VoIP Bandwidth Calculator
Calculate how much network bandwidth your VoIP deployment requires. Accounts for codec bitrate, packetization, and protocol overhead.
Include Protocol Overhead
IP + UDP + RTP headers (40 bytes per packet)
Results
Per Call
160.0 kbps
bidirectional
Total (10 calls)
1.60 Mbps
both directions combined
Recommended
2.08 Mbps
with 30% headroom
Per-Call Breakdown (one direction)
How VoIP Bandwidth Works
VoIP bandwidth is not just the codec bitrate. A voice call involves three layers of processing that each contribute to the total bandwidth consumed.
First, the codec encodes raw audio into compressed digital samples. G.711 uses 64 kbps with no compression, producing high-quality audio at the cost of bandwidth. G.729 compresses aggressively to just 8 kbps, while Opus offers a modern variable-rate approach.
Second, those samples are grouped into packets at regular intervals, typically every 20 milliseconds. This means 50 packets per second in each direction. Each packet carries a small chunk of encoded audio as its payload.
Third, every packet gets wrapped in protocol headers. IP adds 20 bytes, UDP adds 8 bytes, and RTP adds 12 bytes. That is 40 bytes of overhead per packet. At 50 packets per second, the overhead alone consumes 16 kbps, which can be significant for low-bitrate codecs like G.729.
Choosing a Codec
The codec you choose determines the tradeoff between audio quality and bandwidth consumption. Here is how the common codecs compare:
| Codec | Bitrate | With Overhead | Use Case |
|---|---|---|---|
| G.711 | 64 kbps | 80 kbps | PSTN interoperability, highest quality on LAN |
| G.729 | 8 kbps | 24 kbps | Bandwidth-constrained links, WAN deployments |
| G.722 | 64 kbps | 80 kbps | HD voice, wideband audio for internal calls |
| Opus | 24 kbps | 40 kbps | WebRTC, modern VoIP, adaptive bitrate |
G.711 is the safest default. It works everywhere and produces the best audio quality, but uses the most bandwidth. G.729 is the opposite extreme: excellent compression at the cost of some audio fidelity. G.722 provides wideband (HD) audio at the same bandwidth as G.711. Opus is the most flexible modern codec, commonly used in WebRTC applications.
Common Mistakes
Not accounting for overhead. The codec bitrate is only part of the story. Protocol headers add 40 bytes to every packet. For G.729, the overhead triples the actual bandwidth consumption compared to the raw codec bitrate. Always calculate with overhead included for accurate planning.
Not planning for peak usage. Sizing your connection for the average number of concurrent calls means you run out of headroom during busy periods. The calculator above includes a 30% headroom recommendation. For business-critical deployments, consider even more.
Ignoring upload bandwidth. VoIP is bidirectional. Every call consumes bandwidth in both directions. Many internet connections are asymmetric, with significantly less upload than download capacity. Your upload speed is usually the bottleneck. Make sure your connection can handle the total bandwidth in both directions simultaneously.
Forgetting about other traffic. VoIP shares your connection with everything else on the network. Without Quality of Service (QoS) configuration, a large file download or backup can starve voice packets and cause call quality degradation. Understanding how your local network handles traffic is just as important as having enough raw bandwidth.