[Opensim-dev] Voice Module

[Ryan McDougall]

How convenient! There just happens to be a spot of code for just that
purpose: https://sourceforge.net/projects/voipforvw

Although Im thinking of putting it someplace different. I only chose
sourceforge because I had an account from way-back.

It works very well with Asterisk, which is what Im testing with. I
haven't touched it for a week, and it plain doesn't work yet, but its
only a couple steps away from a working solution, and I should be able
to get back to working on it next week.

Its GPL because of the SIP stack being GPL, but really its of no concern
to me. If someone prefers a BSD stack, and can recommend or better yet,
implement it, then Im all for it.

Cheers,

On Wed, 2008-02-27 at 10:26 -0500, Sean Dague wrote:
> I've spent a little bit of time looking at the Voice Module that just
> got moved into trunk from the RealXtend branch, and I think it's headed
> in the wrong direction.
> 
> Embedding voice in the simulator seems like the wrong direction, and
> much less modular than we were heading for.  Plus, I think that under
> any reasonable load, it's going to go useless *real fast*.  Given that
> chat messages can often take 30 seconds in a loaded message, nothing
> like voice really belongs there.
> 
> I think a much better approach would be to carve off voice functionality
> into a seperate VoiceServer, which would let it run on a dedicated
> machine, ensuring low latency.  I also think that the SLVoice approach
> of sending all the streams to the client and letting the client do the
> mixing is a better load approach.  
> 
> Perhaps a hybrid model where after significant avatar moves avatar
> possitions were sent to the Voice Server, so that you don't have the
> issue where your ears can be fully devoid of your camera, would be
> best.  But at the end of the day doing mixing on the server just seems
> wrong.  Even skype offloads this to clients to do.
> 
>      -Sean
> 
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