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我是SIP-WebRTC的初學者,需要知道如何在freeswitch中配置websocket,在/etc/asterisk/http.conf中配置星號,但我不知道配置FreeSWITCH的,波紋管是我sip.jsfreeswitch和sip.js如何配置websocket
(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: '[email protected]',
//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);
}
)();
我的項目使用http://sipjs.com/
非常感謝所有!
大,非常好的,謝謝! –