It is also important to consider factors like sound quality and delay when choosing a codec. The standard measure of sound quality is known as a mean opinion score (MOS), a subjective measurement that ranges in value from 1 (lowest) to 5 (highest). Similarly, because different codecs use different compression schemes, they are subject to processing delays that will ultimately impact the perceived quality of the voice call. The table below outlines the MOS scores and relative delay (in milliseconds) associated with each codec.
When choosing a codec, keep in mind that the two elements that it impacts most strongly are sound quality (determined via the MOS score) and required bandwidth.
It should now be clear that the network designer has a number of factors to consider as part of choose a codec for a voice network. While the G.723.1 codec may result in the greatest overall bandwidth savings, it also has a higher degree of complexity, and results in a higher overall delay. Another important consideration is what happens when a single call needs to use multiple codecs. For example, a call might originate on a packet-switched network, but the destination might be on the PSTN. In this case, assuming that the packet-switched network is using the same codec throughout (say G.728), multiple encoding would need to take place – the call would first be encoded using G.728, but would then need to be transcoded to G.711 for transport over the PSTN. When multiple encoding occurs, it not only introduces additional delay, but also lowers the overall MOS quality of the call. As a general rule, the end-to-end delay of a phone call should not exceed 150 ms (as per ITU recommendations), so keep this in mind.