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WebRTC player has very bad bitrate issue


Jianc
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Hi Selim,

Thanks for your reply. We’re using the Enterprise Edition (WebRTCAppEE).
Yes, publishing and playing with WebRTC. 

We’re running on AWS servers:
Publisher/origin: c5.large
Viewer/edge: t3.small

If it's because of the servers, do you have any suggestions? We're expecting 2 live videos with 30 viewers each, concurrently.

Using the built-in web admin/management app will get you the unpleasant result.
.
Just so you know, using HLS works fine, better quality, etc. ~10seconds delay, which is expected for HLS I guess.

Thanks!
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Hi all
I also have a similar problem.

Yesterday I did tests for a few hours and my conclusion is:

If the Ant Media configuration is higher than 2400 Bitrate (Kbps), the video is broken after going through WebRTCAppEE (OBS rtmp: // - >> WebRTCAppEE / -> 5443 / WebRTCAppEE / player.html), independent of the Bitrate of UpLoad.

The same videos that are broken in WebRTCAppEE look good on YouTube (using the Ant Media UpLoad option to YouTube)

Several tests were carried out for 3hs. Streaming was done in several UpLoad Bitrate with enough bandwidth. Monitoring was done in several locations with more than enough bandwidth.

I attach statistics screenshots in the server (Google Cloud) and Ant configuration.

VM in Google Cloud 4 cores, 15G ram (southamerica-east1-b)
Ant Media Enterprise Edition 1.6.2-SNAPSHOT 20190305_0945




configuration-17-03-19-ant-oscar-parzon.thumb.jpg.11c3acca3da3fada69c31b41f85df290.jpg

statistics-17-03-19-ant-oscar-parzon.thumb.jpg.c40a8e4d05afe66b4ce7ba92157f82cc.jpg

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Hi Everyone!,

Thank you for your interest. This can cause a lot of things. If the broadcast values(Frame drop or etc) and server values (CPU or Ram etc.) are healthy, 3 things that matter to us can be listed below.

- Adaptive Streaming Setting. Here is default Setting in below.

Resolution   Video Bitrate (Kbps)  Audio Bitrate (Kbps)
1080p              2000                           256
720p                1500                           128 
480p                1000                             75
360p                  800                             64
240p                  500                             32

These values changes some different cases. Because everyone's scenario is different, these values are not fixed.

-WebRTC Framerate Setting

Framerate is also a specific parameter. Framerate default parameter is 20. But as I sad above, this values changes your situation.

-Server Location

It is more stable to broadcast physically near servers.

If broadcast quality problems occur, lower these values and select the server close to where you broadcast, I hope your quality problem will go away. Good Luck!

I'm hoping that I was able to answer your queries. If you have any question, please feel free to ask us. Thanks.

Best Regards,
Selim
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Hello Selim, Thanks for your help.

For my case, a Bitrate of 2000Kbps is not an option. I'm looking for good quality.

Where does the limitation arise?
I imagine that in the WebRTCAppEE as an encoder.
I do not know Bitrate limitations in the WebRTC protocol.
The Streaming by RTMP enter and go well (test of reenvio to Youtube).

For my case we searched from the beginning, several months ago, with tests on servers from you in the EU and Canada, and our attempt at a VM in Google Cloud on South America, improve the arrival of data and achieve improvement through the tests. But, what you tell me now discourages us a lot. If I get well with data to the server, can not Ant give me more than 2000 Bitrate?

My need:
make a single audio and video in 1080p with a minimum of 3000Kbps (my ideal is 6000Kbps), and allow only ONE person see on my website that same quality shipment with less than 2s of latency. Basic, not many viewers or adaptive streaming.

Thanks for your help. I await your comments.
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Hi Everyone,

Let's remember the definition of WebRTC from its founders: 
"WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose."

As you may know, the main purpose of WebRTC is Real-Time Communication.
Image quality is an opponent power against real-time (ultra-low latency) communication.
So, there should be a break-even point for the balance of latency and image quality.
The optimum video speed with the current processor and communication platforms is 2500 Kbps.

There are some references to this issue:
- Blog from WebRTC Expert Tashi Levent Levi:  https://bloggeek.me/webrtc-vs-zoom-video-quality/
- Test results for the limits from webrtc-experiment.com 
        Maximum video bitrate on chrome is about 2Mb/s (i.e. 2000kbits/s).
        Minimum video bitrate on chrome is .05Mb/s (i.e. 50kbits/s).
        Starting video bitrate on chrome is .3Mb/s (i.e. 300kbits/s).

As a result, everyone needs to measure the best performant configuration of their infrastructure by changing them step-by-step.
My suggestions are as follows:
- 20 for FPS is optimum; however, 10 and 15 should be examined.
- 720p is good enough for video quality, especially for mobile platforms.
- 1000 Kbps is optimum for 720p, 750 Kbps is also acceptable when FPS is 10.

Have a nice ultra-low latency live streaming experience with WebRTC and sub-second expert Ant Media Server.

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