What is SRTP?

SRTP, defined by RFC 3711, secures real-time RTP audio and video by encrypting media payloads, authenticating packets, and blocking replay attacks while generally leaving RTP headers and traffic metadata visible. WebRTC mandates SRTP through DTLS-SRTP, which derives media keys on the media path and prevents signalling servers from accessing them, whereas SIP deployments may use SDES, where keys travel in SDP and require protected signalling. Common SRTP suites use AES-based encryption with authentication tags, with WebRTC required to support AES-128 counter mode with HMAC-SHA1 and encouraged to prefer AES-GCM where available. SRTP protects media in transit but does not inherently provide end-to-end encryption when a media server, such as an SFU, terminates sessions and holds keys to forward, record, transcribe, or stream calls. True group-call end-to-end encryption requires an additional frame-level layer, such as SFrame or PERC, that keeps media keys from the server, though this limits server-side media processing. SRTP has no fixed port and operates over RTP’s dynamically negotiated network paths, while related protections such as TLS and DTLS secure signalling and data channels.

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