The Full Wiki

RTP Control Protocol: Wikis


Note: Many of our articles have direct quotes from sources you can cite, within the Wikipedia article! This article doesn't yet, but we're working on it! See more info or our list of citable articles.


From Wikipedia, the free encyclopedia

The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in the RTP specification RFC 3550,[1] superseding its original standardization in 1996 (RFC 1889).

RTCP provides out-of-band statistics and control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any media streams itself. Typically RTP will be sent on an even-numbered UDP port, with RTCP messages being sent over the next highest odd-numbered port[2]. The primary function of RTCP is to provide feedback on the quality of service (QoS) in media distribution by periodically sending statistics information to participants in a streaming multimedia session.

RTCP gathers statistics for a media connection and information such as transmitted octet and packet counts, lost packet counts, jitter, and round-trip delay time. An application may use this information to control quality of service parameters, perhaps by limiting flow, or using a different codec.

RTCP itself does not provide any flow encryption or authentication methods. Such mechanisms may be implemented, for example, with the Secure Real-time Transport Protocol (SRTP) defined in RFC 3711.

The Internet Protocol Suite
Application Layer
BGP · DHCP · DNS · FTP · GTP · HTTP · IMAP · IRC · Megaco · MGCP · NNTP · NTP · POP · RIP · RPC · RTP · RTSP · SDP · SIP · SMTP · SNMP · SOAP · SSH · Telnet · TLS/SSL · XMPP · (more)
Transport Layer
TCP · UDP · DCCP · SCTP · RSVP · ECN · OSPF · (more)
Internet Layer
IP (IPv4, IPv6) · ICMP · ICMPv6 · IGMP · IPsec · (more)
Link Layer
ARP/InARP · NDP · Tunnels (L2TP) · PPP · Media Access Control (Ethernet, DSL, ISDN, FDDI) · (more)


Protocol functions

RTCP provides three basic functions expected to be implemented in all RTP sessions:

  • The primary function of RTCP is to gather statistics on quality aspects of the media distribution during a session and transmit this data to the session media source and other session participants. Such information may be used by the source for adaptive media encoding (codec) and detection of transmission faults. If the session is carried over a multicast network, this permits non-intrusive session quality monitoring.
  • RTCP provides canonical end-point identifiers (CNAME) to all session participants. Although a source identifier (SSRC) of an RTP stream is expected to be unique, the instantaneous binding of source identifiers to end-points may change during a session. The CNAME establishes unique identification of end-points across an application instance (multiple use of media tools) and for third-party monitoring.
  • RTCP reports are expected to be sent by all participants, even in a multicast session which may involve thousands of recipients. Such traffic will increase proportionally with the number of participants. Thus, to avoid network congestion, the protocol must include session bandwidth management. This is achieved by dynamically controlling the frequency of report transmissions. RTCP bandwidth usage should generally not exceed 5% of total session bandwidth. Furthermore, 25% of the RTCP bandwidth should be reserved to media sources at all times, so that in large conferences new participants can receive the CNAME identifiers of the senders without excessive delay.

A fourth, optional feature, is the provisioning of session control functions, because RTCP is a convenient means to reach all session participants, whereas RTP itself is not. RTP is only transmitted by a media source.

Message types

RTCP distinguishes several types of packets: sender report, receiver report, source description, and bye. In addition, the protocol is extensible and allows application-specific RTCP packets. A standards-based extension of RTCP is the Extended Report packet type introduced by RFC 3611.[3]

Sender report (SR)
The sender report is sent periodically by the active senders in a conference to report transmission and reception statistics for all RTP packets sent during the interval. The sender report includes an absolute timestamp, which is the number of seconds elapsed since midnight on January 1, 1900. The absolute timestamp allows the receiver to synchronize RTP messages. It is particularly important when both audio and video are transmitted simultaneously, because audio and video streams use independent relative timestamps.
Receiver report (RR)
The receiver report is for passive participants, those that do not send RTP packets. The report informs the sender and other receivers about the quality of service.
Source description (SDES)
The Source Description message is used to send the CNAME item to session participants. It may also be used to provide additional information such as the name, e-mail address, telephone number, and address of the owner or controller of the source.
End of participation (BYE)
A source sends a BYE message to shut down a stream. It allows an end-point to announce that it is leaving the conference. Although other sources can detect the absence of a source, this message is a direct announcement. It is also useful to a media mixer.
Application-specific message (APP)
The application-specific message provides a mechanism to design application-specific extensions to the RTCP protocol.

Scalability in large deployments

In large-scale applications, such as in Internet Protocol Television (IPTV), very long delays (minutes to hours) between RTCP reports may occur, because of the RTCP bandwidth control mechanism required to control congestion (see #Protocol functions). Acceptable frequencies are usually less than one minute. This affords the potential of inappropriate reporting of the relevant statistics by the receiver or cause evaluation by the media sender to be inaccurate relative to the current state of the session. Methods have been introduced to alleviate the problems:[4] RTCP filtering, RTCP biasing and hierarchical aggregation.[5]


Hierarchical aggregation

Hierarchical aggregation (also known as RTCP feedback hierarchy) is an optimization of the RTCP feedback model and its aim is to shift the maximum number of users limit further together with QoS measurement.[6] It is used with Source-Specific Multicast where only a single source is allowed, such as in IPTV. Another type of multicast could be Any-Source Multicast, but it is not so suitable for large-scale applications with huge number of users.

As of 2007, only the most modern IPTV systems use hierarchical aggregation.[citation needed]

See also


  1. ^ RFC 3550, RTP: A Transport Protocol for Real-Time Applications, H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, The Internet Society (July 2003)
  2. ^ RFC 3605, Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP), C. Huitema, Microsoft (October 2003)
  3. ^ RFC 3611, RTP Control Protocol Extended Reports (RTCP XR), T. Friedman (Ed.), R. Caceres, A. Clark (Ed.), The Internet Society (November 2003)
  4. ^ Vít Novotný, Dan Komosný, Large-Scale RTCP Feedback Optimization, Journal of Networks, Vol.3 (3), March 2008
  5. ^ Realtime control protocol and its improvements for Internet Protocol Television
  6. ^ Vít Novotný, Dan Komosný, Tree Structure for Specific-Source Multicast with Feedback Aggregation, ICN07 - The Sixth International Conference on Networking, Martinique, 2007, ISBN 0-7695-2805-8

Additional reading


Got something to say? Make a comment.
Your name
Your email address