Adult Cam Site Live Streaming Hosting: WebRTC and RTMP Infrastructure

Choosing the right live streaming architecture for your cam site:

WebRTC (peer-to-peer). WebRTC enables direct browser-to-browser video connection, reducing server bandwidth requirements for small rooms. Latency under 200ms is achievable. Works natively in all modern browsers without plugins. Requires STUN/TURN server infrastructure for NAT traversal. Does not scale beyond ~50 concurrent viewers per room without Selective Forwarding Units (SFUs) like Janus, mediasoup, or Livekit.

RTMP/HLS (server-side). Traditional cam site architecture. Performers broadcast via RTMP (OBS, XSplit, browser-based encoders). Server transcodes and delivers to viewers via HLS or WebRTC from server. Higher server cost but scales to unlimited viewers per room. 5-30 second latency with HLS, 1-3 seconds with Low-Latency HLS or WebRTC delivery from server.

Recommended approach. Use WebRTC for private shows (2-5 participants, maximum interactivity) and HLS for public rooms (unlimited viewers, acceptable 5-10 second delay). This hybrid architecture gives the best user experience at reasonable server cost.

TURN server requirements:

• 1 CPU core can handle approximately 1,000 TURN relay connections
• 8GB RAM handles approximately 5,000 concurrent connections
• Network bandwidth: each WebRTC stream uses 0.5-2 Mbps. 1,000 concurrent sessions = 500Mbps-2Gbps

SFU (Selective Forwarding Unit) requirements for scaling WebRTC:

• Each Janus/mediasoup server instance handles 200-500 concurrent rooms depending on viewer count
• Horizontal scaling requires session affinity or distributed session management
• CPU allocation: 1 core per 20-50 concurrent rooms depending on video quality

For a cam platform with 1,000 concurrent performers: dedicated server with 32+ cores, 64GB RAM, 10 Gbps unmetered port. Romania or Iceland based for EU latency.