<br><br><div><span class="gmail_quote">On 4/15/06, <b class="gmail_sendername">Adrian Sietsma</b> <<a href="mailto:adrian_groups@sietsma.com">adrian_groups@sietsma.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;">
Matt Ranney wrote:<br> > Somehow the jitter buffering or packet loss concealment or whatever<br>> magic that Skype uses makes it work better than asterisk/IAX over the<br>> same WAN link.<br>Having done some testing of voip over WAN, I would say it is both. I have
<br>seen packet loss of up to 20-30% (using a CDMA wireless card). Skype's<br>codecs, including iLBC, seem very tolerant of packet loss.</blockquote><div><br>I have been dinking with speex a little bit. Doing 30% packet loss (using the iax losspct command) just makes the voice sound metallic, but still quite understandable. speex sounds pretty darn good in normal situations.
<br><br>I note that the latest version of speex 1.1.2, with SSE support, encoding via the show translation claims to be just slightly faster than ilibc with the default asterisk settings for it (and twice as fast as the 1.0.5
release) on a 800Mhz Athlon64... (30 vs 32 millisec)<br><br>that doesn't mean that it's working corectly, however, as at least the last time I tried it, the first bit of audio was consistently cut off. a background(good-bye) for example, was merely "Bye".
<br><br>Lastly, the loop in chan_speex that converts ints to floats is very SSE-able.<br></div></div><br>-- <br>Mike Taht<br>PostCards From the Bleeding Edge<br><a href="http://the-edge.blogspot.com">http://the-edge.blogspot.com
</a>