[asterisk-dev] Meetme/Timing Basics

[Prakash Rao Kanthi]

Thanks Kevin,

I got few things cleared.

Can you tell me how exactly Zaptel converts 2 RTP frames generated from 2 
sources of meetme (with in less than 20ms interval, more like 40 us) into 2 
frames in 20 ms intervals?

Because i see that i am getting 2 frames worth of data back from zaptel in 
less than 1ms. I am not sure what i need to do to time it for a 64kbps 
channel. I did setup all the pseudo channel options just like in MeetMe.

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 15:35:25 -0500
>
>Prakash Rao Kanthi wrote:
>
> > 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.
>
>Zaptel works on audio samples, it has no concept of frames. If you wish
>to learn how the conference mixing works, read the source in zaptel.c
>
> > Also can you point me to the code where incoming audio is converted to
> > SLINEAR?
>
>You said you were experienced with the Asterisk code :) Follow the path
>through ast_set_read_format() and all the things it does, since you can
>see that app_meetme.c sets the read format of the channel to SLINEAR
>before adding it to the conference.
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>asterisk-dev mailing list
>To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev