Glossary

Packet loss: definition and a practical example

Short answer: Packet loss is the fraction of a VoIP call's packets that never reach the destination or arrive too late to be played, causing gaps in the audio.

In Portuguese: Perda de pacotes

What is packet loss?

Packet loss is the share of voice packets that leave one end of the call and never reach the other. On a VoIP call, speech is cut into 20 ms chunks and each chunk travels in an RTP packet. If a packet is lost, that stretch of sound does not exist at the destination. The rate is expressed as a percentage: packets lost divided by packets sent.

Packets get lost through congestion, when a router discards what does not fit in its queue; through errors on radio or Wi-Fi links; and through excessive jitter, when a packet arrives after its time to play and the receiver throws it away. Because RTP runs over UDP, there is no retransmission: in voice, a late packet is no longer useful.

Codecs try to mask the failure with packet loss concealment, which repeats or interpolates the neighboring sound. That works for isolated packets. In bursts, when several packets in a row go missing, concealment has nothing to copy and the listener hears clipped words.

In practice

With 20 ms packets, each direction of the call sends 50 packets per second, or 3,000 per minute.

Loss Packets lost per minute Audio lost per minute
0.5% 15 0.3 s
1% 30 0.6 s
3% 90 1.8 s
5% 150 3.0 s

The same 30 packets lost in a minute sound different depending on how they are spread. Scattered, they are 20 ms gaps that concealment hides. Together, they are 0.6 s of silence all at once, enough to wipe out a word like “confirmed”.

How Monitor uses this

The data circuit agent that 2N5 installs at each site measures packet loss, latency and jitter on that location’s internet link, with charts for the last 100 hours. On a voice call, lost packets show up as choppy audio and pull down the measured MOS-LQO. See data circuits.

Frequently asked questions

How much packet loss can a VoIP call handle?

It depends on the codec and on how the loss is distributed. Codecs with loss concealment mask isolated packets, and 1% spread across the call can go almost unnoticed. The same rate concentrated in bursts, several packets in a row, wipes out whole syllables. That is why the average rate alone is not enough: you have to look at the audio that arrived.

Are packet loss and jitter related?

Yes. When jitter exceeds the receiver's buffer size, packets that arrive late are discarded. To the listener, a packet that arrived too late and a packet that never arrived have the same result: a gap. Many devices count both together in total loss.

Know before your customers do

Monitor calls every voice circuit all day and delivers each failure with the SIP code and the Q.850 cause as evidence.