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VoIP Bandwidth Calculator

Networking
10–60 ms
IPv4 = 20 B header, IPv6 = 40 B header
0 = none, 0.5 = 50% silence reduction

Include Ethernet Framing (LAN)

Total Bandwidth

984.00 kbps

10 calls with VAD

Per-Call Bandwidth

98.40 kbps

without VAD

Packets / Second

50.0 pps

per call

Header Overhead

35.0%

86 B per packet

Payload vs. Protocol Overhead (per packet)

Overhead (35.0%)

Payload (65.0%)

Overhead (86 B)

Payload (160 B)

Protocol Header Breakdown

LayerBytes% of Wire
RTP Header12 B

4.9%

UDP Header8 B

3.3%

IPv4 Header20 B

8.1%

Ethernet Frame + Wire46 B

18.7%

Voice Payload160 B

65.0%

Total Wire246 B100%

Bandwidth Summary

MetricValue
Voice payload bitrate (codec)64.00 kbps
Bandwidth per call (with headers)98.40 kbps
Bandwidth per call (with VAD 0%)98.40 kbps
Total for 10 calls984.00 kbps
Total (kbps)984.00 kbps
Total (Mbps)0.9840 Mbps
RTP packet size (payload + RTP header)172 bytes

About This Tool

VoIP Bandwidth Calculator – Plan Your Call Capacity

Voice over IP (VoIP) seems simple: compress audio, wrap it in packets, send it across the network. Yet every VoIP call consumes significantly more bandwidth than the codec bitrate alone suggests. The VoIP Bandwidth Calculator accounts for the full protocol stack — RTP, UDP, IP, and optional Ethernet framing — so you can accurately size links, plan call-admission control, and compare codecs for capacity planning.

Why Codec Bitrate Understates Real Bandwidth

A G.729 call runs at 8 kbps of compressed voice. But each RTP packet carrying that voice also requires:

  • RTP header (12 bytes): Sequence number, timestamp, and synchronisation source identifier defined in RFC 3550.
  • UDP header (8 bytes): Source and destination port numbers plus checksum.
  • IPv4 header (20 bytes) or IPv6 header (40 bytes): Routing, TTL, and protocol fields. IPv6 doubles the IP overhead, relevant when planning dual-stack or pure-IPv6 deployments.
  • Ethernet framing (46 bytes on the wire): Destination MAC, source MAC, EtherType, FCS (26 bytes) plus preamble, SFD, and IFG (20 bytes). Applicable only for LAN segments; WAN links (MPLS, PPP) use different encapsulations.

For a G.729 call with 20 ms ptime over IPv4 with Ethernet, the voice payload is 20 bytes but total wire bytes per packet is 106 bytes — more than 5× the payload size. Actual bandwidth per call rises from 8 kbps to approximately 42 kbps on a LAN, or about 24 kbps without Ethernet framing on a WAN link.

Understanding Ptime

Ptime (packetization time) controls how many milliseconds of audio are packed into each RTP packet. The default for most codecs is 20 ms. Increasing ptime to 30 ms or 40 ms reduces the packet rate and therefore lowers the relative overhead percentage, but comes at a cost:

  • Higher end-to-end latency (audio algorithmic delay increases with ptime).
  • Greater sensitivity to packet loss — one lost packet means more audio missing when ptime is larger.

ITU-T G.114 recommends keeping one-way delay under 150 ms for acceptable voice quality. For most deployments, 20 ms ptime represents the best balance between overhead efficiency and latency.

Voice Activity Detection (VAD)

Natural speech is active roughly 40–50% of the time. VAD, also called silence suppression, detects pauses between words and phrases and stops transmitting RTP packets during silence, replacing them with lightweight Comfort Noise Generation (CNG) updates. This can reduce effective bandwidth by 40–50% for a typical telephone conversation.

The calculator's VAD suppression factor allows you to model different activity levels. A factor of 0.5 means the call is silent 50% of the time, halving the effective bandwidth. Use 0 for worst-case planning (no suppression), and0.5 for average conversational traffic.

Codec Selection Guide

Choosing the right codec depends on your bandwidth budget, audio quality requirements, and network characteristics:

  • G.711u/a (64 kbps): Toll-quality audio, no compression complexity, standard for PSTN interworking and corporate LANs. High bandwidth but simple implementation.
  • G.729 (8 kbps): Industry standard for bandwidth-constrained WAN and MPLS links. Good quality with moderate CPU load. Licensed in some implementations.
  • G.722 (64 kbps): Wideband (HD) audio, identical bitrate to G.711 but sampled at 16 kHz for noticeably clearer speech. Common in modern desk phones and unified communications platforms.
  • Opus (12–48 kbps shown): Highly adaptive codec used in WebRTC, Zoom, and Teams. Can vary bitrate dynamically from below 10 kbps to over 500 kbps, handles packet loss well, and supports wideband and fullband audio.
  • G.723.1 (5.3–6.3 kbps): Very low bitrate suitable for extremely constrained satellite or cellular links. Noticeable quality trade-off.

