语音聊天的程序，看看或许有帮助

如何成功的运用OPENH323 来开发商业的H.323 协议栈 Openh323 有以下的特征： 1>动态的静音检测算法, 以减小语音包的传输量。 2>支持Windows，Linux 和FreeBSD 的多种客户端命令。 3>包含有MCU，PSTN 网关，和自动语音应答机等多种应用平台。 4>软件支持GSM，LPC-10，G.711 uLaw/Alaw 软件编解码方式。 5>特定硬件（Quicknet/LineXJ）支持G.723.1, G.728 和 G.729 。 6>采用Jitter Buffer 技术对语音信号做接受的缓冲处理。 7>软件产生舒适噪音。 8>采用H.261 视频压缩协议。 9>支持广播的方式查找网守（GateKeeper）。 10>支持H.235 附件D 中和网守的身份认证。(部分地支持H.235) 11>支持部分H.450 补充协议。-How successful the use of OpenH323 to develop commercial H.323

MRCP协议V1版本的JAVA语言实现，做TTS和ASR很有用。-MRCP protocol V1 version of the JAVA language, to do TTS and ASR useful.

企业电话语音录音管理系统,本系统 采用人机对话的操作方式，界面设计美观友好、操作灵活、方便、快捷、准确、数据存储安全可靠。  系统可以进行来电显示和来电接听等功能。  提供电话拨出功能，可以通过程序拨出电话。  提供客户管理功能，记录客户信息，并可以在拨出电话时以电话簿的形式显示客户信息。  提供电话录音功能，当电话接通时开始录音，当电话挂断时结束录音。  提供录音管理功能，可以通过程序修改录音文件的存储位置。  系统最大限度地实现了易维护性和易操作性。  系统运行稳定、安全可靠。 -the use of man-machine dialogue mode of operation, beautiful and friendly interface design, operation flexible, convenient, fast, accurate, secure and reliable data storage.  system can receive Caller ID and call functions.  set aside to provide telephone functions can be allocated through the telephone process.  management capabilities to provide customers, record customer information, and can be allocated at the time of telephone directory information of the client form.  Telephone recording function to provide, when the start recording when connected to phone, phone to hang up when the end of recording time.  manag

voip电话语音信号编码，传输方式，大家顶！-voip telephone speech signal coding, transmission mode, the top U.S.!

Delphi写的IP电话源码-IP phone sourcecode writed by Delphi

RealNetworks公司的Helix 的公共头文件部分源码．-RealNetworks Helix of the public part of the first source document.

搞VOIP的一些原代码，包括MGCP,SIP,RTP/TRCP,SRTP(加密RTP)...等源码，参考价值无限！！这可是我所有的收藏阿，我都献给大家了:=)-engage in a number of VoIP original code, including MGCP, SIP, RTP/TRCP, SRTP (encrypted RTP) ... other source reference value unlimited! ! But it all my collections A, I have dedicated to you : =)

可以直接用VC进入修改编译的基于语音卡上的开发平台，有流程的图形编辑器-can directly enter the VC changes compiler- based voice on the card development platform, a process graphics editor

Delphi写的IP电话源码-IP phone sourcecode writed by Delphi

KPhone is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet. KPhone requires KDE (K Desktop Environment) or at least various libraries from the KDE package. KPhone also requires the OpenSSL libraries to be present in the system. KPhone has been somewhat tested on Red Hat 7.2 and 7.3, Slackware 8.0, SuSE 8.1 and Debian. Requirements on each distribution vary according to the installation layout.