VoIP Bandwidth Calculator

Every RTP packet carries the same fixed header stack regardless of codec — 18 bytes of Ethernet, 20 bytes of IP, 8 bytes of UDP, and 12 bytes of RTP — so a low-bitrate codec like G.729 has a much smaller voice payload per packet relative to that fixed overhead than G.711 does. The calculator's overhead percentage output makes this visible directly: the same header bytes eat a larger share of a small packet than a large one.

The calculator applies a flat 35% reduction to effective per-call bandwidth when VAD is enabled, reflecting the well-established average that roughly a third of a two-way conversation is silence during which no audio needs to be transmitted. Actual savings on a real deployment will vary with call patterns and how aggressively the VAD algorithm detects silence, so 35% is a planning average rather than a guarantee for any specific call.

A longer ptime means fewer, larger packets per second — packets per second is 1000 divided by ptime in milliseconds — so the same fixed per-packet header overhead is paid less often. The tradeoff is that larger ptime values also add more end-to-end latency to the call, since more audio has to be buffered before each packet can be sent, which is why 20ms is the standard default balancing overhead against delay.

It's added on top of the raw calculated bandwidth to reserve extra capacity for jitter buffers and traffic bursts, since real network links rarely see perfectly smooth, evenly-spaced VoIP traffic. The calculator simply multiplies raw bandwidth by (1 + headroom/100) — it doesn't model actual burst patterns or competing traffic on your link, so the appropriate headroom percentage depends on how much non-voice traffic shares the same connection.