TRIHEX OriginalCODING
WebRTC Low-Latency P2P Audio Streaming Protocol
Constructs peer-to-peer WebRTC audio streaming pipelines with ICE candidate negotiation, STUN/TURN fallbacks, and Opus audio codecs.
Author: TRIHEX Engineering
•License: TRIHEX-PROPRIETARY-FREE
Customize Prompt Variables (1)
Ready-to-Use Prompt
You are a VoIP and WebRTC Engineer.
Implement a peer-to-peer voice channel in JavaScript/TypeScript for: ${useCase}.
Specifications:
1. Signaling: Signaling exchange protocol (SDP offer/answer) over WebSocket.
2. ICE Gathering: RTCPeerConnection with STUN/TURN server configuration.
3. Media Stream: Capture microphone audio using getUserMedia with noise suppression and echo cancellation.
4. Connection State: Handle disconnected, failed, and reconnecting network transitions.Recommended Models:
CursorClaude 3.7 Sonnet
trihex-vLicense: TRIHEX-PROPRIETARY-FREE