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RTP Control Protocol : ウィキペディア英語版
RTP Control Protocol

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 RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data, but does not transport any media data itself. 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 transports statistics for a media connection and information such as transmitted octet and packet counts, packet loss, packet delay variation, 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.
==Protocol functions==
Typically RTP will be sent on an even-numbered UDP port, with RTCP messages being sent over the next higher odd-numbered port.〔RFC 3605, ''Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)'', C. Huitema, Microsoft (October 2003)〕
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.
RTCP provides 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.
*Provisioning of session control functions. RTCP is a convenient means to reach all session participants, whereas RTP itself is not. RTP is only transmitted by a media source.
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.
The RTCP reporting interval is randomized to prevent unintended synchronization of reporting. The recommended minimum RTCP report interval per station is 5 seconds. Stations should not transmit RTCP reports more often than once every 5 seconds.

抄文引用元・出典: フリー百科事典『 ウィキペディア(Wikipedia)
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