P2P

sip开源协议 sip开源协议-sip protocol sip open source open-source agreement

一个功能全面的基于H323（视频会议系统协议）的网守-a fully functional based on H323 (Video Conference System agreement) Gatekeeper

Vovida出品的包括RTP、RTCP、SIP等各种基础协议栈和SIP UAC、UAS等功能的IP电话服务器端程序。编译环境为linux-Vovida produced in RTP, RTCP, and other SIP based protocol stack and SIP UAC and UAS capabilities of IP telephony server procedures. Linux compiler environment

应用背景This section shows and describes a call flow for a successful call using the Cisco ATA and MGCP. Note :The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated. 关键技术Figure C-1 on page C-2 illustrates a basic call flow between two Cisco ATAs through a VocalData Call Agent. Table C-1 on page C-3 describes the action the Cisco ATA takes for each step in the call flow illustration and includes the log created by each step.

简单的局域网中用的IP电话程序-A simple IP phone program which used in LAN network

通过此程序可以对所有呼入和呼出的电话进行录音实现实时监控的功能-Through this procedure for all incoming and outgoing telephone recording realize real-time monitoring function

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字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。 使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。 PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC- and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to

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 : =)

系列工具的提供 这行的一定要看！-series of tools that provide a look at the OK!

详细介绍了由（RTC）API创建的音视频会议、共享应用程序、点对点对话和视频聊天优化向导界面。