Securing WebRTC-to-SIP when using FreeSWITCH

Securing WebRTC when using FreeSWITCH involves multiple layers. Here's a step-by-step guide to ensure your WebRTC communications through FreeSWITCH are secure:

1. Transport Layer Security (TLS) for SIP:
   • This ensures that the SIP traffic between your WebRTC client and FreeSWITCH is encrypted.
   • Modify your SIP profile (often located in /etc/freeswitch/sip_profiles/) to use wss (WebSockets over TLS).
   • Use strong, up-to-date ciphers and ensure your TLS certificates are valid and haven't expired.
2. Secure RTP (SRTP):
   • This ensures that the media streams (audio and video) are encrypted.
   • In FreeSWITCH's configuration, ensure that both the rtp_secure_media and rtp_secure_media_outbound are set to enforce encrypted RTP.
3. WebSockets over TLS (WSS):
   • If you're using WebSockets for signaling (e.g., with SIP over WebSockets), make sure you're using WSS, the secure version.
   • This often involves configuring an SSL certificate and key in the WebSocket profile.
4. Firewall Configuration:
   • Restrict which IP addresses can access your FreeSWITCH server.
   • Only open necessary ports (e.g., 5066 for WSS, the RTP port range, etc.).
5. Strong Authentication:
   • Always require authentication for SIP registrations.
   • Use strong passwords and consider rotating them periodically.
6. Keep FreeSWITCH Updated:
   • Always run the latest version of FreeSWITCH.
   • Security vulnerabilities are discovered over time in all software, and keeping FreeSWITCH updated ensures you benefit from the latest security patches.
7. Secure the Verto Communicator:
   • If you're using the Verto Communicator with FreeSWITCH for WebRTC, ensure it's served over HTTPS.
   • Also, keep Verto updated to the latest version.
8. Use a Secure STUN/TURN Server:
   • If you're using a STUN/TURN server for NAT traversal, make sure it's secure. TURN servers, in particular, can be abused as open relays if not properly secured.
   • Use authentication for the TURN server and periodically rotate credentials.
9. Monitor and Audit:
   • Regularly check the server logs for any suspicious activity.
   • Consider setting up alerts for abnormal patterns, such as multiple failed registration attempts.
10. Backup Regularly:
    • Keep regular backups of your configuration and be ready for disaster recovery.
11. Educate Users:
    • If you're providing a service, educate your users about the importance of securing their endpoints, using strong passwords, and being wary of phishing attempts.

Lastly, always stay updated on the latest security recommendations and best practices in the realm of WebRTC and VoIP. The landscape is always evolving, and what's considered secure today might not be tomorrow.