[Asterisk-Users] G.729 Codec on Dialup

[Hemant Kumar]

It is using Annex A but the name is G729B, blame Voice Age for that.


----- Original Message -----
From: "Surajee Ratnayake" <surajee at infotechs.lk>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 17, 2003 10:05 AM
Subject: Re: [Asterisk-Users] G.729 Codec on Dialup


>
> ----- Original Message -----
> From: "Mark Spencer" <markster at digium.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, May 14, 2003 7:02 PM
> Subject: Re: [Asterisk-Users] G.729 Codec on Dialup
>
>
> > > On the local LAN, when we use the SJPhone as the SIP client,
> communication works fine with no disturbances and noices. But when it
comes
> to dialup connection we harldy hear anything except a rough noice.
> > > We have included G.729 Codec (Annex B) with the Asterisk server, and
we
> added the G.729 Codec to the SJPhone too. But it seems like SJPhone is
using
> a version of G.729 other than Codec G.729 (Annex B) , probably 'Annex A'.
> > > What we want is to have a smoother communication when we use a SIP
phone
> on the Dialup connection.
> >
> > If you purchased G729 from Digium, you got the "Annex A" even though the
> > file is, confusingly, called codec_g729b.so
> >
> > Mark
> >
>
> YES, we got the G729 from Digium.
> but, when the asterisk is starting, its giving the following information,
>
> "
>  [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec
Translator)
>   == Detected 2 licensed G.729 transcoders
> WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator
> 'g729tolinb' does not produce sample frames.
>   == Registered translator 'g729tolinb' from format 8 to 6, cost 99999
>   == Registered translator 'lintog729b' from format 6 to 8, cost 46
> "
>
> so, isn't it using "Annex B"? if it is using "Annex B", how to get the
> "Annex A"?
>
> Thanx,
> Surajee
>
>
>
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>