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VoIP Call Quality: Influencing Factors & Ways to Improve The Voice Quality



factors affecting voip call quality

VoIP telephony has significantly evolved over the years. When pandemics hit hard globally, and remote working started to become a thing, its use became more widespread. 

VoIP (Voice over Internet Protocol) is a popular medium of communication through an internet connection. Communication in a VoIP system takes place through the transfer of data packs over the internet. So unlike PSTN telephony, VoIP services are less likely to get affected by signal fluctuations and network disturbances. But that being said, VoIP communications are not entirely bulletproof. 

VoIP call quality drops too, which can adversely affect the user experience. Every year, businesses lose several clients and a ton of revenue due to the poor quality of calls. So if you own an organization that entirely depends on phone calls, it can put your whole business in jeopardy.

So let’s discuss some of the major factors affecting VoIP call quality and then find ways to improve it.

What factors influence voice quality in VoIP calls?

Call quality is crucial for any VoIP service. So, VoIP service providers are always searching for factors that degrade the quality of calls to analyze their service quality. Here are six factors that influence voice quality in VoIP calls:

1. Latency

The term “latency” refers to the measurement of delay. The most prominent latency signals are echo and inadvertent simultaneous talking because of delays in transmitting the data. VoIP call delivery durations are measured in milliseconds, so there isn’t much room for error.

A delay can drop voice quality due to network propagation issues, frame forwarding, or packet queuing. The approach is to leverage traffic prioritization through VoIP-optimized routers. Using Multiprotocol Label Switching (MPLS) lines and merging several types of connections into a software-defined WAN (SD-WAN) is a way to improve quality to lower latency. 

2. Quality of equipment

VoIP may appear to be a simple way to communicate at its most basic level, but it is supported by a complicated networking infrastructure that must work in unison. Routers, modems, and firewalls are all critical components of modern VoIP. The age of these devices and build quality can have a direct impact on call clarity.

3. Loss of packet

Packet loss is another factor that affects voice quality in VoIP calls. It occurs when packets fail to reach their specified destinations. It’s a primary source of jitter and a frequent detriment to the quality of VoIP calls. Callers will detect the impact even if only 1% of packets are dropped. If the loss of packet persists, the VoIP calls can drop too.

It can cause audio to briefly break down or sound distorted. Packet loss can be caused by several factors such as overworked networks, hardware problems, and various routing inefficiencies.

4. Availability of bandwidth

Hosted VoIP’s performance needs network connections that aren’t strained during peak usage to handle the bandwidth. Numerous broadband packages do not meet the requirements. 

They usually have significantly faster upstream speeds than downstream speeds and are shared by several end-users in the local footprint to keep prices down. Bandwidth can be strained to the point that VoIP call quality is compromised.

5. Network Jitter

Jitter occurs when packets arrive out of order. Packets must come sequentially in real-time for optimal clarity in VoIP calls; otherwise, jitter can impair quality significantly. Jitter buffers can preserve call quality. These mechanisms reside between the voice decoder and the incoming packets, collecting and assembling packets correctly.

Many of VoIP’s benefits arise from the fact that it is provided over packet-switched IP networks rather than circuit-switched telephone lines. This allows VoIP to substantially decrease expenses for long-distance calls and is considerably easier to set up for administrators.

6. Codecs and protocols

VoIP relies heavily on codecs. These codecs compress audio signals before sending them over an IP network, then decompress them before playing them back. Not all codecs provide the same level of performance or have the same bandwidth and license needs. G.729, G.711, Speex, and a few additional codecs are the most popular in VoIP telephony. 

Protocols like SIP and H.323 can also influence quality. Although H.323 is still used in some video conferencing systems, SIP has been popular in most VoIP implementations. SIP is often easier to troubleshoot and uses fewer resources, allowing for better call quality consistency.

How to improve VoIP call quality?

Now that we have discussed and likely understood the factors that hinder VoIP Call quality. The next step should always be finding out the way to improve the VoIP call quality. Improving a call quality can be challenging and expensive, but it should not be a blockage for any business to run smoothly. Here are six ways to improve VoIP call quality:

1. Decent Network Setup

If your network isn’t set up properly, call quality can suffer while routing both data and voice through the same internal network. A properly set up VLAN capable switch can solve this issue. Several of these VLAN-capable switches also provide Power over Ethernet (POE), which excludes the need to plug each phone into an electrical outlet.

2. VoIP optimized routers

Latency causes delay during voice transmission. The strategy is to implement traffic prioritization in conjunction with the VoIP-optimized routers. Using MPLS links and merging several types of connections into a software-defined WAN (SD-WAN) is an approach to improving quality by minimizing latency.

3. Eliminate Jitter

The packets are supplied in a steady stream that is equally spaced.  This constant stream might become choppy due to issues such as network congestion, poor queuing, varying packet delays, or setup mistakes. The de-jitter buffer is a method that requires a router or Edge device that collects a Real-Time Protocol (RTP) audio stream for VOIP can eliminate jitter. 

Jitter buffers can assist in preserving the sequence of jitters. These algorithms reside between the voice decoder and the incoming packets by collecting packets and assembling them correctly.

4. Buy Quality Headsets

Headsets are the medium to transmit sound into a person’s ear. The quality of headsets determines the quality of sound the person is hearing. Low-quality headsets have different issues reducing the quality of calls like echo, noise, etc. So, having a good quality headset is necessary to improve call quality.

5. Appropriate Bandwidth

The cause of intervals of silence or robotic voices experienced by call participants is the lack of internet bandwidth. To avoid packet loss and improve overall VOIP calls, there should be at least 100kbps bandwidth in both ends. Slow internet connectivity results in choppy audio. So the internet connection should have enough bandwidth for good quality calls and network activity.

6. Avoid Interference

Interference is more likely to occur with phones that operate at higher GHz frequencies. 2.4 GHz is the frequency used by most VoIP phones. Meanwhile, 5.8-GHz phones and 5-GHz devices are available. 

The 2.4-GHz band has a longer range than the 5 GHz band but has slow data transmission. The 5-GHz band has limited coverage, but data flow is quicker. You should consider an appropriate frequency for you. Using 2.4 GHz with enough bandwidth is an excellent solution to avoid interference.

Taking everything into account,

Good call quality is a focal point for smooth VoIP calls. The factors above are responsible for affecting VoIP call quality. Not everyone will have the same factors influencing voice quality in VoIP calls. So, it is necessary to monitor your network continuously. Remember, constant monitoring of call quality will assist you in identifying the issues. This information helps in searching for possible ways to improve VoIP call quality by eliminating the issues. 

If you want to avoid facing tons of call quality issues in the first place, then the best thing is to get the best telephony service provider in the VoIP market. Recently, a new cloud telephony platform named KrispCall is creating a buzz in the market.

KrispCall has a team of qualified experts who continuously work to give the best user experience during VoIP calls. With the idea of offering seamless VoIP phone calls, KrispCall identifies, analyzes, and eliminates any factors affecting VoIP Call Quality so that users do not have to worry about any issues on the server-side and enjoy quality VoIP calls.

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Dinesh Silwal

Dinesh Silwal is the Co-Founder and Co-CEO of KrispCall. For the past few years, he has been advancing and innovating in the cloud telephony industry, using AI to enhance and improve telephony solutions, and driving KrispCall to the forefront of the field.

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