On 9/3/07, Holger Wirtz <[EMAIL PROTECTED]> wrote:
> Hans,
>
> as I wrote I am writing code for this feature since a year. The first
> implementation was bad (only in Perl). Now I am using C - that's quite
> more stable.
>
> On Sun, Sep 02, 2007 at 05:57:34PM -0600, Hans Fugal wrote:
> [...]
> > First, if you're not intimate with VOIP let me tell you (without
> > discouraging you, I hope) that it won't be as easy as you might think.
> > There's too much going on; it's like herding cats. You have to deal
> > with sound card input, NAT and firewalls, VOIP protocols, and somehow
> > orchestrating it all. Then you have to have someone manage something
> > like Asterisk on a server to provide the conference call capabililty.
> > Certainly doable, but not a weekend project as I'm sure the others
> > working on it are well aware.
>
> In fact thats a real problem. But the solution for some of the problems
> is libiaxclient (a portable softphone with the VoIP-Protocol IAX). I
> don't think that this will solve all problems. I am working with VoIP a
> long time (and with different protocols and manufacturers). But IAX
> has an ALSA/JACK/PortAudio Interface - so the problems for the sound are
> "only" the configuration of ALSA. Also IAX works fine over NAT due to
> use only _one_ port for signalisation and media streaming.

If you're using libiaxclient you're more ambitious than I realized. I
tried to write a client using that library and found it woefully
underdocumented. It seemed like an unmaintained one-off hack, but
perhaps it has improved since. You're right about IAX and NAT.

libiaxclient may even work in Windows with PortAudio, though it might
take some effort.

I think the easiest approach, after some thought, would be to drive an
Asterisk instance with the Asterisk manager API. This isn't very
cross-platform, though. You *could* run Asterisk in Windows on a
virtual machine, or various other workarounds, but nothing
user-friendly comes to mind.

> > My only intent here is to throw out the thoughts that I have about
> > what might trip someone up in doing this, so they can be considered
> > and addressed from the beginning. I don't want to discourage anyone
> > from this, which would be a very cool feature, nor from VOIP in
> > general.
>
> I think that I have solved some of the problems you mentioned - not all.
> But in my opinion (and my hope) the solution for VoIP conferences is closer
> than your thoughts ;-)

Great news! If you need help configuring Asterisk for conferencing or
have any specific ALSA questions drop me a line direct.

-- 
Hans Fugal
Fugal Computing

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