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Video conference network test

Are you about to introduce video conference services in your organization? Then you are perhaps wondering whether your network is capable of carrying this new and quality demanding videoconference traffic? Keep on reading, you have come to the right place.

Introducing video conference in your network, or at a new branch office, will increase the demands on your network in terms of data transport “quality”. In order for video conferencing to work well, your network must guarantee:

  • A sufficiently low level of IP packet loss
  • Low end to end delay
  • Limited jitter (IP packet delay variation)

Most likely, your decision to introduce video conference aims at lowering the cost and to be environmentally friendly. Probably a positive and well founded decision.

But what if this introduction results in lots of problems for your IT operations team? What if videoconference meetings are disconnected and discussions become impossible due to bad video and voice quality? The consequence is then increased costs for your IT operations as well as the organization as a whole, instead of providing all the good benefits that you had hoped for.

Test your network

Before launching your new video conferencing service we strongly advise that you test your network to make sure that the introduction goes smoothly not causing problems for your organization and your IT operations.

Keep on reading for a complete network test suite that we recommend that you should use. All of the network tests are absolutely relevant and can be done in an easy way with Netrounds. You can easily download active measurement probes and get started with your first network tests within 15-30 minutes.

Ping and SNMP are not enough

When it comes to quality demanding services such as video conferencing, knowing that you can ping a host through the network is not really enough. Ping gives a too rough measure, not capturing performance related problems due to for example duplex mismatch or a misconfigured network device that perhaps rewrites the QoS header values.

Likewise, using SNMP statistics based on 5 minute average values of interface counters is not enough either since there can be short peaks in bandwidth that overloads the network which causes packet loss that influences the video conference.

Recommended video conference network test suite

The tests described below use active measurement probes that generate real traffic in your network. This traffic will be generated concurrently with your other existing traffic in your network, just as the new video traffic also will.

Why test my network?

  • Make sure new service launch is successful
  • Reduce risk of increased load on your IT operations team
  • Using ping and SNMP is not enough

How does it work?

  • Deploy active network probes at strategic locations in your network
  • Inject real-world traffic in your network to resemble a real situation
  • Verify that measured quality remains at an acceptable level
Test Purpose of test Description
Connectivity verification Verify that there is connectivity between your offices where video conference service will be introduced Run a one minute UDP test at a low bitrate (100 kbit/s) between selected measurement probes.
DSCP remapping Verify correct QoS DSCP mapping through the network UDP packets marked with different DSCP values are sent between probes and analyzed at the other end to make sure that the QoS encoding is not overwritten.
UDP QoS test Verifies that prioritization works between a prioritized video stream and another lower prioritized UDP stream

Sends two UDP streams that together overloads the network connection: One low priority and one high priority (video stream) and verifies that the high priory stream is unaffected.

Note: this can disturb existing traffic and needs to be considered before performing tests

TCP QoS test Verifies that prioritization works between a prioritized video stream and another lower prioritized TCP stream Sends one high priority UDP video stream and one low priority TCP stream and verifies that the high priority stream is unaffected.
TCP performance test Test the TCP performance of the network connections A TCP stream is sent between selected probes between 1-20 minutes. The acceptance criteria is set by the user depending on the bandwith of the network.
Network stability test Capture possible intermittent quality problems

A VoIP stream is sent between the probes continuously during a period spanning from three days to two weeks.

Parameters such as jitter, delay, packet loss, reordering are measured and evaluated against standardized requirements.

SIP server test Make sure that SIP signalling works Set up media streams between two or more Netrounds probes. SIP signalling functionality and SIP server performance are verified (in case your video conference system uses SIP).

Passing all of the above network tests makes sure that your network won’t cause problems when implementing a video conference solution in your organization.

After your successful implementation you can use Netrounds to make sure that the network quality is not degrading. This is done by using Netrounds to monitor the quality over time, configuring alarm thresholds, and receiving automated weekly SLA reports that visualizes videoconference quality in your network.

Download your free trial of Netrounds, or request a personal presentation and demo for more information on how Netrounds can help you.