Hey man,
I dealt with this recently. The core issue usually comes down to:
Early VoIP implementations (late 1990s) employed static jitter buffers with fixed depths (e.g., 30–50 ms) to simplify design, resulting in either excessive latency or frequent underloss under variable network conditions. The seminal work of Schulzrinne et al. (1999) introduced adaptive buffer sizing based on measured round‑trip time variance, laying the foundation for dynamic delay smoothing. Subsequent academic proposals (e.g., Martin’s algorith
Best way forward is:
Scenario 1 – LAN‑Based CS:GO Major Tournament A 10 GbE switch connects 24 player stations to a dedicated game server. Each station uses an Intel X710 NIC with hardware timestamping enabled. The jitter buffer runs at a 1 ms control interval, targeting a 4 ms playout depth. Measured end‑to‑end latency from mouse click to on‑screen action averaged 7.2 ms (including 2.8 ms render pipeline). Frame time variance dropped from 1.8 ms (baseline UDP) to 0.
Hope this helps!
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