[OpenSIPS-Users] Opensips 3.1 Play Media

[Mehdi Shirazi]

Hi Dhaval
Usually for using rtpengine_play_media we need A and B party,B party
generates Reply and SDP.

If you want to omit B party you should generate Reply and SDP inside
script. please see the workaround mentioned here[1]

and also this feature request[2].
[1] https://github.com/sipwise/rtpengine/issues/870

[2] https://github.com/OpenSIPS/opensips/issues/1996

I do not know exact final script but working on it.

Regards
Shirazi

>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 <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>>\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*
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