Will doubling our internet speed fix choppy calls?

Only if raw capacity is the binding constraint — which the two-point capture will show. On a saturated-but-fast link, more speed shrinks but does not remove the queueing problem; QoS plus sane scheduling fixes most offices at zero bandwidth cost.

Why do calls sound fine in the morning and awful after lunch?

Load. Something fills the uplink or the LAN in the afternoon (backups, syncs, a neighboring tenant, streaming). That time pattern is diagnostic gold — see the peak-hours guide for the full workflow.

What jitter buffer setting should phones use?

Adaptive buffers in the 30–60 ms range suit most networks. Bigger buffers mask jitter but add delay (talk-over); smaller buffers sound tight until the first spike. If you must raise the buffer to hide jitter, you have a network fault to fix, not a setting to celebrate.

Troubleshooting · Symptoms

Choppy Audio and Jitter: A Diagnosis Guide

By the VoIP Cost Guide Editorial Team | Last updated: October 2026

Words arrive scrambled, syllables vanish, voices sound like they're underwater. Choppy audio is not 'the internet being slow' — it is packets arriving at irregular intervals (jitter) or not arriving at all (loss), and the receiver's jitter buffer running dry. The good news: jitter is measurable in any capture, and its cause is almost always a queue somewhere between the two talkers.

This guide shows you how to measure it, locate the congested point, and apply the fixes in the order that works — because 'buy more bandwidth' is the most expensive and least reliable item on the list.

The numbers that define 'choppy'

Voice is sent as a metronome: typically one packet every 20 ms. The receiver holds a small jitter buffer (30–60 ms is common) to smooth arrival variation. Quality collapses when:

Metric	Good	Noticeable	Broken
Jitter (variation in arrival time)	< 10 ms	10–30 ms	> 30 ms sustained
Packet loss (after concealment)	< 0.5%	0.5–2%	> 2–3%
One-way latency	< 100 ms	100–150 ms	> 150 ms (talk-over begins)

Where jitter is born

Packets don't jitter in the middle of an empty network; they jitter in queues — the moment a link is full and a voice packet waits behind a bulk transfer. The usual crime scenes:

Uplink saturation. Asymmetric connections (cable, DSL) have a small upload pipe; a cloud backup or photo sync fills it and voice queues behind it. The tell: calls degrade when someone uploads.
Wi-Fi. Contended airtime, roaming between access points, and a phone at the edge of coverage. Desk phones on weak Wi-Fi are jitter machines.
Bufferbloat. Oversized buffers in cheap routers/modems hold packets for hundreds of milliseconds under load — speed tests look fine while latency balloons.
Missing or broken QoS. DSCP markings stripped at a boundary, or a QoS policy configured on a device the traffic doesn't cross.
A genuinely congested WAN/ISP segment — real, but last on the list; prove it with captures at two points before accepting it.

Measure, locate, fix

Measure first: in a capture of a bad call, open the RTP stream analysis — note jitter and loss figures. Also capture a good call for contrast (different time of day is typical).
Correlate with activity: does choppiness track uploads, backups, or a specific hour? A pattern is a diagnosis half-made (see the peak-hours guide).
Capture at two points (LAN side and WAN side of your router) during the same call window. Jitter present on the LAN side already → internal problem (Wi-Fi, local congestion). Clean on LAN, dirty on WAN → your router's queueing or the ISP path.
Apply QoS properly: mark voice EF (DSCP 46) and signaling CS3, configure priority queuing for voice on the WAN edge, and confirm markings survive to the provider (see the QoS guide).
Tame the elephants: schedule backups and syncs off-hours, or rate-limit bulk classes below voice in the same QoS policy.
Get phones off marginal Wi-Fi: wire them, or fix coverage; no QoS policy repairs bad airtime.
Verify with the same measurement: repeat the capture in the previously-bad window and confirm jitter back under ~10 ms.

Frequently asked questions

Will doubling our internet speed fix choppy calls?: Only if raw capacity is the binding constraint — which the two-point capture will show. On a saturated-but-fast link, more speed shrinks but does not remove the queueing problem; QoS plus sane scheduling fixes most offices at zero bandwidth cost.
Why do calls sound fine in the morning and awful after lunch?: Load. Something fills the uplink or the LAN in the afternoon (backups, syncs, a neighboring tenant, streaming). That time pattern is diagnostic gold — see the peak-hours guide for the full workflow.
What jitter buffer setting should phones use?: Adaptive buffers in the 30–60 ms range suit most networks. Bigger buffers mask jitter but add delay (talk-over); smaller buffers sound tight until the first spike. If you must raise the buffer to hide jitter, you have a network fault to fix, not a setting to celebrate.

Sources and further reading

Related guides

Disclaimer: This guide is educational information based on published VoIP standards and professional practice, not engineering advice for your specific system. VoIP pricing, taxes and fees change and vary by provider and location — verify current figures with providers before purchasing. If your phone system supports emergency calling, work with your provider or a qualified VoIP engineer for changes that affect E911.