SIP.JS 0.11 FREESWITCH 1.6 AUDIO ISSUES











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I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.



APP Code:





  • USERAGENT



    userAgent = new SIP.UA({

    uri: user+'@'+url,

    transportOptions: {
    wsServers: ['wss://' + url],
    },

    authorizationUser: user,
    password: password,
    register : true,
    });

    userAgent.start()



  • MAKE A CALL



    var sessionDescriptionHandlerOptions = {

    constraints: {
    audio: true,
    video: false
    }

    }

    var session = userAgent.invite('sip:'+number+'@freeswitch_server',
    sessionDescriptionHandlerOptions);



  • ATTACH_MEDIA



    session.on('trackAdded', function() {


    var pc = session.sessionDescriptionHandler.peerConnection;


    var remoteStream = new MediaStream();
    pc.getReceivers().forEach(function(receiver) {
    remoteStream.addTrack(receiver.track);
    });

    remoteAudio.srcObject = remoteStream;
    remoteAudio.play();


    var localStream = new MediaStream();
    pc.getSenders().forEach(function(sender) {
    localStream.addTrack(sender.track);
    });

    localAudio.srcObject = localStream;
    localAudio.play();
    });



HTML



<video id="remoteAudio"></video>
<video id="localAudio" muted="muted"></video>


Thanks a lot!










share|improve this question


























    up vote
    0
    down vote

    favorite












    I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.



    APP Code:





    • USERAGENT



      userAgent = new SIP.UA({

      uri: user+'@'+url,

      transportOptions: {
      wsServers: ['wss://' + url],
      },

      authorizationUser: user,
      password: password,
      register : true,
      });

      userAgent.start()



    • MAKE A CALL



      var sessionDescriptionHandlerOptions = {

      constraints: {
      audio: true,
      video: false
      }

      }

      var session = userAgent.invite('sip:'+number+'@freeswitch_server',
      sessionDescriptionHandlerOptions);



    • ATTACH_MEDIA



      session.on('trackAdded', function() {


      var pc = session.sessionDescriptionHandler.peerConnection;


      var remoteStream = new MediaStream();
      pc.getReceivers().forEach(function(receiver) {
      remoteStream.addTrack(receiver.track);
      });

      remoteAudio.srcObject = remoteStream;
      remoteAudio.play();


      var localStream = new MediaStream();
      pc.getSenders().forEach(function(sender) {
      localStream.addTrack(sender.track);
      });

      localAudio.srcObject = localStream;
      localAudio.play();
      });



    HTML



    <video id="remoteAudio"></video>
    <video id="localAudio" muted="muted"></video>


    Thanks a lot!










    share|improve this question
























      up vote
      0
      down vote

      favorite









      up vote
      0
      down vote

      favorite











      I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.



      APP Code:





      • USERAGENT



        userAgent = new SIP.UA({

        uri: user+'@'+url,

        transportOptions: {
        wsServers: ['wss://' + url],
        },

        authorizationUser: user,
        password: password,
        register : true,
        });

        userAgent.start()



      • MAKE A CALL



        var sessionDescriptionHandlerOptions = {

        constraints: {
        audio: true,
        video: false
        }

        }

        var session = userAgent.invite('sip:'+number+'@freeswitch_server',
        sessionDescriptionHandlerOptions);



      • ATTACH_MEDIA



        session.on('trackAdded', function() {


        var pc = session.sessionDescriptionHandler.peerConnection;


        var remoteStream = new MediaStream();
        pc.getReceivers().forEach(function(receiver) {
        remoteStream.addTrack(receiver.track);
        });

        remoteAudio.srcObject = remoteStream;
        remoteAudio.play();


        var localStream = new MediaStream();
        pc.getSenders().forEach(function(sender) {
        localStream.addTrack(sender.track);
        });

        localAudio.srcObject = localStream;
        localAudio.play();
        });



      HTML



      <video id="remoteAudio"></video>
      <video id="localAudio" muted="muted"></video>


      Thanks a lot!










      share|improve this question













      I am trying to use sip.js 0.11 with FreeSwitch 1.6 but when the call is established (ACCEPTED) it has no audio. Audio works when a hold-unhold is made. Tested on: Chrome 70 and electron 3.



      APP Code:





      • USERAGENT



        userAgent = new SIP.UA({

        uri: user+'@'+url,

        transportOptions: {
        wsServers: ['wss://' + url],
        },

        authorizationUser: user,
        password: password,
        register : true,
        });

        userAgent.start()



      • MAKE A CALL



        var sessionDescriptionHandlerOptions = {

        constraints: {
        audio: true,
        video: false
        }

        }

        var session = userAgent.invite('sip:'+number+'@freeswitch_server',
        sessionDescriptionHandlerOptions);



      • ATTACH_MEDIA



        session.on('trackAdded', function() {


        var pc = session.sessionDescriptionHandler.peerConnection;


        var remoteStream = new MediaStream();
        pc.getReceivers().forEach(function(receiver) {
        remoteStream.addTrack(receiver.track);
        });

        remoteAudio.srcObject = remoteStream;
        remoteAudio.play();


        var localStream = new MediaStream();
        pc.getSenders().forEach(function(sender) {
        localStream.addTrack(sender.track);
        });

        localAudio.srcObject = localStream;
        localAudio.play();
        });



      HTML



      <video id="remoteAudio"></video>
      <video id="localAudio" muted="muted"></video>


      Thanks a lot!







      webrtc sip freeswitch






      share|improve this question













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      asked Nov 7 at 18:25









      ElLoko36

      114




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