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| <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> |
Hi ,
I'm beginner with jssip. I run your code. When i make call, after allow mic and camera,I receive:
JsSIP:RTCSession receiveInviteResponse() +1ms
JsSIP.min.js:27083 JsSIP:Dialog new UAC dialog created with status CONFIRMED +0ms
JsSIP.min.js:27083 JsSIP:RTCSession emit "sdp" +1ms
Uncaught TypeError: Failed to set the 'srcObject' property on 'HTMLMediaElement': The provided value is not of type '(MediaSourceHandle or MediaStream)'.
at RTCPeerConnection. (oldEg.html:182:39)
(anonymous) @ oldEg.html:182
JsSIP.min.js:27083 JsSIP:RTCSession session accepted +2ms
JsSIP.min.js:27083 JsSIP:RTCSession emit "accepted" +0ms
oldEg.html:123 accepted
Help me out
Hi @Cruzzhere, what browser are you using?
@echohes Google Chrome Version 108.0.5359.124
@Cruzzhere I can't check right now, I will have to run a test server. Try replacing the method with an object, in add_stream func:
const remoteAudio = document.createElement('audio');
remoteAudio.src = window.URL.createObjectURL(e.stream);
remoteAudio.play();
hi new with jssip while using ur code im getting document is not defined ,window is not defined how do i solve it please help
How can I turn on the webcam on my laptop to send video to subscriber 2?
document is not defined
Hi! Is your code running in the browser?
How can I turn on the webcam on my laptop to send video to subscriber 2?
I understand correctly, you have 2 active calls, and only one need activate video?
Hello
I am using this code and it works on android and desktop
But the sound does not play in the ios and chrome browser
In Safari it doesn't work at all !!!!!
Hello I am using this code and it works on android and desktop But the sound does not play in the ios and chrome browser
In Safari it doesn't work at all !!!!!
Hi!
As far as I remember, the mechanism of working with i/o devices is implemented differently in browsers. It is necessary to read the documentation for Safari. Because I did a different implementation between google chrome and firefox.
Thanks
Hello! I’m new to JSSIP and am exploring ways to improve the handling of outgoing calls. When I make an outgoing call, there’s no response if the callee is busy or their phone is switched off. Is there a way to enable early media or similar functionality in JSSIP to receive feedback in these cases? Any guidance or examples would be much appreciated. Thank you!
You can use method reinvite
or the MUTE mode for video call