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Can the maximum bitrate of a webRTC publish be set higher?


Drew Carroll
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I'm hoping to use an Antmedia server to distribute low-latency webRTC multiviews to our production crews. It seems the default maximum bitrate on the publish page is 2500Kbps. I'm struggling to find a way to increase that. Is it possible to increase that maximum bitrate to somewhere in the range of 5-8Mbps?

Thanks!
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Hi Drew,

Have a good day. You can use "unlimited" parameter for your requirements without any code changes. If you want to increase max bitrate parameter (2500), you need to change some codes in index.html max_bandwidth_apply function. But we don't recommend that. It goes against the nature of WebRTC. 

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).

I hope, I could help you.

Best Regards,
Selim

 

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Hey Selim

Thanks for the assistance! This was definitely the help I was looking for. For a little more context, my use case assumes that both the publishing device and playback device are both quite robust PC's. I don't intend to have anyone watching these streams on a mobile device, rarely even on a laptop. Adjusting the bitrate to 6000kbps has given me the kind of video quality I'm looking for.

 

However, even though I'm publishing and playing on the same device, which is a high-performance PC connected directly to gigabit fiber internet, I am seeing some amount of dropped frames. Since the frames are being "received" but not "decoded" does that mean there is some issue with the decoding process? I'm also getting a "weak network performance" notification every once in a while even though I'm confident that is not the case. Are there some avenues I could follow to troubleshoot this?

Thanks again!
-Drew

 

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Hi Drew,

Have a good day. Thanks for the details.

Are your devices connected via wireless or cable? Because wireless network is not convenient for streaming. Wireless networks cannot prioritize for packaging. Please use an ethernet cable for streaming. Also for the network quality, It's not something like having a fiber, 4G network. The QoS is affected by the whole path. I mean if a switch or router in your path to the server instance, you'll be affected for that and your quality decreases, jitter increases. 

Also could you please check this blog post?

I hope, it's clear.

Best Regards,
Selim

 

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