cisco ivr program introduced the IVR into cisco is cisco ivr basic information

An H.264-Based Solution on the DM642 for Video Broadcast Applications

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 us

RTCP.RTP方面的源码,方便大家学习.-RTCP.RTP source area for everyone to learn.

voip实现的原代码的一个部分 sip协议的实现-voip realize the original code a part of the agreement realize sip

在防火墙情况下,使用stun方式穿越防火墙的音视频代码-Firewall cases, the use of stun way audio and video through the firewall code

国外牛人的毕业论文，英文版本，实时语音传输相关，其中的两个库文件，jrtplib与jvoiplib已经上传。enjoy -foreign elite s graduate paper in english.related to real time voice transimission.it has two library,jrtplib,jvoiplib.enjoy

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.

SIP协议中穿越对称型/非对称型NAT防火墙的多种实现方法，解决私网内部的防火墙穿透问题。-SIP cross symmetric/asymmetric NAT firewall multiple solutions, private internal network firewall penetration.

网络电话例程，可以连接本机做测试！--Network telephone example, and it can connect to local machine to do a test!

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