Logo

MonoCalc

/

Packet Loss Impact Calculator

Networking

Link & Loss Parameters

Nominal Bandwidth

100.00 Mbps

100 Mbps

Effective TCP Throughput

3.58 Mbps

3.58% of nominal

Throughput Reduction

−96.42%

−96.42 Mbps

Packets Lost / s

83.33 pps

of 8.33 Kpps total

TCP vs UDP Throughput Comparison

TCP (Mathis model, 1 flow)

3.58 Mbps

UDP (proportional loss)

99.00 Mbps

Retransmission Overhead

1.01 Mbps

VoIP MOS Score

4.36

Excellent

MOS scale: 1 (worst) – 4.5 (excellent). Business VoIP requires ≥ 4.0. Max tolerable loss for Good VoIP: 5.53%.

Application Quality Dashboard

File Transfer

Fair

VoIP

Fair

Video Streaming

Fair

Gaming

Poor

Throughput vs Packet Loss Sensitivity

Effective throughput at your current settings across the 0–10% loss range. Current loss (1.00%) is highlighted.

Loss %TCP Throughput% of NominalUDP ThroughputMOS
0%100.00 Mbps100.0%100.00 Mbps4.41
0.1%11.31 Mbps11.3%99.90 Mbps4.40
0.5%5.06 Mbps5.1%99.50 Mbps4.38
1%3.58 Mbps3.6%99.00 Mbps4.36
2%2.53 Mbps2.5%98.00 Mbps4.29
3%2.06 Mbps2.1%97.00 Mbps4.22
5%1.60 Mbps1.6%95.00 Mbps4.05
7%1.35 Mbps1.4%93.00 Mbps3.85
10%1.13 Mbps1.1%90.00 Mbps3.51

Input

100 Mbps · 1.00% loss

TCP Effective

3.58 Mbps

MOS / VoIP Quality

4.36 — Excellent

About This Tool

Packet Loss Impact Calculator – TCP Throughput, UDP & VoIP Quality

The Packet Loss Impact Calculator quantifies how dropped packets degrade real-world network performance. Enter your link speed, packet loss rate, and round-trip time to instantly see the effective TCP throughput, UDP throughput, retransmission overhead, VoIP MOS score, and per-application quality ratings. The tool uses the Mathis TCP throughput model and the ITU-T E-Model for accurate, standards-based estimates.

Why Packet Loss Matters More Than You Think

Even a small amount of packet loss causes disproportionate harm to TCP-based applications. TCP's congestion-control algorithm (AIMD — Additive Increase Multiplicative Decrease) interprets every lost packet as a sign of network congestion and halves the transmission window. At 1% packet loss with a 40 ms RTT, a 1 Gbps link delivers only around 13 Mbps of usable TCP throughput — a 98.7% reduction. This non-linear sensitivity is why packet loss is one of the most impactful network quality metrics.

TCP Throughput: The Mathis Formula

The calculator uses the widely cited Mathis formula to estimate maximum TCP throughput:

T = (MSS / RTT) × (C / √p)

Where:

  • MSS — Maximum Segment Size (default: 1460 bytes for standard Ethernet with a 1500-byte MTU)
  • RTT — Round-Trip Time in seconds
  • p — packet loss probability (0–1)
  • C — a constant ≈ 1.22 (√(3/2), standard Mathis value)

For multiple parallel TCP flows, the aggregate throughput scales linearly with the number of flows, each probing independently. The result is clamped to the nominal link speed.

UDP Throughput and Retransmission Overhead

UDP has no retransmission mechanism. The effective UDP throughput is simply nominal × (1 − p), making it far less sensitive to loss than TCP — which is why real-time media (VoIP, video streaming, gaming) commonly uses UDP despite receiving no delivery guarantees.

The retransmission overhead shows how much of your link capacity TCP wastes re-sending dropped packets: overhead = nominal × p / (1 − p). At 5% loss, roughly 5.26% of your bandwidth is consumed by retransmissions alone, on top of the congestion-window throttling.

VoIP Quality: MOS Score via the ITU-T E-Model

The Mean Opinion Score (MOS) is an ITU-T standard for rating voice call quality on a 1–5 scale. The calculator uses a simplified E-Model estimate:

R = 93.2 − (loss% × 2.5)
MOS = 1 + 0.035R + R×(R−60)×(100−R)×7×10⁻⁶

Scores are clamped to the range 1.0–4.5. Business-grade VoIP typically requires a minimum MOS of 4.0. The tool also reports the maximum tolerable packet loss percentage that keeps VoIP at the "Good" level — typically around 0.5%.

