Have you ever experienced a phone call that sounds as if the person on the other end has a tinny, almost robotic, voice? Have you experienced calls that cut out or sound choppy? Or maybe you have experienced two callers continuously talking over one another because the timing of the sound was off. All of this is often the result of a poorly prioritized or overloaded network.
When a call is placed, the real-time audio is transported in network packets. If you have a high level of bandwidth, you can usually expect to have high definition phone calls without all of those aforementioned problems. However, even when bandwidth is high, other issues may still persist in terms of VoIP call quality if the network is improperly prioritized. These issues are even more likely to occur when bandwidth is low.
Network latency is the time it takes to transfer audio packets from point A to point B. To maintain decent sound quality, network latency shouldn’t exceed 150ms (milliseconds). However, for best results, it’s recommended to strive for packet transfer within the range of 75 to 100ms in a VoIP call.
Latency above this general threshold can cause the timing of the sound in a call to be perceptibly delayed. This introduces the issue of two callers talking over one another.
Network Congestion & Prioritization
In most businesses, it is common for bandwidth to be shared among all devices connected to the wireless network, including VoIP calls; this is far more cost-effective and convenient. However, this can introduce problems if the network isn’t properly prioritized. For example, an email may be allocated more bandwidth than an audio packet on a VoIP call. Proper bandwidth allocation should prioritize VoIP calls since they take place in realtime, and increases in latency can cause all of those familiar issues, also known as “jitter.”
Congestion and a lack of prioritization can cause network latency, and can also introduce a number of other issues as well. Many packets traveling at once can use up all available bandwidth, causing the network to create a queue. In other words, an audio packet may be waiting in line behind a software download. This is not good for call quality and leads to jitter.
Jitter is the variation in packet delay and often occurs when a network is congested or if there is not enough bandwidth to support the transit. Jitter can cause severe voice quality issues and sometimes even dropped calls. Common experiences include sounds cutting in and out as well as voice distortions.
Issues associated with jitter will continue to worsen as the network becomes more congested. This means that, with enough congestion, voice packets can be completely missed or tossed out during transit. Losing part of a voice packet causes the audio to be distorted and sometimes even whole pieces of sound are completely lost.
Fragmentation is one potential solution to this problem. If audio packets are attempting to be transferred in whole pieces, rather than several smaller bits, this will put unnecessary strain on the network and can potentially cause and exacerbate packet loss. When a caller’s voice sounds altered, that is because parts of the packet have been lost in transit. Implementing fragmentation allows these packets to be divided up into smaller pieces which puts less strain on the bandwidth, and helps data arrive at its intended destination. The audio packet is then reassembled before arriving at the endpoint. This entire process occurs within milliseconds and is one effective way of retaining audio quality.
What Can QoS Do For a Network?
Another point to consider is Quality of Service (QoS). QoS determines how data traffic is prioritized on a network. Since it is a realtime service, VoIP data needs to be prioritized above all other network traffic. For the best sound quality, you typically want audio packets to be able to move faster than anything else on a network.
For services such as email, downloading software, or web browsing, even differences of a few seconds will probably go unnoticed, but delays in real time services like VoIP calls are much more perceptible and can lead to all of those other problems with quality and connection. Prioritizing VoIP over other traffic, therefore, is a trade that sees benefits on the side of call quality, with tradeoffs on other network tasks that are likely to go unnoticed.
Issues with Network Capacity
If there isn’t enough bandwidth to handle the amount of packets that are being transferred, some of those packets may be dropped altogether. The general tolerance level of packet loss is 0.3 percent. If that threshold is exceeded consistently, even after implementing fragmentation and QoS, then there is simply not enough bandwidth to support VoIP calls along with all other network usage. If that is the case, it is recommended to increase the overall network capacity.