Productivity and Utilities

VoIP Bandwidth Calculator

Enter the codec bit rate, the packet interval and the number of calls.

Your numbers

Example figures throughout. Use the rate of your codec.

How much audio goes in each packet.

For example 18 for an Ethernet header and check sequence.

Result

One call, one direction

87.2 kbps

218 byte packets, 50 a second

25 calls, each direction
2,180.0 kbps
Overhead above the codec rate
36.3%

Build-up

MeasureValue
Audio in each packet160.0 bytes
IP, UDP and RTP headers40 bytes
Link layer header18 bytes
Packet size218.0 bytes
Packets per second50.00
One call, one direction87.2 kbps
One call, both directions174.4 kbps
25 calls, each direction2,180.0 kbps
How this was calculated
  1. Audio in each packet = bit rate x packet interval / 8.
  2. Packet size = audio + 40 bytes of IPv4, UDP and RTP headers + the link layer header.
  3. Bandwidth = packet size x 8 x packets per second, where packets per second = 1000 / packet interval in ms.
  • The 40 bytes assume IPv4 with no options. Encryption, VLAN tags, IPv6 and silence suppression change the real figure, so leave room above the result.

Calculated using SumPanda.com · https://sumpanda.com/voip-bandwidth-calculator/

Results are estimates for general information. Verify important figures before relying on them.

Next step

How to use the voip bandwidth calculator

  1. Enter the codec bit rate in kilobits per second.
  2. Enter the packet interval in milliseconds and the size of the link layer header in bytes.
  3. Enter the number of simultaneous calls.
  4. Read the bandwidth for one call and for all the calls, in one direction.

Formula

Payload per packet = bit rate x packet interval / 8 bits per byte

Packet size = payload + 12 byte RTP + 8 byte UDP + 20 byte IP + link header

Packets per second = 1000 / packet interval

Bandwidth per call = packet size x 8 x packets per second

The result is for one direction. A call uses the same amount each way.

A 64 kbps codec with 20 ms packets

Each packet carries 160 bytes of audio. Adding 40 bytes of RTP, UDP and IP headers and an 18 byte Ethernet header gives 218 bytes, sent 50 times a second. That is 87.2 kbps per call in each direction, and 25 calls need 2,180.0 kbps.

With a codec at 8 kbps the payload is only 20 bytes, so the headers dominate and a call needs 31.2 kbps, nearly four times the codec rate. Using 30 ms packets sends fewer packets and lowers the overhead, at the price of more delay.

Overhead matters

Voice is sent in many small packets, and each packet carries the same headers no matter how little audio is inside. The lower the codec rate, the larger the share taken by headers. That is why the real bandwidth of a call is always higher than the rate in the codec name, sometimes by more than double.

Longer packet intervals reduce the overhead but add delay and make a lost packet more noticeable. Most systems use 20 ms as a balance.

What the tool leaves out

It does not include signaling, silence suppression, header compression, encryption or the extra headers added by a VPN or a wireless link. Leave a margin for those and for other traffic. To size a link by the delay it adds, see the Bandwidth Delay Product Calculator, and for a radio link the Free Space Path Loss and Link Budget Calculator is the next step. The number of lines needed for the traffic can be found with the Erlang B Blocking Calculator or the Erlang C Staffing Calculator.

Assumptions and limits

  • Headers are 12 bytes for RTP, 8 for UDP and 20 for IPv4. IPv6 would add 20 more bytes.
  • The link header is the value you enter, for example 18 for Ethernet.
  • Every call uses the same codec and packet interval.
  • The figures shown are examples. Use the codec and network of your own system.

Formulas reviewed October 10, 2026. See the calculation methodology for how SumPanda rounds, tests and sources its formulas.

Frequently asked questions

Is the result per direction or total?

Per direction. A call is sent both ways, so a link that is full duplex needs this much each way.

Why is it more than the codec rate?

Because every packet carries headers. The codec rate counts only the audio.

Does silence suppression help?

It can reduce the average, but the tool shows the worst case where audio is always being sent.

What packet interval should I use?

Twenty milliseconds is common. Check what your phones or software actually use.