[asterisk-users] Avoid compression with g.729/gsm/etc.

[Anthony Plack]

> 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.