Latency is the delay time for data to travel from sender to receiver. In telephone communications, it is experienced as the time gap between speaking and the other party hearing your voice, making it one of the most important metrics affecting call quality. ITU-T Recommendation G.114 states that one-way latency of 150 ms or less enables comfortable conversation.
The perception of latency degrades in stages. At 0-150 ms, natural conversation flows and delay is barely noticeable. At 150-300 ms, the other party's responses feel slightly delayed, creating an awkward conversational tempo. At 300-400 ms, "talk-over" (both parties starting to speak simultaneously) becomes frequent. Above 400 ms, conversational back-and-forth becomes difficult. Satellite phones experience approximately 540 ms of delay due to the round trip between Earth and geostationary satellites, creating a distinctive conversational "gap."
VoIP call latency is the sum of multiple factors. Codec encoding/decoding adds 5-40 ms, packet transmission across the network adds 10-100 ms (distance-dependent), and the receiving jitter buffer (which absorbs variations in packet arrival order) adds 20-80 ms. Domestic calls typically total 50-100 ms, but international calls can exceed 200 ms due to greater physical distance and more routers along the path.
Jitter is a concept often confused with latency. While latency is the "absolute value of delay," jitter refers to the "variation in delay." If latency is constant, call quality remains stable, but high jitter causes irregular packet arrival intervals, resulting in audio dropouts and distortion. QoS settings must address both latency minimization and jitter suppression. Even with sufficient bandwidth, router processing delays and network congestion can increase latency, making overall network design critical for VoIP deployments.