Jitter
Jitter is the variation in packet arrival times. If packets arrive at inconsistent intervals (5ms, 12ms, 3ms, 20ms), jitter is high. VoIP and video streaming are sensitive to jitter – inconsistent delivery causes choppy audio and video stuttering. Jitter buffers smooth out variation at the cost of added latency.
Definition
Jitter measures the inconsistency of packet inter-arrival times. In a perfect network, packets sent at 20ms intervals arrive at 20ms intervals. Real networks introduce variable queuing delays at each hop, causing some packets to arrive early and others late relative to their expected schedule. Jitter is typically measured as the mean deviation of RTT or one-way delay. VoIP codecs (G.711, Opus) expect packets every 20ms. If jitter exceeds the jitter buffer size, packets arrive too late to be played and are discarded – causing audio dropouts. Jitter buffers add intentional latency (typically 20-80ms) to absorb variation, trading latency for smooth playback. Network causes of jitter include: router queue depth variation, WiFi contention, traffic policing/shaping, and route flapping. QoS marking (DSCP EF for voice) with priority queuing on managed networks reduces jitter.
Examples
- –ping -c 100 host | tail -1 shows mdev (mean deviation = jitter)
- –iperf3 -c host -u -b 1M reports jitter for UDP streams
- –VoIP acceptable jitter: <30ms. Gaming: <15ms
Related Protocols
Related Terms
Latency
Latency is the time delay between a request being sent and the first byte of a response being received. Measured in milliseconds. Network latency is dominated by the speed of light over physical distance (1ms per 100km), plus queuing and processing delays.
Throughput
Throughput is the actual data transfer rate achieved on a network connection, measured in bits per second. Throughput is always less than bandwidth due to protocol overhead, congestion, packet loss, and retransmissions. TCP throughput is bounded by: min(rwnd, cwnd) / RTT.
Packet Loss
Packet loss occurs when transmitted packets fail to reach their destination. Causes include congestion (buffer overflow), CRC errors (physical layer noise), and intentional drops (QoS policing, firewall rules). TCP retransmits lost packets automatically. UDP does not – the application must handle loss.