openh323终端源码

SIP协议资料。收集与SIP协议相关的资料。是不错的参考资料。-information of sip.

使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RTP is to solve real-time streaming media transmission problems the best way, if necessary on the Linux platform for real-time streaming media programming, can consider using open-source library of RTP, as LIBRTP, JRTPLIB so. JRTPLIB is an object-oriented library RTP, which fully comply with the RFC 1889 design, in many occasions is a very good choice, following on to JRTPLIB example, on how the Linux platform on the use of RTP for real-time streaming media programming.

音视频聊天演示程序。聊天双方一个作为服务器，一个作为客户机。客户 机呼叫服务器建立连接之后，即可进行音视频聊天――双方都可以看到对方的图像， 可以听到对方的声音。 要求的系统配置：USB摄像头或数码摄像机，声卡，麦克风等。-Audio and video chat demo. Both sides as a chat server as a client. Client calls the server after establishing a connection, you can conduct audio and video chat- both sides can see each other

一个ip电话的实现我也不太清楚他是用什么语言编的-a telephone realize I did not know what he was part of the language

基于RTP的MPEG_4视频传输研,我的研究生毕业论文，供大家参考-RTP-based video transmission MPEG_4 the inquiry, my graduate thesis, for your reference

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

asterisk 电话未来之路 一本非常有意义的数据 中文asterisk文档 对做voip 和PBX的朋友们有很大帮助-Future of asterisk phone a very meaningful data to make Chinese asterisk documents voip and PBX friends are very helpful

this sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks -this is sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks

一个ip电话的源代码- A ip telephone source code

基于G729语音压缩技术的源码，适当修改就可以加到自己的代码里面，很实用-G729 voice compression technology, based on the DLL source code, with appropriate modifications can be added to your code inside, very practical