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Why VoIP Issues Arise and How to Improve Call Quality

We know that many of you are concerned with VoIP issues. Therefore, we start the series of articles on this topic. Today, we are going to outline the most common reasons which cause echoing, distortion, dropped calls etc.

Everyone knows the advantages of the calls via the Internet. However, it hides many pitfalls. In this post, we’ll unveil most common ones and share troubleshooting tips. But first, let’s learn about how VoIP works.

How VoIP (Voice over Internet Protocol) works 

  1. User 1 speaks via SIP phone, computer or another gadget with User 2. He can use special app or microphone with headsets.
  2. His voice is converted into a digitized stream and compressed into data “packets”.
  3. The data “packets” are transmitted through the User’s device, the router, and the VoIP software’s media server.
  4. The connection between User 1 and User 2 continues through a carrier network via VoIP.
  5. The data “packets” are sent to the device of User 2. In result, he hears the voice of the recipient.
  6. User 2 speaks and this process is repeated in the reverse direction.

Voice journey happens very fast and quietly. Probably, users don’t even notice it. However,  every step of voice journey presents plenty of opportunities for things to go wrong.

voIP, Voice journey, Ringostat telephony


It occurs when the raw audio is being compressed on the old computers. To put it simply, it happens at the time when the voice audio is turning into a digital signal. Also, latency can occur when the compressed audio is traveling through the initial provider network. VoIP opportunities can neutralize the effects of latency. However, when exceeding 250 ms, it affects the quality of voice.  

Causes of latency

  1. Load latency occurs when the audio is being compressed. Its amount is determined by the type of audio codec – a computer program that compresses and decompresses digital audio data. This latency varies from 0, 125 microseconds to milliseconds.
  2. Processing latency occurs when the audio data is being collected and divided into packets for transmission. It depends on the algorithm of processing.
  3. Network latency can be caused by physical environment and protocols used for VoIP. Besides, it can occur because of the wrong work of jitter buffers which will be highlighted below.

As you see, latency causes many issues. It affects not only the quality of voice but the interaction between the two end users. The longer latency is, the more complicated the interaction can be. Call participants start talking on top of each other. Also, latency can result in echo.


Sometimes packets are delivered to a recipient at irregular intervals. That results in choppy voice, temporary glitches and users hear missing audio or lapses of silence. This issue is known as jitter.

That happens if the audio isn’t played at a constant rhythm. To prevent this issue, VoIP service providers build in jiffer buffers.

When jitter occurs, jitter buffers send data that had been collected and stored before. For example, Ringostat jitter buffers collect packets and delay their transmission for 0,1 seconds. It lets avoid lapses of silence when decompressing the voice data.

However, sometimes latency goes beyond the buffer’s capacities. If there’s no new voice data and buffer remains empty, the voice will be choppy. 

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Packet Loss

If packets are delayed or contain errors, they may be lost and not arrive at the destination. This is usually due to the unreliable Internet connection or overloaded network. The result is missing chunks of audio.

People perceive packet loss as a distorted voice and missing audio. Packet loss is considered to be noticeable when 10-15% of data is lost. These indicators depend on algorithms for compressing and decompressing. 


Echo may sound fun in the forest. But when it comes to VoIP, that’s a bad sign. Echo is noticeable when the latency is more than 10ms. There are several types of echo:

  • acoustic echo occurs when talker keeps the phone far from his head or gets the speakerphone. The bad headphones also can be a trouble. In this case, the speaker’s microphone reflects sounds back and the talker hears a delayed copy of his own voice.
  • hybrid echo occurs in the transition point between trunks. The trunk is a channel between switch and another network device. For example, a hybrid echo can occur if a client calls from analog PBX to digital one.
  • data latency echo. Its reasons are similar to the aforementioned reasons of latency.

To remove acoustic echo, you need to reduce the sound volume. To get rid of other types of echo, you may use echo canceller. That’s a specific circuit for removing echo-signals that built into the device.

Echo canceller compares received voice data with sent voice data. If they are similar, it removes extra packets by using a digital filter. However, this method works unless latency is more than 30-200 ms. 

How to avoid VoIP issues

The proper configuration of the network equipment matters. However, this topic is quite comprehensive, so we’ll get back to it in our next post. Now, we’re going to discuss the simplest ways of how to prevent VoIP issues. 

Use devices with the option of network prioritization

This option is called Quality of Service (QoS). It allows to determine bandwidth priorities for different traffic. You can set aside network bandwidth just for VoIP. Why is it essential?

Only one channel is mostly used to access VoIP and Internet. Some in the office can listen to music or chat when those who are speaking over VoIP experience the reducing of quality of the connection. That occurs because packets with encoded voice can’t travel unhindered.

Here’s the scheme which compares bandwidth utilization with and without QoS. 

voIP, the scheme which compares bandwidth utilization with and without QoS
Source — http://blog.pascom.net/

Speaking metaphorically, the bandwidth without QoS is a highway during rush hour. The congestion doesn’t allow an emergency vehicle to move fast.

Quality of Service helps set higher bandwidth priority for voice data packets than for other packets. It’s especially relevant when the connection is slow. 

Avoid WiFi

By using WiFi for VoIP calls you risk to face up the poor quality of connection. The problem is WiFi coverage is spotty. This is because:

  1. Professional routers are expensive and need to be manually configured.
  2. WiFi was never designed for real-time applications.
  3. Office WiFi networks are packed with devices that interrupt each other. 

The good WiFi connection exists but it requires many efforts. Therefore, we recommend you to choose ethernet connection when possible. It’s safer and guarantees the highest quality connection.  

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Determine what interrupts your network 

The VoIP issues occur when other devices such as smartphones, microwaves, and fluorescent lights interfere the phone call. As result, the user hears popping, crackling, or humming noises during the call. 

However, sometimes other devices aren’t the reason for bad VoIP connection. Using WiFi instead of ethernet connection also can cause the interference. 

To avoid this, opt for wire gadgets instead of wireless ones. Try to identify the cause of VoIP issue. Find out what equipment stops working without reason. Turn it off and repeat the phone call to make sure that this equipment causes the interference. 

We hope these simple recommendations will help you improve call quality fast. In the next post about VoIP, we’ll go deeper. Stay in touch.