https://quantum6.blog.csdn.net/article/details/106031371
在这一篇文章中，没有登录成功，自然也无法呼叫。经过痛苦的过程，终于找到了正确办法：
参考：
端口不要改，使用默认的。
不知道从哪里复制来的，非常感谢。确实登录成功。
<!DOCTYPE html>
<title>JsSIP + WebRTC + freeSWITCH</title>
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name="Author" content="foruok" />
<meta name="description" content="JsSIP based example web application." />
<script src="jssip-3.4.4.min.js" type="text/javascript"></script>
<style type="text/css">
</style>
</head>
<div id="login-page" style="width: 424px; height: 260px; background-color: #f2f4f4; border: 1px solid grey; padding-top: 4px">
<table border="0" frame="void" width="418px">
<td class="td_label" width="160px" align="right"><label for="sip_uri">SIP URI:</label></td>
<td width="258px"><input style="width:250px" id="sip_uri" type="text" placeholder="SIP URI (i.e: sip:alice@example.com)"
value="sip:1011@192.168.1.111:5060" /></td>
<td class="td_label" align="right"><label for="sip_password">SIP Password:</label></td>
<td><input style="width:250px" id="sip_password" type="password" placeholder="SIP password"
value="1234" /></td>
<td class="td_label" align="right"><label for="ws_uri">WSS URI:</label></td>
<td><input style="width:250px" id="ws_uri" class="last unset" type="text" placeholder="WSS URI (i.e: wss://example.com)"
value="wss://192.168.1.111:7443" /></td>
<td class="td_label" align="right"><label class="input_label" for="sip_phone_number">SIP Phone Info:</label></td>
<td><input style="width:250px" id="sip_phone_number" type="text" placeholder="sip:3000@192.168.40.96:5060"
value="sip:1001@192.168.1.111:5060" ></td>
<td colspan="2" align="center"><button onclick="testStart()"> Initialize </button></td>
<td colspan="2" align="center"><button onclick="testCall()"> Call </button></td>
<td colspan="2" align="center"><button onclick="captureLocalMedia()"> Capture Local Media</button></td>
</table>
<div style="width: 424px; height: 324px;background-color: #333333; border: 2px solid blue; padding:0px; margin-top: 4px;">
<video id="videoView" width="420px" height="320px" autoplay ></video>
</body>
<script type="text/javascript">
var outgoingSession = null;
var incomingSession = null;
var currentSession = null;
var videoView = document.getElementById('videoView');
var constraints = {
audio: true,
video: true,
mandatory: {
maxWidth: 640,
maxHeight: 360
URL = window.URL || window.webkitURL;
var localStream = null;
var userAgent = null;
function gotLocalMedia(stream) {
console.info('Received local media stream');
localStream = stream;
videoView.src = URL.createObjectURL(stream);
function captureLocalMedia() {
console.info('Requesting local video & audio');
navigator.webkitGetUserMedia(constraints, gotLocalMedia, function(e){
alert('getUserMedia() error: ' + e.name);
function testStart(){
var sip_uri_ = document.getElementById("sip_uri").value.toString();
var sip_password_ = document.getElementById("sip_password").value.toString();
var ws_uri_ = document.getElementById("ws_uri").value.toString();
console.info("get input info: sip_uri = ", sip_uri_, " sip_password = ", sip_password_, " ws_uri = ", ws_uri_);
var socket = new JsSIP.WebSocketInterface(ws_uri_);
var configuration = {
sockets: [ socket ],
outbound_proxy_set: ws_uri_,
uri: sip_uri_,
password: sip_password_,
register: true,
session_timers: false
userAgent = new JsSIP.UA(configuration);
userAgent.on('registered', function(data){
console.info("registered: ", data.response.status_code, ",", data.response.reason_phrase);
userAgent.on('registrationFailed', function(data){
console.log("registrationFailed, ", data);
//console.warn("registrationFailed, ", data.response.status_code, ",", data.response.reason_phrase, " cause - ", data.cause);
userAgent.on('registrationExpiring', function(){
console.warn("registrationExpiring");
userAgent.on('newRTCSession', function(data){
console.info('onNewRTCSession: ', data);
if(data.originator == 'remote'){ //incoming call
console.info("incomingSession, answer the call");
incomingSession = data.session;
data.session.answer({'mediaConstraints' : { 'audio': true, 'video': true, mandatory: { maxWidth: 640, maxHeight: 360 } }, 'mediaStream': localStream});
}else{
console.info("outgoingSession");
outgoingSession = data.session;
outgoingSession.on('connecting', function(data){
console.info('onConnecting - ', data.request);
currentSession = outgoingSession;
outgoingSession = null;
data.session.on('accepted', function(data){
console.info('onAccepted - ', data);
if(data.originator == 'remote' && currentSession == null){
currentSession = incomingSession;
incomingSession = null;
console.info("setCurrentSession - ", currentSession);
data.session.on('confirmed', function(data){
console.info('onConfirmed - ', data);
if(data.originator == 'remote' && currentSession == null){
currentSession = incomingSession;
incomingSession = null;
console.info("setCurrentSession - ", currentSession);
data.session.on('sdp', function(data){
console.info('onSDP, type - ', data.type, ' sdp - ', data.sdp);
//data.sdp = data.sdp.replace('UDP/TLS/RTP/SAVPF', 'RTP/SAVPF');
//console.info('onSDP, changed sdp - ', data.sdp);
data.session.on('progress', function(data){
console.info('onProgress - ', data.originator);
if(data.originator == 'remote'){
console.info('onProgress, response - ', data.response);
data.session.on('peerconnection', function(data){
console.info('onPeerconnection - ', data.peerconnection);
data.peerconnection.onaddstream = function(ev){
console.info('onaddstream from remote - ', ev);
videoView.src = URL.createObjectURL(ev.stream);
userAgent.on('newMessage', function(data){
if(data.originator == 'local'){
console.info('onNewMessage , OutgoingRequest - ', data.request);
}else{
console.info('onNewMessage , IncomingRequest - ', data.request);
console.info("call register");
userAgent.start();
// Register callbacks to desired call events
var eventHandlers = {
'progress': function(e) {
console.log('call is in progress');
'failed': function(e) {
console.log('call failed: ', e);
'ended': function(e) {
console.log('call ended : ', e);
'confirmed': function(e) {
console.log('call confirmed');
function testCall(){
var sip_phone_number_ = document.getElementById("sip_phone_number").value.toString();
var options = {
'eventHandlers' : eventHandlers,
'mediaConstraints' : { 'audio': true, 'video': true ,
mandatory: { maxWidth: 640, maxHeight: 360 }
'mediaStream': localStream
//outgoingSession = userAgent.call('sip:3000@192.168.40.96:5060', options);
outgoingSession = userAgent.call(sip_phone_number_, options);
</script>
</html>
