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FEATURES UC MONITORING HOW TO NETWORK METRICS

How to Measure VoIP Quality & MOS

Score (Mean Opinion Score)
Alyssa Lamberti
Last updated on Apr 7, 2022

Table of Contents
What is VoIP Quality?

What is MOS Score?

Measuring VoIP with MOS

MOS Score vs VoIP Call Quality

How to Measure VoIP Quality with MOS Score

Good VoIP Quality vs. Bad VoIP Quality

High MOS Rate vs. Low MOS Rate

 Most Important VoIP Metrics
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QoE for VoIP

QoS for VoIP Call Quality

Jitter for VoIP

Latency for VoIP

QoS for MOS

Measuring MOS Score with Device Monitoring

A VoIP Monitoring Use Case

Are you tired of constantly dropping calls or struggling to hear your loved ones on
the other end of the line? Fear not, because we're here to talk about the one thing
that can make or break your VoIP experience: MOS score. No, we're not talking
about the fuzzy creature from Star Wars - we're talking about the Mean Opinion
Score, the nifty little metric that can help you measure and improve the quality of
your VoIP calls.

VoIP Quality is highly reliant on network performance, which means that many
network problems like packet loss, latency, and jitter can cause high levels of VoIP
degradation. To avoid embarrassing choppy voice calls, or lagginess during your
next client meeting, we’re running you through how to measure VoIP Quality with
MOS Score (Mean Opinion Score).

So grab a cup of coffee, settle in, and let's dive into the wonderful world of MOS
scores and VoIP quality!
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What is VoIP Quality?
VoIP Quality refers to the overall level of audio and visual performance
experienced during a Voice over Internet Protocol (VoIP) call. This includes factors
such as call clarity, signal strength, delay, echo, and other audio and visual
distortions that can impact the user experience.

VoIP Quality can be influenced by a variety of factors, including the quality of the
internet connection, the type of device used for the call, and the network
infrastructure supporting the call. Measuring and monitoring VoIP Quality is
important to ensure that users are able to communicate effectively and efficiently,
without experiencing frustrating audio or visual interruptions.
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How is VoIP Quality Measured?

VoIP Quality is typically measured using a combination of objective and subjective
metrics. Here are some common methods for measuring VoIP Quality:

1. MOS score: The Mean Opinion Score (MOS) is a subjective metric that is
obtained by having a group of people rate the quality of audio samples. MOS
scores are widely used in the telecommunications industry to assess the quality
of VoIP calls. We'll expand on MOS Score in the next section.

2. R-factor: The R-factor is an objective metric that is used to measure the quality of
VoIP calls. It takes into account factors such as delay, jitter, and packet loss to
produce a score between 0 and 100. A higher R-factor indicates better call
quality.

3. Packet loss rate: Packet loss rate is the percentage of voice packets that are lost
during transmission. A higher packet loss rate can result in poor call quality.

4. Jitter: Jitter is the variation in delay between packets. High jitter can result in
poor call quality.

5. Latency: Latency is the delay that occurs between the time when a voice packet
is sent and when it is received. High latency can result in poor call quality.

6. Echo: Echo occurs when a caller hears their own voice back in their earpiece.
Echo can be caused by network delays, poor echo cancellation, or other factors,
 and can result in poor call quality.
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By measuring these metrics, VoIP service providers can identify areas where call
quality needs improvement and take steps to improve the overall quality of their
services.

What is MOS Score (Mean Opinion Score) for VoIP?

The Mean Opinion Score, usually referred to as MOS Score, has been the most
commonly-used metric to measure the overall voice call quality for decades.
Standardized by the International Telecommunications Union (ITU-T), MOS Score
refers to:

A numerical measure of the human-judged overall quality of voice and video

sessions.

How is MOS Score Measured?

The MOS is obtained by having a group of people listen to a set of standardized
audio samples and rate them on a scale of 1 to 5, with 1 being the lowest quality
and 5 being the highest. The ratings are then averaged to obtain the MOS score.

The MOS score is widely used in the telecommunications industry to measure and
monitor the quality of voice calls, including VoIP calls. By measuring MOS scores,
telecom companies can identify areas where call quality needs improvement and
take steps to improve the overall quality of their services.
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Measuring VoIP with MOS (Mean Opinion Score)
Voice Quality MOS Score is most often judged on a scale from 1 (bad) to 5
(excellent) of the perceived quality of a voice call. Although originally Mean Opinion
Scores were derived from surveys of expert observers, today a MOS Score is often
produced by an Objective Measurement Method approximating a human ranking.

