webrtc接入freeswitch的sip音视频传输

标签：ram   live   log   pem   targe   free   sip   代码   dir   

1、安装freeswitch

https://www.cnblogs.com/dong1/p/10412847.html

我将fs安装到了百度云，按我这个配置就行，默认已经打开了ws和wss

    <!-- for sip over websocket support -->
    <param name="ws-binding"  value=":5066"/>

    <!-- for sip over secure websocket support -->
    <!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
    <param name="wss-binding" value=":7443"/>

2、下载webrtc客户端sipml5(sipjs/jssip也类似)

https://github.com/DoubangoTelecom/sipml5

3、在sipml5根目录启动一个http服务

1）python -m SimpleHTTPServer &

2）访问http://192.168.18.130:8000/

也就是主机ip:8000，http默认是8000端口

如果你想更改http服务的ip和端口，就不能用python默认自带的http服务了，需要写一小段代码

import SimpleHTTPServer
import SocketServer
ip = "127.0.0.1"
port = 8000
Handler = SimpleHTTPServer.SimpleHTTPRequestHandler
httpd = SocketServer.TCPServer((ip, port), Handler)
print("http://%s:%s" %(ip,port))
httpd.serve_forever()

启动 python SimpleHTTPServer_demo.py &

3）进入Enjoy our live demo

4）填写sip账号信息和ws接入信息

5）拨号

end

参考设计

基于freeSWITCH的sip协议利用WebRTC 实现实时视频聊天

https://blog.csdn.net/graceup/article/details/79485976

webrtc接入freeswitch的sip音视频传输

标签：ram   live   log   pem   targe   free   sip   代码   dir   

原文地址：https://www.cnblogs.com/dong1/p/12148731.html