Building Low-Latency WebRTC Video Signaling Infrastructure

During an emergency, a live camera stream from the victim's device provides safety guardians with critical visual context. Since mobile networks in panic scenarios are often highly congested, standard streaming formats (like HLS or RTMP) introduce latency ranges (3 to 15 seconds) that are too high. We require real-time, sub-second latency.

This is why we leverage **WebRTC** (Web Real-Time Communication).

The Signaling Server Architecture WebRTC requires a signaling step to exchange Session Description Protocol (SDP) configurations and Interactive Connectivity Establishment (ICE) candidates. We built our signaling channel using Express, Socket.IO, and Redis. 1. The Android client emits an `emergency-stream-start` WebSocket payload. 2. The server creates a unique room and forwards the SDP offer to authenticated guardians. 3. Once a peer handshake is established, the media flows directly between the device and the guardian browser.

Handling Cellular NAT Firewalls with COTURN In over 65% of mobile cellular networks, symmetric NAT firewalls prevent direct peer-to-peer connections. To solve this: - We deploy a globally distributed network of **COTURN** relay servers. - When direct connection fails, media packets are securely routed through our nearest TURN node. - All media packets are encrypted using **SRTP** (Secure Real-time Transport Protocol), ensuring that even intermediate TURN servers cannot decrypt the stream.

This balance of high-speed signaling and robust media relay ensures live streams work even on remote highways and inside dense concrete structures.