BroadNet
VoIP codecs · runs in your browser

Codec bandwidth calculator.

Pick a voice codec and a packetization interval to see the bandwidth per call — the raw codec bitrate and the real figure once IP, UDP and RTP headers are added. The full arithmetic is shown so you can check it. Nothing leaves your device.

Toll-quality 64 kbps PCM. The VoIP baseline.

Bandwidth / call (one-way)
80.0 kbps
Codec payload only
64 kbps
Payload / packet
160 B
Header / packet
40 B
Packets / second
50
Both directions
160.0 kbps

Overhead adds 16.0 kbps per call at 20 ms — headers are a fixed cost per packet, so shorter ptime means more packets and more overhead.

The math
  • payload = 64 kbps × 20 ms ÷ 8 = 160 bytes
  • packets/s = 1000 ÷ 20 = 50
  • header = 40 = 40 bytes
  • bw = (160 + 40) × 8 × 50 = 80.0 kbps

The short answer

How much bandwidth does a VoIP call use?

More than the codec bitrate, because every packet carries IP, UDP and RTP headers. The codec sets the payload rate — 64 kbps for G.711, 8 kbps for G.729 — but at a 20 ms packetization interval the call sends 50 packets per second, and each one adds a fixed 40 bytes of IP/UDP/RTP header. That overhead is why a G.711 call needs about 80 kbps one-way, not 64, and why a shorter packetization interval costs more bandwidth for the same audio.

The formula
  • payload_bytes = bitrate_kbps × ptime_ms ÷ 8
  • packets_per_second = 1000 ÷ ptime_ms
  • bandwidth = (payload_bytes + header_bytes) × 8 × packets_per_second
  • header_bytes = 40 (IP 20 + UDP 8 + RTP 12), +18 for Ethernet L2

Typical one-way bandwidth (20 ms ptime, IP/UDP/RTP)

Codec bandwidth comparison at 20 ms
CodecPayloadWith IP overhead
G.71164 kbps80 kbps
G.72264 kbps80 kbps
G.726 (32)32 kbps48 kbps
G.7298 kbps24 kbps
AMR-NB (12.2)12.2 kbps28.2 kbps

Figures are one-way; a full duplex call uses roughly double. They exclude compression such as cRTP and any silence suppression (VAD), which reduce the average on real networks.

Frequently asked questions

Why is a G.711 call about 80 kbps, not 64?

The codec produces a 64 kbps payload, but each RTP packet adds 40 bytes of IP, UDP and RTP headers. At 20 ms packetization the call sends 50 packets per second, so the headers add roughly 16 kbps, giving about 80 kbps one-way. Header overhead is a fixed cost per packet, independent of the codec.

How does packetization interval affect bandwidth?

A shorter interval (ptime) means more, smaller packets and therefore more header overhead per second, so bandwidth rises. A longer interval packs more audio per packet and lowers overhead, but adds latency and makes each lost packet more damaging. 20 ms is the common compromise.

Should I count one direction or both?

The main figure here is one-way (per stream). A normal call is full duplex, so end-to-end you provision roughly double for the two directions, plus headroom. The tool also shows the both-directions total.

Does this include silence suppression or cRTP?

No. It shows the steady-state bandwidth of active talk with standard IP/UDP/RTP headers. Voice Activity Detection (silence suppression) and compressed RTP (cRTP) both reduce the real-world average, so treat the result as an upper-bound planning figure.

What bitrate should I use for Opus?

Opus is variable. For voice, common wideband settings sit around 16–24 kbps, while music or high-fidelity audio can use much more, up to 510 kbps. Enter the target bitrate your endpoints negotiate to see its bandwidth with overhead.

Planning voice capacity?

BroadNet runs programmable voice, IVR and Voice OTP on carrier-grade infrastructure across 200+ countries. Talk to our team about codecs, concurrency and routing.