Latency and jitter test
Speed is not everything. Measure round-trip latency and jitter, the numbers that decide whether calls and video hold up.
Measure latency and jitter
Measures round-trip time from your browser to Purple's server over 20 samples. Jitter is the average change between consecutive round trips. Low and stable values matter most for calls, video, and gaming.
Latency, jitter, and why they matter
Latency is how long a round trip takes; jitter is how much that time varies between packets. A connection can post a high download figure and still drop video calls if jitter is high. Real-time traffic, voice, conferencing, and gaming, depends on low and stable latency far more than peak throughput.
Rough guide
- Under 30 ms latency with low jitter is excellent for real-time traffic.
- Jitter above 30 ms tends to cause audible glitches on calls.
- Spiky results often point to WiFi interference or an overloaded uplink.
VoIP and video conference quality benchmarks
Interactive voice and video communication standards (ITU-T G.114) define strict tolerance thresholds for packet transit times:
| Stability tier | Avg latency (RTT) | Packet jitter | VoIP MOS score | Call experience |
|---|---|---|---|---|
| Excellent | < 45 ms | < 10 ms | 4.3 - 4.5 | Crystal clear audio, instantaneous response, zero video lag. |
| Good | 45 - 100 ms | 10 - 20 ms | 4.0 - 4.2 | High quality conversational speech with unnoticeable delay. |
| Acceptable | 100 - 150 ms | 20 - 30 ms | 3.6 - 3.9 | Minor audio delay; occasional talk-over or video resolution drops. |
| Poor / Unusable | > 150 ms | > 30 ms | < 3.5 | Noticeable robotic audio, broken syllables, and dropped video frames. |
Enterprise WiFi QoS and traffic prioritisation classes
To eliminate packet jitter across dense physical venues, wireless access points and network switches rely on Wi-Fi Multimedia (WMM) and Differentiated Services Code Point (DSCP) tagging:
| Traffic category | WMM / 802.11e queue | DSCP value | Handling policy |
|---|---|---|---|
| Voice (VoIP / VoWiFi) | AC_VO (UP 6, 7) | 46 (Expedited Forwarding - EF) | Strict priority queue; minimum latency and jitter guarantee. |
| Interactive video | AC_VI (UP 4, 5) | 34 (Assured Forwarding - AF41) | High-priority queue with bounded delay for Zoom and Teams. |
| Best effort (guest web) | AC_BE (UP 0, 3) | 0 (Default / CS0) | Standard fair queuing for browsing, social media, and email. |
| Background / bulk file sync | AC_BK (UP 1, 2) | 8 (CS1 / Low Priority) | Yields transmission slots whenever higher priority queues are active. |
Causes of high WiFi jitter and network remediation
When latency tests show high standard deviation or jitter spikes on wireless connections, consider these primary engineering causes:
- RF Co-Channel Interference (CCI): Overlapping access points operating on the same 2.4 GHz or 5 GHz channels force devices to wait for clear airtime, creating erratic delays. Remediate by auditing channel plans and adjusting AP transmit power.
- Bufferbloat at the Gateway: Routers with oversized packet buffers hold queues during heavy downloads, delaying small VoIP packets. Remediate with Active Queue Management (AQM) algorithms like FQ-CoDel or CAKE.
- Unconstrained Guest WiFi Bandwidth: A single guest running cloud backups or 4K streams can saturate uplink bandwidth. Purple enforces granular per-user rate limits and dynamic bandwidth tiers to ensure voice traffic remains uninterrupted.
- Sticky Client Roaming: Devices holding onto weak distant access points rather than roaming to closer APs experience severe packet loss and jitter. Remediate by enabling 802.11k, 802.11v, and 802.11r fast transition roaming.
Frequently asked questions about latency and jitter
What is the difference between latency and jitter in WiFi networks?
Latency (ping) measures the total round-trip time (RTT) for a data packet to travel from a client device to a server and back, measured in milliseconds (ms). Jitter measures the statistical variation or irregularity in that latency over time between consecutive packets. High bandwidth alone cannot compensate for high jitter, which causes packet reordering and dropped frames in real-time streams.
What are acceptable latency and jitter thresholds for VoIP and video calls?
For clear VoIP audio and smooth video conferencing (Zoom, Microsoft Teams, Webex), one-way latency should remain under 100 ms (or round-trip latency under 150 ms) and jitter should remain strictly below 20 ms with zero packet loss. When jitter exceeds 30 ms, jitter buffers overflow, leading to robotic voice artifacts, audio clipping, and frozen video frames.
Why does high internet bandwidth not prevent packet jitter on WiFi?
Bandwidth defines maximum data transfer capacity (throughput in Mbps), while latency and jitter define timing consistency. Jitter on WiFi is commonly caused by radio frequency (RF) co-channel interference, bufferbloat in routers, channel contention during peak client density, or lack of Quality of Service (QoS) traffic prioritisation.
How can network administrators reduce latency and jitter on enterprise WiFi?
Network administrators reduce latency and jitter by: 1) Enforcing Quality of Service (QoS) with 802.11e / WMM priority queues (Voice UP 6, Video UP 5); 2) Migrating high-bandwidth devices to clean 5 GHz and 6 GHz channels; 3) Enabling band steering and airtime fairness; 4) Configuring traffic shaping and bandwidth limits on guest SSIDs to prevent bulk downloads from choking interactive traffic.
What is a VoIP MOS score and how does it relate to latency and jitter?
The Mean Opinion Score (MOS) is an ITU-T standard rating scale from 1 (unacceptable) to 5 (excellent) that quantifies perceived voice quality. Low latency (<50 ms) and minimal jitter (<10 ms) achieve a MOS score of 4.3 to 4.5. As latency crosses 150 ms or jitter exceeds 30 ms, the MOS score drops below 3.5, signaling noticeable call degradation.
Enterprise WiFi performance guides and network tools
Explore technical blueprints and interactive calculators for optimizing venue wireless networks:
WiFi analytics guide
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Enterprise WiFi security guide
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