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. > I'm not sure what the best approach would be, but I am inclined to > think it would be somehow talking to an existing VOIP client via IPC > and driving it to join/create the appropriate conference channels. I'm > not aware of any client that can be driven in this way, and I'm almost > sure that there's nothing cross-platform to fit the bill. You could > rip the SIP code out of something like Twinkle, but I'd advise against > that for one simple reason: getting VOIP working (especially SIP) is > hard enough when you've got a full-featured softphone or ATA or IP > phone. Stick things behind a façade like a FlightGear radio and it > will be all the more difficult to troubleshoot and 60%-70% will simply > be unable to get it working. I know that sounds like exaggerated > pessimism, but in my experience there's always *something* that goes > wrong in configuring VOIP. I don't think that all implemenations of FGCOM will work out of the box. The real problem is that I cannot distribute a static binary - it won't work at this time (and I don't know why) - everyone has to compile the sources. But my hope is that it will work for 90% of the users who know gcc and how to install libraries. > 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 ;-) Regards, Holger (sorry for my bad english...) -- ##### #### ## ## Holger Wirtz Phone : (+49 30) 884299-40 ## ## ## ### ## DFN-Verein Fax : (+49 30) 884299-70 ## ## #### ###### Stresemannstr. 78 E-Mail: [EMAIL PROTECTED] ## ## ## ## ### 10963 Berlin ##### ## ## ## GERMANY WWW : http://www.dfn.de GPG-Fingerprint: ABFA 1F51 DD8D 503C 85DC 0C51 E961 79E2 6685 9BCF ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ Flightgear-devel mailing list Flightgear-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/flightgear-devel
