rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt[20]

[入库:2006年2月23日] [更新:2007年3月24日]

本文简介:

Schulzrinne, et al          Standards Track                    [Page 18]

RFC 1889                          RTP                       January 1996


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.

本文关键:rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt
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