<div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">Hi All,</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">I was trying to play with the 3.1 feature specifically media handling capabilities. <br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">I want opensips act as a playing server by answering WebRTC based calls.</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">Here is a scenario I was trying to do.<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">-- Opensips will receive WSS call <br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">-- Process the call</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">-- Play file with 200 OK <br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">-- Sending to Voicemail (recording of file using rtpengine recording module)</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">-- Hangup call by caller or hangup after some time.</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">Here is sample routing I plan to develop. I tried it is not working as 200 OK is not generated with SDP.<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">route[VOICEMAIL]{<br>        xlog("Receiving voicemail");<br>        $var(rtpengine_flags) = "trust-address replace-origin replace-session-connection rtcp-mux-offer ICE=force transcode-PCMU transcode-G722 SDES-off UDP/TLS/RTP/SAVP";<br>        rtpengine_offer("$var(rtpengine_flags)");<br>        rtpengine_start_recording();<br>        append_to_reply("Contact: <sip:voicemail@XXX_XXX_XXX_XXX>\r\n");            <br>        rtpengine_answer("$var(rtpengine_flags)");<br>        t_reply(200, "Ok");<br>        rtpengine_play_media("file=/etc/opensips/sounds/vm-isunavail.wav");<br>        xlog("waiting for voicemail to be recorded");<br>        sleep(30);       <br>        exit;<br>}</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">I want to know how this will be possible? Don't consider Asterisk and Free-switch as a media server.</div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;color:#000000">Any help suggestion would be appreciated.<br></div><br>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div><span style="color:rgb(61,133,198)"><span style="font-family:verdana,sans-serif"><br>Best Regards,<br></span></span></div><b><span style="color:rgb(61,133,198)"><span style="font-family:verdana,sans-serif">Dhaval Indrodiya<br></span></span></b></div><b><span style="color:rgb(61,133,198)"><span style="font-family:verdana,sans-serif">skype: dki123sabse</span></span></b><br></div></div></div>