Why WebRTC beats WebSockets for realtime voice AI (livekit.com)

Recent discussions in the AI/ML community highlight the advantages of WebRTC over WebSockets when developing real-time voice AI agents. While many developers initially opt for WebSockets due to their familiarity and existing integration in web stacks, the limitations of WebSockets in handling real-time audio become apparent in production environments. WebSockets rely on TCP, which ensures ordered delivery but introduces significant latency and head-of-line blocking that disrupts conversational flow. This results in unnatural pauses and delays during voice interactions, diminishing the user experience. In contrast, WebRTC was explicitly designed for seamless real-time media transmission. Utilizing UDP and RTP, WebRTC accommodates packet loss without stalling the audio stream, thereby maintaining a natural conversational rhythm. It includes built-in jitter buffers and media-aware congestion control that allow for smoother playback even under varying network conditions. Furthermore, the architecture of Selective Forwarding Units (SFUs) enhances scalability and efficiency by routing audio without the need for transcoding, enabling optimal performance across diverse network environments. This allows voice AI agents to deliver responsive interactions globally while managing quality and latency effectively, distinguishing WebRTC as the superior choice for real-time audio applications.