Keeping up with our best practices, today we’re offering you more insight into successful implementation and use of your VoIP telephony system, regardless of which communication solution of you are using (direct inward dialing numbers, short duration termination, etc).

Some of the most frequent VoIP quality issues involve what are known as jitter and packet loss. We will discuss what a jitter is, how it can affect your call quality, as well as what factors might cause this connection problem.

So, what is jitter?

From a user’s side, jitter means unstable voice flow in a telephone conversation, i. e. the situation when parts of the conversation are delivered with a delay or not delivered at all. In some cases, their order might be reversed, too, causing confusion and misunderstanding on both ends.

From the technical point of view, jitter occurs when voice data packets do not arrive in a steady flow, required by codecs for sustainable playback. Usually, packets are sent from a caller at the same time intervals, imitating the landline connection, but they do not always arrive in the same order or following the same interval pattern.

In order to avoid any perceivable interruptions in a conversation, the jitter should be 20 milliseconds or less. As it increases, the connection quality drops. Especially high jitter rates will lead to packet loss (that is, when packets are not delivered at all and thus parts of the conversation are missing).

Packet loss occurs either randomly and only by single packets (called “gaps”) or in large numbers at once (called “bursts”).

What are the causes of jitter?

There are three main causes of connection jitter:

  1. Wrong application of queuing. Inappropriate storage of voice packets and the wrong order of their transmission can lead to delays.
  2. Misconfiguration. Faulty configuration of a router or a PVC might easily impede connection quality and cause a jitter.
  3. Network congestion, which might cause irregular spacing between packets.

How to prevent or fix jitter?

In order to reduce jitter and ensure higher quality of connection, networks use jitter buffers – devices that collect packets from the caller and send them to the receiving codec in the right sequence and at even time intervals. In case of packet loss, jitter buffers duplicate missing data or adds comfort white noise.

However, a jitter buffer is not a universal decision. An increase of the buffer length can help reduce stronger jitter, but it will simultaneously lead to delays in voice delivery. If a longer jitter buffer causes a 300ms delay, it will make a regular phone conversation much more difficult.

Besides using a jitter buffer, companies can examine their networks in order to see what causes the jitter and then either correct the instances of wrong configuration or allocate more bandwidth or use Priority or Law Latency Queuing.

Understanding the causes and the impacts of jitter will help you get the most of your VoIP technology provided by See what VoIP solutions we offer and rest assured that no matter which solution you choose, you will receive continuous troubleshooting and maintenance assistance. Stay tuned for more information about VoIP termination and origination services, appropriate solutions for small businesses, and useful security tips.