echohes · August 3, 2018 07:20 · tamtc84 · Nov 5, 2019 · echohes · Nov 5, 2019
diff --git a/jssip_test.html b/jssip_test.html
 <html>
    <head>
        <meta charset="utf-8" />
        <meta name="viewport" content="width=device-width" />
        <title>Sip demo</title>
    </head>

 <body>
 <head><title>WebRT</title>
  <style>
    video { height: 240px; width: 320px; border: 3px solid grey; }
  </style>
 </head>
        <video id="selfView" autoplay muted=true></video>
        <video id="remoteView" autoplay></video>
 </body>

 <script src="jssip-3.2.11.min.js"> </script>
 <script>

 var socket = new JsSIP.WebSocketInterface('wss://${{SERVER}}/ws');
 socket.via_transport = "tcp";

 //Create HTML Audio Object
 var remoteAudio = new window.Audio()
 remoteAudio.autoplay = true;

 const mediaSource = new MediaSource();

 var selfView = document.getElementById('selfView');
 var remoteView = document.getElementById('remoteView');

 var user = "${{USERNAME}}";
 var pass = "${{PASSWORD}}";

 var userAgent = JsSIP.version;

 console.log('sip:%s@${{SERVER}}', user);

 var configuration = {
  'uri': 'sip:'+ user + '@${{SERVER}}',
  'password': pass, // FILL PASSWORD HERE,
  'sockets': [ socket ],
  'register_expires': 180,
  'session_timers': false,
  'user_agent' : 'JsSip-' + userAgent
 };

 var phone;
 if(user && pass){
    JsSIP.debug.enable('JsSIP:*');
    phone = new JsSIP.UA(configuration);
    phone.on('registrationFailed', function(ev){
      alert('Registering on SIP server failed with error: ' + ev.cause);
      configuration.uri = null;
      configuration.password = null;
    });

    phone.on('newRTCSession',function(ev){
        var newSession = ev.session;

        if(session){ // hangup any existing call
            session.terminate();
        }
        session = newSession;
        var completeSession = function(){
                        session = null;
        };


        if(session.direction === 'outgoing'){
                console.log('stream outgoing  -------->');               
        session.on('connecting', function() {
            console.log('CONNECT'); 
                        });
        session.on('peerconnection', function(e) {
            console.log('1accepted');
                        });
        session.on('ended', completeSession);
        session.on('failed', completeSession);
        session.on('accepted',function(e) {
            console.log('accepted')
                        });
        session.on('confirmed',function(e){
            console.log('CONFIRM STREAM');
                        });

                };

        if(session.direction === 'incoming'){
            console.log('stream incoming  -------->');               
        session.on('connecting', function() {
            console.log('CONNECT'); 
                        });
        session.on('peerconnection', function(e) {
            console.log('1accepted');
            add_stream(); 
                        });
        session.on('ended', completeSession);
        session.on('failed', completeSession);
        session.on('accepted',function(e) {
            console.log('accepted')
                        });
        session.on('confirmed',function(e){
            console.log('CONFIRM STREAM');
                        });

                var options = {
        'mediaConstraints' : { 'audio': true, 'video': true },
        'pcConfig': {
          'rtcpMuxPolicy': 'require',
          'iceServers': [
                                                ]
                                        },
                                };
            console.log('Incoming Call');
            session.answer(options);
           }
    });
    phone.start();
 }

 var session;

 function callAsterisk(numTels) {
 var options = {
        'mediaConstraints' : { 'audio': true, 'video': true },
        'pcConfig': {
          'rtcpMuxPolicy': 'require',
          'iceServers': [
          ]
        },
      };

    var numTel = numTels.toString();
    var num = '200';
    console.log(numTel);
    phone.call(numTel, options)
   add_stream();
 };

 function add_stream(){
                session.connection.addEventListener('addstream',function(e) {
                remoteAudio.srcObject = (e.stream);
                remoteView.srcObject = (e.stream);
                selfView.srcObject = (session.connection.getLocalStreams()[0]);
        })
 }

