I would say this more a question for sipjs mailing list. Having said that, do you see rtps arriving on freeswitch? On Mon, Jun 18, 2018, 14:27 Hossein Gholizadeh <hggh96.hg@gmail.com> wrote:
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.
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