> Regarding compression with g.729/gsm/etc. and Asterisk > > If we convert all the voice files to the corresponding format g.729/gsm/etc. > and we send digits using RFC 3261 and we do not need silence detection, is > there still a need to decompress the media stream ? > > If doable how to make sure this will work without compression/decompression ? > >
I believe that Asterisk by default unpackages/repackages the stream. If you are looking for RTP pass-through, you are needing a RTP Proxy or SIP Reinvite and not Asterisk. Look at kamailio.org and RTP Proxy with Asterisk as the VoiceMail/Media Server. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
