Live-Stream Sync Buffer: Multi-Contributor Jitter Model

Stream Setup

Contributors 3

Base network latency 80 ms

Network jitter (σ) 40 ms

Playout buffer delay D 220 ms

Live Sync State

…

Buffer health

100%

Glitches

Contributor sync spread

0 ms

How it works

Each contributor sends timestamped packets toward the shared mixer over an independent network path with its own latency. Every packet's one-way delay is drawn from a Gaussian around the base latency:

latency ~ N(baseLatency, jitter²)
arrival  = sendTime + latency
playout  = sendTime + D          (D = buffer delay)

The mixer commits to playing every contributor's packet at a fixed offset D after it was sent — that shared clock is what keeps a multi-source live stream synchronized. A packet that arrives before its playout time waits quietly in the buffer; one that arrives after its playout time has already passed is dropped, producing an audible/visual glitch:

if arrival ≤ playout:  packet is played on time
if arrival >  playout:  packet is dropped (glitch)

Contributors — how many independent camera/audio feeds are being mixed live.
Base latency & jitter σ — the network's average one-way delay and how much it wobbles packet to packet.
Buffer delay D — the safety margin the mixer waits before playing a packet. Raising D absorbs more jitter but adds real latency to the whole stream.
Sync spread — the gap between the fastest and slowest contributor's typical latency; D must cover the slowest one, so a mismatched network among collaborators forces everyone onto a higher shared delay.

A safe rule of thumb: D should exceed baseLatency + 3σ, covering roughly 99.9% of arrivals under the Gaussian model. This is the same buffering trade-off behind WebRTC, Twitch multi-guest streams, Discord Stage channels and any real-time collaborative broadcast tool.

Packet in flight Played on time Dropped (glitch)

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