[Asterisk-Users] Passing audio stream through Asterisk or not?

[John Todd]

Because your original question asked about G.723, which Asterisk 
cannot transcode because there is no codec support for it within 
Asterisk.

Summary: If all you want to do is have Asterisk "relay" the RTP 
(voice) data between two endpoints, pretty much any codec can be 
used, since Asterisk doesn't have to interpret the data stream for 
any reason - it's just moving the data around, and not "listening" or 
"talking" in the stream.  However, if you want to do something clever 
with that data/sound stream, such as listening for the "#" key (the 
"t" option in a Dial statement) then you'll need to be using a codec 
that Asterisk understands (G711, gsm, iLIBC, etc.)

JT


>I'm not sure that I understand you.
>Why not to do transcoding if sometimes required?
>
>Thanks,
>Dan
>----- Original Message -----
>From: "John Todd" <jtodd at loligo.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Saturday, May 31, 2003 7:35 PM
>Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or not?
>
>
>>  There is one more note: make sure you don't have any options in your
>>  Dial statement that require the Asterisk server to do transcoding.
>>  Such options would be "r", or "m", or "t", which will cause Asterisk
>>  to need to listen and/or insert sounds in an audio stream if I
>>  understand previous conversations here to be correct.  I would just
>>  remove all options from your Dial statments entirely and see what you
>>  get.
>>
>>  JT
>>
>>
>>  >On Sat, 2003-05-31 at 10:51, Dan wrote:
>>  >>  Hi,
>>  >>  > if you turn off the reinvite in the asterisk configs for those
>ata186s
>>  >>  > then it will pass through asterisk even if asterisk doesn't
>understand
>>  >>  > the codec.
>>  >>  So I must have:
>>  >>  canreinvite = no
>>  >>  in sip.conf file for the specific phone?
>>  >
>>  >yes
>>  >
>>  >>  Then the call is passed through Asterisk without any conversion?
>>  >
>>  >yes
>>  >
>>  >>  How can I do to pass all the calls through Asterisk, even if a codec
>>  >>  conversion is required or not?
>>  >
>>  >canreinvite=no
>>  >The whole point is you don't reinvite the phones to talk to each other
>>  >instead of passing through asterisk.
>>  >
>>  >>  ----- Original Message -----
>>  >>  From: "Steven Critchfield" <critch at basesys.com>
>>  >>  To: <asterisk-users at lists.digium.com>
>>  >>  Sent: Saturday, May 31, 2003 5:27 PM
>>  >>  Subject: Re: [Asterisk-Users] Passing audio stream through Asterisk or
>not?
>>  >>
>>  >>
>>  >>  > On Sat, 2003-05-31 at 08:06, Dan wrote:
>>  >>  > > Hi all,
>>  >>  > >
>>  >>  > > One short question.
>>  >>  > > When one extension (let's say ATA-186, SIP image, G.723 codec
>>  >>  > > selected) try to call an external SIP address like:
>>  >>  > > SIP/user at domain.com, where another identical ATA-186 is available
>with
>>  >>  > > G.723 codec selectrd,
>>  >>  > > after the signaling phase, the call is established through
>Asterisk or
>>  >>  > > directly between the two ATAs?
>>  >>  > > There is no G.723 codec available on Asterisk
>>  >>  > > I need to know this because of the firewall.
>>  >>  >
>>  >>  > if you turn off the reinvite in the asterisk configs for those
>ata186s
>>  >>  > then it will pass through asterisk even if asterisk doesn't
>understand
>>  >>  > the codec.
>>  >>  >
>>  >>  > --
>>  >>  > Steven Critchfield <critch at basesys.com>
>>  >>  >
>>  >>  > _______________________________________________
>>  >>  > Asterisk-Users mailing list
>>  >>  > Asterisk-Users at lists.digium.com
>>  >>  > http://lists.digium.com/mailman/listinfo/asterisk-users
>  > >>  >
>  > >>  >
>  > >>
>  > >>
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>>  >--
>>  >Steven Critchfield <critch at basesys.com>
>>  >
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