[asterisk-dev] code-cleanup concerns

[Mike Taht]

On 4/15/06, Adrian Sietsma <adrian_groups at sietsma.com> wrote:
>
> Matt Ranney wrote:
>   > Somehow the jitter buffering or packet loss concealment or whatever
> > magic that Skype uses makes it work better than asterisk/IAX over the
> > same WAN link.
> Having done some testing of voip over WAN, I would say it is both. I have
> seen packet loss of up to 20-30% (using a CDMA wireless card). Skype's
> codecs, including iLBC, seem very tolerant of packet loss.


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.

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
1.0.5release) on a 800Mhz Athlon64... (30 vs 32 millisec)

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".

Lastly, the loop in chan_speex that converts ints to floats is very
SSE-able.

--
Mike Taht
PostCards From the Bleeding Edge
http://the-edge.blogspot.com
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