RTP — Real-time Transport Protocol
What is RTP?
RTP (Real-time Transport Protocol) is the internet standard for delivering real-time audio and video data — the protocol that actually carries a VoIP call's voice packets across the network once SIP has set up the call. RTP is typically paired with SRTP (Secure RTP) for encryption and RTCP for carrying quality-monitoring feedback like jitter and packet-loss statistics back between the two endpoints of a call.
More detail
SIP handles call setup and teardown signaling, while RTP carries the actual audio media once the call is connected — the two protocols work together but do different jobs.
SRTP adds encryption to RTP's media stream, which matters for any call carrying sensitive conversation content over a network segment, including the public internet.
Frequently asked
- What's the difference between SIP and RTP?
- SIP is the signaling protocol that sets up, modifies, and tears down a call; RTP is the media protocol that actually carries the voice (or video) audio once the call is connected — a call uses both together, for different purposes.
- Is RTP traffic encrypted by default?
- Plain RTP is not encrypted; SRTP (Secure RTP) adds encryption and is what a security-conscious VoIP deployment uses instead, especially for calls crossing the public internet rather than a private, already-trusted network.
See also
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