Perhaps a bit of reading here might help: https://webrtc.org/ and https://webrtc.github.io/webrtc-org/ and https://en.wikipedia.org/wiki/WebRTC

"Doesn't zoom use the same protocol as vMix Call?" Probably not....gut feel is no!

A quick way to benchmark the quality you can get is to establish a call between two PC's on your LAN. This will give you an indication of what webRTC is capable of. Now when you have a call to an external location, what you will see is how it adjusts to the prevailing network conditions as it passses through ISP's, the cloud and so on.

Even though a speed test at either end indicates plenty of bandwith it doesn't show where throttling or QOS has been applied in any part of the chain.

The computer is not maxing out. As I mentioned, even doing a stripped back show with only four calls the quality was pretty poor and we were pre recording this show so no streaming was happening at the time.

My guests connect using current model, i5 Microsoft Surface Pro’s or newish ipads.

I know that many external factors are at play when using VMix call but it just seems that zoom gets consistently better results in similar situations.

Thanks for the reply Martin,

In terms of ‘quality’, what I experience is the frame rate dropping to 1- 10 fps rather than compression artifacting or blurriness due to low data rates. Images seem to stay sharpish, but just freeze for over a second fairly often.

I set the return feed to each presenter/ guest to 800kbps as this seems to be the lowest quality I can get away with while presenters can still read the graphics on our output. Any lower, and the text turns to mush and they say they can’t read it.
I have played around with the Advanced settings and switching it from Auto to 2000kbps but neither seemed to make much difference. I’ll try a few in between tomorrow.

I didn’t know Zoom maxes out at 360p! Maybe I do need to lower my incoming feeds in the Advanced settings to 800kbs or so and see how that goes but I assumed that the ‘Auto’ setting should take care of things like that and adjust depending on available bandwidth. It seems to me that rather than adjusting the compression or data rate, it just drops the frames .

All in all, I should say that you have done an amazing job with VMix overall. I’ve been in broadcast TV for twenty years and even now, it’s amazing what VMix is capable of.

Cheers.

I can't help but answer the question, "Does Zoom use WebRTC?"

Yes. And No.

They use a lot of the underlying fundamentals, but they implement it in an installable client application. That's one of the things that WebRTC set out to eliminate.

Zoom does this because they have some secret sauce of their own and they have, they feel, a unique focus on making the process easy for clients. They're not wrong about making things easy. In recent days some of their strategy, like glossing over security, has come back to bite them.

Zoom does have a web client that can be used to join a conference without having the application installed. They don;'t make it especially easy to find this way of connecting. They're prefer people to install the client.

The folks at WebRTC Hack have a great overview of how even the web client isn't normal WebRTC.

https://webrtchacks.com/zoom-avoids-using-webrtc/

I completely get this. The issue that I have observed is the browser like to react very quickly to degrading network conditions. Very slowly to improved network conditions.

And their spouse on a Zoom call for work, and 4 kids on Youtube or playing video games with friends...

vMixCall audio
Great for simplicity and flexibility but I'm also disappointed with vMixCall audio. Can't use in good conscience with musical instruments. FB live stream can do an acceptable job with music from a smartphone. Does FB have some magic tech available nowhere else?

My comment was not intended as a general negative on vMixCall. Only to note that the technology is out there for better audio, e.g. FB Live. I appreciate that implementation is non-trivial.

vMixCall would have a whole world of applications for live music streaming if the audio quality was as good as what can be done with FB Live.
I will be trying beta.vmixcall.com. Thanks.

Hi Mironta, there are many audio settings and controls within vmix, so it would help if you are clear about which direction and what you are hearing. vmix has multiple audio bus, audio bit rate settings, and the possibility of audio being fed back/muted to the speaker as well as them hearing themselves. Also, when talking about vmixcall and comparing it with FB live, are you comparing apples with apples ?. I would have thought you would compare vmix with FB live, and not vmixcall which is a separate component in vmix.

Sorry to say. webRTC has dynamic latency. You'll have hard times getting remote musicians playing together.
In your case you would want to have a locked latency, and the same for all participants. That said SRT seems a viable solution. Larix broadcaster is able to do this (no return feed) or something like medialooks (including return)

If you would like to test give me a ping.

vMix Call is as good as any WebRTC service that uses a browser as a client. The weak link is audio handling in the browser, where most services leave automatic noise reduction, echo cancellation and gain control turned on.

Other services, like SourceConnect, are specifically designed for broadcast contribution. They take more explicit control of the audio settings. They are NOT free.