[Freeswitch-users] Question about Conferencing Capabilities

[Siobhan Hamilton]

My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Freeswitch (the other option being
Asterisk).  I've been poring through the docs, and have even set up a test
server myself, but there are some very specific things we are looking for
that I can't figure out if Freeswitch can do or not.

We want to be able to do the following:
- Create dynamic, on-the-fly conferences that can remain active even when
initiating user leaves
- Within a conference, give users the ability to mute and/or deaf individual
users (which I know can already be done with the "relate" command)
- Give users the ability to enter a "whisper" mode with another user - where
they are holding a private conversation that can only be heard by the two of
them
- Allow users to be in two conferences at once; the user would most likely
have one muted at any given time so as to hear the other one, but we want
them to be able to switch back and forth easily

Could anyone advise me on whether Freeswitch can accomplish these needs, or
perhaps what it might take to do so?  We are not averse to doing some
customization if we can find the people who know how to make it happen!

Thanks,
Siobhan Hamilton
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