[sldev] SLDev-Traffic #24

[Chance Unknown]

There are a number of full rate codecs that are compatible with VOIP
telephony. One that is used quite frequently, and that you can find
implemented in open source VOIP products is GSM729 and the ilk. A
resonable codec needs to have adequate noise suppression and suitable
bandwidth for its intended use. Full rate codecs are both wasteful of
data bandwidth and generally do not offer better sound quality for
speech.

I would be much more interested on the documentation of LL's
integration effort and documentation of the API where the viewer
connects to the voice infrastructure, and what messaging is with
respect to the SIM is passed to the viewer (and on to the voice stream
if required). This would allow open source people to be able to better
understand the requirements, and to possibly make more sophisticated
optimizations within the viewer. -- Example: running the daemon under
wine, but the SL client native on linux...

==

On 8/15/07, Callum Lerwick <seg at haxxed.com> wrote:
> On Wed, 2007-08-15 at 11:47 -0700, Chance Unknown wrote:
> > The magic of analog to digital conversion of voice is performed by the
> > Codec. There are a number of opensource alternative codecs available,
> > but they are far from optimum for very specific use cases.
>
> Is there something wrong with Speex?
>
> The problem is not simply a matter of open/closed source, from what
> we've been told the primary codec in use is patented encumbered, thus
> even if we had a 100% open source, clean-room engineered implementation
> of the codec, we still couldn't distribute it in Fedora unless the
> patent holder gives a legally binding, written, irrevocable, perpetual,
> royalty free license grant for use by any and all Open Source software.
>
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