關於linphone手機端開發參考https://blog.csdn.net/Java_lilin/article/details/80539116
配置幫助羣:261074724
sipjs 官網http://sipjs.com/
參考官網說明 寫法如下:
<body>
<div id="id_msg"></div>
<input id="id_to" value="10000"/>
<button οnclick="call()">call</button>
<button οnclick="gua()">gua</button>
<br/>
<video id="remoteVideo" style="width:200px;height:400px;"></video>
<video id="localVideo" muted="muted"></video>
<script src="sip-0.13.6.min.js"></script>
<script src="vconsole.min.js"></script>
<script type="text/javascript">
var vConsole = new VConsole();
var config = {
// Replace this IP address with your FreeSWITCH IP address
uri: '10001@xx:9060',
// Replace this IP address with your FreeSWITCH IP address
// and replace the port with your FreeSWITCH ws port
transportOptions: {
wsServers: ['wss://xx:7443']
},
// FreeSWITCH Default Username
authorizationUser: '10001',
// FreeSWITCH Default Password
password: 'test1'
};
var userAgent = new SIP.UA(config);
var remoteVideo = document.getElementById('remoteVideo');
var localVideo = document.getElementById('localVideo');
var sipsession = null;
userAgent.on('registered', function () {
document.getElementById('id_msg').innerText="ok";
});
userAgent.on('invite', function(session) {
var url = session.remoteIdentity.uri.toString()+"--->call";
var isaccept = confirm(url);
if(isaccept)
{
//接受來電
session.accept({
sessionDescriptionHandlerOptions: {
constraints: {
audio: true,
video: true
}
}
});
sipsession = session;
session.on('accepted', function() {//
// We need to check the peer connection to determine which track was added
var pc = session.sessionDescriptionHandler.peerConnection;
console.log(pc);
console.log(pc.getLocalStreams());
// Gets remote tracks
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);
});
remoteVideo.srcObject = remoteStream;
remoteVideo.play();
if(pc.getSenders() ){
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localVideo.srcObject = localStream;
localVideo.play();
}
});
}
else
{
//拒絕來電
session.reject();
}
} );
function gua(){
sipsession.terminate();
}
function call( ){
var to =document.getElementById('id_to').value;
sipsession = userAgent.invite(to+'@xx:7443',{
sessionDescriptionHandlerOptions: {
constraints: {
audio: true, video: true
}
}
});
sipsession.on('accepted', function() {
// We need to check the peer connection to determine which track was added
var pc = sipsession.sessionDescriptionHandler.peerConnection;
// Gets remote tracks
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
remoteStream.addTrack(receiver.track);261074724
});
remoteVideo.srcObject = remoteStream;
remoteVideo.play();
// Gets local tracks
if(pc.getSenders() ){
var localStream = new MediaStream();
pc.getSenders().forEach(function(sender) {
localStream.addTrack(sender.track);
});
localVideo.srcObject = localStream;
localVideo.play();
}
});
}
</script>
</body>
................
測試結果如下
| 局域網 | 外網(nat ok) | 音視頻 | linphone | 正常接聽 | |
| pc瀏覽器 | 支持 | 支持 | 支持 | 支持 | ok |
| 手機瀏覽器 | 支持 | 支持 | 支持 | 支持 | ok |
| 微信網頁 | 支持 | 支持 | 支持 | 支持 | ok |
手機端須改動一點地方 可以加羣討論:261074724
測試挺好的這個
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jssip 官網https://jssip.net/
頁面參考
https://blog.csdn.net/foruok/article/details/74321214
經個人測試 jssip 在內網效果很好
在外網配置turn後呼叫有點慢 接通好像有點問題
和linphone集成的交流羣 :261074724