[Freeswitch-users] sip.js has no audio

[David Villasmil]

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 at 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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