RTP (Real-time Transport Protocol)

RTP (Real-time Transport Protocol) is a protocol that delivers audio and video in packets that carry a sequence number and a timestamp. These two fields let the receiver detect lost or reordered packets and play the media at the correct pace. It is almost always sent over UDP.

Why it matters

Live audio is useless if it arrives late, so retransmitting lost packets the way TCP does would make things worse. RTP accepts occasional loss and leaves recovery to the application, for example by concealing a missing 20 ms of audio. It is the media carrier for VoIP, video conferencing and WebRTC. Its companion protocol RTCP reports loss, delay and jitter back to the sender.

Example

A phone call using the Opus or G.711 codec sends an RTP packet every 20 ms, so 50 packets per second. If packet 1041 arrives and then 1043, the receiver knows 1042 is missing and can hide the gap. A jitter buffer of 40 to 60 ms smooths out uneven arrival times.

Common misconceptions

• RTP does not set up calls; that is the job of SIP or another signaling protocol.
• It does not guarantee delivery or quality by itself.
• Plain RTP is unencrypted; the secure variant is SRTP.

Related

• VoIP
• Jitter
• UDP
• Packet loss

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