[Asterisk-Users] Drops due to codecs?

[Daniel Flickinger]

Thank you for your help Steven.

> Message: 8
> Subject: Re: [Asterisk-Users] Drops due to codecs?
> From: Steven Critchfield <critch at basesys.com>
> To: asterisk-users at lists.digium.com
> Date: 03 Jul 2003 15:26:45 -0500

> Reply-To: asterisk-users at lists.digium.com

> 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>