Real-Time Message Jitter Buffer

Network Conditions

Send rate 6 msg/s

Network jitter σ 80 ms

Packet loss 3%

Playout buffer delay 220 ms

Delivery Stats

On-time delivery

—

Buffer occupancy

Dropped (too late)

Lost in network

Avg jitter absorbed

0 ms

Delivered total

In flight (network)

Waiting in playout buffer

Delivered in order

Lost / dropped as late

How it works

Real-time mobile channels (WebSocket, SSE, push streams) deliver messages over an unreliable network: each one takes a different amount of time to arrive, so they can reach the phone out of the order they were sent. A jitter buffer is the standard fix used by VoIP, video calls and live chat clients — it holds arrivals briefly so they can be replayed in the correct sequence instead of showing them the instant each one lands.

on first arrival (seq s0, arrival A0):
  playoutBase = A0 - s0·Δt + bufferDelay

target_playout(seq) = playoutBase + seq·Δt

at every scheduling tick:
  while now ≥ target_playout(nextExpected):
    if buffer has nextExpected: deliver it, nextExpected++
    else: mark DROPPED (too late), nextExpected++  ← gap skip

The schedule is fixed the moment the very first message arrives, using the round-trip send interval Δt = 1 / send-rate. Every later message has a target playout time on that same clock. Increase the buffer delay and more out-of-order arrivals make their deadline (fewer red drops) at the cost of extra latency before anything is shown; shrink it and the app feels snappier but jittery messages get skipped as gaps.

Send rate / jitter σ / loss — shape the network: how often messages are sent, how much their travel time varies (Gaussian spread), and what fraction never arrive at all.
Playout buffer delay — the deliberate lag the receiver adds before replaying, trading responsiveness for order-correctness.
A message that finally lands after its playout deadline has already passed is discarded on arrival — real-time delivery never rewinds to show something late.

source → network → playout buffer → delivered

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