VoIP Bandwidth Calculator

Calculate real VoIP bandwidth per call including RTP, UDP, IP and Ethernet overhead - which is roughly double the codec rate for G.711.

Calculator Bandwidth & Data Runs in your browser
Try:

Result

Bandwidth per call
96.8 kbit/s
one direction
Per call bidirectional
193.6 kbit/s
Total for 20 calls
3.87 Mbit/s
Link utilisation
3.9%
Maximum calls on this link
516
at 100% utilisation - plan for less
Overhead ratio
1.51×
160 B payload + 82 B overhead

Codec comparison

CodecCodec rateReal per callMax callsTypical MOS
G.711 (PCM)64 kbit/s96.8 kbit/s5164.4
G.7298 kbit/s40.8 kbit/s1,2253.9
G.722 (HD)64 kbit/s96.8 kbit/s5164.5
Opus (typical VoIP)32 kbit/s64.8 kbit/s7714.5
G.726 (ADPCM 32k)32 kbit/s64.8 kbit/s7714
Overhead dominates
Voice packets are small and frequent, so overhead exceeds payload for low-rate codecs. G.729 at 8 kbit/s actually consumes about 31 kbit/s on tagged Ethernet - four times the codec rate. Compressing the codec saves far less than the marketing suggests.
Quality, not just capacity
Bandwidth is the easy part. Voice needs jitter under 30 ms, loss under 1% and one-way latency under 150 ms (ITU-T G.114). Give voice a strict-priority queue rather than more capacity.

About VoIP Bandwidth Calculator

A G.711 call is 64 kbit/s of audio and roughly 110 kbit/s on the wire once RTP, UDP, IP and Ethernet framing are counted at 50 packets per second in each direction. Planning voice capacity on codec rate alone undersizes the link by half.

Queueing matters more than headroom

Voice does not need much bandwidth; it needs predictable bandwidth. Put voice in a strict-priority (LLQ) class sized at roughly the calculated peak plus signalling, and police it so a runaway stream cannot starve everything else. Mark at the source with DSCP EF (46) for media and CS3 (24) for signalling, and make sure the marking survives every hop — a switch that rewrites DSCP to zero on ingress undoes the entire design silently.

Where the overhead comes from

Each packet carries 12 bytes of RTP, 8 of UDP, 20 of IPv4, 18 of Ethernet including FCS, 20 of preamble and inter-frame gap, plus 4 more if the voice VLAN is tagged. That is 82-86 bytes wrapping a 160-byte G.711 payload, or a 20-byte G.729 payload. Raising the packetisation interval to 30 or 40 ms cuts overhead but adds latency and makes each lost packet more damaging.

Common use cases

  • Sizing a WAN link for a branch office phone system.
  • Deciding whether a codec change actually saves meaningful bandwidth.
  • Setting a bandwidth reservation for a voice priority queue.

Edge cases and gotchas

  • Reserve capacity for signalling (SIP) and for the codec upshift that happens when calls go to conferencing or video.
  • Silence suppression reduces average bandwidth but you must plan for peak, not average.

Frequently asked questions

Does silence suppression reduce the bandwidth I need to provision?
It reduces average consumption by roughly a third but not the peak, and provisioning is a peak problem. Treat it as a bonus, not a planning input.
How much bandwidth for 30 concurrent G.711 calls?
About 6.6 Mbit/s bidirectional on tagged Ethernet. Provision 10 Mbit/s of guaranteed capacity to leave room for signalling and burst.