hi all ... i'm using sip.js for my IVR project ... it could connect to freeswitch correctly but there is no audio between call legs ... first of all i have to say i really cannot use Verto . this is my function of when call button pressed * phoneCallButtonPressed : function(sessionid) {* * var s = ctxSip.Sessions[sessionid],* * target = $("#numDisplay").val();* * if (!s) {* * $("#numDisplay").val("");* * ctxSip.sipCall(target);* * } else if (s.accept && !s.startTime) {* * s.accept({* * media : {* * stream : ctxSip.Stream,* * constraints : { audio : true, video : false },* * render : {* * remote : { audio: $('#audioRemote').get()[0] }* * },* * RTCConstraints : { "optional": [{ 'DtlsSrtpKeyAgreement': 'true'} ]}* * }* * });* * }* * },* ----------------------------------------------------------------------------------------- also i have to say that i tested it between my phone and x-lite ... X-lite could receive the voice but my phone could'nt ... i was using Zoiper in my phone.