Understanding VoIP Call Quality

Factors Affecting VoIP Networks And Call Quality

So, you’ve decided to implement a VoIP solution but concerned about the quality of your calls or you’ve already implemented a VoIP solution and have quality issues - you’ve come to the right place.

Many believe the myth that a high-speed internet connection solves all quality issues instantaneously (we’re not saying it won’t) however, there are many technology and bandwidth factors that need to be considered and assessed before making that all important call to your ISP to increase your line speeds.

Below we’ll go through some of the most important factors that need to be considered:

Codec (also known as Coder-Decoder)

Codec is the all-important technology component that converts the analog speech into a digital format that can be transmitted over a data network. Different codecs will most certainly produce different results. Some might offer better call quality at the expense of data size while others will keep the size down but sacrifice call quality. Two of the most common codecs used is G711U and G729. Its high quality is achieved by not using any data compression, making it also the most bandwidth-hungry codec. This codec requires 90 Kbps of bandwidth for each conversation. The G729 codec also provides excellent bandwidth utilization and is very error-tolerant. You would however need a VoIP device or hardware that supports this codec.

Packet Loss

There’s a reason we’re covering this early on as it’s probably the biggest factor when it comes to VoIP quality. Speech is submitted over a network in loads of tiny packets. These packets contain the audio that needs to be delivered from Point A to Point B. For normal data packets, there might be time for Point B to request Point A to resend a packet that was lost in transmission however as voice is a real-time transmission, there is no time for this and results in audio lost. If it happens every now and again, it might go completely unnoticed but the higher the percentage of packet loss the higher chance there is that this will have a significant impact on your call quality.

Bandwidth

As mentioned previously, having a higher line speed could aid a call quality problem however bandwidth utilisation on the line is a very important factor when troubleshooting call quality issues.

For example, the use of LAN switches with 1 Gbps ports will give you a better call quality than one using switches with 100 Mbps ports.

Although a conversation uses an average of less than 100 Kbps of bandwidth, it can, as you’ll soon discover by reading through the other factors, suffer greatly from congestion. And since networks are very seldom dedicated to VoIP or your Voice Service, any over-utilisation will have an immediate impact on call quality.

Jitter (also known as packet delay variation)

Jitter refers to an irregularity in the delivery delay of data packets from point A to point B. Voice is transmitted over the network broken down into tiny packets (What we call RTP packets). These packets should arrive at Point B in the same order it left Point A. The reality however is that this is not always the case and results in what we call Jitter.

Very small percentages of jitter is acceptable and will most likely not affect the call quality however as the data needs to be transmitted from point A to point B in real-time a larger percentage of Jitter will result in the speech to be chopped up and you asking that dreaded question, “I’m sorry, would you mind repeating that?“.

Latency

Latency refers to any delay that data suffers from point A to point B. In an ideal situation, data should be transmitted almost instantaneously however in reality this is not the case. There are several reasons packets will be delayed which includes network congestion, route congestion or even equipment that is not functioning as it should. Latency can cause unnecessary delays in voice transmissions and is typically identified by end-users when they are talking over each other.

MOS Quality - An objective score

Call quality is a highly subjective concept. For instance, one user can find some degree of audio degradation acceptable while another might insist on crystal-clear audio. In order to add some degree of objectivity to an otherwise subjective concept, the telephony industry has come up with the concept of a Mean Opinion Score or MOS.

Mean Opinion Score gives VoIP testing a number value as an indication of the perceived quality of received voice after being transmitted and compressed using codecs. This measurement is the result of underlying network attributes that act upon data flow and is useful in predicting call quality and in determining issues that can affect VoIP quality.

MOS values range from 1 to 5 with values of 4 or higher indicating a generally satisfying call quality. Values between 3 and 4 generally indicate some level of insatisfaction among users while values below 3 indicate a mostly unsatisfactory call quality.