SRTP

Secure VoIP and WebRTC Deployments Securing VoIP and WebRTC means protecting signaling, media, and the systems that run them. Calls travel across networks you don’t control, so a layered approach helps prevent eavesdropping, tampering, and abuse. Start with a simple security checklist and build it as your setup grows. Key areas to protect Signaling channels must be private and authenticated. Use TLS for SIP over TLS or WebSocket signaling with WSS. Enforce certificate validation and rotate certificates regularly. Media paths should be encrypted. Use SRTP or DTLS-SRTP for all audio and video. Disable unencrypted RTP and set strong cipher preferences. Identity and access matter. Require strong passwords, enable MFA for admin portals, and apply least-privilege roles for operators. Infrastructure security. Harden servers, patch software, and isolate VoIP components from the public internet when possible. Use a firewall to limit open ports and monitor for unusual activity. NAT traversal. WebRTC often uses STUN and TURN. Require authenticated TURN with TLS and restrict relay use to trusted networks. Secure signaling Signaling should be treated like a critical control plane. Use certificates, enforce mutual authentication where possible, and keep signaling servers behind a secure proxy with rate limits. Regularly review access to signaling endpoints and rotate keys on a fixed cadence. ...