[asterisk-dev] Meetme/Timing Basics

[Prakash Rao Kanthi]

Alright guys, i got it working. The mistake was to take the output on psuedo 
channels created for 2 RTP streams and dumping it on to the caller channel. 
Instead what worked is to create another channel (3rd) for the caller and 
just receive the output on that channel so that it gives the RTP mix of 
first 2 channels.

Sounds simple now. But when i was debugging, did not had a clue. Thanks to 
Kevin for his help and sharp remarks, it finally occured to me.

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: Sat, 20 May 2006 14:13:53 -0500
>
>Prakash Rao Kanthi wrote:
> > Does any one else know how zaptel, streams samples out at 64 kbps rate?
> > Because when i send 2 chunks of 160 bytes at zaptel psuedo device back
> > to back (with in less than 20ms), two 160 byte chunks are thrown back at
> > my application with in the same time duration on psuedo device
> > descriptor. It almost seems like zaptel is not timing the data out.
> > Because of this the listener on the other end hears a scratchy voice
> > unlike a single stream which produces smooth voice.
>
>If you send two frames of audio in, you are going to get two frames of
>audio out.
>
>If you are trying to mix two streams together, then you need to open
>_two_ pseudo channels and send each source's frames down separate
>channels; the read side of each channel will then return back the
>'mixed' audio (although with only two channels, each side will only hear
>the other side, since the channel's audio is never returned back to it).
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