VoIP VOICE QUALITY

For several decades, telephone users have been experiencing public switched telephone network (PSTN) based voice communication quality. Subjectively, this experience is used as the main reference for comparing the voice quality from other voice communication systems. In PSTN-based systems, voice has good intelligibility, acceptable speaker identification, naturalness, and only minor disturbing impairments. The PSTN uses G.711 |>law and A-law as compression. Compressed bits are sent on synchronous times-division multiplexed (TDM) interfaces. In VoIP voice communication, a similar G.711 is also used end to end as one of the compression codecs. In VoIP, instead of TDM bits, groups of compressed bytes are sent as a packet on the Internet Protocol (IP) network. The VoIP quality closely matches with PSTN quality under the correct conditions of end-to-end G.711 IP packet transmission. VoIP will also use many high-compression codecs like G.729AB, G.723.1A, and iLBC, which causes VoIP voice quality to be lower than PSTN quality even in best end-to-end packet delivery networks. The goal here is how to reach and finally to exceed PSTN quality through VoIP.
VoIP will have to take care of additional impairments to achieve the quality comparable with the PSTN quality. The four main voice impairment contributors popularly referred for VoIP telephony according to TIA-810A [TIA/EIA-810A (2000)] are delay, echo, voice compression, and packet loss. In a recent upgrade of TIA 116A [TIA/EIA-116A (2006)], additional influencing factors such as G.711 Packet Loss Concealment (PLC), transcoding, gateway loss plan,
and terminal coupling loss (TCLw) are added as other contributors to voice quality. From the deployment perspective, the following major contributors to voice quality can also be considered for VoIP customer premises equipment
(CPE) products.
• Analog front ends that can meet compliance with the PSTN specifications like TR-57 transmission
• Call establishing, call progress tone timing matching to PSTN switching specifications, and avoiding audible ticks, hollowness, and voice transients at call establishing and termination phases
• Country- and deployment-specific deviations and parameters matching
the local PSTN
• Ability to provide lifeline PSTN capabilities or emergency services
• Diagnostics of front end and total system, voice quality monitoring, and incorporating dynamic voice quality feedback
• Supporting right interfaces, signal level combinations for subscriber end-devices like telephones, cordless phone, DECT, and Bluetooth devices
• Front end ability to support wide bandwidth and processors that can support wideband codec modules
In the PSTN, voice quality also degrades from the use of 32-kbps adaptive differential pulse code modulation (ADPCM) to double the transmission channels on the existing digital channels. Regular channels with G.711 use 64 kbps. At some stage of terminal points, digital circuit multiplication equipment (DCME) performs this operation of creating overload channels. The principal application of ADPCM 24 and 16 kbps are also used in overload channels carrying voice in DCME. The 40-kbps channel carries data modem signals in DCME, especially for modems operating at greater than 4800 kbps. VoIP G.711 can deliver better quality than the PSTN channels created through a DCME operation. VoIP also provides several benefits when compared with PSTN – based services as listed in Table 20.1 .