How much bandwidth does a VoIP call use?

G.711 produces 64 kbps of audio. At 20 milliseconds per packet, each packet holds 160 bytes of speech, and the phone sends 50 packets a second.

Every packet also carries addressing. Cisco's voice bandwidth calculation counts 40 bytes of IP, UDP and RTP headers, plus 18 bytes of Ethernet framing. That makes each packet 218 bytes on the wire, or 1,744 bits. Fifty of those a second is 87,200 bits per second, which is 87.2 kbps.

So about a quarter of a G.711 call is packaging. The headers stay the same size when the audio gets smaller, and that limits what compression can save. G.729 squeezes the audio to 8 kbps, an eighth of G.711. Each packet then carries 20 bytes of speech and the same 58 bytes of headers and framing. The call uses 31.2 kbps, which is a little over a third of the G.711 figure.

A call is two streams of audio, one from you and one to you. Cisco's contact center design guide adds the two together. It puts a G.711 call at 174.4 kbps with 20-millisecond packets and 159.0 kbps with 30-millisecond packets, and a G.729 call at 62.4 and 47.0 kbps. The 30-millisecond rows in the table are half of those two-way totals.

That total is easy to misapply. An internet connection has separate capacity for upload and download, and each stream uses only one of them. The guide makes the point with a 100 Mbps full-duplex link, which has 100 Mbps in each direction. On that link a G.711 call takes 87.2 kbps of the upload and 87.2 kbps of the download.

So compare the one-way figure with each direction of your connection. If your service has less upload than download, the upload figure sets the limit on how many calls fit.

Packet size is the first variable. Putting more speech into each packet means fewer packets a second, and fewer packets means fewer headers. That's why the 30-millisecond rows in the table are lower. Cisco notes the cost: a larger payload reduces bandwidth and increases overall delay. RFC 3551, which sets 20 milliseconds as the default for audio over RTP, adds that longer packets make packet loss more noticeable, because each lost packet takes more speech with it.

The link is the second. The 18 bytes of framing belong to Ethernet. Cisco's formula uses the header of whichever link carries the call, so the same codec gives a different figure on a different kind of circuit. Cisco's own sample shows how far that can go. With G.729 on a link that has a 6-byte header and compresses the IP, UDP and RTP headers down to 2 bytes, the call comes to 11.2 kbps. Cisco also notes that this header compression isn't available over Ethernet.

Encryption and VPN tunnels move the number the other way, because they wrap each packet in more bytes. The amount depends on the method, so add it from your own equipment's documentation.

Silence is the last variable. Some systems stop sending packets when nobody is talking. Cisco says this can save up to 35 percent on average across more than 24 calls, and it also says the saving must not be counted when you size a network. A link has to carry the moments when everyone talks at once.

The per-call figure applies to each call in progress at the same moment, so multiply by calls and not by employees. Suppose ten people in your office are on G.711 calls at once. They need 872 kbps of upload and 872 kbps of download, on a connection that's also carrying everything else the office does. With G.729 the same ten calls need 312 kbps each way.

Working out how many calls happen at once is the harder estimate. We cover it in how many simultaneous calls you need on a SIP trunk. If the calls are part of a wider setup with video and screen sharing, see what UCaaS bandwidth requirements to plan for.

The same arithmetic gives a rough figure for data use. At 87.2 kbps, a G.711 call moves about 39 megabytes an hour in each direction, or roughly 78 megabytes for both. At 31.2 kbps, a G.729 call moves about 14 megabytes an hour each way. Those totals carry the same Ethernet assumptions as the rates, so a mobile or metered connection with different framing will count somewhat differently.

A call needs capacity on every stretch between the phone and the provider's voice equipment, and the office connection is only the first stretch. Each stretch has its own framing, so use the allowance that fits each link. The routes differ too.

An Atlantech Online article from 2016, BYOB vs. dedicated connection, sets out two of them. A business can buy internet access from one company and phone service from another, and then each call crosses the public internet to reach the phone provider's switch. Or the office connection can sit on the same network as the voice switch. In that case, Atlantech wrote, the provider manages the routing equipment on both sides and can prioritize voice traffic.

The arrangement doesn't change how the per-call figure is worked out. It changes who controls the stretches where voice competes with other traffic. On your own office network that's you, and a large upload or a video meeting can crowd a call even when the monthly plan looks generous.

Bandwidth is also only one requirement. RTP, the protocol that carries call audio, doesn't guarantee delivery or timely arrival, according to RFC 3550. A link with room to spare can still deliver packets late or out of order. The receiving end can absorb some of that, and too much delay or loss hurts the audio. We explain how the audio stream works in how VoIP works and what to look for in a connection in what type of internet connection you need for VoIP.