VoIP jitter is the variation in the arrival time of voice packets. For good call quality, the common targets are jitter under 30 ms, one-way latency under 150 ms (the ITU-T G.114 guideline) and packet loss under 1%. Above those levels, callers start to hear choppy audio, gaps and talk-over.
What is jitter in VoIP?
A VoIP call sends audio in small packets, usually one every 20 ms. If the network delivered every packet exactly 20 ms apart, there would be no jitter. In reality packets queue in routers and take different paths, so some arrive early and some late. Jitter measures that variation. RTP endpoints calculate it with the interarrival jitter formula in RFC 3550 and report it in RTCP, which is why most phones, PBXs and monitoring tools can show it per call.
What is acceptable jitter, latency and packet loss for VoIP?
| Metric | Good | Noticeable | Poor |
|---|---|---|---|
| One-way latency | Under 150 ms | 150 to 400 ms | Over 400 ms |
| Jitter | Under 30 ms | 30 to 50 ms | Over 50 ms |
| Packet loss | Under 1% | 1 to 3% | Over 3% |
The latency figures come from ITU-T G.114, which says most applications are not significantly affected below 150 ms one way and that 400 ms should not be exceeded for general network planning. The jitter and packet loss thresholds are widely used industry targets rather than formal standards. Codecs also matter: G.729 has built-in packet loss concealment, while G.711 relies on the endpoint adding it (see G.711 vs G.729).
What is a jitter buffer?
A jitter buffer holds incoming packets for a short time and plays them out at a steady rate, so variable arrival times turn into smooth audio. The trade-off is delay: a 60 ms buffer absorbs 60 ms of jitter but adds 60 ms of latency. Most phones and gateways use an adaptive buffer that grows when jitter rises and shrinks when the network is calm. Packets that arrive after the buffer has played their slot are discarded, so very high jitter looks like packet loss to the listener.
What causes VoIP jitter and latency?
- Congested links: uploads, backups or video calls on the same internet line fill router queues.
- No QoS: voice packets are not prioritised over data (DSCP EF marking is the usual setting for RTP).
- Wi-Fi: retransmissions and interference add variable delay; wired phones are more stable.
- Distance and routing: every 1,000 km of fibre adds about 5 ms one way, and indirect paths add more.
- Transcoding and long carrier chains: each media hop adds processing delay. Wholesale routes that pass through several carriers can add tens of milliseconds.
How do you test VoIP jitter and packet loss?
Start with the call statistics your PBX or phone already records: most report jitter, loss and an estimated MOS score per call. A packet capture of RTP (Wireshark has an RTP stream analysis) gives exact values for one call. To separate your network from the route, run a speed and jitter test toward the provider's media IP, then place test calls on two different routes to the same destination. If both suffer, the issue is local; if only one does, it is the route. Our guide on how to test a SIP trunk lists the checks.
How do you reduce jitter and latency?
- Enable QoS on your router and switches and mark RTP as EF.
- Give voice its own VLAN or a dedicated link, and limit large uploads during call hours.
- Use wired connections for phones and call center agents where possible.
- Connect to the provider's nearest point of presence.
- Avoid unnecessary transcoding by agreeing one codec end to end.
- Ask the supplier about the route path: a direct or short route usually has lower latency than one passing through several carriers.
Network problems such as NAT and firewalls cause a different symptom, audio that is missing entirely in one direction; see one-way audio and dropped calls.
Routes with stable voice quality
If your own network is clean and one destination still suffers from latency or loss, the route is the problem. VoIPMatcher sources VoIP termination routes from its carriers and manages them, including quality checks. Request a VoIP route and tell us the quality you need.
Frequently asked questions
Is 50 ms jitter bad for VoIP?
It is at the edge. Most networks aim for jitter below 30 ms. At around 50 ms an adaptive jitter buffer can usually still smooth the audio, but it adds delay, and spikes beyond the buffer size cause choppy speech. If jitter stays at 50 ms or more, look for congestion or Wi-Fi problems on the path.
What is the difference between latency and jitter?
Latency is how long a voice packet takes to travel from speaker to listener, which causes talk-over when it is high. Jitter is how much that travel time varies from packet to packet, which causes choppy or broken audio. A link can have high latency but low jitter, such as a stable satellite connection.
How much packet loss is acceptable for VoIP?
Below 1% is the usual target for business calls and is rarely noticed. Between 1 and 3% many listeners hear clipped words, especially in bursts. Above 3% calls become hard to follow. Packet loss concealment in the codec or the phone helps with isolated losses but cannot fix sustained loss.
Does a VPN increase VoIP latency?
Usually a little. A VPN adds encryption overhead and may send traffic through a server far from both callers, which can add tens of milliseconds and sometimes more jitter. If calls get worse on a VPN, try a split tunnel that sends voice traffic directly, or a VPN server closer to the provider.
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