VoIP Bandwidth Calculator – Plan Your Call Capacity Right

March 18, 2026

📞 VoIP Bandwidth Calculator

Calculate exact bandwidth requirements for concurrent VoIP calls with codec overhead and QoS planning

Quick Presets
⚙️ Calculator Inputs
✔ Bandwidth Results
📶 Codec Quick Reference
87
G.711 kbps w/ headers
31
G.729 kbps w/ headers
87
G.722 kbps w/ headers
40
Opus kbps (typical)
160
RTP header bytes/pkt
20ms
Standard packet interval
50
pkts/sec at 20ms
20%
Recommended overhead
📋 Codec Bandwidth Comparison Table
Codec Raw Bitrate Headers Added Total per Call Calls per 1 Mbps Quality
G.711 (a-law/u-law)64 kbps~23 kbps~87 kbps~11Excellent
G.7298 kbps~23 kbps~31 kbps~32Good
G.722 (HD Voice)64 kbps~23 kbps~87 kbps~11HD
G.722.1 (Siren)24 kbps~23 kbps~47 kbps~21HD
G.726 (32 kbps)32 kbps~23 kbps~55 kbps~18Good
G.72816 kbps~23 kbps~39 kbps~25Good
Opus (NB)~12 kbps~28 kbps~40 kbps~25Good
iLBC15.2 kbps~23 kbps~38 kbps~26Fair-Good
🗂 Deployment Type Overhead Reference
Deployment Protocol Overhead Signaling Additional Overhead Recommended Buffer
SIP TrunkingRTP + UDP/IP = ~23 kbpsSIP (low)RTCP ~5%10–15%
H.323 PBXRTP + UDP/IP = ~23 kbpsH.225/H.245GK queries ~5–8%15–20%
WebRTCSRTP + DTLS = ~28 kbpsICE/STUN/TURNSTUN overhead ~8–12%20–25%
Unified CommunicationsRTP + video RTPSIP + presenceVideo + presence ~30–50%20–30%
📏 VoIP QoS Standards Reference
Parameter Acceptable Recommended Standard
One-way Latency<150ms<100msITU-T G.114
Jitter<50ms<30msITU-T G.114
Packet Loss<5%<1%ITU-T G.113
MOS Score (Good)3.54.0+ITU-T P.800
DSCP Marking (Voice)EF (46)EF (46)RFC 3246
DSCP Marking (Signaling)CS3 (24)AF31 (26)RFC 4594
Jitter Buffer50–200ms50–150msVendor dependent
🏗 Common VoIP Deployment Bandwidth
Scenario Concurrent Calls Codec Bandwidth Needed With 20% Buffer
Home Office2G.7110.17 Mbps0.21 Mbps
Small Business10G.7290.31 Mbps0.37 Mbps
Mid-Size Office30G.7112.61 Mbps3.13 Mbps
Call Center (100 seats)80G.7292.48 Mbps2.98 Mbps
Enterprise UC200G.72217.4 Mbps20.9 Mbps
Large Call Center500G.72915.5 Mbps18.6 Mbps
💡 Tip 1 — Count Concurrent Calls, Not Users: VoIP bandwidth is determined by the number of simultaneous active calls, not the total number of phones or users. A 100-seat office may only have 20–30 concurrent calls during peak hours. Use your busiest hour call volume for accurate planning.
💡 Tip 2 — Header Overhead Is Significant for Low-Bitrate Codecs: With G.729 at 8 kbps raw, the RTP/UDP/IP headers add ~23 kbps — nearly tripling the actual bandwidth per call. Always calculate total per-call bandwidth including headers, RTCP (~5%), and your deployment's signaling overhead. The 20% buffer is your safety net for burst traffic and retransmissions.

 

voip requires internet connection to work, so the bandwidth matters, but here the best part: voip calls almost do not use a lot of bandwidth. It eats only around 80 to 100 kbps for one call, what really is little. To have a notion streaming YouTube videos require almost six times more, we talk about 3 to 4 Mbps for download.

When your net can handle Netflix or browse social nets without problems, you already have more than enough space for voip.

How much internet speed does VoIP need

The real bandwidth that a call uses depend fully on the codec that you choose. Codecs pack and unpack voice data, so that they easily cross the internet. G.711 use only around 80 kbps, 64 kbps for the outgoing voice and another 16 kbps for the incoming.

Then come G.729 that is much less needy with only 24 kbps. Opus sits a bit in the middle at 40 kbps, although it adjusts according to the settings. Interesting fact is, that many new HD codecs adapt instead of have fixed speed, so they often consume less than expected.

Widely, the more bandwidth a codec requires, the clearer sounds your call.

From my tests with real setups, plan around 85 to 100 kbps for one call at a time works well. Simple DSL connection already can work with minimal 0.384 Mbps up and down. I found one system with more then 150 users on 10 Mbps fiber, and that was only for voip.

It shows how little those calls really need at home.

Here the main point however: only the speed does not decide the quality of call. Jitter, packet loss and buffer delay affect more than the basic speed, according to my experience. If you book bandwidth specially for voip in your router, it does big difference.

Put limit on all other stuff, while you keep space for the phone, and the choppy audio will disappear. Set QoS to prioritise voip traffic above other data is clear advantage. Without turned on QoS, someone that downloads big files can destroy your call quality.

Running around 16 calls at once with G.729 take almost half a megabit, like this it is.

Fixed phones depend on physical setup with limited options, while voip works on modern internet tech, that is much more flexible and cheap. Platforms like Microsoft Teams, Zoom Phone and Google Voice give big IP-based voice network power for businesses. Toll-free voip numbers are easy to install and manage, so contact centers and new clients prefer them.

Emergency services are available through voip in around 40 countries with automatic setting. You get also main features like caller ID and support for SMS, that already is included.

Good net at 10 to 15 Mbps download and 5 to 15 Mbps upload wellserve most homes, even during use of HD codecs.

VoIP Bandwidth Calculator – Plan Your Call Capacity Right

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