dota游戏软件开发

代码来自sourceforge，在linux下的实时语音聊天程序，使用了adpcm编码，同时还使用了RTP实时传输协议，是一个很好的学习实时传输协议的程序。- The code comes from sourceforge, chats the procedure under the linux real-time pronunciation, has used the adpcm code, meanwhile has used the RTP real-time transmission agreement, is a very good study real-time transmission agreement procedure.

symbian平台下的基于SIP的简单游戏，可以方便地扩展成VoIP通话。-symbian platform simple SIP-based games, can be easily extended into a VoIP call.

这是最新的IETF有关P2PSIP的研究的文档，拿来给大家共享一下-This the latest IETF P2PSIP of the documents, used to share!

本文提供了在自己的系统中嵌入视频点播的技术，希望对各位有所帮助。-in their embedded systems video-on-demand technology, you want to help.

用于Voip的MySQL数据库接口函数库，很有用处。-for the MySQL database interface functions, the very useful.

using G.729 language real-time communication [G729b_v14Code.rar] - G729B the realization code, and G729A difference is he could support multithreading codec, multi-channel audio simultaneously decode [MChat.rar] - a LAN can video chat source code, voice compression using G729. H.263 video compression and network transmission using RTP. The procedures of the basi -using G.729 language real-time communication [G729b_v14Code.rar]- G729B the realization code, and G729A difference is he could support multithreading codec, multi-channel audio simultaneously decode [MChat.rar]- a LAN can video chat source code, voice compression using G729. H.263 video compression and network transmission using RTP. The procedures of the basi

在智能平台下使用PHONECALL 控件。实现拨号连接-Platform in the smart use phonecall control. Realize a dial-up connection

视频会议系统，使用RTP协议。可以实现多人视频语音会议-video conferencing systems, the use of RTP. People can achieve video voice conferencing

VoIP正在arapidly不断发展的技术，可以彻底改变可能的电信业。当wirelessdata网络实施，网络电话可能被证明是有助于收敛ofexisting固定和移动电话网络与快速增长的有线andwireless数据网络。服务的能力andQuality是两个需要无线VoIP的商业部署之前beresolved最重要的问题。本文开头的数据adetailed解释网络与问候网络建模与流量分布在一起。的VoIP，WLAN和VoIP在WLAN已经解释的优势，劣势和实施基于WLAN的VoIP的问题。不同的VoIP传输系统：RTP，RTSP，SIP和H323也进行了审查。一个8节点的移动电话系统hasbeen使用NS2和实验设计已经进行测量thesystem在提供服务的质量方面通过一系列的因素，包括多个节点，包大小和客流率。如延迟，吞吐量和packetlost参数已经测量通过WLAN来评估的VoIP的服务质量。基于一个NS2代码和differentparameters如延迟，吞吐量和数据包丢失的打印输出包括以supportthis论文。

跨平台（Linux，Windows）的开源SIP服务器端-Cross-platform (Linux, Windows) open-source SIP server-side