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Overview

The Qualify Test is a snapshot of the current Internet connection between you and LiveVox. The test is intended to gauge the health of your Internet connection. This document provides an overview of the test and answers to some frequently asked questions.

You must run the Qualify Test only when an issue occurs (as opposed to running it when an issue does not currently exist or if the issue occurred in the past).

Qualify Test Site

The following table contains links to the Qualify Test site for the LiveVox environments.

EnvironmentQualify Test Link
NA3

https://qualify.na3.livevox.com/

  • This link is applicable to only versions earlier than U12.
  • If you are using U12 or a later version, your environment is NA3.VA2. The Qualify Test site does not exist for NA3.VA2.
  • If you encounter an issue in U12 or a later version, see the following section: What to Do If You Cannot Run the Qualify Test?
NA4https://qualify.na4.livevox.com/
NA5https://qualify.na5.livevox.com/
NA6https://qualify.na6.livevox.com/

What to Do If an Issue Occurred Earlier?

  • Keep monitoring the connection.
  • Provide the following information about the occurrence (at a minimum) to LiveVox:
    • Timestamps
    • Call examples
    • Number of affected agents
    • Number of affected locations
    • How the agent was establishing the audio path

Run the Qualify Test only when an issue occurs.

How LiveVox Troubleshoots

If you send the results of the Qualify Test to LiveVox via email (default subject line: VoIP Quality Report NA3 - JV Test.msg), depending on the results, LiveVox performs the following steps.

The results cannot show more than 20 hops. 

Where to see this? In the email you will see the following section. Click on the link:

After clicking on it, you will be routed to a page where you have to look for the below section and there you will see the amount of hops, in this case this is not an issue because the total amount is 14.



What to do if the # of hops is greater than 20 hops? Don't escalate the case, just recommend the client to check with their ISP why the routing is taking this amount of hops. If the client continues pushing back, escalate and make sure you highlight this result.

  1. The result cannot show more than 200ms of RTT (Average)

    Additional note about the RTT

    This average could be bigger if the agents are in locations such as India or Phillipines. The maximum here could be 300ms.


  2. The result cannot have more than 20 ms of Jitter.
  3. The result cannot have less than 3.0 MOS score → Min value: 0 (worst) - Max value: 4.2 (best)
  4. The result cannot have 3-4% packet loss on the upstream or downstream since this could cause audio quality issues.

    Where to see this?     In the email you will see the following section. You can click on the link of see this specific item in the email:



    If you click on the link, the information will be available in the next section:



    What to do if one or more of the above results appear? Don't escalate the case, just recommend the client to check with their ISP. If the client continues pushing back, escalate and make sure you highlight this result. 


If some logical port is blocked the test will show it (TCP 80, TCP 443, TCP 8080, UDP 5060, TCP 5061, TCP 5071, UDP 15001-15005, UDP 20001-20005 and UDP 29991-29995).

However, it is important to remark that the port may be or not the cause of the issue depending on the implementation that the client has (how they establish the audio path):

  • TCP 443 → Used for Web traffic (HTTPS)

  • TCP 8080 → Used for WebRTC traffic (web and audio paths established simultaneously)

  • UDP 5060 → Used for SIP-Trunks when the client's PBX does not support encryption with TLS version 1.2
  • TCP 5061 → Used for SIP-Trunks when the client's PBX supports encryption with TLS version 1.2. Also used for sLVC (99.9% of the clients use the Secured accounts). If a client just says LVC, probably they are referring to sLVC since currently 99% of our clients are using this type of configuration.
  • TCP 5071 → Used for sLVT (hard-phones supported: Polycom VVX-310 and SoundPoint-331)
  • UDP 15000-30000 → Used for RTP streams (audio)


How to understand the above? If a client is establishing the audio path via WebRTC and the result shows that the port TCP 5061 is blocked but the TCP 8080 is enabled, then, this is not the cause of the issue.


When asking the client to enable ports

Ports MUST BE ENABLED only in the OUTBOUND direction. So, if you are recommending the client to enable a port, make sure you tell them this.

What to Do If You Cannot Run the Qualify Test?

You cannot run the Qualify Test because of security or if you are using the NA3.VA2 environment.

Perform the following steps:

  1. Have your IT department run the following commands from the affected network:
  2. Send the results of the commands to LiveVox.
  3. Escalate the case with this information.
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