RTPRTCP

RTPRTCP refers to the combined use of Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) in the transport and monitoring of real-time multimedia over IP networks. RTP provides end-to-end delivery of real-time data such as audio and video, with features like payload type identification, sequence numbering, and timestamps, but it does not guarantee delivery.

RTCP operates alongside RTP to provide control and feedback. It allows participants to monitor quality, synchronize multiple streams, and manage session state. RTCP packets include Sender Reports, which carry timing information (NTP and RTP timestamps) and counts of packets and octets sent; Receiver Reports, which convey statistics on packet loss, jitter, and timing; Source Description items (SDES), such as a canonical name (CNAME); and application-defined APP packets for custom data.

In practice, RTP and RTCP share the same transport session and often the same ports. RTCP typically uses a small portion of the session bandwidth—often around 5%—to minimize impact on media throughput, though this can vary with session size and network conditions. RTCP reports provide ongoing quality metrics that enable adaptive streaming, congestion management, and synchronization of multiple media streams (for example, audio and video) within a session.

Security considerations for RTPRTCP include protection through SRTP (Secure Real-time Transport Protocol), which provides confidentiality and integrity for both RTP and RTCP packets. Standards and related extensions are described primarily in RFC 3550 (RTP and RTCP), with additional RFCs addressing payload types, security, and interoperability.

Limitations include additional protocol overhead and potential reporting cadence issues in large or low-bandwidth sessions. Nevertheless, RTPRTCP remains a foundational framework for real-time media delivery in applications such as VoIP, video conferencing, and streaming.