[Asterisk-Users] Drops due to codecs?

[Steven Critchfield]

On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote:
> Hello,
> 
>    It is my understanding that on the softphone side, asterisk is only 
> responsible for establishing the session between two phones.  If this is the 
> case, does it matter what type of audio codecs the two phones are using?  And 
> if it does matter, are there any codecs that cause problems with asterisk 
> bridging two SIP connections?  Thanks for your helpful input,

This depends. If a SIP phone must use the IVR feature of asterisk to get
routed to another SIP phone, then codecs matter. If asterisk is
listening on the line to hear when you issue commands via DTMF for it to
do something like transfer, then yes it matters. Also some SIP phones
don't handle reinvite properly, and then asterisk is stuck redirecting
the audio from place to place for you, here it doesn't matter unless on
of the above comments apply.
-- 
Steven Critchfield  <critch at basesys.com>