How Much Bandwidth Does VoIP Need? (Per Call, With Real Numbers)
The short answer: a standard G.711 call uses about 87 kbps in each direction on Ethernet, and a G.729 call uses about 31 kbps. The codec's bitrate alone (64 kbps or 8 kbps) understates the real figure, because every voice packet also carries headers. This guide shows where the numbers come from, so you can size a link with confidence.
Why the codec bitrate isn't the real number
A VoIP call is a stream of small RTP packets, usually one every 20 milliseconds (50 packets per second). Each packet carries a slice of compressed audio, the payload, wrapped in several layers of headers:
- RTP: 12 bytes (sequence numbers and timestamps)
- UDP: 8 bytes
- IPv4: 20 bytes (40 bytes for IPv6)
- Ethernet: 18 bytes (header plus frame check sequence), or 22 bytes with an 802.1Q VLAN tag
That is 58 bytes of overhead on every packet on a typical LAN. With G.711 at 20 ms the payload is 160 bytes, so overhead adds about 36%. With G.729 the payload is only 20 bytes, so the headers are almost three times bigger than the audio.
Bandwidth per call by codec
Figures are for one direction, at 20 ms packetisation (30 ms for iLBC), over IPv4 and Ethernet. A call uses this much upstream and the same amount downstream.
| Codec | Codec bitrate | Payload | On the wire (Ethernet) |
|---|---|---|---|
| G.711 (PCMA/PCMU) | 64 kbps | 160 B | 87.2 kbps |
| G.722 (HD voice) | 64 kbps | 160 B | 87.2 kbps |
| Opus @ 32 kbps | 32 kbps | 80 B | 55.2 kbps |
| GSM-FR | 13.2 kbps | 33 B | 36.4 kbps |
| G.729 | 8 kbps | 20 B | 31.2 kbps |
| iLBC (30 ms) | 13.33 kbps | 50 B | 28.8 kbps |
Want different settings, such as VLAN tags, PPPoE, SRTP or a VPN tunnel? Use the VoIP bandwidth calculator. It shows every codec side by side.
How to calculate it yourself
- Payload bytes = codec bitrate (kbps) × packet interval (ms) ÷ 8. For G.711 at 20 ms: 64 × 20 ÷ 8 = 160 bytes.
- Packet size = payload + 12 (RTP) + 8 (UDP) + 20 (IP) + layer-2 overhead. On Ethernet: 160 + 40 + 18 = 218 bytes.
- Packets per second = 1000 ÷ packet interval. At 20 ms that is 50.
- Bandwidth = packet size × 8 × packets per second. 218 × 8 × 50 = 87,200 bps = 87.2 kbps.
Add 10 bytes per packet for SRTP (the authentication tag) and roughly 60 bytes for a WireGuard tunnel over IPv4. VPN overhead hurts low-bitrate codecs most: a G.729 call over WireGuard uses nearly double its normal bandwidth.
Sizing an office connection
Size for peak concurrent calls, not the number of users. A 30-person office rarely has 30 calls at once. A common rule of thumb for general office use is that about a third of users are on the phone at the busiest moment. Contact centres are different: plan for every agent on a call.
Then add headroom. Voice is sensitive to congestion, so allow 20–30% spare capacity on top of the voice traffic, and keep in mind that the rest of the office is using the same link. The upload side is usually the bottleneck, especially on asymmetric connections such as ADSL, some fibre packages and LTE.
Bandwidth is only half the story. A link with plenty of capacity can still give poor calls if latency, jitter or packet loss are high. Check those with the MOS call quality estimator.
Ways to reduce VoIP bandwidth
- Use a lower-bitrate codec such as G.729 or Opus at a modest bitrate, if your provider supports it. Expect a small quality trade-off.
- Increase packet interval from 20 ms to 30 or 40 ms. Fewer packets means less header overhead, at the cost of slightly more delay.
- Avoid unnecessary tunnels. Running voice through a full VPN adds tens of bytes to every packet.
- Prioritise rather than reduce. Often the real fix isn't less bandwidth but QoS: mark RTP as DSCP EF (46) and give it priority on the upload queue.
Try the VoIP Bandwidth Calculator
Bandwidth per call and for your whole office, for G.711, G.729, Opus and more, including all overhead.