<div dir="ltr">I would say this more a question for sipjs mailing list.</div><span></span>Having said that, do you see rtps arriving on freeswitch?<div><br><div class="gmail_quote"><div dir="ltr">On Mon, Jun 18, 2018, 14:27 Hossein Gholizadeh <<a href="mailto:hggh96.hg@gmail.com" target="_blank">hggh96.hg@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">hi all ...
<span style="color:rgb(51,51,51);font-family:"Helvetica Neue",Helvetica,Arial,sans-serif;font-size:14px;background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">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 </span><div><span style="color:rgb(51,51,51);font-family:"Helvetica Neue",Helvetica,Arial,sans-serif;font-size:14px;background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></div><div><span style="background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline;font-size:14px"><font face="Helvetica Neue, Helvetica, Arial, sans-serif"><div><font color="#333333"> </font><b><font color="#333333"> phoneCallButtonPressed : </font><font color="#0000ff">function</font><font color="#333333">(</font><font color="#666666">sessionid</font><font color="#333333">) {</font></b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> var s = ctxSip.Sessions[sessionid],</b></div><div><b><font color="#333333"> target = $(</font><font color="#e69138">"#numDisplay"</font><font color="#333333">).val();</font></b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> if (!s) {</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div><b><font color="#333333"> $(</font><font color="#ff9900">"#numDisplay"</font><font color="#333333">).val(</font><font color="#ff9900">""</font><font color="#333333">);</font></b></div><div style="color:rgb(51,51,51)"><b> ctxSip.sipCall(target);</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> } else if (s.accept && !s.startTime) {</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> s.accept({</b></div><div style="color:rgb(51,51,51)"><b> media : {</b></div><div style="color:rgb(51,51,51)"><b> stream : ctxSip.Stream,</b></div><div><b><font color="#333333"> constraints : { audio : </font><font color="#0000ff">true</font><font color="#333333">, video : </font><font color="#0000ff">false </font><font color="#333333">},</font></b></div><div style="color:rgb(51,51,51)"><b> render : {</b></div><div><b><font color="#333333"> remote : { audio: $(</font><font color="#ff9900">'#audioRemote'</font><font color="#333333">).get()[0] }</font></b></div><div style="color:rgb(51,51,51)"><b> },</b></div><div><b><font color="#333333"> RTCConstraints : { </font><font color="#ff9900">"optional"</font><font color="#333333">: [{ </font><font color="#ff9900">'DtlsSrtpKeyAgreement'</font><font color="#333333">: </font><font color="#ff9900">'true'</font><font color="#333333">} ]}</font></b></div><div style="color:rgb(51,51,51)"><b> }</b></div><div style="color:rgb(51,51,51)"><b> });</b></div><div style="color:rgb(51,51,51)"><b> }</b></div><div style="color:rgb(51,51,51)"><b> },</b></div><div style="color:rgb(51,51,51)"><br></div><div style="color:rgb(51,51,51)">-----------------------------------------------------------------------------------------</div><div style="color:rgb(51,51,51)">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.</div><div style="color:rgb(51,51,51)"><br></div></font></span></div></div>
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