[asterisk-users] any codec passthru mode

[Jaswinder Singh]

Yes it might be dumb but since asterisk is a pbx and not a sip proxy
it has to perform many other functions as well .  But i do think that
asterisk should act little smart in this case

> SIP wrote:
> > That just seems really, REALLY dumb for a program of this magnitude.
> >
> > I know this has been patched here and there by one person or another,
> > but does anyone know if any of these patches to make CODEC negotiation
> > actually, you know, negotiate a CODEC will ever make it into the core
> > src?
> >
> >
> > Jaswinder Singh wrote:
> >> Asterisk by default uses the codec preferred by other device/client  .
> >> Asterisk 1.2 ( dunno abt 1.4 specifically)  is not intelligent enough
> >> to check if it can avoid transcoding by forcing same codec on other
> >> side of conversation . If both sides prefer g729 then asterisk does
> >> not do transcoding but if one side prefer gsm and other prefers g729
> >> and the gsm side can also support g729 still asterisk will transcode .
> >> Someone posted a patch to this in mantis bug tracking system at digium
> >> for 1.2 .. google for it and maybe you can find  .
> >>
> >> On 31/05/07, Rizwan Hisham <rizwanhasham at gmail.com> wrote:
> >>> Does anybody has any documentation on codec negotiation within
> >>> asterisk?
> >>>
> >>> Well im using free g729 codec for testing purposes. i mentioned g729
> >>> just as
> >>> an example. whatever codec is mentioned in user perefernce, asterisk
> >>> uses
> >>> ulaw to throw out the call.
> >>>
> >>>
> >>> On 5/30/07, Marco Mouta <marco.mouta at gmail.com> wrote:
> >>> > so you r sure you have g729 licences installed and ur * is
> >>> transcoding
> >>> your RTP streaming?
> >>> >
> >>> > Test the work flow with disallow=all and allow=g729, can be my
> >>> mistake but
> >>> I remember to read somewhere on the net any issue about codec
> >>> negotiating
> >>> precedence when you use allow=all.
> >>> >
> >>> > good luck
> >>> >
> >>> >
> >>> >
> >>> > On 5/30/07, Rizwan Hisham < rizwanhasham at gmail.com> wrote:
> >>> > >
> >>> > > Hi all,
> >>> > > My configuration is:
> >>> > > USER ----(connects to)----> ASTERISK---(connects to)--->CARRIER-OUT
> >>> > >
> >>> > > i want the user preffered codec to pass thru asterisk to
> >>> carrier-out.
> >>> what i mean is:
> >>> > > USER ----(user uses g729)----> ASTERISK---(asterisk should use
> >>> g729 for
> >>> dialing out)--->CARRIER-OUT
> >>> > >
> >>> > > instead, this is what happens
> >>> > > USER ----(user uses g729)----> ASTERISK---(asterisk uses
> >>> g711u)--->CARRIER-OUT
> >>> > >
> >>> > > How can i force asterisk to use user preffered codec for dialing
> >>> out so
> >>> that my asterisk machine saves time by no conversion
> >>> > > USER PREFERENCE IS
> >>> > > disallow=all
> >>> > > allow=g729
> >>> > >
> >>> > > CARRIER PREFERENCE IS
> >>> > > allow=all
> >>> > >
> >>> > > Anybody who can help?
> >>> > >
> >>> > > --
> >>> > > Rizwan Hisham
> >>> > > Software Engineer
> >>> > > AXVOICE Inc.
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> >>> > >
> >>> >
> >>> >
> >>> >
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> >>>
> >>> --
> >>> Rizwan Hisham
> >>> Software Engineer
> >>> AXVOICE Inc.
> >>> _______________________________________________
> >>> --Bandwidth and Colocation provided by Easynews.com --
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