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What Is Internet Jitter? VoIP, Latency, & Jitter

by Margaret Joy

Jitter, latency, and packet loss are a natural part of WebRTC, but what are they and how can you manage them in hosted VoIP calls?

There are a few natural enemies to streaming media in real-time (aka WebRTC) and chief among them are jitter and latency. Before you panic, these nuisances are easily dealt with. 

So what is jitter, and what is latency? And for that matter, what is packet loss? The former terms often come as a packet deal (yes, we fully intended that pun).

Whenever you’re streaming something in real-time, whether it’s your local game or, more to the point of this blog, a video call for work, you’ve probably noticed some hiccups in your feed. 

These hiccups appear as though your monitor blinked in the middle of the call and missed a blip of the stream. The technical term for these blips is packet loss. Insufficient bandwidth leads to packet loss. Packet loss leads to jitter. Jitter leads to the dark side—oops, we mean latency. Now, you’re going to experience a little bit of these all the time, and that’s normal! It only becomes an issue when they add up to noticeable levels.

Lag is another term for jitter.
  • Packet: A collection of bytes of digital data that can be transmitted over the Internet.
  • Packet Loss: When packets are lost during transmission of a VoIP call, call quality degrades.
  • Jitter: Variation in packet transit delays. As the speed of transmission fluctuates, individual packets of data can arrive out of sync, harming call quality (e.g., choppy sound).
  • Latency: Amount of time it takes for sound to travel from one call participant to another. A latency of 150ms is barely noticeable, so it is acceptable.

Acceptable Jitter and Latency for VoIP

We know, it feels odd to say there are acceptable levels of call issues, but that’s what we mean when we say they’re normal—to a point. Jitter and latency are measured in milliseconds. And while the odd dropped packet is typical, packets are so small that the human senses honestly don’t pick up on a few drops here and there.  

A key step in setting up your business VoIP service is to run a voip latency and jitter test on your office network to double-check it’s capable of handling VoIP traffic. Luckily today, decades after Internet communications debuted, it’s practically guaranteed that any system can smoothly handle cloud VoIP without sacrificing quality. We’re in the 5G era after all, when streaming HD movies on your phone is pretty commonplace. 

On the bright side, since these are well-known issues, the fixes are equally well known. The quickest fix in the moment is to just turn off your video if a call gets choppy. But to check on a deeper level we have solidly documented a few tips and tricks which you can find in our blog:

VoIP Troubleshooting: 4 Common VoIP Problems & How to Solve Them

Poor VoIP Call Quality? Try a Ping Test

Quick Tips on How to Improve VoIP Call Quality

Latency in Business VoIP - Why It's So Important


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