RTP and playout delay compensation

1
RTP and playout delay compensation

• Henning Schulzrinne
• Dept. of Computer Science
• Columbia University
• Fall 2003

2
RTP packet header
0 1 2
3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7
8 9 0 1 2 3 4 5 6 7 8 9 0 1 -----------
---------------------
V2PX CC M PT
sequence number ------------
--------------------
timestamp
---------------------
-----------
synchronization source (SSRC) identifier

contributing
source (CSRC) identifiers
....
-----------------------
---------
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RTP timestamp

• Timestamp measured in sample units
• reflects nominal sampling time of first sample in
  packet
• e.g., 20 ms block size of 8,000 Hz audio ? 160
  timestamp units per packet
• always 90 kHz for video
• e.g., 3000 timestamp units per packet for 30 fps
• 3600 for 25 fps
• 3750 for 24 fps
• even if real system clock is slower or faster
• note 32 bit integer ? may wrap around
• if start at 0, after about 6 days for audio, ½
  day for video
• but starting value is supposed to be random

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RTP sequence number

• Counts packets actually sent
• Wraps around much quicker
• e.g., for 20 ms packets, in about 22 minutes
• Also uses random starting value

5
RTP timestamp vs. sequence number

• Related, but different purposes
• timestamp for timing reconstruction
• playout delay compensation (later)
• synchronization with other sources (later)
• sequence number for loss measurements and gap
  detection
• t sb c
• where t timestamp
• s sample units per packet
• offset c is constant within a talkspurt, but
  changes after each talkspurt or after
  transmission gap

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Playout delay

• Converts variable network delay (jitter) into
  fixed delay
• thus, end-to-end delay is max(jitter)
  propagation delay
• or, if willing to tolerate some late packets
• delay lt 95 of jitter propagation delay
• Propagation delay is invisible
• and hard to measure without synchronized clocks
• about 5 ms/1000 km one way
• Total delay should be less than 150 ms one-way
• End-to-end delay must remain constant within a
  talkspurt
• otherwise gaps

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Playout delay
playout delay
packet jitter
late lost
time
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Playout buffer
silence

• Logically infinite buffer
• Implemented as circular buffer, with wrap
  around
• Takes care of jitter and re-ordering based on RTP
  timestamp t
• Playout point p tb c
• p buffer position, measured in samples
  (typically, 16 bits if decoding is done before
  playout)
• b buffer positions per sample (usually, 1)
• c offset

decoder (G.729 ? L16)

• Usually, best to think of each talkspurt as an
  independently schedulable unit
• p p0 (t t0) b
• t0 timestamp for first packet in talkspurt
• p0 position for first packet in talkspurt

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Playout buffer, contd.

• Thus, hard part is computing insertion point for
  first packet in talkspurt
• Trying to predict future
• late loss vs. excessive delay
• Conceptually, two approaches
• look at current playout point when first packet
  arrives
• then, leave some margin of error
• may be too conservative
• compute based on last talkspurt and change c
• avoids overestimation due to slow first packet
• deals less well with jumps in delay after long
  pauses

insert
play
t
t140
t100

• Simple method assume roughly normal distribution
  and take n times the variance of the delay (
  jitter)
• this becomes the extra delay
• Other mechanisms
• spike detection
• optimal value for last talkspurt