搜索资源列表
jrtplib-2.8.rar
- 一个外国牛人用C++写的实时传输协议RTP,源代码的编写风格很好，可读性很强。可以用于网络视频和音频的传输，enjoy it！！！
siproxd-0.5.7.tar
- 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.
astguiclient_1.0.3
- This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32.
MyPhone12b
- 一个基于Openh323的视频会议终端，基于VC开发环境，可以与netmeeting互联。-It is a video meeting terminal based on Openh323 protocol, which developed with VC and can communicate with netmeeting.
microsoft-v2[1].5-040207-NTU
- microsoft的视频会议程序源码，非常的复杂，但是功能强大-Here is the SourceCode of MicroSoft s VideoMeeting,It s really Complicated,but good use
回声消除算法
- 回声消除算法的C＋＋程序，运行在unix/linux上， 也可以在vc中被编译， 测试， 效果还行-The program file of the algorithm for echo cancellation written by C++, can be runed on unix/linux, and also can be complied in VC It efficient good by testing.
vis_h323_dll_2.5r
- H.323协议源码开发包，可以很方便地开符合标准的H.323终端或MCU，内容已经过测试-H.323 source development kits, it is easy to open sub-standard H.323 terminals or MCU, which has been tested
MS语音聊天程序VB
- 一个ＭＳ的语音聊天程序，是用ＶＢ写的，很不错，不是用控件实现的，不过不是实时语音，有点像对讲机-a voice chat program is written in VB, very good, not with the controls to achieve, but it is not real-time voice, a bit like a walkie-talkie
Bank Demo
- 这是一个银行电话系统,用api做的.对与刚进入这个行业的开发人员有用,大家看看吧-This is a telephone banking system, with api do. With just entering the industry to develop useful, we look at it
sipXportLib
- sipXportLib is a library of general purpose utilities and OS abstraction APIs. Being a library, it, itself has no end user application however this project also contains end user utilities related to this library s functionality.-sipXportLib is
sipXcallLib
- sipXtapi is a comprehensive client library and software development kit (SDK) for SIP-based user agents. It includes SIP signalling support as well as a media framework. A complete and very feature rich softphone can be built easily by adding a graph
zlib123
- The deflation algorithm used by gzip (also zip and zlib) is a variation of LZ77 (Lempel-Ziv 1977, see reference below). It finds duplicated strings in the input data. The second occurrence of a string is replaced by a pointer to the previous
ChatRoom+G
- 这是关于语音聊天软件的一个程序代码 当运行该软的时候，至少需要有一方按下“建立服务器”按钮，表示该台机子用于做服务器，成功时文本显示区会显示“Server Has Been Set OK!”，才有可能接受来自其他机子的连接，当然客户端要输入服务器的IP地址和指定端口后，然后按“连接服务器”按钮，若成功则会在本地文本显示区显示“Client Connection Succeed”，双方建立了连接后，就可以开始通信了。你可以在文本输入区输入你想说的话，按回车，就会在本地文本显示区显示你刚才说的话
eXosip-0.7.6.tar
- eXosip is a GPL library that extend the capability of the oSIP library. It aims to implement a simple high layer API to control the SIP for sessions establishements and common extensions.
IPphone2.0
- 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
TalkG726
- 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
video_capturevfm-matlab
- matlab的视频捕获程序，希望对大家有用。谢谢了。-Matlab Video Capture procedures in the hope that it may be useful right. Thank you, the.
iad132e
- IAD （Integrated Access Device ）132E(T) 综合接入设备（下文简称IAD132E(T) ）是华为技术有限公司下一代网络NGN（Next Generation Network ）解决方案中的重要部件，用以向公司等用户提供小容量VoIP （Voice over IP）/FoIP（Fax over IP ）解决方案。IAD132E(T) 作为VoIP/FoIP 媒体接入网关，应用于NGN 用户接入层，完成模拟话音信号与IP 包之间的转换，并通过包交换网络传送数据的功能
eXosip-0.7.6
- SIP Stacks GNU eXosip: C Linux+Windows+... License: GPL Homepage: http://savannah.nongnu.org/projects/exosip/ The extensible osip: \"...It aims to implement a simple high layer API to control the SIP for sessions establishements and common extensio
VOIP_SRC_ALL
- 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,