VoIP Bandwidth per Call Calculator

Understanding the VoIP bandwidth calculator — concept and principle

A VoIP call needs more bandwidth than its codec's nominal bit rate, because every small voice packet carries IP, UDP and RTP headers plus link-layer framing. This calculator adds up that overhead for common codecs and tells you the bandwidth per call in one direction and the total for a number of simultaneous calls in both directions. It is used to size internet uplinks, WAN links and QoS queues for phone systems and contact centres.

A 64 kbps G.711 call needs 95.2 kbps on the wire because of headers.

How it is calculated

Packets per second: with 20 ms of audio per packet, 1 ÷ 0.020 = 50 packets per second.
Payload per packet: codec rate × 0.020 s ÷ 8. G.711 and G.722 (64 kbps) give 160 bytes, G.729 (8 kbps) 20 bytes and Opus at 24 kbps 60 bytes.
Packet size on the wire = payload + 40 bytes (IPv4 20 + UDP 8 + RTP 12) + layer-2 overhead.
Per call, one direction (kbps) = packet size × 8 × 50 ÷ 1,000.
Total (Mbps) = per call × number of calls × 2 directions ÷ 1,000.

The default layer-2 overhead of 38 bytes is full Ethernet on the wire: header 14, FCS 4, preamble 8 and inter-frame gap 12.

The principle

Voice is sent in many tiny packets to keep delay low, so headers form a large share of each packet. For G.711 the 40 bytes of IP/UDP/RTP add 25% to a 160-byte payload, and for G.729 they are twice the payload. A call is full-duplex: each side streams continuously, so capacity is needed in both directions.

Worked example

Inputs: G.711 at 20 ms, 10 calls, 38 bytes layer-2 overhead.
Packet size: 160 + 40 + 38 = 238 bytes.
Per call, one direction: 238 × 8 × 50 = 95,200 bit/s = 95.2 kbps.
Total: 95.2 × 10 × 2 = 1,904 kbps = 1.904 Mbps.
With G.729 the same call needs (20 + 40 + 38) × 8 × 50 = 39.2 kbps.

Each point is the calculator re-run with only “Layer-2 overhead (bytes)” changed; the gold point is the default example. Change the inputs above and this chart follows your numbers.

Layer-2 overhead (bytes)	Per call, one direction (kbps)
9.5	83.8
19	87.6
28.5	91.4
38	95.2
47.5	99
57	102.8
76	110.4

How to read the result

The total counts both directions added together. On a symmetric link, provide about half of it (here 952 kbps) in each direction, plus headroom for signalling and other traffic. Many vendor tables quote 87.2 kbps for G.711 over Ethernet because they count only the 18-byte header and FCS, not preamble and gap. Plan for peak concurrent calls, not average.

Common mistakes

Budgeting the codec's 64 kbps and ignoring headers.
Forgetting that VPN, VLAN tags or PPPoE add further bytes per packet.
Comparing totals with an upload speed without dividing by two directions.

Keep in mind: Results assume constant-rate audio without silence suppression and a 20 ms packet interval. Encryption (SRTP), tunnels and variable-rate codecs change the figures, so verify with your provider and equipment.

Frequently asked questions

How much bandwidth does one VoIP call use?

Roughly 80 to 100 kbps each way for G.711, and about 30 to 40 kbps for G.729, depending on link overhead.

Does a longer packet interval save bandwidth?

Yes. At 30 ms there are fewer packets and less header overhead, but delay increases.

Is bandwidth enough for good call quality?

No. Latency, jitter and loss matter as much, which is why voice traffic is usually prioritised with QoS.

Sources: IETF RFC 3550, RTP: A Transport Protocol for Real-Time Applications · ITU-T G.711, Pulse code modulation (PCM) of voice frequencies · ITU-T G.729, Coding of speech at 8 kbit/s using CS-ACELP · IETF RFC 6716, Definition of the Opus Audio Codec

Checked 8 Nov 2026. Standard formulas and RFC/IEEE definitions, computed locally in your browser — nothing you type is sent anywhere.

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