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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