[asterisk-users] G729 Confusion

[Andres]

Matt wrote:

>Well does g729 have to run on both legs of a call? 
>
If the call is established and there is audio both ways then yes.  If 
the call has not been answered yet then you will only see 1 encoder 
used.  In your case somebody is using up 5 encoders and it is probably 
from calls coming into the box from the PSTN side since Asterisk is 
having to use 5 encoders to 'encode G729 from ulaw or another codec'.  
If this is a test platform then you might have a loop.  If this is a 
production system with a lot of users then try to track down the 
offenders with a 'sip show channels' and look at the 'Form' column to 
see who is using G729.

> For instance, when
>I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't
>hear any audio on my SIP phone, however if I call someone they can
>hear me.
>  
>
That is expected since when you tried to make the call there were no 
encoders left.  Your phone encodes the call and Asterisk is able to 
decode it and deliver the audio to the other party but not the other way 
around.

>On 8/28/07, Andres <andres at telesip.net> wrote:
>  
>
>>>and reload, strange things begin to happen.  A show g729 shows this:
>>>5/0 encoders/decoders of 5 licensed channels are currently in use
>>>
>>>
>>>      
>>>
>>I think you have a loop of some kind.  As you can see none of those call
>>are actually established since no decoders are in use.  Try to debug and
>>see why those 5 calls are acually not connected in the first place.
>>
>>    
>>
>>>and suddenly I can not hear anything if I try to make a call.    From
>>>observation, it almost seems like other units on the network are using
>>>the g729 codecs, but doesn't my sip.conf prohibit g729 unless
>>>expressly allowed?!  Why would allowing g729 under one extension allow
>>>everyone else to suddenly start using g729?
>>>
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>>>      
>>>
>>Andres
>>http://www.telesip.net
>>
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>>    
>>
>
>
>  
>