Question

我使用freeswitch 1.6并跟随cookbok实现webrtc。为此，我也下载了sip.js@0.7.0。并创建了call.html，call.js和answer.html，answer.js页面。我的call.html包括js是

    <html>
        <body>
             <button id="startCall">Start Call</button>

             <button id="endCall">End Call</button>
             <br/>
             <video id="remoteVideo"></video>
             <br/>
             <video id="localVideo" muted="muted" width="128px" height="96px"></video>
             <!--<script src="js/sip-0.7.0.min.js"></script>-->
             <!--<script src="call.js"></script>-->
    </body>

    <HEAD>
    <script src="js/sip-0.7.0.min.js"></script>
    <script>

                var session;
                console.log('hiiiiiiiiiiii')
                var endButton = document.getElementById('endCall');
                endButton.addEventListener("click", function () {
                             session.bye();
                             alert("Call Ended");
                             }, false);
                console.log('hiiiii2')

                var userAgent = new SIP.UA({
                                uri: 'sip:anonymous@gmaruzz.org',
                                wsServers: ["ws://call.sia.co.in:5066"],
                                authorizationUser: 'anonymous',
                                password: 'welcome'
                });

                console.log('hiiii3')
                var startButton = document.getElementById('startCall');
                startButton.addEventListener("click", function () {
                    session =userAgent.invite('sip:1010@139.59.17.63', options);
                    alert("Call Started");
                }, false);

                console.log('hiiii4')
                var options = {
                        media: {
                                    constraints: {
                                                        audio: true,
                                                        video: true
                                                },
                                    render: {
                        remote:document.getElementById('remoteVideo'),
                        local: document.getElementById('localVideo')
                                                }
                                }
                };
</script>
</HEAD>
</html>

请在我出错的地方纠正我。提前谢谢。

Answer 1

您必须将wsServers放置在transportOptions中，例如：

var userAgent = new SIP.UA({
uri: 'sip:anonymous@gmaruzz.org'

transportOptions: {
  wsServers: "ws://call.sia.co.in:5066"
},

authorizationUser: 'anonymous',
password: 'welcome'

参数wsServers的值错误

1 个答案: