[Asterisk-video] Videomixing in MeetMe

[Lorenzo Miniero]

On Wednesday 20 June 2007 18:14:00 Klaus Darilion wrote:
> Always - IMO a problem inside eyebeam or Windows (i have a "System"
> process which takes up to 70% CPU)


Ah ok, I was afraid that it could be fault of the mixed frames, or a wrong 
header.


> > This could be related to two aspects:
> >
> >         * the H.263 codec used by eyebeam could be not-100%-compliant
> > with the ffmpeg H.263 codec (and this is something we'd have to live
> > with); * there might be problems with the H.263 payload header and how I
> > handle it.
> >
> > I'm still investigating the H.263 payload header issue, also considering
> > that there are three different modes for it. In fact, different settings
> > give different results according to the client (for example, a setting
> > that works fine with our client gives strange behaviour with Wengophone,
> > while settings that work fine with Wengophone don't work at all with our
> > client, and the same applies with some other clients we tested). I'll try
> > eyebeam myself as soon as I can to see if I can sort it out. Of course,
> > if you have advices for this, let me know!
>
> Sorry - no advices. But interesting to know that wengophone works too.
> Which version of wengophone do you use? I never had success with using
> wengophone at all (lots of crashes).


I've tried with Wengophone classic. It allows you to specify another SIP 
account that is not their one, even if it forgets it each time you close the 
client... however, the video calls to MeetMe worked fine. I'll give you more 
details about the client tomorrow (I'm not at work now), since I can't 
remember if I installed it with a rpm or by compiling it.

>
> > Thanks a lot for your precious feedback, it's much appreciated!
> > If you have any other question and/or advices/suggestions/criticisms
> > about this work, I'll be glad to hear them.
>
> Where should the discussion happen - here or on another mailing list? (I
> prefer here - asterisk-dev).


It's the same for me, even if I think that the asterisk-dev wouldn't be very 
interested at the moment, since they probably have, and rightly so, other 
priorities over this one. So maybe keeping the discussion on asterisk-video 
would be better.


>
> btw: some feature requests for the next version ;-)
> - a MeetMe option which automatically activates this caller as video
> source (an implicit "videoswitch x y")


I'll work on some basic DTMF mechanism to automize this: something like the 
mute/unmute that already exists for audio.


> - A overlay in each video showing the name of the caller. e.g. by
> setting a certain variable before calling MeetMe.


That would be a really cool feature! Which will require some more work, of 
course... I'll study a bit how it could be done.


> - a option to have only all other peers mixed, but not yourself. E.g.
> currently if all sources are activiated, using eyebeam you see yourself
> two times: 1: in the eyebam video preview, 2: in the received mixed
> video. Of course this requires that each participant gets a dedicated
> mixed stream which probably requires much more load for the mixing.


Yes, the problem is that currently the layout is fixed for everyone. Besides, 
there's not an encoding context for each user, but only one for each involved 
codec. However, such a step was already in our head (each user should get a 
customized profile, and not only for layout, but also for bitrate, fps and 
all that is related to content adaption) so sooner or later we'll try to get 
this done too.


> -more debug outpout, e,g, in "meetme list 123" with details about the
> video (which source is activated, codec, ...)


You're right, this can be added without much efforts, and I'll surely do it.


>
> btw: When activating video I always get logs like:
> Incoming call: Got SIP response 415 "Unsupported Media Type" back from
> 83.136.33.3
>
> Is this only a warning that my SIP client does not support some
> conference extension (I saw INFO requests with XML payload)


As soon as a source gets enabled, I request a VIDUPDATE to the source as 
suggested on this mailing list when I uploaded the videoswitching some time 
ago. It is used to ask the user to send a full frame, which is useful when 
you add a new source to mixing. In SIP this is done with an INFO message, 
even if not all clients support it, so I guess Eyebam is one of them.


>
> regards
> klaus


Thanks again for your great help, cheers,
Lorenzo

-- 
Lorenzo Miniero, Junior Researcher
Dipartimento di Informatica e Sistemistica
Università degli Studi di Napoli "Federico II"
Via Claudio 21 -- 80125 Napoli (Italy)
Phone: +390817683821 - Fax: +390817683816
Email: lorenzo.miniero at unina.it