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5 Issues That Disrupt Call Quality in VOIP.

Updated: July 24, 2013

Don’t make the mistake of choosing an On-Premise VOIP system before understanding the technical planning and maintenance that it calls for. Proper configuration and monitoring are requisites for a hassle-free On-Premise VOIP experience. If too many technical variables go unchecked, your advanced telecom can quickly become an advanced problem.

Here are 5 of the most common causes of voice or call disruption. While each one is manageable with the right expertise, it can take constant effort to keep your communications running smooth.

1. Delay or Latency - Delay in voice.

Delay is the time that it takes for sound to travel from one end of a network to  another—the time between when someone speaks and when it is heard. Surprisingly, it only takes milliseconds of delay to negatively affect call quality. VOIP systems are so sensitive to delay since conversations are had in real time, unlike text or image data.

Handling delay describes the various changes that a sound must undergo before reaching the listener. Converting an analog sound into digital information takes a few steps that add milliseconds to the overall transmission. These processes include the packetization, compression and packet switching. When the various nodes in a network get backed up each of these processes can take longer.

Packetization delay is the time is takes to ‘fill’ a packet with voice data. This process is crucial to voice over IP since it builds the main unit of data. Abnormal delays in the process are related to a system’s configuration—packet size, for example, must be optimized over time to minimize latency. If an inefficient packet size is programmed, it can have a rippling effect throughout the system.

Queuing delay is the time that a packet waits while other packets are being processed. As packets reach a node, they must wait for packets further ahead in the transmission to ‘play out.’ Optimizing this queuing process takes routine configuration as well.

2. Jitter - Broken up or jumbled voice.

Jitter is the discrepancy between when data packets are expected to arrive at a given intercept or node and when they actually do arrive. Data packets can arrive late if other steps and nodes are overburdened or poorly configured.

Jitter gets disruptive when the discrepancy between the expected intervals and actual intervals is considerably off. For instance, if the system is configured to receive packets every 20 milliseconds and the rare packet arrives a few milliseconds later, the problem may not affect call quality. But if the packets are consistently off, users will notice strange sounds during calls.

While jitter can be minimized by adding a buffer that stops and sends the packets uniformly, this will only work if the jitter is moderate and predictable.

3. Voice Compression - Sound interference.

Compression is required to make voice data more manageable. By compressing data into smaller packets, it can travel through a network without causing as much strain to bandwidth. When it reaches its destination it is decompressed and converted back into sound.

Compression software is important since an improperly compressed or decompressed packet can cause a distortion in the sound. Problems can also arise if the compression process is slower than normal.

Part of the compression process erases unimportant sound data like white noise, background sound and frequencies that the human ear can’t hear. This won’t affect voice quality unless the process is erasing more than just these irrelevant sounds.

4. Echo - Voice echoes.

Echo is created by impedance, which is the electric resistance or opposition to a signal from a device or network segment. Impedance mismatch is when different segments of a network have different levels of opposition, causing further problems for the signal. The resistance is then heard by VOIP users as voice echo. The severity depends on the frequency and discrepancy between levels of impedance.

Echo cancellation is the process of matching up an inverse copy of the voice message with the echo. The system uses the copy as reference for what to cancel out during a call. It can take time to configure the appropriate amount of echo cancellation since impedance levels can vary.

5. Packet Loss - Crackling & popping sounds during calls.

Packet loss can be particularly disruptive to a conversation and is one of the more important problems to correct. It has many causes, but is most commonly attributed to an overburdened network. When a low-bandwidth system encounters a mass of data, some of the information never reaches its destination and the rest is slowed significantly.

For regular data, packet loss isn’t much of an issue since the packet can be resent in almost any order. Order is crucial for voice data, so the packet loss has to be avoided. The goal of 1% packet loss or less is achievable through a high-quality internet connection and a well configured VOIP system that prioritized voice data.

Hosted VOIP vs On-Premise VOIP

Let experts take care of your telecom.
One way around these On-Premise configuration and maintenance issues is to go with a Hosted VOIP service. Hosted VOIP technology gives your company all the advantages of VOIP without the extra costs or responsibilities. When something disrupts voice or call quality, your provider deploys highly specialized technicians and support to fix it. Ongoing monitoring, configuration and optimization is all part of your monthly fee.

Before you have an On-Premise VOIP system installed, be ready to deal with a slew of possible issues. Enterprise IT department may be able to contend with the demands of On-Premise VOIP, but it’s not a good fit for just any company.


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