[OpenSIPS-Users] Opensips 3.1 Play Media

[DHAVAL INDRODIYA]

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:voicemail at XXX_XXX_XXX_XXX>\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*
*skype: dki123sabse*
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