[asterisk-dev] meetme and zaptel mixing algorithm

[Takashi Ohashi]

Hi,

Can somebody show me about the working of Asterisk Meetme audio mixing.

Now, I have found out the working of Meetme audio mixing shown below.
It seems that softphone of some conference users and asterisk build RTP 
session
separately(per user).
Each time asterisk(app_meetme.c) receive RTP packet from
the conference user, traslate the codec of payload data to signed linear
and write to pseudo device.

I want to know  how to zaptel mix frames(audio data) from users,
after asterisk write to pseudo device.

I confirmed that in case that 3 users participate in Meetme conference and
only one user speek on the softphone, this users voice is recognized and
heard on the softpone of the other two users.
But 2 user speek on the softphone at the same time, I can not hear both of
their voice on the other user's softphone.

I want to know the mixing algorithm of asterisk and zaptel.
One of mixing algorithm is that every sampling time(20ms),
mixer sums the encoded audio value (signed linear value) of the other users.
I think this is the most popular audio mixing algorithm.
Otherwise, how asterisk and zaptel are working?

Best regards,
Takashi. 
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