How is the call quality of a VoIP phone?

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illustration of someone speaking into a microphone on a VoIP call

When making a Voice over IP (VoIP) phone call, the sound of your voice is broken into thousands of packets. These packets travel various paths on the Internet to your service provider, and on to their final destination, where they are reassembled. Many factors can affect packets on this journey, and thus impact the quality of the call. The three most common: Latency, Jitter, and Packet Loss.

  • Latency – The time it takes a voice packet to reach its destination is called Latency. Latency is measured in milliseconds (ms)—thousandths of a second. Latency of 150ms is barely noticeable and generally acceptable. Latency higher than 150ms adversely affects VoIP quality, while latency higher than 300ms is generally unacceptable.
  • Jitter – Jitter measures the variation in the arrival time of those individual packets making their way along various routes over the Internet. Jitter can be caused by Internet congestion, timing drift, or Internet route changes. Jitter is measured in milliseconds (ms)—thousandths of a second. Jitter greater than 50ms can result in packet loss and degraded voice quality. 
  • Packet Loss (Data Loss) – Packets are sent over the Internet and reassembled at their destination. Packet loss occurs when some packets are dropped by congested network routers or switches, or discarded by the jitter buffer. If you miss one out of every 10 words, or 10 words all at once, chances are you won’t understand the conversation.

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