The MOS Score (Mean Opinion Score) for Voice Quality was originally developed for
traditional voice calls but has been adapted to Voice over IP (VoIP) in the ITU-T
PESQ P.862.

I. How MOS Measures VoIP

The standard defines how to calculate MOS Score for VoIP Quality based on
multiple factors such as the specific codec used for the VoIP call. Each VoIP codec
(ex: G.711, G.722, G.723.1, G.729) behaves differently. Some codecs such as G.711
are uncompressed for higher quality but use more bandwidth than compressed
codecs such as the G.729.

The MOS Score we measure is for the G.711 codec, which is by far the mostly used
codec for VoIP calls. The maximum MOS Score for a G.711 call is 4.4.

II. VoIP MOS Test

During the MOS test, human participants are asked to listen to a series of pre-
recorded speech samples or participate in live VoIP calls. After each sample or call,
the participants are asked to rate the quality of the voice on a scale of 1 to 5, with 5
being excellent and 1 being unacceptable. The participants' ratings are then
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averaged to calculate the overall Mean Opinion Score for the VoIP call or service.

The MOS test considers several factors that can influence the voice quality,
including:

Clarity: How clear and distinct the voice sounds without any distortion or noise.

Loudness: Whether the voice is at an appropriate volume level, not too loud or
too quiet.

Delay: The time it takes for the voice to be heard after speaking, also known as
latency.

Jitter: Variability in the delay between packets, which can cause choppy audio.

Packet Loss: The percentage of voice packets that are lost during transmission,
leading to gaps in audio.

Background Noise: Any unwanted noise or interference during the call.

The MOS test results in a single score that provides a concise summary of the VoIP
call quality. It gives VoIP service providers and network administrators valuable
feedback on how users perceive the quality of their VoIP service. A higher MOS
score indicates better voice quality and a more satisfactory user experience.

Typically, MOS scores are categorized as follows:

5: Excellent

4: Good
 3: Fair
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2: Poor

1: Unacceptable

VoIP MOS testing is essential for maintaining and improving VoIP services. It helps
service providers identify and address issues that may negatively impact call
quality, leading to a better overall user experience. By consistently conducting MOS
tests and monitoring the scores, providers can make data-driven decisions to
optimize their VoIP networks and ensure high-quality voice communication.

MOS Score vs VoIP Call Quality

When deciding how we would be measuring VoIP Quality in the Obkio app, we
decided to measure it based on the MOS Score. Here are the details for how we
measure VoIP MOS Score. The table below lists the different qualities and the
lowest MOS Score limit for each of them. The limit values are from the ITU-T
standards.
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We also realized that most of our customers were not very familiar with the MOS
Score and the interpretation of MOS in VoIP Quality. To help all our users
understand more easily, we redesigned the MOS Score graph to create the VoIP
Quality graph.

Obkio’s MOS VoIP Quality graph categorizes, for every minute, the MOS Call Quality
as

Best

High

Medium

Low

or Poor

The exact MOS Voice Quality Score is always available in the graph tooltip.
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This feature is a great measure of the Quality of Experience (QoE) for users using
VoIP applications over their network and helps IT Pros evaluate the impact of
Network Performance on VoIP Applications.

How to Measure VoIP Quality with MOS Score

VoIP issues make up 50% of IT problems!

VoIP travels a long distance from your network, through the Internet, and up to your
Service provider. That means that, when a problem occurs, no matter where it is,
you need to manage it.

Step 1. Deploy A Network & VoIP Quality Monitoring Tool

A VoIP monitoring tool is the best way to ensure accurate and continious VoIP
Quality measurement - and Obkio Network Performance Monitoring software can
help with that.
Obkio is a Network Monitoring and Troubleshooting tool with it's own VoIP
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Monitoring feature.

Network performance can have a significant impact on VoIP Quality. When making
a VoIP call, the voice data is transmitted over the internet, and any issues with the
network can cause delays, packet loss, and other disruptions that can affect call
quality. Which is why it's important to monitor network and VoIP Quality all together.

Obkio's VoIP Monitor measurse VoIP Quality with MOS Score and start to proactive
identify and troubleshoot VoIP issues before your users even know they exist!

Step 2. Deploy Monitoring Agents for End-to-End VoIP Quality

Monitoring
To monitor VoIP Quality in all your network locations, for all your core VoIP apps,
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you need to deploy Network Monitoring Agents which continuously measure
network performance in key network locations like offices, data centers and clouds.

