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6.2 RTCP Transmission Interval Connected: An Internet Encyclopedia
6.2 RTCP Transmission Interval

Up: Connected: An Internet Encyclopedia
Up: Requests For Comments
Up: RFC 1889
Up: 6. RTP Control Protocol -- RTCP
Prev: 6.1 RTCP Packet Format
Next: 6.2.1 Maintaining the number of session members

6.2 RTCP Transmission Interval

6.2 RTCP Transmission Interval

   if encrypted: random 32-bit integer
    |
    |[------- packet -------][----------- packet -----------][-packet-]
    |
    |             receiver reports          chunk        chunk
    V                                    item  item     item  item
   --------------------------------------------------------------------
   |R[SR|# sender #site#site][SDES|# CNAME PHONE |#CNAME LOC][BYE##why]
   |R[  |# report #  1 #  2 ][    |#             |#         ][   ##   ]
   |R[  |#        #    #    ][    |#             |#         ][   ##   ]
   |R[  |#        #    #    ][    |#             |#         ][   ##   ]
   --------------------------------------------------------------------
   |<------------------  UDP packet (compound packet) --------------->|


   #: SSRC/CSRC

              Figure 1: Example of an RTCP compound packet

RTP is designed to allow an application to scale automatically over session sizes ranging from a few participants to thousands. For example, in an audio conference the data traffic is inherently self- limiting because only one or two people will speak at a time, so with multicast distribution the data rate on any given link remains relatively constant independent of the number of participants. However, the control traffic is not self-limiting. If the reception reports from each participant were sent at a constant rate, the control traffic would grow linearly with the number of participants. Therefore, the rate must be scaled down.

For each session, it is assumed that the data traffic is subject to an aggregate limit called the "session bandwidth" to be divided among the participants. This bandwidth might be reserved and the limit enforced by the network, or it might just be a reasonable share. The session bandwidth may be chosen based or some cost or a priori knowledge of the available network bandwidth for the session. It is somewhat independent of the media encoding, but the encoding choice may be limited by the session bandwidth. The session bandwidth parameter is expected to be supplied by a session management application when it invokes a media application, but media applications may also set a default based on the single-sender data bandwidth for the encoding selected for the session. The application may also enforce bandwidth limits based on multicast scope rules or other criteria.

Bandwidth calculations for control and data traffic include lower- layer transport and network protocols (e.g., UDP and IP) since that is what the resource reservation system would need to know. The application can also be expected to know which of these protocols are in use. Link level headers are not included in the calculation since the packet will be encapsulated with different link level headers as it travels.

The control traffic should be limited to a small and known fraction of the session bandwidth: small so that the primary function of the transport protocol to carry data is not impaired; known so that the control traffic can be included in the bandwidth specification given to a resource reservation protocol, and so that each participant can independently calculate its share. It is suggested that the fraction of the session bandwidth allocated to RTCP be fixed at 5%. While the value of this and other constants in the interval calculation is not critical, all participants in the session must use the same values so the same interval will be calculated. Therefore, these constants should be fixed for a particular profile.

The algorithm described in Appendix A.7 was designed to meet the goals outlined above. It calculates the interval between sending compound RTCP packets to divide the allowed control traffic bandwidth among the participants. This allows an application to provide fast response for small sessions where, for example, identification of all participants is important, yet automatically adapt to large sessions. The algorithm incorporates the following characteristics:

  • Senders are collectively allocated at least 1/4 of the control traffic bandwidth so that in sessions with a large number of receivers but a small number of senders, newly joining participants will more quickly receive the CNAME for the sending sites.

  • The calculated interval between RTCP packets is required to be greater than a minimum of 5 seconds to avoid having bursts of RTCP packets exceed the allowed bandwidth when the number of participants is small and the traffic isn't smoothed according to the law of large numbers.

  • The interval between RTCP packets is varied randomly over the range [0.5,1.5] times the calculated interval to avoid unintended synchronization of all participants [10]. The first RTCP packet sent after joining a session is also delayed by a random variation of half the minimum RTCP interval in case the application is started at multiple sites simultaneously, for example as initiated by a session announcement.

  • A dynamic estimate of the average compound RTCP packet size is calculated, including all those received and sent, to automatically adapt to changes in the amount of control information carried.

This algorithm may be used for sessions in which all participants are allowed to send. In that case, the session bandwidth parameter is the product of the individual sender's bandwidth times the number of participants, and the RTCP bandwidth is 5% of that.


Next: 6.2.1 Maintaining the number of session members

Connected: An Internet Encyclopedia
6.2 RTCP Transmission Interval

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