[sldev] SLDev-Traffic #24

Dana Fagerstrom Dana.Fagerstrom at Sun.COM
Wed Aug 15 14:56:45 PDT 2007


I'd like to second (or third) the request for more details on how voice 
is "strapped" into the viewer.
D

Chance Unknown wrote:
> 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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