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titleVoIP Test Results

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
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MOS is a metric to assess the overall quality of a call. 

Objectives:

  • Average round-trip time (RTT) is not more than 200 ms.

    Note
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    If the agents are in locations such as India or the Philippines, the maximum RTT should not be more than 300 ms.


  • Upstream and downstream jitter is not more than 20 ms.
  • MOS is not less than 3. 

    Info
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    • Minimum MOS: 0 (worst)
    • Maximum MOS: 4.2 (best)


  • Upstream and downstream packet loss is not 3%-4% 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.
 Summary Section

Info

If you insist that LiveVox pursue the case, the case will be escalated while ensuring that the test result is noted.

Issue Resolution: If an objective is not met, LiveVox will not escalate the case and will ask you to contact your Internet service provider to identify the cause of the issue. 

Info

If you insist that LiveVox pursue the case, the case will be escalated while ensuring that the VoIP result is noted.



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titleFirewall Test Results

Firewall test checks if outbound communication to the LiveVox application ports is permitted by your network devices. 

Info
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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.


Note
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If the result of this test shows Blocked, contact the LiveVox Customer Care Team.

Objective: Ports are not 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
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The following table describes the use of each port.

PortUse
TCP 443Web traffic (HTTPS)
TCP 8080WebRTC traffic (web and audio paths established simultaneously)
UDP 5060Session InIP-Trunks when your Private Branch Exchange (PBX) does not support encryption with TLS version 1.2
TCP 5061SIP-Trunks when your PBX supports encryption with TLS version 1.2. Also used for sLVC (99.9% of the clients use the Secured accounts) 
TCP 5071sLVT (hard-phones supported: Polycom VVX-310 and SoundPoint-331)
UDP 15000-30000RTP streams (audio)



Note
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A blocked port can be considered to be the cause of an issue depending on your implementation of the port (that is, how you establish the audio path). For example, if you are establishing the audio path via WebRTC and the test result shows that the port TCP 5061 is blocked but TCP 8080 is enabled, this is not the cause of the issue.


Warning
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Ports must be enabled only in the outbound direction. 


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