BYTETOOLS

VoIP Bandwidth Calculator

Work out VoIP bandwidth per call for G.711, G.729, Opus and more, including RTP/UDP/IP, Ethernet, PPP and IPsec overhead, with and without cRTP.

50
Packets per second
218 B
Bytes on the wire
87.2 kbps
Per call
0.087 Mbps
Total (1 call)

How the packet is built

ComponentBytesNotes
Voice payload16064 kbps × 20 ms ÷ 8
RTP + UDP + IP header4012 + 8 + 20
Layer 2 framing18Ethernet — 18 bytes (14 header + 4 FCS)
Total per packet218× 8 bits × 50 pps = 87.2 kbps

Bandwidth summary

Per call1 callBoth directions
Uncompressed headers87.2 kbps87.2 kbps174.4 kbps
With cRTP (2 B)72 kbps72 kbps144 kbps

A call is full duplex, so a WAN link that carries both directions of the media needs the “both directions” figure. Voice activity detection can cut real usage by roughly a third on conversational speech, but never size a link on that assumption.

What is the VoIP Bandwidth Calculator?

The VoIP Bandwidth Calculator works out what a call really costs on the wire. Pick a codec and a packetisation interval and it derives the payload size and packets per second, adds the RTP, UDP and IP headers, then the Layer 2 framing and any IPsec tunnel overhead you select, and multiplies out for the number of concurrent calls.

  • Thirteen codec presets from G.711 and G.722 down to G.723.1, iLBC and low-rate Opus
  • Derives payload bytes and packets per second from the packetisation interval
  • Layer 2 presets for Ethernet, 802.1Q, MPLS and PPP, plus a custom byte value
  • IPsec transport, tunnel and NAT-T overhead options, and IPv4 or IPv6 headers
  • Side-by-side comparison with cRTP header compression
  • Byte-by-byte breakdown of the packet so you can check the arithmetic yourself

How to use the VoIP Bandwidth Calculator

  1. 1

    Choose your codec, or pick “Custom bit rate” and type the encoded rate in kbps.

  2. 2

    Set the packetisation interval — 20 ms is the usual default, and larger values trade delay for bandwidth.

  3. 3

    Select the Layer 2 overhead for the link (Ethernet, VLAN-tagged, PPP or your own value) and any IPsec tunnel.

  4. 4

    Enter the number of concurrent calls and, if you use header compression, the cRTP header size.

  5. 5

    Read the per-call figure, the total, and the both-directions column for sizing a WAN link.

About the VoIP Bandwidth Calculator

The VoIP Bandwidth Calculator works out what a call really costs on the wire. Pick a codec and a packetisation interval and it derives the payload size and packets per second, adds the RTP, UDP and IP headers, then the Layer 2 framing and any IPsec tunnel overhead you select, and multiplies out for the number of concurrent calls.

The headers dominate the answer, which is why the numbers surprise people. G.711 encodes at 64 kbps, but at 20 ms packets over Ethernet each call actually uses 87.2 kbps in each direction, because 58 bytes of header ride along with every 160 bytes of audio. G.729 at 8 kbps costs 31.2 kbps the same way.

A breakdown table shows exactly where every byte goes, and a second table compares uncompressed headers against cRTP for one call, for the whole call count, and for both directions of a full-duplex link. All arithmetic runs in your browser — nothing is uploaded.

Frequently asked questions

How much bandwidth does a G.711 call use?

About 87.2 kbps per direction over Ethernet at the default 20 ms packetisation. The codec itself is 64 kbps, but each 160-byte payload carries 40 bytes of RTP/UDP/IP header and 18 bytes of Ethernet framing, and there are 50 packets a second.

How do I calculate packets per second for VoIP?

Divide 1000 by the packetisation interval in milliseconds. A 20 ms ptime gives 50 packets per second, 30 ms gives about 33, and 10 ms gives 100. The payload size in bytes is the codec bit rate multiplied by the interval and divided by eight.

Does a larger packetisation interval save bandwidth?

Yes, because the fixed 40-byte header is amortised over more audio. Going from 20 ms to 40 ms roughly halves the header cost. The trade-off is latency and loss impact — one lost packet now takes twice as much speech with it.

What is cRTP and how much does it save?

Compressed RTP squeezes the 40-byte RTP/UDP/IP header down to two to four bytes on a single link by sending only what changed. On low-rate codecs the saving is dramatic — a G.729 call drops from 26.4 kbps to about 11.2 kbps on a 6-byte PPP link — but it is per-hop and costs CPU.

Should I size a link for one direction or two?

A voice call is full duplex, so a WAN link carrying both directions needs twice the per-call figure. The calculator shows a both-directions column for exactly this reason. Do not subtract anything for silence suppression when sizing — treat any saving as headroom.

Does this include signalling and jitter buffer overhead?

No. It covers the RTP media stream only. SIP or H.323 signalling is bursty and small compared with the media, and a jitter buffer costs delay rather than bandwidth. For capacity planning, media is what fills the link.

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