eans that voice IP packets are replayed a certain time after they are received in order to pro- vide smooth reproduction of the digitalized voice into its analogue form that can be heard by humans. Network equipment. For provision of voice services, PSTN uses the telephone network with exchanges as network nodes and overlay signaling network (for call management and QoS), while VoIP in general can utilize more intelligent terminals (e.g. smart- phones, computers) and generic IP network nodes (e.g. routers) that are used for all IP traffic. However, again in this case QoS requires an overlay signaling network. In the case of the operator's VoIP, signaling is also used to provide QoS-enabled VoIP pro- vision, in the same (or a similar) manner as voice service previously found in PSTN. However, the protocols used for signaling are different in IP networks (e.g. SIP is used for VoIP, while SS7 is used in PSTN [1]). Besides the different aspects of data transfer and signaling in the case of PSTN (digital telephony networks) and operator's VoIP services (with QoS support), the voice service is the same toward end-users (i.e. they are not likely to note any difference between PSTN and NGN-based VoIP services). Overall, voice has been the primary form of communication between humans since the birth of civilization, and voice services will continue to be important in the future. However, because of the single-dimension nature of the voice-only service (unlike video, as an example), it requires lower bitrates (e.g. compared with video services and most data services). Bitrates are decreasing on one hand due to the higher processing power of user terminals (to use different voice source coding codecs, for compression of voice bits), while on the other hand voice bitrates may increase with the extension of the used frequency band for voice itself (the human ear can hear on frequencies between 20 Hz and 20 kHz), such as using not only 300-3400 kHz frequency band for voice as in PSTN, but gradually up to 20 kHz. Additionally, 