<div dir="ltr">Hi Mahmood,<br><br><br>Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call.<br>Define ICE servers to be used by JsSip.<br><br>               var options = {<br>    'eventHandlers'          : your eventHandlers,<br>    'mediaConstraints' : { 'audio': true, 'video': true },<br>    'pcConfig'           : { 'iceServers': [<br>        {'urls': 'stun:stun.stunprotocol.org:3478'},<br>        {'urls': 'stun:stun.l.google.com:19302'}, ] },<br>             };<br><br>          var session = ua.call(uri, options);<br><br><br>kind regards,<br>Piotr<br clear="all"><div><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif"><img src="https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX8-AIlZuCq&export=download" width="96" height="49"><br></font></div><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif"><br></font></div><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif">Piotr Gregor</font></div><div><span style="background-color:rgb(255,255,255)"><font size="2" face="arial, helvetica, sans-serif">Software Engineer</font></span></div><div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small"><br></span></div><div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">M: (+44) 07483 866 525 </span><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">    </span><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">www: </span><a href="http://dataandsignal.com" style="font-family:arial,helvetica,sans-serif;font-size:x-small" target="_blank">dataandsignal.com</a></div><div><br></div><div><br><br><br></div></div></div></div></div></div></div></div></div></div></div><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil <<a href="mailto:mahmood.alkhalil@outlook.com">mahmood.alkhalil@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">




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Hi Everyone,</div>
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I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with <span style="background-color:rgb(255,255,0)">SIP/2.0 488 Not Acceptable Here.</span></div>
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<span style="background-color:rgb(255,255,255)">​The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found.</span><br>
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<span style="background-color:rgb(255,255,255)">I'v attached the call trace, appreciate any help.</span></div>
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<span style="background-color:rgb(255,255,255)">Thanks,</span></div>
<div id="gmail-m_838301763522276059Signature">
<div id="gmail-m_838301763522276059divtagdefaultwrapper" style="font-size:12pt;font-family:Calibri,Arial,Helvetica,sans-serif;color:rgb(0,0,0);background-color:rgb(255,255,255)">
Mahmood Alkhalil.</div>
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