open IVR platform can be directly based on the use of cards DIALOGIC

H.323规范的一部分，规定的信令部分的控制，是开发H.323不可缺少的文档。-part of the H.323 standard, signaling the part of the control, development of H.323 is an indispensable document.

可以用vc编译的一个sip 客户端,使用一个基于osip,exosip,ortp和mediastreamer的高层库.有该库的使用文档-Vc compiler can use a sip client, based on the use of a osip, exosip, ortp and high-level Treasury mediastreamer. The use of the library documents

2.0说明 1.0支持系统的录音多种格式,有一定的延时,没有过滤杂音功能. 2.0已取消支持系统的录音多种格式,使用PCM采集数据G711A压缩格式(8000HZ单声道16位格式录音每秒以8K完成数据,16000HZ单声道16位格式录音每秒以11K完成数据,音质相当好),延时降低到最小100-500MS以内,不会随时间增加而增加延时(如果是说话测试一直保持200MS的延时,如果是用播放歌曲来测试,有自动校正延时功能,恢复成200MS的延时,恢复过程中不会中断歌曲的播放,只是小小加快唱歌的速度,听觉不会觉察出来),加入了过滤杂音功能. 2.0安装 将ACMG711.dll复制到系统文件夹里 再安装控件 -2.0 1.0 Note recording system to support multiple formats, a certain delay, no noise filtering function. 2.0 support system has been abolished recording formats, the use of PCM data acquisition G711A compression format (8000HZ 16 mono recording format to 8K per second complete data 16000HZ mono recording format 16 to 11K per second complete data quality is very good), Delay reduced to the minimum 100-500MS within not with time delay increases (if word has been tested 200MS delay, If it is used to test play the songs, the automatic correction function of delay, 200MS resume into the delay, the recovery process will not be interrupted play songs, singing just a small speed up

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

分网络传输部分、音频输入、音频输出、视频输入、视频输出、音频视频压缩、音频波形显示、界面。共创建个线程，音频输入线程、音频输出线程（这两个有WINAPI隐式创建）、音频视频压缩传送线程和主线程。因为音频的采样率比视频采样率高得多，这里的音频的采样率为8000Hz而视频仅为20左右，所以从宏观来看，音频是连续的，而视频是非连续的，这也就是无须为视频输入输出创建独立线程的原因。- Minute network transmission part, audio frequency input, audio frequency output, video input, video output, audio frequency video frequency compression, audio frequency profile demonstration, contact surface. Altogether founds the line regulation, the audio frequency input line regulation, the audio frequency output line regulation (these two has WINAPI to hide type to found), the audio frequency video frequency compression transmission line regulation and the master line regulation. Because the audio frequency sampling rate is much higher than the video frequency sampling rate, here audio frequency sampling rate is 8000Hz but the video frequency is only about 20, therefore from macroscopic looked, the audio frequency is continual, but video frequency right and wrong continual, this also

gsm算法最新源码，适用于语音压缩，如gsm和voip系统-gsm latest source algorithm applicable to voice compression, such as GSM and voip system

Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router.

OpenJSIP 使用JAIN实现了一个完整的  SIP注册服务器和SIP代理服务器，并采用分布式架构提高性能和并发数。                                                                        At the current stage of development the project includes: 请点击左侧文件开始预览 ！预览只提供20%的代码片段，完整代码需下载后查看 加载中 侵权举报

视频压缩编解码ITU h.263源代码.经典的视频会议和远程监控算法.- The video frequency compression arranges decodes ITU the h.263 source code Classical video frequency conference and long-distance monitoring algorithm

RealNetworks公司的Helix 的协议部分源码．-RealNetworks Helix source part of the agreement.