The Agents exchange synthetic traffic between each other to measure core network
metrics that are essential to VoIP Quality, like MOS score.

To collect the data you need to translate, you’ll need:

Local Agents: Installed in the targeted office location experiencing VoIP Quality
issues. There are several Agent types available (all with the same features), and
they can be installed on MacOS, Windows, Linux and more.

Public Monitoring Agent: These are deployed over the Internet and managed by
Obkio. They compare network and VoIP performance up to the Internet and can
be used to monitor VoIP Quality for apps like Microsoft Teams and Zoom. They
can also help identify if the network and VoIP issue is global or specific to a
destination.

Step 3. Measure VoIP Quality with MOS Score

As we explained earlier, a tool like Obkio will automatically measure your VoIP
Quality for you! Once your Monitoring Agents are deployed, they will start
exchanging synthetic traffic to measure VoIP Quality.

Obkio’s MOS VoIP Quality graph categorizes, for every minute, the MOS Call Quality
as

Best
 High
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Medium

Low

or Poor

Step 4. Measure VoIP Quality using Other Core Metrics

Once you’ve deployed Obkio Monitoring Agents in your key network locations, they
will start measuring other key network metrics that essential to your network and
VoIP performance.

Here are some metrics you can measure with Obkio that are detrimental to VoIP
Quality:

Here are some ways that network performance can affect VoIP Quality:
1. MOS score: As discussed earlier, the Mean Opinion Score (MOS) is a subjective
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metric that is used to assess the quality of human speech and other forms of
audio. MOS scores can provide an overall measure of VoIP Quality and can help
identify areas where call quality needs improvement.

2. Bandwidth: VoIP calls require a certain amount of bandwidth to maintain call

quality. If the network is congested or there is not enough bandwidth available,
the call quality may suffer.

3. Latency: Latency is the delay that occurs between the time when a voice packet
is sent and when it is received. High latency can cause a delay in the
conversation, making it difficult to communicate effectively.

4. Jitter: Jitter is the variation in delay between packets. High jitter can cause the
voice quality to degrade, resulting in a choppy or garbled conversation.

5. Packet loss: Packet loss occurs when voice packets are dropped during
transmission. Even a small amount of packet loss can result in poor call quality
and affect the ability to understand the conversation.

To ensure high-quality VoIP calls, it is important to have a network that is capable of

handling the required bandwidth and has low latency, jitter, and packet loss.
Network monitoring and optimization can help identify and resolve any issues that
may be affecting VoIP Quality.
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Step 5. Identify VoIP Quality issues

With this setup, you now have everything you need to identify and troubleshoot VoIP
Quality issues. Obkio will automatically alert you of any VoIP performance
degradation, MOS score changes, and VoIP Quality issues.

Here are some common VoIP Quality issues you may encounter There are several
common VoIP quality issues that can affect the user experience. Here are some of
the most common VoIP quality issues:

Jitter: High jitter can result in choppy or distorted audio, which can make it
difficult to understand the conversation.

Latency: High latency can cause delays in the conversation, making it difficult for
 users to have a natural conversation.
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Packet loss: High packet loss can result in gaps in the conversation, making it
difficult to understand what is being said.

Echo: Echo occurs when a user hears their own voice after speaking. This can be
distracting and make it difficult to concentrate on the conversation.

Noise: Noise can occur due to a variety of factors, such as network interference
or equipment issues. This can make it difficult to understand what is being said
and can be very distracting.

Call drops: Call drops occur when a call is unexpectedly disconnected. This can
be frustrating for users, especially if it happens frequently.

Poor call setup time: Poor call setup time can make it difficult for users to make
and receive calls quickly, leading to a poor user experience.

By identifying and addressing these common VoIP quality issues, VoIP service
providers can improve the user experience and ensure that their systems deliver
high-quality audio that meets the expectations of users.
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What is Good VoIP Quality vs. Bad VoIP Quality?
When it comes to VoIP quality, there's a big difference between crystal-clear calls
and garbled, robotic-sounding conversations. To help you distinguish between the
two, we're breaking down what constitutes good VoIP quality versus bad.

Good VoIP quality is generally considered to be when voice calls are clear and have
a natural sound that is similar to a traditional phone call. In general, good VoIP
quality is characterized by the following:

High MOS score: A MOS score of 4 or above is generally considered good VoIP
quality.

Low packet loss: A packet loss rate of less than 1% is generally considered good
VoIP quality.

Low jitter: Jitter of less than 30 milliseconds is generally considered good VoIP
quality.

