[asterisk-dev] Asterisk and video conferencing

[Johan Wilfer]

2014-03-31 13:41, Olle E. Johansson skrev:
>
> On 31 Mar 2014, at 12:47, Johan Wilfer <lists at jttech.se> wrote:
 >
>> 4. Jitsi Videobridge - multiple video streams from server, but send
>> only your stream to the server. The jitsi videobridge the
>> distributes the stream to all other clients. This will eat a lot of
>> bandwidth for the server, but not for the clients. This is also how
>> Google Hangouts works. So if you are 10 participants you will send
>> one stream to the server with your audio/video and receive 9
>> streams from the server for the other participants.
>>
>> What you lose with option 4 is everything asterisk excels at: pstn
>> connectivity, fine-grained control of each participant in the
>> bridge.
>>
>> What are your thoughts on adding Jitsis approach in regards to
>> video to Asterisk for confbridge or even ARI? No composing of
>> video, just relaying the other participants streams to each other
>> in the bridge. Then it's up to the client in the other end to
>> display these streams in a reasonable way (like Google Hangout, and
>> https://meet.jit.si/).
>
> Why? The jitsi video bridge exists and work fine :-)

I think it's hard to integrate them and keep the functionality from 
both, maybe I'm wrong. But Asterisk does the audio mixing really great, 
and you have very fine-grained control on recording / muting / menu's 
etc. You may want the audio in asterisk to do video follow speaker.

So that's my question - could this fit into asterisk architecture to get 
multiple videos instead of one. I find the Jitsi approach very clever to 
avoid hitting the cpu too much.

> What you are forgetting here is the thing that has stopped us from
> doing really cool stuff with video - patents and licensing. The jitsi
> video bridge is a nice workaround, but not optimal if you have a lot
> of different devices. You put the load on the device and in
> bandwidth-constrained environments that's not good.

I think this is a related but different issue. In a 1-1 call you may 
want to transcode to a different codec, or downsample to save bandwidth 
I think with VP8 you can do that by just dropping packets, if I got it 
correctly.

On a conference-call however you may want all video streams (or just 
one, but asterisk can already do this). Regarding bandwidth people tend 
to have a lot more bandwidth downstream than upstream.

>
> Video is heavily dependend on peer2peer negotiation and doesn't
> really fit well in a PBX b2bua architecture... The jitsi model could
> work - but the SDP o/a handling would be really hard to get right in
> Asterisk.

This is really the question, if this fits into the Asterisk 
architecture? Because I expect the clients to be able to decode and play 
the video streams. Transcoding would be cool also, but is resource 
intensive.

>
> I've been lobbying hardware manufacturers to provide video cards for
> Asterisk where we can have licenses to do transcoding and
> reformatting, so far with no success. Cisco's H264 codecs recently
> became available for us in the Open Source world thanks to a generous
> solution by Cisco. I guess funding is needed to add anything cool to
> Asterisk using them. We can do MCU-style stuff, reformatting - but to
> do transcoding we need another codec :-)
>
> Google VP8 is around, I don't know what Digium's legal team have to
> say about us using it.
>

Given how cpu intensive transcoding video I wouldn't like that to hit my 
cpu anyway.. :-)

> Random thoughts...

Thanks Olle!


-- 
Johan Wilfer