freeswitch 使用sipjs 、jssip webrtc网页音视频通话

原创已于 2022-02-19 14:59:26 修改 · 1.6w 阅读

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文章标签：

#freeswitch #freeswitch webrtc #音视频通话 #linphone #sipjs jssip

于 2019-03-12 11:57:09 首次发布

freeswitch 专栏收录该内容

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关于linphone手机端开发参考基于linphone android sdk 的voip语音、视频通话教程一、注册_Java_lilin的专栏-优快云博客

配置帮助群:261074724

呼叫系统参考https://item.taobao.com/item.htm?id=653611115230

sipjs 官网SIP Signaling JavaScript Library for WebRTC Developers | SIP.js

参考官网说明写法如下:

<body>
<div id="id_msg"></div>
<input id="id_to" value="10000"/>
<button onclick="call()">call</button>
<button onclick="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)
{
   //接受来电
   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
{
   //拒绝来电
   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>

................

测试结果如下

	局域网	外网(nat ok)	音视频	linphone	正常接听
pc浏览器	支持	支持	支持	支持	ok
手机浏览器	支持	支持	支持	支持	ok
微信网页	支持	支持	支持	支持	ok

手机端须改动一点地方可以加群讨论:261074724

测试挺好的这个

/////////////////////////////////

jssip 官网JsSIP - the Javascript SIP library

页面参考

WebRTC + JsSIP + freeSWITCH一对一视频聊天_安晓辉生涯——聚焦程序员的职业规划与成长-优快云博客_freeswitch webrtc

经个人测试 jssip 在内网效果很好

在外网配置turn后呼叫有点慢接通好像有点问题

和linphone集成的交流群 :261074724

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