use H.323 voice calls for the procedure

G726局域网语音通话程序和源代码 这是使用G726语音压缩（16kbps）和RTP进行传输的程序，因为我没有带WIFI的PPC，所以每个程序都是单独测试的，PC端和PPC端分别都工作正常。 G726编解码算法来自OpenH323.传输使用的RTP可以在RTP程序中找到讲解，这个程序主要是G726的函数。将整个 G726封装为g726_Encode和g726_Decode两个函数，参数为压缩和解压数据存储的地址指针，可以将960字节压缩到120字节和将 120字节解压为960字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。 使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。 PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC- and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to

ITU g723标准参考代码-ITU standard reference source

QtPhone is a cross-platform telephone emulation application designed to be used with a PC, speakers, microphone, V.92 voice modem and a standard phone line. ISDN, xDSL and VoIP will be supported. Features include Caller Identification and Voice Mail. -QtPhone is a cross-platform telephone emu IBM application designed to be used with a PC , speakers, microphone, V.92 voice modem and a standard phone line. ISDN , xDSL and VoIP will be supported. Features inc lude Caller Identification and Voice Mail.

国外牛人用C++编写的voip的库文件，多媒体实时流传输协议（RTP/RTCP）lib包,平台要求(linux,unix,windows)用于网络音频视频传输-foreign cattle with a C voip prepared by the Treasury, the documents, multimedia real-time streaming protocol (RTP/RTCP) lib packages, platform requirements (Linux, Unix, windows) for audio and video transmission network

introduce to sip h323

openh323终端源码-openh323 terminal source

用于Voip的MySQL数据库接口函数库，很有用处。-for the MySQL database interface functions, the very useful.

大名鼎鼎的speex语音压缩源码，应用于voip系统-well known speex voice compression souce code,apply to voip system

在开发 VoIP 系统时，主要考虑的一个方面就是连接到模拟电话的接口。设计人员必须了解 PSTN 中存在的电话接口要求，因为它们必须在 VoIP 系统中得到支持。本文将重点介绍连接到标准 POTS 电话的两种最常用接口：Foreign eXchange Subscriber (FXS) 与 Foreign eXchange Office (FXO)。重点说明了设计人员在为 VoIP 小区网关的模拟电话接口提供支持时可能面临的某些挑战。-In the development of VoIP system, one aspect of the main considerations is connected to the analog telephone interface. Designers must understand the existing PSTN telephone interface requirements, because they are in the VoIP system must be supported. This article will focus on the standard POTS telephone connected to the two most commonly used interface: Foreign eXchange Subscriber (FXS) and Foreign eXchange Office (FXO). Focusing on the design staff in the Residential Gateway for VoIP analog telephone interface may provide support to some of the challenges faced.

版本V4.1.1,Linux上的开放源代码的IP电话软件SIP协议，主要用于语音通话，但它也支持发送文本的即时消息，即可以像ICQ那样地用文字聊天。 如果网络带宽足够的话，它还支持可视电话 Kphone是GPL的开放源代码的软件。-version 4.1.1, It is IP phone software of Linux open source code based on SIP protocol, it can be used to voice communication, and surpport text IM, same function as ICQ used to text chat. In broad band circumstance, it surpport video conference. Kphone is GPL open source software .