RTCP (RTP Control Protocol)
RTCP is the companion protocol to RTP. It carries periodic reports on packet loss, jitter and timing so endpoints can monitor and adapt call quality.
Published 8 November 2026 · 1 min read · By SpaGreen Creative
Short answer
RTCP is the control protocol paired with RTP, defined in RFC 3550, that exchanges periodic sender and receiver reports about packet loss, jitter and timing for a media stream.
RTCP travels alongside every RTP stream. Where RTP carries the audio, RTCP carries information about how well the audio is arriving. It is defined in the same document as RTP, RFC 3550, section 6.
Each participant sends RTCP packets at intervals, usually every few seconds. Sender reports include how many packets and bytes have been sent and a timestamp that links RTP time to wall-clock time. Receiver reports include the fraction of packets lost since the last report, the cumulative loss, the highest sequence number received and an estimate of interarrival jitter. A source description packet carries a canonical name for the participant, and a BYE packet signals that a stream is ending.
By convention RTCP uses the UDP port one above the RTP port, so audio on 10000 has RTCP on 10001. Some systems multiplex both on one port instead.
Endpoints and PBX systems use RTCP to display call quality statistics and, in adaptive codecs such as Opus, to adjust bitrate when loss rises. RTCP also has to cross NAT, so the same STUN and ICE mechanisms that fix RTP apply to it. When SRTP is in use, RTCP is protected as SRTCP.
Sources and further reading
This article is published by SpaGreen Creative, the maker of SIPdia. Product names mentioned belong to their owners. Where it describes other software, treat the steps as general guidance and check the official documentation linked above.
