RTCP

RTCP, or Real-Time Control Protocol, is a companion protocol to the Real-Time Transport Protocol (RTP) used for delivering audio and video over IP networks. Defined in RFC 3550, RTCP carries control and feedback information about RTP streams, enabling participants to monitor delivery quality, synchronize multiple media streams, and identify sources within a session.

RTCP defines several packet types. Sender Report (SR) packets are issued by active transmitters and include wall-clock timestamps and RTP timestamps to aid synchronization. Receiver Report (RR) packets are sent by receivers and report reception quality metrics such as packet loss, jitter, and transit delay. Source Description (SDES) packets carry metadata about participants, including a CNAME that uniquely identifies a source across the session. APP packets are application-defined, allowing custom control data. A BYE packet signals that a source is leaving the session.

CNAME and synchronization are central to RTCP's function. The CNAME provides a stable identifier for a source across RTCP reports, facilitating correlation between RTP streams and their control information. RR and SR packets supply data used by receivers and senders to assess network performance and adjust transmission behavior if needed.

Bandwidth, timing, and reporting habits are managed to minimize overhead. RTCP traffic is kept small relative to the media stream, typically a few percent of the session bandwidth, and is shared between senders and receivers. The RTCP reporting interval is dynamically adjusted based on the number of participants and the allocated RTCP bandwidth, becoming less frequent as session size grows. This mechanism supports timely quality feedback without excessive control traffic.

Security considerations: RTCP itself does not provide encryption. In secure deployments, RTCP is carried alongside SRTP (Secure Real-time Transport Protocol) to deliver confidentiality and integrity for media and control data.