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Product Info

VoIP Readiness

A key component to owning a network phone system is the quality and capacity of your internet connection. To better assist you in evaluating your broadband connection, the following tests will provide feedback on your network bandwidth and line quality.

Note: The VoIP Readiness Test may prompt you to Allow applets from virtualpbx.com to run.

Click 'Begin Test' below to get started.

The Virtual PBX Line Quality Test requires at least version 8 of Flash. Please update your client.

Please make sure you are NOT downloading files, streaming content or connected to a VPN while performing the test.

Bandwidth

The upload and download speeds reported in the Speed Test reflect the bandwidth of your internet condition. When calculating bandwidth required for the VirtualPBX phone system you should allocate 80-100 Kbps of bandwidth per active phone. Keep in mind that not all phones in a business are active at one time. The phone bandwidth requirements are in addition to your current network bandwidth requirements from activities such as email, web surfing and data uploads/downloads. Bandwidth calculations should be made using the smaller of the upload or download speeds.

Firewalls

In order for VoIP data to be received by your network, proper firewall settings must be made. The client UDP firewall check must be open. Please check with your network administrator or check with VirtualPBX support for further help.

Latency

Latency refers to the measurement of delay or time lag between your phone and the server. If there is too much latency in the network, voice quality can become garbled and unclear. Latency can be caused by many factors including router delays, number of routers in the path and congestion caused by insufficient bandwidth. Latency delays under 50 ms. Is considered sufficient for good quality VoIP service. If your latency exceeds the maximum recommendations you should consult your network administrator.

Jitter

Jitter is the measurement of variable delay between the arrival of data packets. Too much jitter can noticeably degrade voice quality. Jitter can be caused by differences in packet processing, router forwarding and congestion of the network. For good voice quality, jitter should not exceed 20 ms. If your jitter exceeds the maximum recommendations you should consult your network administrator.

Packet Loss

Packet Loss refers to voice packets that are not received due to network issues. Since VoIP does not retransmit lost packets, any lost packet will affect voice quality. Packet loss in a voice network should be addressed by the network administrator.

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