On 10/3/07, Tilghman Lesher <tilghman@mail.jeffandtilghman.com> 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!
On Tuesday 02 October 2007 16:55:52 Brian West wrote:
> On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
> > anyway still if there's a hack for meetme to work with g729 codec
> > this won't be an issue. So is there a hack or patch that i can use
> > any codec for meetme? tnx
>
> You still do not understand. It doesn't matter if the call coming in
> is g729 you must transcode it to signed linear, mix the frames and
> then code it back into g729 you end up with quality loss doing that.
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential for signal degradation.
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!
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