Re: Conference call with mixing

[Frederic Tardif]

Hi,

I had the same use case and I was wondering if WebRTC was able to do that peerconnection mixing? 

The idea is to mimic the scenario of a 3way conference call done by bridging 2 endpoints directly in a phone without an external conference bridge. 

Here's an example:

- Alice is a normal SIP phone

- Bob is a webRTC browser app

- Carol is another SIP phone

1. Alice calls Bob through a SIP/HTTP+WebSocket+WebRTC Gateway (so far so good, a peerconnection is mounted and the RTP flows)

2. Bob (the WebRTC dude) wants to consult Carol. It can create a new independent PeerConnection with Carol (through the sip gateway).

3. Alice can then talk to Bob and Bob can talk to Carol. But Alice is not bridged with Carol.

4. Is it then possible for Bob through the WebRTC api to mix Alice's and Carol's peerConnections in order to conference the 3 participants together?

thanks

Frédéric Tardif

On Friday, March 15, 2013 9:36:33 AM UTC-4, Vincent Séguin wrote:

Hi people,

We already implemented webRTC in our application successfully. We are now looking to do multi-users conference calls.
We looked at Veckon (http://webrtc-multivc.appspot.com/) and it seems not so hard.
Problem is, in our application, users aren't (will be probably not in fact) web browsers.

On one side, there will be one browser, but other sides are two regular phones.... we got some code to transform SIP calls into SDPs offer and all.
We understood that we would need some kind of mixing for the video/audio in the browser side, since we can't maintain multiple peer connections
on each client..

Question is : is it possible in webrtc? How? Does webRTC support mixing ? Do any example exist?

Thanks a lot!

[Sergio Garcia Murillo]

Hi Vincent

Check this

http://www.medooze.com/products/mcu/webrtc-support.aspx

Best regards
Sergio

El 15/03/2013 14:36, Vincent S�guin escribi�:

> Hi people,
>
> We already implemented webRTC in our application successfully. We are
> now looking to do multi-users conference calls.
> We looked at Veckon (http://webrtc-multivc.appspot.com/) and it seems
> not so hard.
> Problem is, in our application, users aren't (will be probably not in
> fact) web browsers.
>
> On one side, there will be one browser, but other sides are two
> regular phones.... we got some code to transform SIP calls into SDPs
> offer and all.
> We understood that we would need some kind of mixing for the
> video/audio in the browser side, since we can't maintain multiple peer
> connections
> on each client..
>
> Question is : is it possible in webrtc? How? Does webRTC support
> mixing ? Do any example exist?
>
> Thanks a lot!

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[Frederic Tardif]

Medooze MCU sounds interesting (I'll definitely have a glimpse at it), but it involves an external media server to mix the RTPs. Could be an option, but I would like to first confirm if it is feasible (or plan on the roadmap) to mix peerconnections directly in the browser with the webRTC api?

Frederic Tardif

On Friday, March 22, 2013 5:35:21 PM UTC-4, Sergio Garcia Murillo wrote:

Hi Vincent

Check this

http://www.medooze.com/products/mcu/webrtc-support.aspx

Best regards
Sergio

El 15/03/2013 14:36, Vincent S�guin escribi�:

[Caragea Silviu]

Hello,

Some while ago I have build voice conference (for the moment) with
mixing on the client side as skype. Basically one of the clients is
the mixer (the one who initiate the conference in my case ) . From my
tests is working very nice (comparable with skype) . But I'm sure
webrtc is not interested in this as time they have removed this
features from GIPS.

The advantages that I'm seeing is that the conference is totally free
from a cost perspective and no additional server to be scaled and
maintain. I tested with over 8 participants (including SIP, android,
ios) and the quality was very good.

From what I see in skype for video conference (which is a paid system)
only the video stream is coming from a 3rd party server . the audio
stream is mixed on the client which initiate the conference . I'm
still working on this, the main problem is how to sync the audio and
video as time they are coming from different location.

Any idea ?

Silviu

[Frederic Tardif]

Hi,

thanks for sharing the info. What I hear from the comments above is that mixing on the client side using webRTC could be a compelling use case. 

Could someone from the dev team comment on the feasibility of such a scenario? Is it on the dev roamap? If not, is there a steering process to vote for this feature in the upcoming versions?

thanks

Frédéric Tardif

[Justin Uberti]

I think you could do this with the Web Audio API, once WebRTC supports using remote MediaStreams as input to a PeerConnection. (Receive N streams, mix using Web Audio, send output to a PeerConnection).

That said, once video gets involved, this architecture isn't a good fit, so I don't know how useful this really is.

[Frederic Tardif]

Thanks for the clarification

Frederic

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[Lee McKenzie]

"once WebRTC supports using remote MediaStreams as input to a PeerConnection. (Receive N streams, mix using Web Audio, send output to a PeerConnection)."

Does this already work?  Or is there a time estimate for when this might be available?  Or how can I help?

[Vikas]

Not yet please see issue :https://code.google.com/p/webrtc/issues/detail?id=383

/Vikas