asterisk users - Mar 2009 - Avoid compression with g.729/gsm/etc.

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 ?
 
 
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> 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.