Paul W. Frields
stickster at gmail.com
Tue Feb 12 18:47:57 UTC 2008
On Tue, 2008-02-12 at 09:33 -0600, Jeffrey Ollie wrote:
> On 2/12/08, Jon Stanley <jonstanley at gmail.com> wrote:
> > On Feb 12, 2008 9:42 AM, Jeffrey Ollie <jeff at ocjtech.us> wrote:
> > > After writing the message last night I dug a bit more into the details
> > > of hooking up Asterisk and Flumotion and I definitely think that it'll
> > > be possible. I think that combining Asterisk and Flumotion give us the
> > > best of both worlds - use Asterisk to bring together the board members
> > This sort of eliminates the possibility of a "town hall" type meeting
> > though (which I thought was the whole point), where there's
> > interaction between the Board and the community
> > Unless questions are accepted via IRC, for instance....but you still
> > lose the verbal interaction. May or may not be a big deal.
> Having tens or hundreds of people in an Asterisk conference call would
> not be feasible I think. The management interface for the
> conferencing isn't great so it'd be difficult to moderate who has the
> floor, etc. I'd have to look at the code to be sure, but I'm not sure
> if the mixing code optimizes out silent frames, and getting various
> SIP clients to stop transmitting frame is problematic because many NAT
> implementations need the two-way RTP flow to keep the ports open -
> Asterisk has RTP timeouts as well.
> If some sort of verbal interaction was desired I think that you'd need
> to conduct the meeting more like I've seen various school board and
> city council meetings conducted. All members of the board would be
> connected via a SIP client and would have full-duplex audio. Members
> of the general public would be able to listen in on the audio
> streaming site. If someone had something to present or a question to
> ask would need to request the floor, probably though a IRC channel.
> Once granted permission by the chair of the meeting, you'd be sent a
> private SIP URL to connect to which would give you access to the
> conference call. Once your turn at the "microphone" was over the SIP
> URL would be disabled.
> Yes, this does limit somewhat a more free-flowing discussion, but it
> also keeps chaos at bay.
CC:ing f-a-b since this relates to its readers as well.
This sounds like a workable solution if we want call-in questions -- do
we have the technical bits to support it?
Paul W. Frields http://paul.frields.org/
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