Quoting Luke Curley

📰 Simon Willison's Blog

Learn how WebRTC prioritizes latency over audio quality in conference calls and how it affects user experience

intermediate Published 9 May 2026

Action Steps

Investigate WebRTC's packet loss and latency settings to understand their impact on audio quality
Configure WebRTC to prioritize audio quality over latency in scenarios where it's acceptable to introduce slight delays
Test the effects of different WebRTC settings on conference call audio quality
Compare the trade-offs between latency and audio quality in different real-time communication applications
Apply WebRTC's principles to other applications that require rapid back-and-forth communication

Who Needs to Know This

Developers and product managers working on real-time communication applications can benefit from understanding WebRTC's trade-offs between latency and audio quality

Key Insight

💡 WebRTC prioritizes latency over audio quality, which can result in distorted audio on conference calls

Key Takeaways

Learn how WebRTC prioritizes latency over audio quality in conference calls and how it affects user experience

Full Article

WebRTC is designed to degrade and drop my prompt during poor network conditions. wtf my dude WebRTC aggressively drops audio packets to keep latency low. If you’ve ever heard distorted audio on a conference call, that’s WebRTC baybee. The idea is that conference calls depend on rapid back-and-forth, so pausing to wait for audio is unacceptable. …but as a user, I would much rather wait an extra

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