协议栈运用到商用 VoIP软件的设计开发中进行探讨。
Openh323 is for the development of the use of H. 323 IP Internet protocol for multimedia communications applications specially designed full-featured protocol stack. The protocol stack Packaging a lot H. 323 entities, the functional improvement of the area v0IP a very important infrastructure projects. First introduced Openh323 project. Then describe the protocol stack library structure, focusing on such an agreement and the demarcation process Implementation Mechanism, Gatekeeper and integration technology to achieve the basic ideas from the process model, Class of areas, including inheritance to how to Openbh323 protocol stack used in commercial VoIP software design development explored. (2007-05-09, Unix_Linux, 135KB, 下载88次)
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)