Free Tool

SIP Trunk & VoIP Bandwidth Calculator

Calculate how many SIP channels you need and how much internet bandwidth your VoIP deployment requires — using the Erlang B formula and real codec bandwidth figures. Supports G.711, G.729, and Opus.

No email required Instant results Erlang B formula

Your SIP trunk configuration

Peak concurrent calls

Calls per hour at peak

Average call duration (minutes)

Voice codec

Target Grade of Service (blocking %)

Include T.38 fax channels

Your SIP trunk requirements

Based on Erlang B formula and codec bandwidth figures

SIP Channels Required

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— Mbps bandwidth required

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Erlang B channels

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Total channels (inc. fax)

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Voice bandwidth

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Recommended internet speed

Calculation breakdown

Erlang B assumes no queuing (calls are blocked, not held). Recommended internet speed is 2× voice bandwidth to leave headroom for data traffic. Use DSCP QoS to prioritize voice packets.

Codec Reference

Codec comparison

Choose the right codec for your network constraints and quality requirements. Bandwidth figures include RTP headers and RTCP overhead.

Codec	Bandwidth/call	Quality	Best for
G.711 PCMU / PCMA	90 Kbps	Excellent (PSTN)	Standard SIP trunks
G.729 Compressed	30 Kbps	Good	Bandwidth-constrained sites
Opus Adaptive	30 Kbps	Excellent (adaptive)	WebRTC, UCaaS clients

How it works

How SIP channel and bandwidth requirements are calculated

SIP trunk sizing uses classical telecom engineering — the Erlang B formula — combined with codec-specific bandwidth figures. Here's what each step means.

Traffic in Erlangs

Erlang is the unit of telephone traffic intensity. Traffic (A) = calls per hour × (average duration in hours). A value of 1 Erlang means 1 channel is occupied continuously. At 100 calls/hour with 4-minute average, traffic = 100 × (4/60) = 6.67 Erlangs.

Erlang B channels

The Erlang B formula calculates the channels needed to achieve your target Grade of Service. For 6.67 Erlangs at P.01 (1% blocking), about 13 channels are required. Erlang B assumes blocked calls are lost — appropriate for most SIP trunk scenarios without queueing.

Peak concurrent override

If you know from call records that your peak simultaneous calls exceed the Erlang B result, the calculator uses the higher value. This guards against under-provisioning when traffic has high variance or burst patterns not captured by average call rates.

Codec bandwidth

Each active call uses bandwidth equal to the codec rate plus RTP/UDP/IP packet headers. G.711: ~90 Kbps/call. G.729: ~30 Kbps/call. Opus: ~30 Kbps/call at 24 Kbps audio with headers. Voice requires symmetrical bandwidth — your upload speed matters as much as download.

20% signaling overhead

SIP signaling (INVITE, ACK, BYE), RTCP quality reports, and network packet headers add approximately 20% on top of raw RTP bandwidth. Always size your connection for this overhead to prevent voice quality degradation during INVITE storms or heavy signaling bursts.

Recommended internet speed

The recommended speed is 2× the total voice bandwidth. This headroom accounts for data traffic running on the same connection and allows for traffic spikes without impacting voice quality. For dedicated VoIP links, 1.5× overhead may be sufficient with strict DSCP QoS enforcement.

FAQ

Common questions

How many SIP channels do I need?

The number of SIP channels equals the number of simultaneous calls your business needs to support at peak. Use the Erlang B formula for pure overflow (no wait queue): for 100 calls/hour at 3-min average call duration, you need about 9–11 channels to maintain a Grade of Service (blocking probability) below 1%. The formula: A = calls_per_hour × (avg_duration_min / 60) gives traffic in Erlangs, then Erlang B tables give the channels needed for target GoS.

How much bandwidth does a VoIP call use?

Bandwidth per call depends on codec: G.711 (PCMU/PCMA) uses ~90 Kbps per call including RTP headers and SIP signaling — the highest quality, uncompressed. G.729 uses ~30 Kbps per call — compressed, 3× more efficient, slight quality reduction. Opus uses ~20–50 Kbps depending on bitrate settings — best for WebRTC and variable-bandwidth scenarios. Always add 20% overhead for SIP signaling, RTCP, and packet headers.

What is Grade of Service (GoS)?

Grade of Service is the probability that a call will be blocked (receive a busy signal) because all channels are in use. The telecom standard is P.01 — a 1% blocking probability, meaning 99 out of 100 calls connect immediately. Very conservative deployments use P.005 (0.5%). The Erlang B formula calculates the channels needed to achieve a target GoS at a given traffic load.

Should I use G.711 or G.729?

G.711 is preferred when bandwidth is not a constraint: it delivers PSTN-quality audio and requires no transcoding at most carriers, which reduces latency and complexity. G.729 saves 66% of bandwidth per call and is ideal for remote offices, low-bandwidth WAN links, or high-density deployments. Use G.729 only if your carrier supports it natively — forced transcoding adds latency and degrades quality. For modern SIP trunks, G.711 is the safe default.

What internet speed do I need for VoIP?

As a rule: each G.711 call needs ~90 Kbps symmetric (up and down). Add 20% margin for jitter buffer and packet loss recovery. For 10 concurrent G.711 calls: 10 × 90 Kbps × 1.20 = 1.08 Mbps. A business with 50 users and 10 concurrent calls at peak needs at least a 10 Mbps symmetric connection dedicated to VoIP, separate from general internet traffic. Use QoS (DSCP) to prioritize voice packets.

Can I share or download my results?

Yes — inputs are encoded in the URL for sharing/bookmarking. Use "Download PDF report" for a branded printable report with your channel requirements, bandwidth calculations, and codec comparison. No email required; all calculations happen in your browser.

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