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To ensure the best sound quality during calls, ThreePBX uses uncompressed audio codecs (e.g. G.711-ulaw) delivered through Real-time Transport Protocol (RTP); which allows the audio to stream in real-time. As a result, your internet connection is the most important component to call quality.

The following are important internet requirements that can affect the quality of your calls.

For those who like a more hands-on approach, you can test for these metrics by using sites like Test to a nearby location to get an idea of what your connection looks like during peak hours. (ThreePBX will host your services in a cloud location near you! )


Requirement: ≈100kbps up/down for each active call

The rate at which data is carried over the internet from one point to another in a given time period (usually a second). Each active call uses approximately 100kbps of bandwidth for uploading and downloading call data.


What the other person hears. Audio is uploaded in real-time from your phone to ThreePBX using your internet connection's upload speed.

Note: The upload connection is more prone to call quality issues because upload bandwidth is less available on most internet connection types.

What you hear. Audio is downloaded in real-time from ThreePBX to your phone using your internet connection's download speed.

While the bandwidth requirement is minimal, you have to take into account how many calls are taking place simultaneously at any given time to ensure call quality is maintained. For example, if there were 10 phone calls happening at the same time, your bandwidth requirement would be 1000kbps up/down. Not only that, consider what other types of internet activity (file transfers, video streaming, internet browsing, etc.) are taking place on every computer or mobile device within your network.

Because each device is fighting for bandwidth, Quality of Service (QoS) can be implemented to dedicate bandwidth to phones during periods when internet usage may be saturating your connection.


Requirement: <100ms

The average time it takes packets (audio) to travel from Point A (phone) to Point B (ThreePBX) and back. Many people, including internet service providers (ISP), only consider bandwidth when evaluating internet speeds. However, that is only half the picture. Bandwidth only shows how much internet traffic can be pushed through; where latency shows how fast that traffic arrives at its destination.

Think about driving on the freeway. Bandwidth represents the number of lanes that are available—if you have more lanes, more traffic can be pushed through and the likelihood of a traffic jam is reduced. Latency represents how fast you drive—it doesn't matter how many lanes there are if other things are slowing you down (inclement weather, gravel, potholes, etc.).

To avoid issues with call delay and call quality, it is recommended that latency does not exceed 100ms on average (use a ping test to analyze latency).

Jitter (Packet Delay)

Requirement: <10ms

The change in the amount of time it takes for one packet (audio) to move from Point A (phone) to Point B (ThreePBX). When you are checking your email or casually browsing the web, it doesn't really matter when packets arrive or if they arrive in order—in most cases you will never notice. But when you are streaming media, like a phone call, this packet precision becomes extremely important. If there is excessive jitter, packets will be dropped and call quality will be affected.

Jitter close to 0ms is ideal, but it should not exceed 10ms.

Packet Loss

Requirement: <0.5%

The percentage of packets (audio) lost while traveling from Point A (phone) to Point B (ThreePBX). If packets are lost, audio will be dropped and the sound quality will be compromised.