Is your network ready for voice, video, and Unified Communications? High quality delivery for voice, video, and UC services does not happen by accident. To ensure top performance, it is important to plan the implementation, including: baselining present network performance, validating carrier service level agreements, measuring traffic thresholds, validating QoS settings, and resolving underlying problems before they impact the users.
A network assessment to prepare for voice, video, and UC traffic will require more than throughput tests and capacity planning on the network. For example, most VoIP calls generate under 100Kbps per call, so ensuring that the network can handle high throughput end to end doesn’t test the unique requirements of voice. Rather, metrics such as packet loss, jitter, and network delay have a major impact on call quality, and should be validated during the network assessment. Normal network traffic will compete with these services for bandwidth and forwarding priority, requiring a solid QoS configuration across the network. Testing and validating QoS settings with simulated traffic should be a part of the network assessment.
Once the network has been assessed for problems that will impact the performance of voice and video, a VoIP Audit should be run to simulate traffic streams and fully evaluate network readiness. This test will also bring to light any other unexpected issues that will impact these services before they are deployed. This assessment applies in principal to video and UC as well, and can be adjusted to simulate these services as well.
For this task, engineers will need a tool (such as the OptiView XG) that can simulate the traffic load of multiple voice and video streams to remote endpoints throughout the network, while simultaneously measuring traffic performance with respect to loss, delay, jitter, and throughput. The test should compare the actual performance of the test traffic against configured thresholds to provide a simple pass/fail summary result. Appropriate thresholds can be determined with help from the VoIP system vendor, who will typically provide recommended metrics for optimum system performance and call quality.
Testing network readiness for voice and video differs from standard throughput testing on the network. Each call generates a relatively low amount of bandwidth, depending on the codec in use. The chart below gives the approximate Ethernet bandwidth in use for each codec and number of simultaneous calls.
Using the performance testing tool, a traffic stream is configured to simulate these levels of call loads. The traffic can be configured using a packet size of around 200 bytes which accurately represents a real RTP packet. In the test, settings for QoS can be configured, which will then validate both local devices and service provider infrastructure, ensuring they are being handled properly from end to end.
In addition to simulating RTP streams, additional traffic streams should be layered in by the performance test to simulate call setup traffic. If packet loss occurs during call setup, the call may not establish at all. The thresholds in this individual stream of the test can be configured to be more loss sensitive. A third stream should also be configured to simulate background user traffic, loading the network to normal usage levels. This will create more realistic traffic levels for measuring simulated voice and video streams. Testing in this way gives engineers a clear picture of how real application traffic will perform once it is deployed, while unearthing problems which may impact performance.