shipped voice features over both plain websockets and webrtc this year, some notes for anyone wiring real-time audio into a web app. websockets are great for prototypes. server-side everything, easy to debug, works everywhere. the pain shows up on real networks: packet loss and jitter turn into audio gaps, you end up hand-rolling buffering and timing recovery, and interruption detection feels late because user speech takes 300 to 500ms to arrive. webrtc earns its complexity the moment the product needs real barge-in. jitter buffers, echo cancellation, noise suppression and media-aware congestion control come built in, which sounds boring until you A/B it against raw websocket audio on a lossy connection. for the voice-agent project i used agora as the rtc layer instead of self-hosting TURN

  • 情报分类：技术价值
  • 命中依据：实时语音方案对比，实用技术经验
  • 来源：Reddit · SideProject
  • 原作者：/u/BruceWayne4607 https://www.reddit.com/user/BruceWayne4607
  • 发布时间：2026/9/10 19:28:42