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3.2. Codec Selection
Codec (Coder / Decoder)
Codecs are used to convert analog signals to a digital bit stream, and another identical
codec at the far end of the communication converts digital bit stream back into an
analog signal. Codecs vary in the sound quality, the bandwidth required, the
computational requirements, etc. Codecs generally provide a compression capability
to save network bandwidth. Some codecs also support silence suppression, where
silence is not encoded or transmitted. In the VoIP world, codec's are used to encode
voice for transmission across IP networks.
Micronet VoIP gateway supports several different codecs, G.711A/µ law, G723.1,
G729, and when talking to each other, negotiate which codec they will use.
Codec Description
Bit Rate
(Kb/s)
Remark
G.723.1 G.723.1 is an ITU-T
standard codec. Its
reasonably low bit rate
(6.3Kbps or 5.3Kbps). Use
of this codec in a product
requires licensing by Sipro
Lab Telecom
5.6 / 6.3 It encodes speech or other
audio signals in frames using
linear predictive
analysis-by-synthesis coding.
G.729 Coding of speech at 8 kbit/s
using conjugate-structure
algebraic-code-excited
linear-prediction
(CS-ACELP)
8 Low delay (15 ms)
G.711 G.711 is the international
standard for encoding
telephone audio on an 64
kbps channel. It is a pulse
code modulation (PCM).
This is most
64 mu-law (US, Japan) and A-law
(Europe) companding
All VoIP packets are made up of two components: voice samples and IP/UDP/RTP
headers. Although the voice samples are compressed by the Digital Signal Processor
(DSP) and may vary in size based on the codec used, these headers are a constant
40 bytes in length. This table shows the nominal Ethernet bandwidth consumption.
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