Network Assessment Instructions

Updated at April 20th, 2026

Prerequisites: Release notes April 21: Installing the Network Assessment Tool The software can be downloaded from here: Network Assessment Mac App Rebuilt from the Ground Up - March 2026 Running the Assessment What does this test do? Network Assessment Hosted Voice Quality Call Testing Packet Prioritization 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.

Release notes April 21:

Mac Version: The Mac application has been completely rebuilt, with many bug fixes, feature enhancements, and support for light and dark themes. This new version will bring parity between our Mac and PC versions globally.

Packet Priority: We have built A/B testing into both the Mac and PC versions of the Network Accessibility Test suite. This ensures that customers allow Hosted Voice packets through their gateways ahead of other media types, improving QoS confidence.

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	The purchased software key
Location address	Required	123 Anywhere Ave S, Calgary, AB, Canada T1Y-4G5

Click the "Download for Windows" or "Download for Mac" Button based on your operating system.

Mac App Rebuilt from the Ground Up - March 2026

Version 2.0 (5) of the Mac application is here, bringing full parity with the PC version. This rebuild includes extensive bug fixes, feature updates, and support for light and dark themes.

If you are already using the Mac app, you will be prompted to update automatically the next time you open it.

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 when 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 by 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 to test not only router compatibility but also network strength and whether 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 can 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 the location name for the location you want 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.

Hosted Voice Quality

This test runs a continuous quality monitoring test while the application is open.
This is useful for determining whether network issues at specific times of day could affect call quality.

Path Analysis

MOS Score Voice Quality (Mean Opinion Score)

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 Load"
Click "Run".

Packet Prioritization Testing

(New Functionality Added April 2026)

The goal of prioritizing Hosted Voice traffic is to avoid delaying it by moving it ahead of larger or less time-sensitive packets. This is done within a router that implements Priority Queuing. The router places Hosted Voice packets in the "priority queue" and transmits them before other packets that may have arrived at the router before the Hosted Voice packets arrived. The router can identify the markings on the Hosted Voice packets, place them in the priority queue, and then transmit them first.

This is tested as part of the Network Assessment for a LAN, with results being rendered in the Firewall test results grid.

Test Methodology

The Reply packet prioritization test performs two tests. The first test is a baseline test in which all packets are sent with a DSCP marking of 0x00. The second test sends large 1500-byte packets with DSCP 0x00 marking and small 128-byte packets with DSCP 0xEF marking representing the Hosted Voice packets. All packets in both tests are numbered and transmitted in sequence by the Network Assessment Tool.

During the second test, the expected result is that the router (running a Priority Queuing algorithm) will place the 128-byte 0xEF-marked packets in the priority queue and transmit them ahead of the larger 1500-byte 0x00-marked packets. If those "Hosted Voice" packets arrive out of order or after the 0x00-marked packets at the test receiver, it is assumed that prioritization is not being used.

In a normal case where a SIP Client has a private IP address, and NAT is performed by a CPE edge router, only the IP address in the IP header is replaced by the NAT. In the example, that would be 10.4.12.35 -> 155.166.40.93 being replaced in the IP header. With SIP-ALG = OFF, the original private IP address will still be visible end-to-end in the SIP INVITE message at the receiving HPBX. With SIP-ALG = ON, the ALG Gateway (often also the CPE edge router) will place its address in the SIP INVITE as the VIA, From, and Contact fields.

Remedy

There are several ways to prioritize Hosted Voice traffic with your router:

Quality of Service (QoS) settings: Many routers offer QoS settings that let you prioritize certain types of traffic, such as Hosted Voice, over others. You can typically access these settings through the router's web-based interface. Many routers have a "Hosted Voice Prioritization or VOIP" setting predefined and visible.
Port forwarding: If your Hosted Voice service uses a specific port, you can set up port forwarding to prioritize traffic on that port.

Traffic shaping: You can use it to limit bandwidth for certain types of traffic, such as file sharing or streaming, and allocate more bandwidth to Hosted Voice traffic.
Virtual LAN (VLAN): If your router supports VLANs, you can create a separate VLAN for your Hosted Voice traffic and assign it a higher priority than other types of traffic.
Differentiated Services Code Point (DSCP): Some routers allow you to set a DSCP value for different types of traffic, which can be used to prioritize certain types of traffic

It's important to note that these techniques may not work with all routers and may require some configuration. If you're not comfortable making these changes yourself, you may want to consult a network administrator or contact your router's manufacturer for assistance.

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 is adapted to hosted voice packets) in the ITU-T PESQ P.862. The standard defines how to calculate the MOS score for Hosted Voice calls based on multiple factors, including the specific codec used. 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 based on the G.711 codec, which is by far the most commonly used for Hosted Voice calls. The maximum MOS for a G.711 call in Hosted Voice is 4.4 (even though the standard sets 5 as the maximum).

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 states that the Applicable Network performance will not drop below 3.8, with MOS calculated using the standards-based E-model (ITU-T G.107).

Jitter: Also known as delay variation, jitter is the variation 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 exceeds 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 as the time a packet takes to travel from its source to its destination. The primary elements of digital networking and packet-switched networks are the transmission media and the processing at intermediate switching nodes. All media, 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 reach the destination than larger ones. Once a full or partial packet reaches a switching node, it must be processed for consumption or retransmission 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 by network congestion, in which a packet is discarded by an intermediate switching node due to queue limitations or a queuing policy. Packet loss is measured as the percentage of packets lost relative to the total number of 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.

KB Knowledge