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6 msgCould two Asterisk servers connect through VPN
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8 msgProblem with the loading of the cards in Debian

RTP Mixer
\ Akashdeep Dutta (7 May 2007)
. \ Akashdeep Dutta (7 May 2007)
. \ Akashdeep Dutta (7 May 2007)
. . \ Joshua Colp (7 May 2007)
. . \ Akashdeep Dutta (8 May 2007)
. . . \ Alex Balashov (8 May 2007)
. . . \ Eric \ManxPower\ Wieling (8 May 2007)
. \ Kapil Dhawan (8 May 2007)
. \ Kapil Dhawan (12 May 2007)
. . \ Kapil Dhawan (14 May 2007)
. . \ Tzafrir Cohen (14 May 2007)
. . . \ Kapil Dhawan (14 May 2007)
. . . . \ Joshua Colp (14 May 2007)
. . . . . \ Kapil Dhawan (14 May 2007)

1 msgRhino Analog Cards - Questions
6 msgSend SIP Re-invite.
22 msgHPEC audio clipping
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Subject:Re: RTP Mixer
Group:Asterisk-users
From:Kapil Dhawan
Date:14 May 2007


 
Perfect Josh...but if i am running an application which has a capability
of showing number or participants depending upon CC value, that doesn't
work. Secondly, Asterisk can show on CLI about current "talking" users
where it is maintaining talking status but not sending it down the line
to be used by other apps.

Anyways, i will go with your statement and leave it on core developers
to comment.

Joshua Colp wrote:
> Kapil Dhawan wrote:
>> I was reading an article on RTP Mixer so started studying about the
>> mixing done by Asterisk in MeetMe. Read that CC should contain the
>> no of participants ifupto 15 and CSRC should come, but not getting
>> any by asterisk.
>>
>>
>
> I'll just leave it at this so we can all move on with our lives:
> Asterisk isn't totally an RTP Mixer in the sense you are reading
> about. It is an audio mixer. Frame of audio comes in over RTP, gets
> sent in (only the audio portion) to be mixed, frame comes out, gets
> turned into RTP again. The RTP part has no idea that multiple sources
> were mixed together, 'nor should it care. The sources could have been
> Zaptel channels for example in which case they couldn't be added to
> the list.
>
> Joshua Colp
> Software Developer
> Digium, Inc.
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