Stream video and audio to browser via WebRTC with low latency

Use This When

• Building live monitoring dashboards for surveillance or analytics
• Creating interactive demos where users see real-time processing results
• Implementing remote inspection tools accessed via web browser
• Developing customer-facing applications needing low-latency video streaming

What It Does

• Publishes WebRTC stream from pipeline video/audio tracks to browser clients
• Manages WebSocket signaling for peer connection establishment
• Converts incoming frames to I420 format for WebRTC encoding
• Supports real-time mode (drop old frames) or non-real-time (process all frames)
• Supports optional local WebRTC UDP port range constraints via webrtc_port_range

Works Best With

• input-browser-webcam → detectors → visualize → this component for interactive analytics
• Camera feeds → this component for live monitoring dashboards
• Integration with detect-objects or track-object for real-time visualization
• Voice assistant pipelines combining video and audio in browser interface

Caveats

• Network latency and jitter affect streaming quality; WebRTC adapts but not magical
• Real-time mode prioritizes immediacy and drops frames; non-real-time ensures all frames processed
• Requires HTTPS in production for browser security policies
• Multiple concurrent browser clients multiply encoding load; GPU encoding recommended