[Asterisk-video] MCU architecture

Sat Jan 31 04:34:31 CST 2009

Hi,

The architecture is obviously very influenced by IMS but I've not tested 
in a IMS environment but it should
work without problem (note that it is not a MRFC or MRFP so you'll have 
to use as an independent service).

The source code is at

https://sourceforge.net/projects/mcumediaserver/

Best regards
Sergio

telecomtom at vedatel.com escribió:
> sergio, mitul, just a couple general questions: 1. can you indicate again
> where the code is? 2. are you applying this in an IMS environment?
>
> -- TT
>
> Mitul Limbani, wrote:
>   
>> Sergio,
>>
>> This sounds pretty interesting, well I m actually not terrified when u
>> said JAVA :p, we can help you over there since we have java developers
>> on board.
>>
>> I would also want to revisit this entire project coz the last time I
>> did that I had to install A LOT (ye a lot) of dependent libraries.
>>
>> Also as I mentioned earlier my goal is to make the whole install
>> SIMPLE, so we might have to build in some sorts of shell script based
>> installer or mebbe gotta build relevant RPMs( yeah terror for debian
>> fans like me) but I clarified myself above "SIMPLE".
>>
>> So let's coordinate n make things moving,
>>
>> Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt Ltd,
>> The Enterprise Linux Company(r),
>> http://www.enterux.com/
>>
>>
>> On 30-Jan-09, at 15:48, Sergio Garcia Murillo <sergio.garcia at fontventa.com
>>  > wrote:
>>
>>     
>>> Hi everyone,
>>>
>>> Currently the mcu solution has two main components, the VideoMixer and
>>> the mcuWeb.
>>>
>>> The videomixer component handles ONLY media, i.e. it receives rtp,
>>> unpack audio and video, performs audio/mixing video, encoding, packing
>>> and rtp sending. It is completely controlled by a xmlrpc interface and
>>> has no service logic at all.
>>>
>>> The current xmlprc api has the following methods:
>>>
>>> -Create/Destroy conference
>>> -Add/Remove participant to conference
>>> -Set conference parameters like video size and number and distribution
>>> of participants on screen
>>> -Set audio/video send/receive ports per participant
>>> -Set audio/video codec and parameters (size,fps) etc per participant
>>> -(Un)Mute participant
>>> -Set conference mosaic positions: lock slot, assign slot to
>>> participant,
>>> etc..
>>> -Add watch only participant to conference (experimental: flash video
>>> broadcasting in web)
>>>
>>> It currently supports only h263p but should not be too difficult to
>>> add
>>> support to h264. The rest of the functionalities are completed (except
>>> flash support) and only a bit of testing is needed. In the near
>>> future I
>>> would like to convert the videomixer in a MediaServer, been able not
>>> only to offer multivideo conference services, but also transcoding,
>>> flash casting, etc..
>>>
>>> As I said before everything is controlled by an xmlrpc api, so a
>>> component handling the service logic and signalling is needed. That
>>> component could app_conference and the confiance project did integrate
>>> the video mixer as an external unit.
>>>
>>> I decided to implement it as a complete external unit from Asterisk.
>>> Why? I think it was easier quicker and easier to develop, avoid the
>>> monolithic and sometimes obscure architecture of asterisk and could
>>> provide much more functionalities. And the chosen technology was..
>>> java
>>> (I feel a great disturbance in the Force, as if millions of voices
>>> suddenly cried out in terror and were suddenly silenced).
>>>
>>> Yes, Java, using the Sailfin Sip Application Server
>>> (https://sailfin.dev.java.net/) which allows to create an application
>>> that handles SIP and HTTP request (an application like click to dial
>>> is
>>> just a few lines of code
>>> http://wiki.glassfish.java.net/Wiki.jsp?
>>> page=SipClickToDialExample2). If
>>> you start with your prejudges about java, speed and show on, just
>>> think
>>> that it is a telco grade Sun and Ericsson development.
>>>
>>> The mcuWeb component implement the service logic, handles all the SIP
>>> signalling (receiving invite request from asterisk), controls the
>>> Video
>>> Mixer with the xmlrpc and offers a WEB UI to manage the conferences.
>>>
>>> This part is also fully functional, but I think that is where more
>>> work
>>> is needed in order to customize the service with the functionalities
>>> needed by the customers. In particular questions like the following
>>> need
>>> to be answered:
>>>
>>> - Is it required to create the conference before the user calls? or it
>>> get created when it calls in?
>>> - Are there private conferences? How are the participants allowed to
>>> get
>>> in, by password or by invite only?
>>> - Is there always a default public room?
>>> - etc...
>>>
>>> Any thoughts are welcome
>>>
>>> Best regards
>>> Sergio
>>>
>>>
>>>
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>
>
>
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