Low latency: Latency of less than 150 milliseconds is generally considered good
VoIP quality.

On the other hand, bad VoIP quality is characterized by poor audio quality that is
often characterized by the following:

Low MOS score: A MOS score of less than 3 is generally considered poor VoIP
quality.
 High packet loss: A packet loss rate of more than 5% is generally considered
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poor VoIP quality.

High jitter: Jitter of more than 50 milliseconds is generally considered poor VoIP
quality.

High latency: Latency of more than 300 milliseconds is generally considered

poor VoIP quality.

Ultimately, what constitutes good or bad VoIP quality can depend on individual
users' expectations and preferences. However, the general guidelines above can
provide a useful benchmark for evaluating VoIP quality.

VoIP Monitoring 101: Keeping Your Calls

Crystal Clear
Unlock the secrets of VoIP monitoring! Discover tools,
techniques, and the power of Obkio for unbeatable call
quality. Begin your adventure today!

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The Most Important VoIP Metrics to Measure VoIP Quality

Measuring VoIP (Voice over Internet Protocol) quality is crucial to ensure optimal
performance and user satisfaction. There are several important VoIP metrics to
consider when evaluating VoIP quality. These metrics can help identify and
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diagnose potential issues affecting call quality.

Some of the metrics we've already touched on in this article, but some may be new
to you!

I. MOS (Mean Opinion Score) for VoIP

We've already discussed this very important metrics many time in this article, but
it's worth presenting again!

MOS is a widely used metric to assess overall voice quality. It's typically rated on a
scale of 1 to 5, with 5 being excellent and 1 being unacceptable. MOS is obtained by
collecting subjective user feedback about call quality.

What is High MOS Rate vs. Low MOS Rate?

So we now know that a high MOS score means that you have great VoIP Quality!
But what is considered a high MOS score rating?

MOS (Mean Opinion Score) measures the perceived quality of VoIP audio on a scale
from 1 to 5, with 5 being the best possible score. A high MOS rate indicates that the
audio quality is good, while a low MOS rate indicates poor audio quality.

In general, a MOS score of 4 or above is considered to be high-quality audio and

indicative of a good VoIP experience. This means that users perceive the audio
quality to be similar to that of a traditional phone call.
On the other hand, a MOS score of less than 3 is considered to be low-quality
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audio and indicative of a poor VoIP experience. This means that the audio quality is
distorted, choppy, or robotic-sounding, making it difficult for users to understand the
conversation.

Overall, aiming for a high MOS rate is important for delivering a good VoIP
experience and ensuring that users are satisfied with the audio quality of their calls.

II. What is QoE for VoIP?

When it comes to measuring the performance of any technologies or application,
the user-perspective is your most important metric. Users identify good
performance because essentially, good performance is the ability to meet a user’s
expectations.

Quality of Experience (QoE) is a metric that allows you to measure performance

from the end-user perspective to gain a better understanding of human quality
metrics and be able to actually measure whether performance met a user’s
expectations.

QUALINET (The European Network on Quality of Experience in Multimedia Systems

and Services), describes QoE as:

“The degree of delight or annoyance of the user of an application or service. It results

from the fulfillment of his or her expectations with respect to the utility and / or
enjoyment of the application or service in the light of the user’s personality and
current state.”
QoE allows you to measure the user’s perception of the effectiveness and quality of
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a system or service to essentially give you a performance standard. In fact, users
base their opinions about the network exclusively on their perception of QoE.

QoE and VoIP Quality are two metrics that really work hand-in-hand. You can
actually use VoIP Quality as a way to measure QoE, because after all, if poor VoIP
Quality is leading to choppy calls, and robot voices, you can be pretty certain that
your user is not having a pleasant experience.

III. QoS for VoIP Call Quality (Quality of Service for VoIP)
Nowadays, IP networks are used to transport various types of applications which
are a lot more sensitive to network performance and quality. One of those
applications is VoIP.

This is why network engineers implement QoS (Quality of Service) to prioritize some
traffic on the network in order to reduce latency, jitter and packet loss. In case of a
network congestion, this will ensure that performance sensitive applications are
always running without degradation and that only the less critical applications
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(such as web browsing) are impacted.

Quality of Service (QoS) for VoIP is a set of techniques and mechanisms used to
prioritize and manage network resources to ensure optimal and reliable
performance for Voice over Internet Protocol (VoIP) traffic.

