Voice quality, or Quality of Service, refers to the clarity of your phone call over a network. The "how good do I sound" approach to measuring the VoIP network. There are many factors that go into the end quality of a phone call, including Latency, Jitter, Packet Loss, and an overall combination of network equipment and bandwidth. We'll attempt to discuss each of these terms in detail. How a VoIP provider deals with these issues can be the difference between the "clearest" phone call you've ever made and one that sounds like the person is driving through a wind tunnel.
The three most common quality issues affecting VoIP are:
- Jitter, and
- Packet Loss
Latency generally refers to the physical distance that your phone call must travel to reach your service provider. When you make a phone call with VoIP, the sounds you make are cut into thousands of little pieces, called Packets, and then sent through the Internet to your service provider. These packets travel so fast that the process of traveling and reassembling them to the phone at the other end of the conversation generally takes milliseconds.
Usually, most US residents are not affected by Latency with their VoIP providers. If the roundtrip travel time of the packet takes more then 250 milliseconds the quality of the communication may experience some issues due to Latency. Most commonly, this occurs when trying to connect to a US service provider from an International location. Latency can occur in both VoIP and traditional phone systems.
Many VoIP providers have established multiple hosts to reduce Latency and provide a quick connection from any location. One of the benefits of using VoIP over traditional phone systems is that Internet speed is constantly increasing, helping to keep Latency down. Additionally, many VoIP companies provide service centers located in specific areas to ensure Latency is low, regardless of your location.
When packets are received with a timing variation from when they were sent, a quality issue of Jitter may be noticed. When Jitter occurs, participants on the call will notice a delay in phone conversation. You may have experienced this with your traditional phone service from time to time. Many VoIP providers reduce or eliminate Jitter by controlling for Jitter and time issues within their networking equipment.
In VoIP systems, Packet Loss can take place when a large amount of network traffic hits the same Internet connection. When talking on a VoIP system, Packet Loss can be identified with an echo or tin-like sound. Packet Loss is most commonly measured in percentages. For VoIP use, packet loss should not exceed 1%. A one percent packet loss will result in a skip or clipping approximately once every three minutes.
VoIP customers can help to reduce Packet Loss by reducing "high traffic" tasks (such as uploading files or sending emails with large attachments) while on the telephone.
VoIP and Sound Quality
With VoIP, the transmission of you voice is conducted digitally over the Internet. Digital transmissions most often result in increased sound clarity and quality. Many users of VoIP immediately notice the reduction of background hissing sounds or humming experienced with traditional telephone lines. Some quality issues, such as Packet Loss, can be controlled and prevented by customer activity. You can also improve your VoIP quality by choosing a service provider who has committed resources to combat Latency, Packet Loss, and Jitter.