[asterisk-dev] Meetme/Timing Basics

[Prakash Rao Kanthi]

Thanks Kevin. You said Zaptel does the mixing of SLINEAR data sent to it. Is 
it byte level mixing or RTP packet level mixing? And also i still have my 
questions on how the RTP from 2 users is streamed to 3rd user in the 
conference as a signle 64 Kbps channel without buffering.

Also can you point me to the code where incoming audio is converted to 
SLINEAR?

Thanks,
Prakash



>From: "Kevin P. Fleming" <kpfleming at digium.com>
>Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Subject: Re: [asterisk-dev] Meetme/Timing Basics
>Date: Fri, 19 May 2006 15:19:13 -0500
>
>Prakash Rao Kanthi wrote:
>
> > I see that meetme application is setting up pseudo device
> > /dev/zap/pseudo with a conference number for the first participant and
> > then on open the deivce for read/write for later participants. But it is
> > not clear how the RTP is mixed by this pseudo device.
>
>There is no RTP mixing; these are Asterisk channels, not related to any
>particular technology.
>
>The incoming audio is converted to SLINEAR, then passed to Zaptel over
>the pseudo channels, where it is then mixed and returned.
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