[Asterisk-video] Need to implement Multi point Video Conferencing

[Matt Riddell]

On 29/01/2009 10:34 p.m., Sergio Garcia Murillo wrote:
> Matt Riddell escribió:
>> On 27/01/2009 9:58 p.m., Sergio Garcia Murillo wrote:
>>  
>>> Hi everyone,
>>>
>>> From time to time the multi video conference topics comes up so this is
>>> quite an old issue.
>>>
>>> First, the self-advertisment, I already have available (for a looong
>>> time) a multi video conference unit
>>> with video mixing, SIP based, no participant limit (apart of cpu
>>> resources), continius presence,
>>> different mosaic modes and sizes, web ui, etc, etc..
>>>
>>> The main problem here is none has a commercial plan or is willing to
>>> invest a cent in supporting the
>>> development or even give feedback ideas. So, I think that until someone
>>> shows some real interest
>>> in deploying a commercial mcu service nothing is going to change and
>>> this thread will be a dead end
>>> again.
>>>     
>>
>> You didn't realize everyone wants everything for free (with free
>> support)?
>>
>> :)
>>
>> But seriously, what are we looking at?
>>
>> Funding?  How much?  What time period?  What method?
>>
>> End result?  Commercial/Open Source/mixed license etc?
>>
>> Support/help?  Are there other things people could help you with?
>>
>>   
> 
> Hi again Mat,
> 
> The main problem with the multi video conference is that most people
> thinks about the service and just thinks,
> "hey, that's cool" and want to have it. The problem is that it's not an
> easy issue (even with my help) and
> is not as easy as plug&play, so as soon as they have to spend some time
> working and setting everything up,
> they get bored and realize that they don't really need the service at all.
> 
> So what it's needed? Just commitment, so the time I waste is worthy.
> Obviously being willing to spend some
> money is a very good sign of commitment.. :)

:) Ok, I can offer time, a testing lab, possibly money (although I would
rather contribute to a pool rather than paying for it all).

> Also, time and patience are needed, and lots of testing..

Can help here as well.

So, I guess we start with the components.

I suppose it would be best to run this outside of Asterisk so that
patches don't need to be kept up to date.

So, as I see it the components are:

Video SIP Client
Asterisk
Video Mixer

User calls Asterisk and reaches a dialplan entry:

exten => s,1,Dial(SIP/my_mixer)

I guess the problem with this is implementing a SIP stack in the mixer.

Is there a possibility to use Sofia or something?

Maybe what is actually wanted is to just reinvite the rtp via the video
mixer so that there is a stream.

Not sure, but I thought there was some capability in 1.6 to process rtp
externally?  Or was that just an idea that never happened?

I'm happy to get one of the people in our offices to write up
documentation, and I can possibly help with a bit of ANSI C, installers,
Debian packages, running in valgrind, running against Asterisk 1.4 and 1.6.

Obviously I can also help to promote the idea of a multi party video
conferencing solution whether it be a commercial or Open Source project.

GPL licensing would make it easier to reuse external components, but you
may want to create a solution which you can sell after creation.

If we break the project down into manageable pieces we can possibly
garner support, and assign different pieces to different people.

Thoughts?

-- 
Kind Regards,

Matt Riddell
Director
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