Jitter vs. Latency: Key Differences and How They Impact Performance

Jitter in Audio and Video: Why It Matters for Real-Time Streaming

What jitter is

Jitter is short-term variability in packet arrival times on a network (timing variation). In audio/video streaming, jitter means audio or video packets arrive earlier or later than expected relative to a steady playout schedule.

How jitter affects real-time streaming

• Audio glitches: dropped or repeated samples, pops, or momentary silence when packets miss their playout window.
• Video stuttering: frames freeze, skip, or display out of order when frame packets arrive irregularly.
• Lip-sync issues: inconsistent delays between audio and video streams cause noticeable desynchronization.
• Increased buffering: to hide jitter, players add buffer delay, which raises end-to-end latency—problematic for live interactive use (e.g., calls, gaming, live broadcasts).

Common causes

• Network congestion: queues and retransmissions introduce variable delays.
• Route variability: different network paths have different delays.
• Wireless interference: retransmits and variable airtime on Wi‑Fi or cellular.
• CPU scheduling and I/O delays: on sender or receiver devices, causing irregular packet processing.
• QoS not applied: lack of prioritization lets real-time packets be delayed behind bulk traffic.

How streaming systems handle jitter

• Jitter buffer: temporarily stores incoming packets and delivers them at steady intervals. Size trades off between dropouts (too small) and added latency (too large).
• Adaptive bitrate (ABR): reduces bitrate under poor conditions to lower packetization and retransmits.
• Forward error correction (FEC): sends redundant data so missing packets can be reconstructed without retransmission.
• Packet pacing and traffic shaping: sender spaces packets evenly to avoid bursts.
• Network QoS / DiffServ: marks real-time packets for priority handling in routers and switches.

Practical mitigation steps (sender, network, receiver)

1. Prioritize traffic: enable QoS/DSCP for RTP/RTCP/QUIC streams on routers.
2. Increase jitter buffer moderately: tune buffer size for acceptable latency vs. smoothness.
3. Use FEC or retransmission strategies: e.g., RTX for selective retransmit, FEC for one-way streams.
4. Reduce network congestion: limit competing bulk transfers, use bandwidth reservation if available.
5. Prefer wired or higher-quality wireless: Ethernet or 5 GHz/6 GHz Wi‑