Free tool ยท VoIP bandwidth
How Much Bandwidth Do Your VoIP Calls Need?
This calculator works out the bandwidth your phone calls need from three inputs: how many calls happen at once, which codec your phones use, and which packet headers your network adds. It shows the result per call and in total, each direction, with the packet math written out and the codec bit rates cited to the RFCs.
01 Calculator
Enter your calls and codec
Count the calls that happen at the same time at your busiest hour. The codec is in your phone system or trunk settings; G.711 is the usual default.
02 Your result
What your calls need
Enter the number of concurrent calls above and press Calculate to see the bandwidth per call and in total.
- Per call, each direction
- Voice payload plus headers; the same again in the other direction.
- All calls, each direction
- Reserve this upstream and downstream at the busiest moment.
- Packets per second
- All calls, each direction. Routers and firewalls are rated in packets too.
- Size of each packet
- Payload plus the headers you chose.
- Share of your upload speed
- Voice alone, before everything else that shares the line.
Headroom
Voice needs the figure above upstream and the same again downstream, at the busiest moment of the day, with low delay and almost no packet loss. Treat the total as a floor to reserve for voice with quality of service (QoS) on the router and switches, not as the size of the line: everything else your office does has to fit beside it, and the upload side is usually the smaller one. When voice would take a large share of the line, size up the circuit or add a second one.
04 How it is calculated
How is VoIP bandwidth calculated?
The standard packet model: a slice of voice plus its headers, sent many times a second.
- Voice payload per packet codec bit rate (kbit/s) x packet interval (ms) / 8
- Packet size on the wire payload + IP, UDP and RTP headers (+ Ethernet on a LAN)
- Packets per second, per call 1000 / packet interval (ms)
- Bandwidth per call, each direction packet size x 8 x packets per second / 1000
- Total for your calls, each direction bandwidth per call x concurrent calls
- Share of your upload speed total / upload speed (optional input)
Assumptions
- Bandwidth is per direction. A call uses the figure shown upstream and the same again downstream.
- Silence suppression and RTP header compression are off, so the result is the ceiling for the codec, not an average.
- Header sizes are IP 20, UDP 8, RTP 12 and Ethernet 18 bytes (58 bytes in all on a LAN), as in Cisco's reference calculation. VLAN tags, IPv6 and the WAN circuit's own framing are not included.
- Opus is a variable-rate codec; we use 20 kbit/s, the top of the range RFC 7587 recommends for wideband speech at 20 ms frames.
- G.722 samples at 16 kHz but still carries 64 kbit/s, so it costs exactly what G.711 costs.
- SIP signaling is not included; it is a few small messages at the start and end of a call next to a steady voice stream.
Sources
- Cisco, "Bandwidth Consumption Calculation for Voice Calls" (document 7934)
The header sizes (IP 20, UDP 8, RTP 12, Ethernet 18 bytes), the formula, and the reference results of 87.2 kbit/s for G.711 and 31.2 kbit/s for G.729 at 20 ms on Ethernet.
- IETF RFC 3551, "RTP Profile for Audio and Video Conferences with Minimal Control"
G.711 PCMU/PCMA at 64 kbit/s (section 4.5.14), G.722 at 64 kbit/s (section 4.5.2), G.729 at 8 kbit/s (section 4.5.6), and the 20 ms default packetization interval (section 4.2).
- IETF RFC 7587, "RTP Payload Format for the Opus Speech and Audio Codec"
Recommended Opus bit rates for 20 ms frames: 8 to 12 kbit/s narrowband speech, 16 to 20 kbit/s wideband speech, 28 to 40 kbit/s fullband speech; ptime default 20 ms.
- IETF RFC 6716, "Definition of the Opus Audio Codec"
Opus runs from 6 to 510 kbit/s (section 2.1.1) and encodes frames of 2.5 to 60 ms, with 20 ms "a good choice for most applications" (section 2.1.4).
05 Codecs compared
How do the four codecs compare?
One call at 20 ms packets on an Ethernet LAN, computed from the cited bit rates and header sizes.
| Codec | Payload bit rate | Voice per 20 ms packet | Per call on Ethernet, each direction | Notes |
|---|---|---|---|---|
| G.711 (PCMU or PCMA), 64 kbit/s | 64 kbit/s | 160 bytes | 87.2 kbit/s | Uncompressed narrowband voice, the default on most SIP trunks and desk phones. Handles fax and music on hold. |
| G.722 (HD voice), 64 kbit/s | 64 kbit/s | 160 bytes | 87.2 kbit/s | Wideband (7 kHz) audio at the same bit rate as G.711, so it costs nothing extra on the wire. |
| G.729, 8 kbit/s | 8 kbit/s | 20 bytes | 31.2 kbit/s | Compressed voice with an eighth of the payload of G.711. Lower audio quality; not for fax or music. |
| Opus (wideband speech), 20 kbit/s | 20 kbit/s | 50 bytes | 43.2 kbit/s | Variable-rate codec used by WebRTC and softphones. RFC 7587 recommends 16 to 20 kbit/s for wideband speech; we use the top of that range. |
06 What to do about it
What to do with the figure
- Connect
Business internet, fiber and SD-WAN
Size the circuit for voice plus everything else, add a backup path and let SD-WAN steer the calls over the healthy line.
- Phones
Phone systems and SIP trunks
Codec choice, call limits on the trunk and quality of service are set here, with Yealink, Nextiva, Avaya, Yeastar and Polycom.
- Network
Switching, firewalls and managed Wi-Fi
Voice VLANs and QoS on Fortinet, Ubiquiti and Cisco Meraki gear keep the calls ahead of the backups.
- Talk
Book a free 30-minute consultation
Bring your call counts and your current line speeds; we tell you what to change first.
07 Questions
Questions about VoIP bandwidth
Why does G.711 need 87 kbit/s if the codec is 64 kbit/s?
Because every 20 milliseconds of voice travels in its own packet, and each packet carries headers: 20 bytes of IP, 8 of UDP, 12 of RTP and 18 of Ethernet. For G.711 that is 58 bytes of headers on 160 bytes of voice, fifty times a second, which is the 87.2 kbit/s Cisco publishes for the same case.
Does the calculator cover calls between phones in the same office?
Only if you want it to. Calls between two desk phones in one building usually stay on the local network and never touch the internet line, so for sizing an internet circuit count only the calls that leave the building: SIP trunk calls, remote workers and softphones at home. For sizing switches and Wi-Fi, count every simultaneous call.
How much headroom should I leave for voice?
Enough that voice never competes. The total is what the calls consume at the busiest moment, each direction; everything else your office does must fit beside it. Reserve the voice total with quality of service on the router and switches, watch the upload side, usually the smaller one, and size up the circuit when the share grows large.
Which codec should a small office choose?
G.711 is the safe default: phones and SIP trunks support it, it carries fax and music on hold, and the bandwidth is modest on any business line. Choose G.722 when your phones and provider both support HD voice, since it costs the same. Keep G.729 for thin links such as cellular backup, and expect Opus on softphones and WebRTC.
Next step
Need a Line That Carries Every Call?
We size, source and manage business internet, SD-WAN and phone systems for Orange County offices, with 24/7 support for everything we install. Start with a look at the connectivity options, or at the phone systems the calls run on.
Last reviewed: . Published by Telxpress, Irvine, CA.