[Opensim-dev] Status of voice

[Stefan Andersson]

> >> - Asterisk is a PITA to work with> >> - One would need to learn SIP telephony networking details to do a> >> proper and secure job of making a VoIP network> >> - using the hopefully simpler an better supported XMPP protocol suite> >> for IM, voice, and video> >> > >> > I really think this will only be interesting if SIP is still supported.
Yay for SIP - we've implemented spatial (server-side mixed) voice for OpenSim using an unaltered SL client communicating with an voice client using an embedded SIP Client based on pjsip[1]
 
For us, Freeswitch[2] turned out to be the best alternative on the server side. We have also implemented a full conference system with a custom freeswitch module, allowing us to do very nifty stuff with creating sub-groups and moving people between them.
 
As this is a full voice and moderated conferencing implementation, we are offering it as a commercial license - but of course, if anybody would initiate an open source initiative based on the same set of technologies, we would be more than willing to contribute our experience.
 
> > As Voice in Virtual worlds gets really interesting if you can also> > transparently bring in other sources of voice (call in numbers, or> > individuals). It's not just asterix that supports SIP, it's every major> > new commercial phone switch on the market.
 
As I said, yay for SIP - I also think we should think a bit about using SIP in other areas of OpenSim, as for example as the.. uhh.. protocol to initiate sessions.
> SIP was used in rexcom prototyping in December. Worked, even.> > The Telepathy lib(s), which are now considered for IM and voice etc, > support both XMPP and SIP (and IRC and Skype and whatnot).
We will follow your development with great interest!
 
/Stefan
 
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