Call quality and reliability has always been used by critics of Voice over Internet Protocol (VoIP) technology to discredit VoIP and discourage consumers from jumping into the technology. While it was indeed a problem during the earliest days of the technology, VoIP has vastly improved over the years and the benefits it brings are well-known.
But like any internet technology, VoIP still isn’t perfect. Users may experience call quality issues, albeit easily solved and corrected. In this post, we identify some of the most common factors that may affect the quality of VoIP calls.
Bandwidth, which refers to the rate at which data is transmitted over a wired or wireless connection, can greatly affect the quality of your phone conversations. It’s similar to a water pipe. If the pipe is larger (or if the bandwidth is bigger), more water (or more data) can flow at a given time.
If you have a shared network and don’t have enough bandwidth, you can expect choppy calls and delays, or even a system that can’t make or receive calls. To avoid such problems, it’s important that you secure an internet plan that has sufficient bandwidth.
- Audio codec
The type of audio codec (which stands for code-decoder) used is also one factor that can affect voice quality. Audio codecs are used to convert analog sound waves into compressed digital form and back again. There many different codec types, which vary in sound quality and bandwidth required, among others.
Generally, the codecs that provide the best call quality consume the most bandwidth. G.729 is the most common codec. It has low bandwidth requirements and provides medium audio quality. G.722, meanwhile, offers superior audio quality but requires bigger bandwidth.
Also called lag, latency is the amount of time it takes for a voice packet to be transmitted and reach its destination. Latency slows down your phone conversation, causes an echo, and results in garbled speech or overlapping noises. Various factors can cause delay in VoIP conversations. These include the distance between the calling parties, the VoIP router used, as well as compression algorithms.
For a VoIP call to be clear, the latency must be 150 milliseconds or less. Anything over than that could lead to a poor call quality. Some suggestions to solve latency include prioritizing VoIP traffic over the network, and getting a quality VoIP router.
Jitter refers to the variation in the arrival times of voice packets over a network. This happens when voice packets arrive at their intended destination in a different order as they were sent. Often caused by network congestion, high levels of jitter can lead to long delays in the conversation and an overall poor call quality. To mitigate the effects of jitter, getting a jitter buffer helps. It collects arriving packets temporarily and rearranges them in the proper order before sending them for decompression.
These days, every business expects to enjoy high quality VoIP calls, and understandably so. Clear business communications help companies become more productive and competitive. Although VoIP phone systems aren’t exempt from call quality issues, such problems can easily be solved and corrected through regular monitoring.