Speech coding methods – 2N VoiceBlue Lite - Manual v2.6 User Manual

Page 43

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2N - VoiceBlue Lite

5.1. Speech Coding Methods

Voice and signal transmission is strictly separated in VoIP
networks. The RTP (Realtime Transport Protocol) is mostly used
for voice transmission in modern VoIP networks. The purpose of
the RTP is to transmit data from the source to the destination and
ensure the real time data (voice) transmission.

Codecs are used to save capacity of data channels, processing the
voice signal using various algorithms with the aim to minimise the
volume of user data. The degree of compression used by codecs
influences the quality of voice to be transmitted. This means that
the higher voice quality is required, the wider data band
(transmission rate) should be used. The voice transmission quality
is evaluated using the Mean Opinion Score (MOS) where 1 means
the lowest and 5 the best quality. A survey of codecs supported by
VoiceBlue Lite is included in Tab. 3.

Codecs Supported by 2N - VoiceBlue Lite

Transmission

rate [kb/s]

Standard

Algorithm

MOS

*

G.711 PCM

64 4.1

G.726 ADPCM

32 3.85

G.729 CS-ACELP

8

3.92

G.723.1 ACELP

5.3

3.65

Tab. 3 Table of Codecs Supported by 2N

®

- VoiceBlue Lite

To achieve a high-quality voice transmission requires not only
keeping of a constant necessary transmission rate during the whole
connection but also a constant low data packet transmitting time.

*

Multiple the above mentioned rate by eight (four full duplex calls) and add the transmission rate

necessary for the TCP and IP header to the resultant value for to get a successful 2N

®

- VoiceBlue

Lite Lite connection.

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