 </script>
 <p>
 <a href="javascript:callAsterisk(123)">Test</a>
 <a href="javascript:callAsterisk(777)">Echo</a>
 <a href="javascript:callAsterisk(501)">Call to 501</a>
	<html>
	<head>
	<meta charset="utf-8" />
	<meta name="viewport" content="width=device-width" />
	<title>Sip demo</title>
	</head>

	<body>
	<head><title>WebRT</title>
	<style>
	video { height: 240px; width: 320px; border: 3px solid grey; }
	</style>
	</head>
	<video id="selfView" autoplay muted=true></video>
	<video id="remoteView" autoplay></video>
	</body>

	<script src="jssip-3.2.11.min.js"> </script>
	<script>

	var socket = new JsSIP.WebSocketInterface('wss://${{SERVER}}/ws');
	socket.via_transport = "tcp";

	//Create HTML Audio Object
	var remoteAudio = new window.Audio()
	remoteAudio.autoplay = true;

	const mediaSource = new MediaSource();

	var selfView = document.getElementById('selfView');
	var remoteView = document.getElementById('remoteView');

	var user = "${{USERNAME}}";
	var pass = "${{PASSWORD}}";

	var userAgent = JsSIP.version;

	console.log('sip:%s@${{SERVER}}', user);

	var configuration = {
	'uri': 'sip:'+ user + '@${{SERVER}}',
	'password': pass, // FILL PASSWORD HERE,
	'sockets': [ socket ],
	'register_expires': 180,
	'session_timers': false,
	'user_agent' : 'JsSip-' + userAgent
	};

	var phone;
	if(user && pass){
	JsSIP.debug.enable('JsSIP:*');
	phone = new JsSIP.UA(configuration);
	phone.on('registrationFailed', function(ev){
	alert('Registering on SIP server failed with error: ' + ev.cause);
	configuration.uri = null;
	configuration.password = null;
	});

	phone.on('newRTCSession',function(ev){
	var newSession = ev.session;

	if(session){ // hangup any existing call
	session.terminate();
	}
	session = newSession;
	var completeSession = function(){
	session = null;
	};


	if(session.direction === 'outgoing'){
	console.log('stream outgoing -------->');
	session.on('connecting', function() {
	console.log('CONNECT');
	});
	session.on('peerconnection', function(e) {
	console.log('1accepted');
	});
	session.on('ended', completeSession);
	session.on('failed', completeSession);
	session.on('accepted',function(e) {
	console.log('accepted')
	});
	session.on('confirmed',function(e){
	console.log('CONFIRM STREAM');
	});

	};

	if(session.direction === 'incoming'){
	console.log('stream incoming -------->');
	session.on('connecting', function() {
	console.log('CONNECT');
	});
	session.on('peerconnection', function(e) {
	console.log('1accepted');
	add_stream();
	});
	session.on('ended', completeSession);
	session.on('failed', completeSession);
	session.on('accepted',function(e) {
	console.log('accepted')
	});
	session.on('confirmed',function(e){
	console.log('CONFIRM STREAM');
	});

	var options = {
	'mediaConstraints' : { 'audio': true, 'video': true },
	'pcConfig': {
	'rtcpMuxPolicy': 'require',
	'iceServers': [
	]
	},
	};
	console.log('Incoming Call');
	session.answer(options);
	}
	});
	phone.start();
	}

	var session;

	function callAsterisk(numTels) {
	var options = {
	'mediaConstraints' : { 'audio': true, 'video': true },
	'pcConfig': {
	'rtcpMuxPolicy': 'require',
	'iceServers': [
	]
	},
	};

	var numTel = numTels.toString();
	var num = '200';
	console.log(numTel);
	phone.call(numTel, options)
	add_stream();
	};

	function add_stream(){
	session.connection.addEventListener('addstream',function(e) {
	remoteAudio.srcObject = (e.stream);
	remoteView.srcObject = (e.stream);
	selfView.srcObject = (session.connection.getLocalStreams()[0]);
	})
	}

	</script>
	<p>
	<a href="javascript:callAsterisk(123)">Test</a>
	<a href="javascript:callAsterisk(777)">Echo</a>
	<a href="javascript:callAsterisk(501)">Call to 501</a>