Sizing a Link for Concurrent VoIP Calls

To size a WAN link or MPLS circuit for VoIP, multiply the per-call bandwidth (including headers) by the maximum concurrent call count, then add:

  • Signalling overhead: SIP traffic for call setup and teardown typically adds 5–10 kbps per active gateway.
  • Headroom (10–20%): VoIP is highly sensitive to congestion. Reserve capacity for traffic bursts and re-transmissions.
  • QoS marking: Mark RTP packets with DSCP EF (Expedited Forwarding,46) to prioritise voice over background data traffic.

As a worked example: 50 concurrent G.711 calls over IPv4 without Ethernet (WAN scenario) at 20 ms ptime require approximately 40.8 Mbps. Adding 20% headroom and SIP signalling, a 50 Mbps dedicated voice circuit is adequate. For the same call count over LAN with Ethernet headers, budget closer to 58 Mbps.

IPv4 vs IPv6 Overhead

Moving to IPv6 adds 20 bytes per packet compared to IPv4 (40-byte vs 20-byte IP header). For G.729 at 20 ms ptime, this increases per-call bandwidth on a LAN from roughly 42 kbps to 50 kbps — a 19% increase in per-call bandwidth. At scale (100+ concurrent calls), this difference is material and should be factored into dual-stack migration planning.

Frequently Asked Questions

Is the VoIP Bandwidth Calculator free?

Yes, VoIP Bandwidth Calculator is totally free :)

Can I use the VoIP Bandwidth Calculator offline?

Yes, you can install the webapp as PWA.

Is it safe to use VoIP Bandwidth Calculator?

Yes, any data related to VoIP Bandwidth Calculator only stored in your browser (if storage required). You can simply clear browser cache to clear all the stored data. We do not store any data on server.

How does the VoIP Bandwidth Calculator work?

Select a codec, set the packet time (ptime) and number of simultaneous calls, then choose whether to include Ethernet framing and IPv4 or IPv6 headers. The tool stacks all protocol headers — RTP (12 B), UDP (8 B), IP, and optional Ethernet — on top of the voice payload to compute per-call and total bandwidth, packets per second, and header overhead percentage.

What is ptime and how does it affect bandwidth?

Ptime (packetization time) is the amount of voice audio packed into each RTP packet, typically 10–30 ms. Shorter ptime means more packets per second, higher overhead percentage, and slightly increased bandwidth consumption. Longer ptime reduces packet rate and overhead but increases end-to-end latency and sensitivity to packet loss. 20 ms is the standard default for most codecs.

Why is actual VoIP bandwidth higher than the codec bitrate?

The codec bitrate only covers compressed voice payload. Every RTP packet also carries at minimum 40 bytes of protocol headers: 12 bytes RTP + 8 bytes UDP + 20 bytes IPv4. For a G.729 call at 8 kbps with 20 ms ptime, payload is 20 bytes but total IP packet is 60 bytes — meaning protocol overhead more than doubles the actual bandwidth required to roughly 24 kbps.

What is Voice Activity Detection (VAD) and how much bandwidth does it save?

VAD (also called silence suppression) detects pauses in speech and stops sending RTP packets during silence, replacing them with Comfort Noise Generation (CNG) packets. A typical telephone conversation is active roughly 50% of the time, so enabling VAD can reduce bandwidth by 40–50%. However, aggressive VAD settings can clip the beginning of words; a 50% suppression factor is a conservative estimate.

Which codec is best for low bandwidth versus high quality?

G.729 (8 kbps) and G.723.1 (5.3–6.3 kbps) offer the lowest bandwidth per call and are popular for WAN and cellular links. G.711u/a (64 kbps) and G.722 (64 kbps, wideband) deliver higher quality and are standard on LAN and broadband connections. Opus is highly flexible, supporting 4–510 kbps and adapting to network conditions, making it ideal for WebRTC and modern UC applications.

How do I size a link for a given number of VoIP calls?

Use this calculator with your target codec, enable VAD if your system supports it, and add 10–20% headroom on top of the total bandwidth result for signalling traffic (SIP, H.323) and network jitter buffers. For example, 100 concurrent G.711 calls with Ethernet headers and no VAD require approximately 87 Mbps; with 50% VAD and 20% headroom, a 50 Mbps link is sufficient.