QoS (Quality of Service), is important for all VoIP Quality, because, at some point,
voice and data will ‘mix’ on your network, or on the Internet. VoIP Call Quality can be
measured using a variety of metrics, but it is more so perceived in regard to clarity
of the voice quality heard at both ends of a call. The human ear is very sensitive,
and will pick up on anything that may sound abnormal.

Quality of Service for VoIP allows us you set different priorities for different types of
data Services on your network. To ensure that phone calls are good quality, we need
to give VoIP traffic a higher priority than, for example, a download of a Windows 10
upgrade. That way, even with several computers doing upgrades on your network,
calls are still crystal-clear.

VoIP Quality is about being able to get high-fidelity audio at each end of the phone
call, without unwanted distortions. VoIP QoS will be affected by:

Bandwidth

Packet Loss

Jitter

Latency
Obkio allows you to continuously monitor QoS with DSCP features - specifically the
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DSCP code located in the IP header.

The Objectives of Quality of Service for VoIP

Prioritization: QoS prioritizes VoIP packets over other types of non-real-time data
traffic on the network. This ensures that voice packets receive preferential
treatment and are delivered with minimal delays, even when the network is
congested.

Low Latency: Reducing latency is critical for VoIP to maintain real-time

communication. QoS mechanisms prioritize VoIP packets to minimize the time it
takes for voice data to traverse the network from the sender to the receiver.

Jitter Management: QoS helps control jitter by using techniques like jitter buffers,
which temporarily store incoming packets and release them at a steady rate. This
 helps mitigate the effects of packet arrival variations and ensures smoother
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playback of voice data.

Packet Loss Mitigation: QoS mechanisms reduce the likelihood of packet loss by
giving priority to VoIP packets and ensuring that they are less likely to be
dropped, even in situations of network congestion.

IV. Jitter for VoIP

Jitter is a phenomenon that occurs in data networks, including the Internet, where
data packets experience varying delays in their arrival times at the destination. In
the context of VoIP, jitter refers to the inconsistency in the arrival of voice packets,
which can negatively impact call quality.

In a VoIP call, voice data is divided into small packets before being transmitted over
the network. These packets need to arrive at the destination in a steady and
consistent manner to reconstruct the voice signal properly. However, due to
network congestion, routing inefficiencies, or other factors, the packets may
experience different travel times, causing jitter.

Here's how excessive jitter can affect VoIP call quality:

Choppy Audio: When packets arrive with significant variations in delay, the
receiving end may experience interruptions in audio playback, leading to choppy
or broken voice. It can make the conversation difficult to understand, resulting in
a frustrating user experience.

Out-of-Order Packets: In severe cases of jitter, packets might arrive at the

destination out of order. The VoIP application needs to reorder them correctly,
 which can cause additional delays and further degrade voice quality.
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Packet Loss: If jitter is extreme, it can lead to packets arriving too late or being
dropped altogether. Packet loss in VoIP can result in missing parts of the
conversation, leading to gaps in audio or complete drops in speech.

Echo: Jitter can also cause echo, where parts of the sender's voice are played
back to them as an echo. This can happen when packets containing the sender's
voice arrive out of order and are played back later than expected.

Acceptable Jitter for VoIP

Acceptable jitter for VoIP is typically kept within a narrow range to ensure good call
quality. The goal is to keep jitter at a level that minimizes its impact on the VoIP
call without introducing noticeable degradation in voice quality. While there is no
universally defined threshold, a general guideline is to aim for jitter to be below 30
milliseconds (ms) for optimal VoIP performance.

Here's a breakdown of jitter levels and their impact on VoIP quality:

Low Jitter (Below 10 ms): This is considered excellent and should result in very
minimal or no noticeable impact on call quality. VoIP calls should be clear and
reliable at this level of jitter.

Moderate Jitter (10 ms to 20 ms): While still generally acceptable, jitter in this
range may cause occasional minor disruptions in call quality, but users are
unlikely to experience significant issues.
 High Jitter (20 ms to 30 ms): At this level, users may start to notice occasional
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choppiness or brief interruptions in the audio during calls. While still somewhat
acceptable, efforts should be made to reduce jitter to improve call quality.

Excessive Jitter (Above 30 ms): Jitter beyond 30 ms can significantly degrade

VoIP call quality. Users are likely to experience frequent audio disruptions,
choppy voice, and potential gaps in communication.

Max Jitter for VoIP

The maximum jitter for VoIP should generally be kept as low as possible to ensure
optimal call quality. While there is no fixed universally accepted maximum jitter
threshold, the commonly recommended maximum jitter for VoIP is around 30
milliseconds (ms) or less.

