rtsp协议相关之-rfc1889(RTP 实时应用传送协议文档).txt[28]

[入库:2006年2月23日] [更新:2007年3月24日]

本文简介:

   interarrival jitter: 32 bits
        An estimate of the statistical variance of the RTP data packet
        interarrival time, measured in timestamp units and expressed as
        an unsigned integer. The interarrival jitter J is defined to be
        the mean deviation (smoothed absolute value) of the difference D
        in packet spacing at the receiver compared to the sender for a
        pair of packets. As shown in the equation below, this is
        equivalent to the difference in the "relative transit time" for
        the two packets; the relative transit time is the difference
        between a packet's RTP timestamp and the receiver's clock at the
        time of arrival, measured in the same units.

Schulzrinne, et al          Standards Track                    [Page 26]

RFC 1889                          RTP                       January 1996


   If Si is the RTP timestamp from packet i, and Ri is the time of
   arrival in RTP timestamp units for packet i, then for two packets i
   and j, D may be expressed as

                 D(i,j)=(Rj-Ri)-(Sj-Si)=(Rj-Sj)-(Ri-Si)

   The interarrival jitter is calculated continuously as each data
   packet i is received from source SSRC_n, using this difference D for
   that packet and the previous packet i-1 in order of arrival (not
   necessarily in sequence), according to the formula

                    J=J+(|D(i-1,i)|-J)/16

   Whenever a reception report is issued, the current value of J is
   sampled.

   The jitter calculation is prescribed here to allow profile-
   independent monitors to make valid interpretations of reports coming
   from different implementations. This algorithm is the optimal first-
   order estimator and the gain parameter 1/16 gives a good noise
   reduction ratio while maintaining a reasonable rate of convergence
   [11].  A sample implementation is shown in Appendix A.8.

   last SR timestamp (LSR): 32 bits
        The middle 32 bits out of 64 in the NTP timestamp (as explained
        in Section 4) received as part of the most recent RTCP sender
        report (SR) packet from source SSRC_n.  If no SR has been
        received yet, the field is set to zero.

   delay since last SR (DLSR): 32 bits
        The delay, expressed in units of 1/65536 seconds, between
        receiving the last SR packet from source SSRC_n and sending this
        reception report block.  If no SR packet has been received yet
        from SSRC_n, the DLSR field is set to zero.

   Let SSRC_r denote the receiver issuing this receiver report. Source
   SSRC_n can compute the round propagation delay to SSRC_r by recording
   the time A when this reception report block is received.  It
   calculates the total round-trip time A-LSR using the last SR
   timestamp (LSR) field, and then subtracting this field to leave the
   round-trip propagation delay as (A- LSR - DLSR).  This is illustrated
   in Fig. 2.

   This may be used as an approximate measure of distance to cluster
   receivers, although some links have very asymmetric delays.

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

本站最佳浏览方式为 分辨率 1024x768 IE 6.0(或更高版本的 IE浏览器)

go top