RTP and playout delay compensation Henning Schulzrinne Dept. of Computer Science Columbia University Fall 2003

RTP packet header |V=2|P|X| CC |M| PT | sequence number | | timestamp | | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | contributing source (CSRC) identifiers | |.... |

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

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

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 = s*b + 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

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 < 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

Playout delay time playout delay late = lost packet jitter

Playout buffer 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 = t*b + 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) silence Usually, best to think of each talkspurt as an independently schedulable unit p = p 0 + (t – t 0 ) * b t 0 = timestamp for first packet in talkspurt p 0 = position for first packet in talkspurt

Playout buffer, cont’d. 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 t=100 t=140 play insert t 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