[asterisk-users] G.729 and Voicemail

[Jeff LaCoursiere]

On Fri, 9 Oct 2009, Moises Silva wrote:

>>
>> I would be very surprised if that were true.  Your phones speak many
>> codecs, but they negotiate with asterisk on registration which one they
>> will be using.  They don't switch codecs based on the remote channel
>> (which they don't even know about).  Today, if your phones are negotiating
>> 729 on registration, you are definitely transcoding calls to/from the
>> PSTN.
>>
>
> Assuming we're talking about SIP (and any other voip protocol I know of for
> that matter), that is incorrect, codec negotiation is done during the call
> setup based on preferences stored in both Asterisk and the phone (that is
> what the SDP is for among other things). However at that point Asterisk does
> not know that the dial plan is going to call the voicemail application to
> play a file in g729 format (how can possibly now that), and therefore when
> the file is being played the phone already expects the audio for the call in
> the format negotiated during call setup which may or may not be g729. Not
> sure if a re-invite could be issued to change the codec type in the middle
> of the call, but I suppose it should be possible to implement.

Of course you are correct - not during registration but during call setup. 
The main idea, though, is that the negotiation doesn't change based on the 
far leg of the call inbound or outbound.  If the phone is set to 
negiotiate G729 it will always negotiate G729 regardless of the far end.

j

>
> -- 
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>