Measuring jitter with a Network Monitoring tool, like Obkio, will help you understand
if your jitter levels are increasing above the desired thresholds, and negatively
impacting the most sensitive applications, like VoIP.
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How to Measure Jitter & Keep Your

Network Jitterbug Free
Learn how to measure network jitter using Obkio’s
Network Monitoring software to identify network
problems & collect data to troubleshoot.

Learn more

V. Latency for VoIP

Latency is a critical metric in data networking and has a significant impact on the
performance of real-time applications like VoIP. It refers to the time it takes for a
data packet to travel from the sender to the receiver across a network. In the
context of VoIP, latency directly affects the delay experienced by users during a
conversation.

There are generally three types of latency in VoIP:

1. Transmission Latency: This is the time it takes to encode the voice data into
packets and transmit them over the network to the destination. This includes the
time it takes for the data to leave the sender's device and reach the network
router or gateway.

2. Propagation Latency: Propagation latency is the time it takes for the data
packets to travel across the physical distance between the sender and the
 receiver. This type of latency is influenced by the physical medium used for
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transmission, such as copper wires, fiber-optic cables, or wireless links.

3. Processing Latency: Processing latency occurs at network devices like routers

and switches, where the packets may experience some delay due to queuing,
buffering, and decision-making processes.

High latency in VoIP can lead to the following issues:

Delays in Conversations: High latency causes noticeable delays between when a

person speaks and when the other party hears the response. This delay can be
frustrating and disruptive during real-time communication.

Talk-Over Effects: In some cases, when latency is extremely high, the

conversation may become difficult to manage, as people might unintentionally
speak over each other due to the delayed audio.

Negative User Experience: Latency negatively impacts the user experience,

leading to reduced efficiency and potentially affecting the clarity of
communication.

Impact on VoIP Call Quality: Latency can lead to audio disruptions, choppiness,
and gaps in voice communication, resulting in poor call quality.

The VoIP Latency Requirements

VoIP latency requirements can vary based on the specific use case and user
expectations. Generally, the goal is to keep latency at levels that provide a seamless
and natural conversation experience, similar to traditional landline telephone calls.
The acceptable VoIP latency requirements depend on the type of VoIP application
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and the user's tolerance for delays. Here are some common latency requirements
for different VoIP scenarios:

Consumer VoIP: For residential or consumer VoIP services, the acceptable

latency typically ranges from 100 to 150 milliseconds (ms). Most users can
comfortably conduct conversations with this level of latency without noticing
significant delays.

Business VoIP: In business environments, where call quality is crucial for

professional communication, the target latency is usually lower, around 50 to 100
ms. Lower latency ensures that business calls are crisp and responsive.

Contact Centers: In call centers and customer support environments, where

efficient communication is essential, the target latency is often below 50 ms.
This is to enable real-time interactions between agents and customers without
noticeable delays.

Video Conferencing: For VoIP-based video conferencing applications, including

web conferencing and virtual meetings, the acceptable latency is around 150 to
250 ms. Lower latency is preferred, but some tolerance is given due to the
additional data processing involved in video transmission.

It's important to note that while lower latency is generally desired, achieving
extremely low latency can be challenging, especially in networks with multiple hops
and varying levels of congestion. Additionally, latency is influenced by factors like
network distance, routing efficiency, and the type of network connection (wired vs.
wireless).
To meet VoIP latency requirements, network administrators and service providers
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can take the following measures:

1. Quality of Service (QoS): Implement QoS mechanisms to prioritize VoIP traffic

and ensure that it receives preferential treatment over other types of data traffic
on the network.

2. Network Optimization: Optimize the network infrastructure to minimize the

number of hops and reduce packet processing times.

3. Jitter Buffering: Use jitter buffers at VoIP endpoints to manage and smooth out
variations in packet arrival times, helping to reduce latency and jitter-related
issues.

4. Bandwidth Management: Ensure sufficient bandwidth for VoIP traffic to prevent

congestion and excessive delays.

By continuously measuring latency and maintaining acceptable VoIP latency levels,

service providers and businesses can provide high-quality communication services,
that will undoubtably increase user satisfaction and improve productivity during
your business' VoIP calls!

