VoIP jitter is variation in packet arrival timing. RTP packets are sent at regular 20ms intervals but arrive at irregular intervals due to network queuing. Too much jitter — and the jitter buffer can't compensate — causes choppy audio identical to packet loss. Here's how to measure it, find the cause, and fix it.
Jitter is the variation in packet arrival time. If packets are sent every 20ms but arrive at 18ms, 24ms, 20ms, 31ms intervals, the interarrival variation is the jitter. The jitter buffer at the receiving endpoint holds packets briefly to smooth out this variation before playing them out — at the cost of added latency.
Jitter is measured in milliseconds and reported in RTCP Receiver Reports as the interarrival jitter value. High jitter doesn't cause audio problems directly — the jitter buffer handles it. Problems occur when jitter exceeds the jitter buffer depth, causing buffer overflow and dropped packets.
| Jitter | Impact | Status |
|---|---|---|
| < 20ms | Imperceptible, jitter buffer handles it easily | Excellent |
| 20–40ms | Acceptable, may require deeper jitter buffer | Good |
| 40–80ms | Noticeable latency increase, occasional clipping | Degraded |
| 80–150ms | Choppy audio if buffer is shallow, high latency | Poor |
| > 150ms | Severe audio issues, buffer overflow causing loss | Unacceptable |
These thresholds apply to one-way jitter. Total one-way latency for acceptable VoIP is under 150ms — jitter buffer depth contributes directly to this budget, so very deep jitter buffers that fix jitter may break latency requirements.
Jitter — packets arrive but at irregular intervals. The jitter buffer can compensate at the cost of latency. Jitter alone does not cause audio problems unless it exceeds the buffer depth.
Packet loss — packets don't arrive at all. The jitter buffer cannot compensate. The codec's PLC algorithm fills gaps but quality degrades above 1-2% loss.
High jitter without packet loss = tune the jitter buffer deeper. Packet loss without high jitter = network congestion or physical layer issue. Both together = severely congested path, QoS and capacity are the fix.
RTCP RR packets contain the interarrival jitter field — the estimated jitter at the receiver in RTP timestamp units. For audio codecs with 8000Hz clock rate, divide by 8 to get milliseconds. Wireshark displays this in the RTCP analysis window.
Use Telephony → RTP → Stream Analysis. Wireshark plots per-packet jitter over time. Spikes indicate momentary congestion events. Sustained high jitter indicates a systemic path problem.
Ping the SIP server with small packets and measure RTT variation: ping -i 0.02 -c 1000 sip.server.com. High RTT variation correlates with jitter. Not a substitute for RTCP measurement but useful for quick network path assessment.
QoS: Mark RTP packets DSCP EF (46) and configure routers to give EF traffic strict priority queuing (LLQ). This eliminates queuing jitter caused by competing traffic. Single highest-impact fix.
Separate VLANs: Put VoIP on a dedicated VLAN with its own uplink QoS policy. Prevents a large data transfer on the same switch port from competing with active calls.
Jitter buffer tuning: Increase jitter buffer target depth to 60-80ms for high-jitter paths. Use adaptive jitter buffers that automatically adjust. Don't increase maximum buffer depth above 150ms or you'll introduce noticeable latency.
Fix Wi-Fi: Use 5GHz, select non-overlapping channels, ensure adequate AP coverage, and consider dedicated VoIP-optimized APs for high-density environments.
Dedicated SIP trunking: Use a SIP trunk provider with direct exchange points rather than internet transit. MPLS or SD-WAN with traffic shaping provides more consistent path latency than best-effort internet.
Acceptable jitter for VoIP is under 40ms. Below 20ms is ideal and imperceptible. Between 40-80ms users notice degraded quality. Above 150ms jitter causes severe audio issues as the jitter buffer overflows and packets are dropped.
The most common causes of VoIP jitter are network congestion on a shared link, Wi-Fi contention and retransmissions, variable routing via ECMP, server-side CPU scheduling issues, and deep packet inspection appliances adding variable processing delay.
Fix VoIP jitter by implementing QoS with DSCP EF (46) marking to prioritize RTP traffic, separating voice onto a dedicated VLAN, increasing the jitter buffer target depth to 60-80ms, switching to 5GHz Wi-Fi, and using a SIP trunk provider with direct peering rather than internet transit.
Paste your SIP trace or upload your PCAP. SIPSymposium extracts RTCP jitter reports, analyzes per-packet interarrival variation, and identifies whether jitter is network-side, server-side, or Wi-Fi related.