Question

我是SIP-WebRTC的初学者，需要知道如何在/etc/asterisk/http.conf中配置星号中的freeswitch配置websocket，但我不知道在freeswitch中配置，下面是我的sip。 JS

＆＃13;

( function()
  {
     var session;
     var endButton = document.getElementById('endCall');
     endButton.addEventListener("click", function (){
           session.bye();
           alert ("Call Terminated");
           }
           , false
                               );


     //Registration via websocket 
     var config = {
                        // my extension and ip of freeswitch
                        uri: '4009@10.20.11.10',

                        //in asterisk i used some how this. here is my problem :( how to do it in freeswitch?
                         wsServers: 'ws://192.168.0.3:8088/ws',

                        //here is my 4009
                        authorizationUser: '4009',

                        // my password
                        password: 'testsip',

                        
                        traceSip: true,


                        stunServers: 'null',
                 };


   
     var userAgent = new SIP.UA (config);

     var options = {

                     media: {
                              constraints: {
                                             audio: true,
                                             video: false,
                                           },
                              render: {
                                        remote: {
                                                   audio: document.getElementById('remoteAudio')
                                                },

                                        local:  {
                                                   audio: document.getElementById('localAudio')
                                                }
                                      }
                           }
    };



    function onAccepted ()
    {
        alert("Call Connected");
    }

    function onDisconnected ()
    {
        alert("Call Terminated");
    }


    //makes the call
    session = userAgent.invite('1000', options);
    session.on('accepted', onAccepted);
    //session.on('disconnected', onDisconnected);

  }

)();

＆＃13;

我的项目使用http://sipjs.com/

非常感谢所有人!!!

Answer 1

我假设您已经安装并运行了FreeSwitch实例。在定义用于侦听的套接字的conf文件中，您需要取消注释用于侦听的ws和wss端口。这应该让实例从sip.js侦听WebSocket消息。

<param name="ws-binding"  value=":80"/>
<param name="wss-binding"  value=":443"/>

了解更多信息 - https://wiki.freeswitch.org/wiki/Webrtc

freeswitch和sip.js如何配置websocket

1 个答案: