RTCP stands for Real-time Transport Control Protocol. It is defined in IETF RFC 3550.
RTCP is used alongside RTP, offering a lightweight control mechanism for sending statistic reports and flow control messages. These enable the receiver to provide feedback to the sender, who can then deduce the network’s status and adjust (by changing bitrate or adding FEC).
RTCP message types
The most important RTCP message types in WebRTC include:
- Sender Report (SR): Statistics from the media sender (bytes sent, packets sent, timestamps)
- Receiver Report (RR): Feedback from the receiver (packet loss rate, jitter, RTT)
- NACK: Negative acknowledgement requesting retransmission of lost packets
- PLI / FIR: Requests for a new keyframe
- REMB: Receiver Estimated Maximum Bitrate feedback
- Transport-CC: Transport-wide congestion control feedback
- RRTR: Receiver Reference Time Report for RTT calculation
RTCP in WebRTC
In WebRTC, RTCP is multiplexed on the same port as RTP (RTCP-MUX) and encrypted as SRTCP. RTCP feedback is essential for BWE (Bandwidth Estimation), enabling the sender to adapt its encoding in real time.