Max Latency for VoIP

The maximum acceptable latency for VoIP depends on the specific use case and
user expectations. Generally, for most VoIP applications, the maximum latency
should be kept below 150 milliseconds (ms) to provide a satisfactory user
experience. However, the ideal target for maximum latency is typically lower, around
100 ms or less.
Here are some guidelines about how different latency levels impact VoIP quality:
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Low Latency (Below 50 ms): Excellent for VoIP, with minimal perceptible delay.
Calls feel immediate and responsive.

Acceptable Latency (50 ms to 100 ms): Generally considered acceptable for

most business and consumer VoIP applications. Users may notice a slight delay,
but it is not usually disruptive to conversations.

Marginal Latency (100 ms to 150 ms): At this level, users may experience
noticeable delays in conversations. While still tolerable, call quality may be
slightly impacted.

High Latency (Above 150 ms): Latency beyond 150 ms can significantly affect
the quality of VoIP calls. Users may experience delayed and disjointed
conversations, leading to reduced call clarity and communication difficulties.

How to Measure Latency

Learn how to measure latency with Obkio’s Network &
Latency Monitoring tool. Check for latency in your
network & analyze latency measurements.

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VI. Packet Loss for VoIP

Packet loss is a common phenomenon in data networks, including the internet,
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where data packets may fail to reach their intended destination. In the context of
VoIP, packet loss is a critical metric as it directly affects the quality of voice
communication during calls.

When a VoIP call is made, the voice data is divided into small packets and
transmitted over the network to the receiver. Ideally, all these packets should reach
their destination intact and in the correct order. However, due to various network
issues, some packets may be lost along the way.

Here's how packet loss impacts VoIP call quality:

Audio Gaps: When packets are lost, parts of the voice conversation go missing,
leading to gaps or silence in the audio. This can make the call sound choppy or
disjointed, making it difficult for users to understand each other.

Reduced Voice Clarity: Packet loss can result in distorted or garbled voice
quality. As voice packets are dropped, the audio may become unclear or even
unintelligible, making it challenging to carry on a conversation.

Conversation Disruption: Repeated or prolonged packet loss can cause

significant disruptions in the conversation flow. Users may have to repeat
themselves, leading to frustration and a less efficient communication experience.

Echo and Jitter: Packet loss can also contribute to echo and jitter issues. Echo
occurs when the sender's voice is played back as an echo to them, while jitter is
caused by variations in packet arrival times. Packet loss can exacerbate these
problems and make them more noticeable during calls.
Even a small amount of packet loss can have a noticeable impact on VoIP call
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quality, especially during real-time communication where immediate feedback is
essential. That's why measuring packet loss with a network monitoring tool like
Obkio is important to monitoring and solving packet loss in your network, before it
affects sensitive applications like VoIP and UC.

Acceptable Packet Loss for VoIP

In VoIP, any amount of packet loss can have a negative impact on call quality, as it
can lead to audio disruptions and gaps in communication. However, it is
challenging to achieve a network with absolutely zero packet loss due to the nature
of data transmission over the internet. Therefore, acceptable packet loss for VoIP
is typically set at or near 0%.

VoIP is a real-time communication application, and even a small percentage of

packet loss can result in noticeable call quality degradation, such as choppy audio
and reduced voice clarity. In an ideal scenario, VoIP systems should strive for zero
packet loss to ensure optimal call quality.
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QoS for MOS

QoS (Quality of Service) for MOS (Mean Opinion Score) refers to the ability of a
VoIP system to deliver high-quality audio that meets the expectations of users.
QoS for MOS is typically achieved through a combination of network design,
equipment selection, and network management practices.
To achieve QoS for MOS, VoIP service providers may implement various strategies,
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such as:

1. Prioritizing voice traffic over other types of network traffic to ensure that voice
packets are given priority over data packets.

2. Implementing Quality of Service (QoS) mechanisms such as traffic shaping,

traffic policing, and traffic prioritization to manage network traffic and ensure that
voice packets are delivered with minimal delay, packet loss, and jitter.

3. Selecting high-quality VoIP equipment such as routers, switches, and IP phones

that are designed to handle the demands of VoIP traffic.

4. Monitoring network performance using tools such as network monitoring tools

like Obkio, network analyzers, packet sniffers, and other monitoring tools to
identify issues that could affect call quality and take corrective actions.

By implementing QoS strategies, VoIP service providers can ensure that their
systems deliver high-quality audio that meets the expectations of users, as
reflected in MOS scores. This can lead to improved user satisfaction, better
business outcomes, and increased customer loyalty.

Measuring MOS Score with Device Monitoring

When looking at how to calculate MOS Score, network metrics give you the most
value and detail when they’re combined to give you a full overview.

