[Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[andy]

Hi all,

I'm interested in using asterisk WITHOUT codec support: I work in a LAN, with no 
bandwidth, delay, ... problems; I use a Cisco GW as PSTN interface and when I 
use asterisk the overall delay is to high and the quality drops.

In particular, I'm interested in using asterisk as h323 to sip translator (and 
viceversa).
do you have any suggestion?
thanks

Andres


Quoting Mike Ciholas <mikec at ciholas.com>:

> 
> On Tue, 9 Sep 2003, Eric Wieling wrote:
> 
> > Transcoding would be required for access to ANY of the asterisk
> > sound files, voicemail and PSTN via Zap interfaces.
> 
> If you are using G711 ulaw from the SIP phones, and that is what
> you are getting from the T1 PSTN link, would * have to transcode
> that?  Is there more to it than digital to digital copy?  Perhaps 
> echo canceling?
> 
> Can we also store sound files in ulaw?  I know that takes more 
> space, but perhaps it is less CPU work to move the bits around 
> than to codec them.
> 
> -- 
> Mike Ciholas                            (812) 476-2721 voice
> CIHOLAS Enterprises                     (812) 476-2881 fax
> 2626 Kotter Ave, Unit D                 mikec at ciholas.com
> Evansville, IN 47715                    http://www.ciholas.com
> 
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