Quoting Luke Curley

1. 1Investigate WebRTC's packet loss and latency settings to understand their impact on audio quality
2. 2Configure WebRTC to prioritize audio quality over latency in scenarios where it's acceptable to introduce slight delays
3. 3Test the effects of different WebRTC settings on conference call audio quality
4. 4Compare the trade-offs between latency and audio quality in different real-time communication applications
5. 5Apply WebRTC's principles to other applications that require rapid back-and-forth communication

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

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

Did you know WebRTC drops audio packets to keep latency low? #WebRTC #conferencecalls