Network Device Monitoring is a feature that allows you to monitor the performance
of networking devices such as firewalls, routers, switches and wifi access points is
crucial for IT teams.
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The Device Monitoring feature inside the Obkio Software is a fast and easy solution
to get detailed information about the health of devices using Ultra-Fast Polling
(every 30 seconds).
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Measuring MOS Score with Network Monitoring

Along with Device Monitoring, using a complete end-to-end Network Monitoring
tool, monitoring VoIP MOS Score with Obkio’s Network Monitoring solution and
Network Device Monitoring allows you to truly understand VoIP Quality from the
end-user perspective.

Obkio measures performance from the end-user perspective by monitoring

synthetic traffic (with no packet capture required) every 500ms!

This allows you to perform a network assessment of your entire network

infrastructure to identify exactly:

1. What network problems?: It’s important to find out what network problems are
affecting VoIP Quality to be able to begin network troubleshooting. As I
mentioned earlier, there are a variety of network problems that can affect VoIP
performance, such as packet loss, jitter, and bandwidth.

2. Where problems are located: Modern networks are vast infrastructures that can
span over a variety of locations. When a problem arises, it is important to
pinpoint where VoIP issues are located along your network so you know where to
focus your troubleshooting efforts.

3. Who the owner of the problem is: Your business may only have one IT specialist,
or you may have a large IT department with different employees responsible for
different parts of the network. Identifying who is responsible for dealing with a
 problem affecting VoIP Quality (user, application, network, or ISP), will help you
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decide who is responsible for fixing it.

4. How to solve problems: Once you’ve collected all the information from the three
previous points, you can assess the data to come to a resolution. It’s a given that
having as much information as possible will lead to a quick and efficient solution.

A VoIP Monitoring Use Case

You can check out the Network Performance Monitoring case study with Station 22,
an international beverage manufacturer and distributor, to find out how they
pinpointed undetected network problems on their firewall using MOS Score,
Network Performance Monitoring, and Network Device Monitoring.
He’s a little summary about their problem and their solution.
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Station 22 began experiencing VPN network application slowness and instability for
their Microsoft Teams and GoToMeeting application.

Using Obkio’s Network Performance Monitoring software, they identified that their
network latency was constant but the packet loss kept increasing. Specifically, there
was a lot of packet loss occurring over the Internet as the bandwidth usage
increased during the day, likely due to increased activity from users. They
pinpointed the exact percentage of packet loss in the Obkio app with the graph
tooltip.

They looked into the MOS Score chart to find that it reported thresholds well above
what is normally acceptable.

Finally, they moved to Obkio's Network Device Monitoring feature on their firewall
and saw that the CPU usage was well above the 40% threshold shown on their
Firewall's GUI.
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By using all three features, Station 22 was able to get a complete overview of all of
the metrics affecting their network performance, and realized that the type of
problem they were facing was known and could be fixed with a firmware update of
the firewall. They immediately performed emergency maintenance and the problem
was resolved in no time.

Start Measuring VoIP Quality with Mean Opinion Score for Crystal
Clear Call Quality!
Congratulations, you're now a VoIP quality measuring expert! By understanding the
metrics and factors that contribute to VoIP quality, such as MOS score, packet loss,
jitter, and latency, you're well-equipped to diagnose and troubleshoot any issues that
arise with your VoIP system.
Network performance, specifically VoIP Quality, can be affected by a variety of
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different factors, which is why it’s important to get a complete, end-to-end overview
of all these factors to truly understand what is wrong, and how to fix it.

Whether you're a business looking to improve your customer service or an individual

seeking a high-quality VoIP experience, these tools and tips will help you ensure
that your calls are crystal-clear and enjoyable. So go forth, make some VoIP calls,
and experience the magic of high-quality audio!

Put It to the Test: Trying Is the Ultimate Way to Learn!

Start monitoring VoIP Quality and MOS Score in minutes using Obkio's end-to-end
network monitoring tool!

14-day free trial of all premium features

Deploy in just 10 minutes

Monitor performance in all key network locations

Measure real-time network metrics

Identify and troubleshoot live network problems

 You can rest assured that we're not like those pushy Sellsy people - there's no catch
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here. We firmly believe in the excellence of our product, but if it's not the right fit for
you, we understand and want what's best for you.

These might interest you

VoIP Monitoring 101: Keeping Your How to Measure Jitter & Keep Your
Calls Crystal Clear Network Jitterbug Free

Say goodbye to network headaches.

Get a live demo of Obkio now!
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