最新的asterisk源码，pbx服务器，sip网络电话服务器，可实现模拟电话和网络电话互通-Asterisk source, pbx server, sip network telephony server, analog phones and VoIP interoperability

基于pjsip 1.10的开源代码实现的测试mediastream 功能的测试代码，支持流媒体的收发。此文件是模拟代理服务器端代码。-The open source code based on pjsip 1.10 the test code test mediastream functions to support streaming transceiver. The project is proxy server-side code.