[asterisk-users] G723 pass-through and codec negotiation

[Ray Jackson]

Hi Andrew,

All that happens is Asterisk sets up the call using G723 as the codec 
and calls then fail due to transcoding problems (since it cannot 
encode/decode g723).  The softphone always prefers g723 over gsm, so if 
you allow both in sip.conf Asterisk always select g723 due to the 
preference of the client.  What I need is for Asterisk to understand 
what the 'other end' is capable of before selecting a codec. i.e. if the 
call terminates to the Voicemail then select gsm and do NOT allow g723.  
Codec negotiation is usually something that occurs in the call setup 
between 2 endpoints, but Asterisk seems to do this in isolation for each 
client without understanding where the call is going first to make that 
decision.  What I really need is something that forces the 'allow' 
codecs in the dial plan per call.  Does anybody know if this is this 
possible?

Cheers,
Ray

Andrew Joakimsen wrote:

> What happens if in your sip.conf you set
>
> disallow=all
> allow=g723,gsm
>
> And then allow both codec in the phone?
>
> On 11/19/06, *Ray Jackson* < ray at jacksonz.net 
> <mailto:ray at jacksonz.net>> wrote:
>
>     All,
>
>     Our users have a softphone client that supports the G723 Codec
>     which we
>     want to use for bandwidth reasons, however we do not wish (or have the
>     funds) to license the codec on our Asterisk servers.  We have G723
>     pass-through working between the clients just fine, however calls fail
>     when terminating with Asterisk itself (i.e. Voicemail) or out to the
>     PSTN due to transcoding issues.
>
>     If it possible to build the config into our Asterisk servers so that
>     calls between the softphones defaults to G723 pass-through, whilst
>     all
>     other calls (PSTN, Voicemail etc.) default to GSM as their preferred
>     codec?  Is there a way of getting Asterisk to be smart with Codec
>     negotitation and figure out which codec the other end of the call is
>     capable of before negotitating back to the Softphone with the
>     selected
>     codec?  I assume you would have to do something in the dial
>     plan?  I saw
>     the SIP_CODEC variable, but couldn't make it work.
>
>     Any advice would be very welcome!
>
>     Cheers,
>     Ray
>
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