How to Test VoIP Devices

A VoIP telephone uses voice over internet protocol to transport voice from one point to another. It has several interesting components that work together to provide convenient and economical communication with a large number of features. Each of these components is necessary, but each also presents challenges for measurement methods and standardisation.

We will discuss signal flow from phone to test system in the sending direction. The receiving direction is basically the reverse (please see the Figure).

In the sending direction, a user interacts directly with a handset, headset or speakerphone device when talking and listening. Current best practices for using a head and torso simulator (HATS) and other transducers handle these acoustic challenges well.

The long delays typical of a VoIP call makes the user extremely sensitive to echo from the phone at the far end. VoIP phones, therefore, have to have very small amounts of echo. Echo cancellers can reduce echo greatly, but the residual echo is sometimes in bursts or spurts that can be especially annoying. Methods in common use today are often not adequate to measure small amounts of echo, or to evaluate it in a subjectively meaningful way. This subject has already been addressed in some standards, but implementation is not yet widespread.

A VoIP phone is likely to have extensive speech processing. This can include voice activity detection, single or multiband compression, acoustic echo cancellers, and other processes with nonlinear and/or adaptive characteristics. Some traditional test signals, such as sine waves, do not activate these processes normally. Real or artificial speech generally does activate these functions, so they are a good choice for measuring frequency response.

Distortion measurements are difficult without the use of sine waves. A test signal consisting of a band of noise presented in a pulsed pattern has shown promising results. Methods using such signals may soon appear in published standards. The issue of test signals and measurement methods is an ongoing subject of investigation and revision in many standards.

Most standards specify the use of linear or nearly linear codecs for testing. This is to avoid problems with reduced performance or difficulty in passing test signals which can occur in low bit rate codecs.

The most frequently asked question about testing VoIP phones is "how do I interface a VoIP phone to my test equipment?" First, the interface has to be able to establish a call between itself and the phone. Then the packetized data in the network has to be reassembled into a continuous signal stream. It then has to be presented to the test system in a usable and calibrated analog or digital format, as required. The delay of the interface has to be known, stable, and free from jitter or dropouts.