IP telephony network test

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

Introducing IP telephony 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 IP telephony to work flawlessly, 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 IP telephony aims at reducing the cost for your telephony service or to be able to increase the number of subscribers without having to make heavy investments in legacy equipment.

But what if this introduction results in lots of problems for your IT operations team? What if phone calls are disconnected and conversations become impossible due to bad voice quality? The consequence is then increased costs for your IT operations, instead of providing all the good benefits that you had hoped for.

Test your network

Before launching your new IP telephony 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.

Below is a complete and recommended 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 IP telephony.

Recommended IP telephony 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 IP telephony 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 speech quality (Mean Opinion Score, MOS) remains at an acceptable level

Test	Purpose of test	Description
Connectivity verification	Verify that there is connectivity between your offices where IP telephony 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 high priority VoIP stream and another lower prioritized UDP stream	Sends two UDP streams that together overloads the network connection, one low priority, and one high priority (VoIP), 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 high priority VoIP stream and other lower prioritized TCP stream	Sends one high priority UDP stream (resembling the VoIP stream) and one low priority TCP stream, and verifies that the high priory 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	Two or more Netrounds probes sets up real SIP calls to verify SIP signalling functionality. SIP server performance is also verified.
Network security test	Make sure that it is not possible to intercept phone calls	Verify that the network blocks ARP poisoning attempts that can cause man-in-the-middle attacks.

Passing all of the above network tests makes sure that your network won’t cause problems when implementing IP telephony.

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