<div dir="ltr">
<pre>Hi Dhaval
Usually for using rtpengine_play_media we need A and B party,B party generates Reply and SDP.<br></pre><pre>If you want to omit B party you should generate Reply and SDP inside script. please see the workaround mentioned here[1]<br></pre><pre>and also this feature request[2].<br>[1] <a href="https://github.com/sipwise/rtpengine/issues/870">https://github.com/sipwise/rtpengine/issues/870</a></pre><pre>[2] <a href="https://github.com/OpenSIPS/opensips/issues/1996">https://github.com/OpenSIPS/opensips/issues/1996</a></pre><pre>I do not know exact final script but working on it.<br></pre><pre>Regards<br>Shirazi <br><br>>Hi All,
>I was trying to play with the 3.1 feature specifically media handling
>capabilities.
>I want opensips act as a playing server by answering WebRTC based calls.
>Here is a scenario I was trying to do.
>-- Opensips will receive WSS call
>-- Process the call
>-- Play file with 200 OK
>-- Sending to Voicemail (recording of file using rtpengine recording module)
>-- Hangup call by caller or hangup after some time.
>Here is sample routing I plan to develop. I tried it is not working as 200
>OK is not generated with SDP.
>route[VOICEMAIL]{
> xlog("Receiving voicemail");
> $var(rtpengine_flags) = "trust-address replace-origin
>replace-session-connection rtcp-mux-offer ICE=force transcode-PCMU
>transcode-G722 SDES-off UDP/TLS/RTP/SAVP";
> rtpengine_offer("$var(rtpengine_flags)");
> rtpengine_start_recording();
> append_to_reply("Contact: <sip:<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">voicemail at XXX_XXX_XXX_XXX</a>>\r\n");
> rtpengine_answer("$var(rtpengine_flags)");
> t_reply(200, "Ok");
> rtpengine_play_media("file=/etc/opensips/sounds/vm-isunavail.wav");
> xlog("waiting for voicemail to be recorded");
> sleep(30);
> exit;
>}
>I want to know how this will be possible? Don't consider Asterisk and
>Free-switch as a media server.
>Any help suggestion would be appreciated.
>--
>Best Regards,
>*Dhaval Indrodiya*</pre>
</div>