问使用Jssip的Webrtc客户端-没有音频，两种方式都使用免费开关和铬

EN

var localStream, remoteStream = null;

var remoteVideo = document.getElementById('remoteVideo');
var ua, session = null;

var eventHandlers;
var configuration = {
    'ws_servers': '******',
    'uri': '******',
    'password': '*****'
};

// Register callbacks to desired call events 

eventHandlers = {

    'peerconnection': function (e) {

        console.trace("fired for outgoing calls but before sdp generation in peerconnection ");

    },
    'connecting': function (e) { 

    },
    'progress': function (e) {

        console.trace('call is in progress', e);

    },
    'failed': function (e) {
        console.trace('call failed with cause: ', e);
    },
    'ended': function (e) {

        console.trace('call ended with cause: ', e);
    },
    'confirmed': function (e) {
    },
    'accepted': function (e) {
        console.trace(" call accepted ");
    },
    'addstream': function (e) {

 if(session.connection.getRemoteStreams().length > 0)
 {

    console.trace('remote stream added ' +e.stream.getAudioTracks().length);

    console.trace('remote stream added ' + e.stream.getTracks());

   remoteVideo = JsSIP.rtcninja.attachMediaStream(remoteVideo,e.stream);
        }
      }
};

var options = {

    'eventHandlers': eventHandlers,
    'extraHeaders': ['X-Foo: foo', 'X-Bar: bar'],
    'mediaConstraints': {'audio': true, 'video':false},
    'rtcOfferConstraints' : {'offerToReceiveAudio' : true } ,

    mandatory: [{
                OfferToReceiveAudio: true,
                OfferToReceiveVideo: false
            },{'DtlsSrtpKeyAgreement': true} ]

};
init();

function init() {

    console.trace("intializing user agent");
    ua = new JsSIP.UA(configuration);
    ua.start();
    console.trace("is registered : " + ua.isRegistered());
    uaEventHandling();
}
;


function uaEventHandling() {

    //events of UA class with their callbacks
    ua.on('registered', function (e) {
        console.trace("registered", e);
    });

    ua.on('unregistered', function (e) {
        console.trace("ua has been unregistered periodic registeration fails or ua.unregister()", e);
    });

    ua.on('registrationFailed', function (e) {
        console.trace("register failed", e);
    });
    ua.on('connected', function (e) {
        console.trace("connected to websocket");
    });
    ua.on('disconnected', function (e) {
        console.trace("disconnected");
        ua.stop();
    });

    ua.on('newRTCSession', function (e) {
        console.trace('new rtc session created - incoming or outgoing call');
        session = e.session;
        if (e.originator === 'local') {
            console.trace(e.request + ' outgoing session');

        }
        else {
            console.trace(e.request + ' incoming session answering a call');
            e.session.answer(options);
        }
    });

    ua.on('newMessage', function (e) {
        if (e.originator === 'local')
            console.trace(' outgoing MESSAGE request ', e);
        else
            console.trace(' incoming MESSAGE request ', e);
    });
};

ua.call('sip:********', options);