Looking for real-world WebRTC experiences

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Hi everyone!

I’m working on a privacy-focused peer-to-peer messenger built on WebRTC, and I’m researching how WebRTC behaves in real-world conditions.

Rather than benchmarks or lab tests, I’m interested in hearing about your actual experience using WebRTC-based applications (Signal, Element, Jitsi, PeerTube, Brave Talk, browser P2P apps, or any others).

Some questions I’d love to hear your thoughts on:

  • Have you experienced random disconnects during calls or chats?
  • Do connections fail more often on mobile networks, public Wi-Fi, or behind strict firewalls?
  • Have you noticed problems when switching between Wi-Fi and cellular data?
  • Do you frequently end up using relay servers instead of direct P2P connections?
  • Have you encountered NAT or firewall issues that made WebRTC unusable?
  • Have you ever had a WebRTC application work perfectly for one person but fail completely for another?
  • Are there any recurring issues that you think developers tend to overlook?

If you’re a developer, I’d also be interested in hearing about the most difficult networking problems you’ve encountered while building or maintaining WebRTC applications.

I’m especially interested in reliability under poor or restrictive network conditions, since one of my goals is to improve connection resilience while preserving privacy.

Thanks in advance to everyone willing to share their experiences!

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I think overall, WebRTC works reasonably well and is quite reliable. It’s technology, though and technology will fail. I had issues in larger videoconferences with people on a super slow and unreliable connections. They’d intermittently drop out, pop back in again etc. And I had issues with people (me) using a Libre version of Firefox and some codecs weren’t supported. I also had issues with people having their microphone set to weird sound devices.

Other than that, I generally had a good time with WebRTC. Especially the 1:1 direct peer calls. They’re awesome and generally well supported. Peertube etc also work flawlessly here.

I guess 80% of the experience depends on how you implement it. And what code you write to handle edge cases like a poor internet connection. Or people who are bad with computers and can’t figure out why their microphone doesn’t work.

And if you’re looking for a more bleeding edge Web API and data transport channel… I recently learned about WebTransport. It’s a W3C draft for the more recent HTTP versions.

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