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1. Go to the Qualify Test site for your environment.
   The VoIP Qualification Test page appears.
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2. Download Visualware Browser Compatibility Service (BCS) from by using the page here link, and then run it on your computer.

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   To understand what the test parameters such as jitter or packet loss mean, click the Understand results link.

   
   

3. In the Site Information section of the page, specify values in the fields, and then click START TEST.
   The test results appear on the page.

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   The test results may take up to five minutes to appear on the page.

   
   

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   You can reload the page by clicking STOP TEST.

   
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4. If you want to share the test results with LiveVox, click EMAIL RESULTS.
   An email containing the test results is sent to LiveVox. 

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   The subject line of the email begins with the following text: VoIP Quality Report

   
   

If an Issue Occurred In the Past

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LiveVox Troubleshooting Methodologies

You can send the results of the Qualify Test to LiveVox by email by using the EMAIL RESULTS button.

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By default, the subject line of the email begins with the following text: VoIP Quality Report

The following sections describe the steps that LiveVox performs depending on the test results displayed in the VoIP Quality Report email. 

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title Route Test Results
Introduction: Route test measures the speed and route a data packet takes to a destination server. Info icon false This test works by sending test packets of data to a specific destination address and records each device (represented as hops) passed by the data on its journey. A minimum, maximum, and average response time measurement is recorded per hop. Objective: Number of hops must not be more than 20.  Test: In the Route Results section of the email, click the Detailed Route Results link. The My Connection Server page appears. In the Route Test section on the page, check the number of hops. Info icon false In this image, the number of hops is 14, which is not an issue. Issue Resolution: If the number of hops is more than 20, LiveVox does not escalate the case. You must contact your Internet service provider to identify the cause of the issue.  Info If you want LiveVox to pursue the case, the case is escalated while ensuring that the route result is noted.

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title VoIP Test Results
Introduction: VoIP test emulates VoIP calls by using the selected codec and number of lines to provide a Mean Opinion Score (MOS) based on a combination of results.  Info icon false MOS is a metric to assess the overall quality of a call.  Objectives: Average round-trip time (RTT) must not be more than 200 ms. Note icon false If the agents are located in countries outside the United States of America, such as India or the Philippines, the average RTT must not be more than 300 ms. Upstream and downstream jitter must not be more than 20 ms. MOS must not be less than 3.  Info icon false Minimum MOS: 0 (worst) Maximum MOS: 4.2 (best) Upstream and downstream packet loss must not be more than 3% because this could affect audio quality. Test: In the VoIP Results section of the email, click the Detailed VoIP Results link. The My Connection Server page appears. In the Summary section on the page, check the values for the aforementioned parameters.   Issue Resolution: If an objective is not met, LiveVox does not escalate the case. You must contact your Internet service provider to identify the cause of the issue.  Info If you want LiveVox to pursue the case, the case is escalated while ensuring that the route VoIP result is noted.

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title Firewall Test Results
Introduction: Firewall test checks if outbound communication to the LiveVox application ports is permitted by your network devices.  Info icon false Outbound communication to the LiveVox application ports is required for the platform to function correctly. Blocked ports may result in the loss or degradation of service. Objective: Ports must not be blocked. Test: In the Firewall Results section of the email, check if the ports are reported as blocked. Issue Resolution: If the ports are reported as blocked, contact the LiveVox Customer Care Team. Info icon false The following table describes the standard use of each port. Port Use TCP 443 Web traffic (HTTPS). TCP 8080 WebRTC traffic (web and audio paths established simultaneously). UDP 5060 SIP-Trunks when your PBX does not support encryption with TLS version 1.2. TCP 5061 SIP-Trunks when your PBX supports encryption with TLS version 1.2. Also used for sLVC.  TCP 5071 sLVT (hard-phones supported: Polycom VVX-310 and SoundPoint-331). UDP 15000-30000 RTP streams (audio). Note icon false A blocked port can be considered to be the cause of the issue depending on your voice implementation (that is, how you have established the audio path). If the port that is intended for establishing the audio path is reported as blocked, you must enable itare asked to enable the port. For example, if you cannot establish the audio path via WebRTC and the test result shows that the port TCP 8080 is blocked, you must are asked to enable the port. Warning icon false If you are asked to enable a port, ensure that the port is enabled only in the outbound direction.

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