Application Quality Thresholds

Different applications tolerate packet loss differently. The dashboard rates four traffic types against standard industry thresholds:

  • File Transfer (TCP) — Good below 1%, Poor above 2.5%. TCP eventually delivers all data, but throughput collapses severely at high loss rates.
  • VoIP — Good below 1%, Poor above 3%. Listeners hear clipping, dropouts, and robotic artefacts above 1%.
  • Video Streaming — Good below 0.5%, Poor above 2%. Buffering, pixelation, or frozen frames appear above the threshold.
  • Online Gaming — Good below 0.5%, Poor above 1%. Rubber-banding, teleporting characters, and disconnections are common at even moderate loss rates.

Using the Sensitivity Table

The Throughput vs Packet Loss Sensitivity table shows TCP and UDP throughput at standard loss percentages (0%–10%), computed with your current bandwidth and RTT settings. This lets you quickly gauge the full impact curve — for example, seeing how a link that performs adequately at 0.5% loss becomes critically impaired at 2%.

Practical Guidance for Network Engineers

When diagnosing network problems, target less than 0.1% packet loss for all production applications. Investigate loss above 0.5% immediately. Common causes include congested uplinks, faulty cables or transceivers, misconfigured QoS policies, wireless interference, and buffer bloat. Tools like ping, mtr, and iperf3 can help localise which hop in the path is dropping packets.

Use the Number of Flows advanced option to model multi-stream downloads (e.g., web browsers opening 6–8 parallel connections, or BitTorrent-style transfers). Parallel flows partially recover aggregate throughput because each independently probes the congestion window.

Frequently Asked Questions

Is the Packet Loss Impact Calculator free?

Yes, Packet Loss Impact Calculator is totally free :)

Can I use the Packet Loss Impact Calculator offline?

Yes, you can install the webapp as PWA.

Is it safe to use Packet Loss Impact Calculator?

Yes, any data related to Packet Loss Impact 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 Packet Loss Impact Calculator work?

The tool uses the Mathis TCP throughput formula — T = (MSS / RTT) × (C / √p) — to model the severe non-linear impact of packet loss on TCP performance. For UDP, it applies a simpler linear model: effective throughput = nominal bandwidth × (1 − p). It also estimates VoIP quality using a simplified ITU-T E-Model MOS score and rates per-application quality against standard industry thresholds.

Why does 1% packet loss reduce TCP throughput so dramatically?

TCP's congestion-control algorithm (AIMD) treats every lost packet as a signal of network congestion and halves the transmission window. With RTT of 40 ms and 1% loss, the Mathis model predicts TCP throughput of roughly 13 Mbps on a 1 Gbps link — a 98.7% reduction. This is normal TCP behaviour; it is not a flaw in the formula.

What is a MOS score and what values are acceptable?

MOS (Mean Opinion Score) is a 1–5 scale standardised by the ITU-T for rating voice call quality. Scores of 4.3–5.0 are Excellent, 4.0–4.3 are Good, 3.6–4.0 are Fair, and below 3.6 are Poor. For business-grade VoIP, a minimum of 4.0 is typically required. Packet loss above 1–3% generally pushes scores into the Poor range.

What are typical acceptable packet loss values for different applications?

For TCP file transfer, less than 1% loss is considered Good. VoIP tolerates up to 1% before quality degrades noticeably. Video streaming is sensitive above 0.5%. Online gaming is most stringent — even 0.5% loss can cause visible rubber-banding or lag spikes. Packet loss above 5% makes most real-time applications unusable.

Does the number of TCP flows affect the result?

Yes. Each parallel TCP flow operates its own congestion-control window. Adding more flows can partially recover aggregate throughput because each flow independently probes for bandwidth. The tool multiplies the single-flow Mathis result by the number of flows, clamped to the nominal link speed. This approximation works well for a moderate number of independent flows.

What is the difference between the TCP and UDP throughput results?

TCP actively retransmits lost packets and throttles its sending rate in response to loss, leading to a severe throughput reduction modelled by the Mathis formula. UDP has no retransmission mechanism; the effective throughput is simply reduced proportionally to the loss rate, making it better for real-time media where latency matters more than reliability.