Network Assessment Instructions

Updated at November 29th, 2024

Prerequisites: Installing the Network Assessment Tool The software can be downloaded from here: Network Assessment Running the Assessment What does this test do? Network Assessment VoIP Quality Call Testing Network Assessment Scores - Interpreting the Assessment Measuring Hosted Voice Call Quality with MOS (Mean Opinion Score) Performance Measurement Definitions

Prerequisites:

Software license key

Please note you will only ever see this Activation code once, so please save it somewhere safe.

Installing the Network Assessment Tool

The software can be downloaded from here: Network Assessment

Before downloading the application, you will be prompted to enter some information about the test site. Please see the table below for an example.

Company name	Required	Demo Company Name
Your company email address	Required	demo@company.com
Company phone domain	Required	company.com
Location name	Required	Calgary South
Security key	Required	Your purchased software key
Location address	Required	123 Anywhere Ave S, Calgary, AB Canada T1Y-4G5

Click the "Download for Windows" button.

Your download will then begin, and your activation code will be displayed. Please copy this code to the clipboard. You will need the code to proceed.

Select the file that you downloaded.

When you are finished installing the application, please launch it and enter the activation license key once you are prompted.

Enter the activation code that you copied into the clipboard in the previous step.

Click on "Run Network Assessment."

The Network Assessment will load working through the eight tests depicted below. Let the process run its course to review the results.

Done! The app is now installed and ready to test the network for SIP compatibility. Please run this test continuously for 3 DAYS so that it tests not only the router compatibility but also the network strength and if it drops. Additional information is provided below if you are interested in diving into the details. If you have started the test and let it run, our team will be able to review the results.

Running the Assessment

To run a Hosted Voice Network Assessment, first navigate to the location where you want to run it by clicking the Locations tab on the right-hand menu.

Then, click on the location name for the location you would like to run the assessment. This will take you to that location's dashboard.

Once on the location dashboard, click the "Run" button to the right of the Hosted Voice Network Assessment line.

Once completed, the test results can be exported to a .pdf and shared.

What does this test do?

This test allows you to perform a variety of tests on your client's sites; some of the tests include:

The network passed the test in the sample below, which indicates that it is suitable for voice traffic. However, keep in mind that the results can change. For example, the test result would differ if your network resources are saturated by another process competing for the same bandwidth.

VoIP Quality

This test runs a continuous quality monitoring test as long as the application is open.
This is useful for determining if network issues at specific times of the day could impact call quality.

Path Analysis

MOS Score Voice Quality

Call Testing

This test set allows you to run concurrent test calls under different scenarios.
This is useful for checking if the network can handle the call volume the site requires.

The Concurrent Call Test Simulates and measures the performance of a specified number of concurrent calls on the network. It will return the MOS, Jitter, Latency, Packet Loss, and R-factor for each call.

Select "Concurrent Call Test" from the options in the drop-down.
Input the number of concurrent calls
Select the codec G7.11, G7.22, G7.29 (List of Codecs).
Click Run

The Concurrent Call Test Under Data Load: Simulates and measures the performance of a specified number of concurrent calls on the network under load. It will return the MOS, Jitter, Latency, Packet Loss, and R-factor for each call.

Select "Concurrent Call Test Under Data Load" from the options in the drop-down.
Input the number of concurrent calls
Select the codec G7.11, G7.22, G7.29 (List of Codecs).
Select The bandwidth Loadd
Click Run.

Network Assessment Scores - Interpreting the Assessment

After completing the Network Assessment, the system will return a score of either Passed, Moderate, or Issues Found. When a moderate score is reflected, the location returns higher or lower scores against key metrics. The following are the thresholds for each component measured:

MOS is below 3.8
Download Speed is below 10 Mbps
Concurrent Call Test: Latency > 80ms and/or Jitter > 30 and/or Packet Loss >3
Packet Prioritization: If not detected, a Moderate score is given
Double NAT: If detected, a Moderate score is given.

Measuring Hosted Voice Call Quality with MOS (Mean Opinion Score)

The Mean Opinion Score (MOS) has been used for decades to measure overall voice call quality. It is a rating from 1 to 5 of the perceived quality of a voice call, with 1 being the lowest score and 5 the highest for excellent quality. The International Telecommunications Union (ITU-T) has standardized it.

MOS was originally developed for traditional voice calls but has been adapted to Voice over IP (VoIP) in the ITU-T PESQ P.862. The standard defines how to calculate the MOS score for Hosted Voice calls based on multiple factors, such as the specific codec used for the Hosted Voice call. Each codec (e.g., 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 G.729.

The MOS score we measure is the G.711 codec, which is by far the most commonly used codec for Hosted Voice calls. The maximum MOS in Hosted Voice for a G.711 call is 4.4 (even though the standard sets 5 as the highest).

The following table lists the different qualities and the lower MOS limit. The limit values are from the ITU-T standards.

Performance Measurement Definitions

Mean Opinion Score (MOS): MOS is a measure (score) of a voice call's audio fidelity or clarity. It is a statistical measurement that predicts how the average user would perceive the clarity of each call. The Hosted Voice MOS SLA provides that the Applicable Network performance will not drop below 3.8, where MOS is calculated using the standards-based E-model (ITU-T G.107).

Jitter: Also known as delay variation, jitter is defined as the variation or difference in the end-to-end delay between received packets of an IP or packet stream. When certain packets of information arrive out of order, the conversation becomes jumbled. If jitter creates a delay of more than 50ms, your call quality will degrade significantly, resulting in choppy voice or temporary glitches.

Latency. In the context of Hosted Voice latency, all latency of concern is one-way latency. One-way latency is measured by counting the time a packet travels from its source to its destination. The primary elements for digital networking and packet-switched networks are the transmission media and intermediate switching node processing. All mediums, from fibre optics to coaxial cables, take some time to transmit a packet from a source to a destination. Transmission delays depend on packet size; smaller packets take less time to transmit the complete packet to the destination than larger packets. Once a full or partial packet reaches a switching node, it must be processed to be consumed or re-transmitted to the next destination. Normally reflected in milliseconds from source to destination. Generally, we consider scores below 70ms healthy, 70-120ms moderate and possibly impactful and anything over 120ms alarming.

Packet Loss. Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. It is either caused by errors in data transmission or network congestion, whereby a packet is discarded by an intermediate switching node due to queue limitation or queuing policy. Packet loss is measured as a percentage of packets lost with respect to packets sent. Packet loss is one possible contributor to one-way audio.

Causality

Poor Jitter, Latency or Packet Loss can be the byproduct of one or any combination of the following:

Firewall/gateway TCP or UDP port settings.
Low or oversubscribed bandwidth.
Non-prioritized Hosted Voice packets from the gateway router.
SIP ALG (Application Layer Gateway) activation
Public/private Internet services and dependencies,
Generation of large SIP packet payloads from paging or all-call paging practices,
VPN routing may not be optimal for Hosted Voice SIP traffic to the provider.
Distance between a customer endpoint and the Hosted Voice provider data center.