FOSDEM - Open Source VoIP

[Diana Cionoiu]

Hi Kilian,

It sounds very nice to have just one application for VoIP but it doesn't 
really work. Take a look on what happend with Asterisk. Was the only one 
media gateway we had in the begining, and we all hoped it will be as 
good as Apache. But things didn't work like that. Asterisk doesn't 
handle everything. In the same way Ekiga doesn't do everything you will 
need from a VoIP client. sipX is not the ultimate sip PBX, and ser is 
not the ultimate sip proxy since is so hard to use.
The beauty of open source is that we learn from mistakes and we can 
build better systems. Unified solutions are good in theory but since 
computers are made to serve humans, and not the other way around (humans 
to serve computers), and people are so complicated and different, having 
a unique solution doesn't work.
Think how many VoIP protocols, and pstn protocols exists.
I really doubt that we want to lose the diversity the open source provides.

Diana Cionoiu

Kilian Krause wrote:
> Daniel,
>
> On Fri, Feb 16, 2007 at 08:34:00PM +0000, Daniel Pocock wrote:
>   
>> As some of you know, I'll be running an open source VoIP session in the 
>> Debian devroom at FOSDEM.
>>
>>    http://www.fosdem.org/2007/schedule/speakers/daniel+pocock
>>
>> The theme will be quite generic and wide-ranging, looking at:
>>
>> - What is VoIP, codec comparison, etc
>>     
>
> Good. Please try to also make people understand that a 64kBit codec
> needs 80kBit/s bandwidth. And of course all of this "Skype uses iLBC
> too, why can't I just use /AppXwhatever/ just as a drop in replacement
> with my line like on Windows and have same A/V quality results" etc.. I
> guess you know what the various answers and most common questions are,
> but still, if unusre send me a copy of the presentation and we can
> discuss it. 
>
>
>   
>> - Survey of the products already out there, particularly the Debian 
>> packaged ones
>>     
>
> Ok.
>
>
>   
>> - As it is a developers conference, I will then proceed to look at how a 
>> SIP stack can be linked into an existing non-VoIP application, to give 
>> that application VoIP capabilities (e.g. messaging, click-to-dial)
>>     
>
> This point should especially point out that one doesn't need to fully
> implement one's own VoIP app, but doing remote control (like DBUS with
> Ekiga) is the much more sensible thing to do. If we'll get some hundret
> client apps out there in the long run, we'll just duplicate the same
> problems all over and over again. That means lost time and more dead
> upstreams when people loose interest. Interaction and synergy is what
> counts here.
>
>