<br><br><div><span class="gmail_quote">On 10/3/07, <b class="gmail_sendername">Tilghman Lesher</b> <<a href="mailto:tilghman@mail.jeffandtilghman.com">tilghman@mail.jeffandtilghman.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On Tuesday 02 October 2007 16:55:52 Brian West wrote:<br>> On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:<br>> > anyway still if there's a hack for meetme to work with g729 codec<br>> > this won't be an issue. So is there a hack or patch that i can use
<br>> > any codec for meetme? tnx<br>><br>> You still do not understand. It doesn't matter if the call coming in<br>> is g729 you must transcode it to signed linear, mix the frames and<br>> then code it back into g729 you end up with quality loss doing that.
<br><br>Or, in other words, you cannot mix compressed data. You must first<br>decompress the data for mixing, then recompress it for transmission.<br>During both operations, there is a potential for signal degradation.</blockquote>
<div><br><br>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.
<br><br>do you get what i want to do? Thanks!<br></div><br></div>