In article <[EMAIL PROTECTED]>, Mark Quitoriano <[EMAIL PROTECTED]> wrote: > > yeah i still don't understand. this is what i want to do. I want asterisk > not to compress and decompress codecs. so either i can use SLIN as my codec > for my SIP or IAX. or i can remove SLIN codec in meetme and change it to > g729a so there's is no compression and decompression. > > do you get what i want to do? Thanks!
Yes, but it can't be done. In order to allow each conference participant to hear all the others at once, it is necessary to mix the audio by adding the contents of each channel. It is impossible to mix G.729 compressed because there is not a simple mathematical relationship between the output data and multiple input data. The mathematical way to do it would be what you are trying to avoid: convert each incoming stream to signed linear samples, then perform the mixing by adding those samples together, and then convert the outgoing mixed stream back to G.729 or whatever. This is what Asterisk does with any kind of codec that talks to Meetme, whether it be uLaw, ALaw, GSM, G.729, ILBC, and it doesn't need all participants to be using the same codec. Why were you so set on mixing G.729 without decoding/encoding? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
