Sip 参数wsServers的值错误
我正在使用freeswitch 1.6并遵循cookbok来实现webrtc。为此,我下载sip。js@0.7.0也并创建了call.html、call.js和answer.html、answer.js页面。包含js的my call.html是Sip 参数wsServers的值错误,sip,sip-server,Sip,Sip Server,我正在使用freeswitch 1.6并遵循cookbok来实现webrtc。为此,我下载sip。js@0.7.0也并创建了call.html、call.js和answer.html、answer.js页面。包含js的my call.html是 <html> <body> <button id="startCall">Start Call</button> <but
<html>
<body>
<button id="startCall">Start Call</button>
<button id="endCall">End Call</button>
<br/>
<video id="remoteVideo"></video>
<br/>
<video id="localVideo" muted="muted" width="128px" height="96px"></video>
<!--<script src="js/sip-0.7.0.min.js"></script>-->
<!--<script src="call.js"></script>-->
</body>
<HEAD>
<script src="js/sip-0.7.0.min.js"></script>
<script>
var session;
console.log('hiiiiiiiiiiii')
var endButton = document.getElementById('endCall');
endButton.addEventListener("click", function () {
session.bye();
alert("Call Ended");
}, false);
console.log('hiiiii2')
var userAgent = new SIP.UA({
uri: 'sip:anonymous@gmaruzz.org',
wsServers: ["ws://call.sia.co.in:5066"],
authorizationUser: 'anonymous',
password: 'welcome'
});
console.log('hiiii3')
var startButton = document.getElementById('startCall');
startButton.addEventListener("click", function () {
session =userAgent.invite('sip:1010@139.59.17.63', options);
alert("Call Started");
}, false);
console.log('hiiii4')
var options = {
media: {
constraints: {
audio: true,
video: true
},
render: {
remote:document.getElementById('remoteVideo'),
local: document.getElementById('localVideo')
}
}
};
</script>
</HEAD>
</html>
开始通话
结束通话
var会议;
console.log('hiiiii')
var endButton=document.getElementById('endCall');
endButton.addEventListener(“单击”),函数(){
session.bye();
警报(“通话结束”);
},假);
console.log('hiiii2')
var userAgent=new SIP.UA({
uri:'sip:anonymous@gmaruzz.org',
wsServers:[“ws://call.sia.co.in:5066”],
authorizationUser:“匿名”,
密码:“欢迎”
});
console.log('hiiii3')
var startButton=document.getElementById('startCall');
addEventListener(“单击”,函数(){
session=userAgent.invite('sip:1010@139.59.17.63",选择),;
警报(“呼叫启动”);
},假);
console.log('hiiii4')
变量选项={
媒体:{
限制条件:{
音频:是的,
视频:真的
},
呈现:{
远程:document.getElementById('remoteVideo'),
local:document.getElementById('localVideo')
}
}
};
请纠正我哪里出错了。提前感谢。您必须将wsServers置于传输选项中,如:
var userAgent = new SIP.UA({
uri: 'sip:anonymous@gmaruzz.org'
transportOptions: {
wsServers: "ws://call.sia.co.in:5066"
},
authorizationUser: 'anonymous',
password: 'welcome'
你的语法是正确的。但是,尝试将
wsServers
作为字符串wsServers:“ws://call.sia.co.in:5066”
或wsServers:“ws://call.sia.co.in:5066/ws”
是的,我做了相同的操作,但错误仍然与参数“wsServers”的“无效值”ws://call.sia.co.in:5066/ws”相同。粘贴您在控制台中得到的准确错误<代码>异常代码:1条消息:“无效值[{“ws_uri”:”wss://call.sia.co.in:7443“}]对于参数“wsServers”“name:“CONFIGURATION\u ERROR”参数:“wsServers”Dude这很奇怪,你做错了什么!在您的代码中粘贴了wsServers:[“ws://call.sia.co.In:5066”]
,但错误显示[{“ws\u uri”:wss://call.sia.co.in:7443“}]
如何将ws
变成wss
并更改端口?