VoIP Bandwidth Calculator (per call and per link)
Bandwidth per VoIP call and for many calls: codec payload, packet rate, IP/UDP/RTP and layer-2 overhead (Ethernet, 802.1Q, PPP/Frame Relay), cRTP and silence suppression.
Plan for both directions: a 10-call trunk at 87.2 kbit/s needs 0.872 Mbit/s up and the same down, plus headroom for signalling and jitter.
| Interval (ms) | Payload (bytes) | Packets/s | kbit/s per call | kbit/s for all calls |
|---|---|---|---|---|
| 10 | 80 | 100 | 110.4 | 1,104 |
| 20 | 160 | 50 | 87.2 | 872 |
| 30 | 240 | 33.333 | 79.467 | 794.667 |
| 40 | 320 | 25 | 75.6 | 756 |
| 60 | 480 | 16.667 | 71.733 | 717.333 |
A VoIP call needs much more bandwidth than its codec rate suggests, because every small voice packet carries 40 bytes of IP, UDP and RTP headers plus the layer-2 frame around them. This calculator adds it all up: payload per packet from the codec and packet interval, packets per second, header and link overhead, cRTP, VPN or SRTP overhead and silence suppression, for one call and for all concurrent calls.
How to use this tool
- Choose the codec (or enter the bitrate for Opus or another codec).
- Pick the packetization interval — 20 ms is the usual default.
- Choose the layer-2 type of the link and whether cRTP is used, and add any VPN or SRTP overhead.
- Enter the number of concurrent calls and read the bandwidth per call and in total, for each direction.
Formula
payload = codec rate × interval ÷ 8; packet = payload + 40 (IP 20 + UDP 8 + RTP 12) + layer 2; packets/s = 1000 ÷ interval; bandwidth = packet × 8 × packets/s
Bytes per packet are converted to bits (× 8) and multiplied by the packet rate. Layer-2 overhead is 18 bytes for Ethernet (14-byte header + 4-byte FCS), 22 with an 802.1Q VLAN tag, and 7 for PPP, MLPPP or Frame Relay FRF.12 (6 bytes + 1 flag byte). cRTP shrinks the 40-byte header to 2 or 4 bytes on serial links. Figures are per direction.
Example
G.711 at 20 ms sends 64,000 × 0.02 ÷ 8 = 160 bytes of voice per packet, 50 packets a second. Each Ethernet packet is 160 + 40 + 18 = 218 bytes, so one call uses 218 × 8 × 50 = 87.2 kbit/s in each direction — not 64. G.729 at 20 ms needs (20 + 40 + 18) × 8 × 50 = 31.2 kbit/s. Ten G.711 calls need 872 kbit/s up and 872 kbit/s down.
Frequently asked questions
Why does a 64 kbit/s codec use 87.2 kbit/s?
Headers. At 20 ms each packet carries 160 bytes of voice and 58 bytes of IP, UDP, RTP and Ethernet overhead, so about a quarter of the traffic is overhead. With G.729 the overhead is over 70%.
Does a longer packetization interval save bandwidth?
Yes: fewer packets mean fewer headers, so 30 ms uses less than 20 ms. The cost is extra delay (each packet waits for more audio) and a bigger gap when one packet is lost. 20 ms is the common compromise.
How much do I need for a SIP trunk?
Multiply the per-call figure by the most calls you expect at once, in both directions, and leave headroom for signalling, other traffic and jitter. Most teams also mark voice traffic for priority (QoS) on busy links.
Sources
Last updated:
Page address with these inputs: https://kakao.mobile/tools/voip-bandwidth-tool/
Cite or embed this tool
kakaoCalc. (2026, November 8). VoIP Bandwidth Calculator (per call and per link). https://kakao.mobile/tools/voip-bandwidth-tool/
“VoIP Bandwidth Calculator (per call and per link).” kakaoCalc, 8 Nov. 2026, kakao.mobile/tools/voip-bandwidth-tool/.
Paste this code into your page. The widget runs in your visitors' browsers, shows no ads and keeps a link back to this page. Embedding terms