User Manual

VoIP Settings
7-8
7
Codecs
A codec (coder/decoder) is the way a voice analog signal is converted into a digital
bitstream to send over the network, and how it is converted back into an analog
signal at the receiving end. Codecs differ in the type of data compression that is
used to save network bandwidth and in the time delay caused in the signal. This
results in different voice quality experienced by the user.
The voice codecs in common use today have been standardized by the International
Telecommunication Union Telecommunication Standardization Sector (ITU-T) and
are identified by a standard number, such as G.711 or G.726. The same codec must
be supported at each end of a VoIP call to be able to encode and decode the signal.
Since devices in other networks may want to use different codecs, the OD200
provides support for several common standards.
Figure 7-5 Codecs
Codec – Lists the codecs supported by the OD200. You can enable specific codecs
to use, or enable all. Alternatively, you may want to disable certain codecs, such as
high-bandwidth codecs, to preserve network bandwidth.
• PCMA (G711.aLaw): The ITU-T G.711 with A-law standard codec that uses Pulse
Code Modulation (PCM) to produce a 64 Kbps high-quality voice data stream. This
standard is used in Europe and most other countries around the world.
• PCMU (G711.uLaw): The ITU-T G.711 with mu-law standard codec that uses
Pulse Code Modulation (PCM) to produce a 64 Kbps high-quality voice data
stream. This standard is used in North America and Japan.
• G723.1: The ITU-T G.723.1 standard low bitrate codec that uses Multi-Pulse
Maximum Likelihood Quantization (MP-MLQ) and Algebraic Code Excited Linear
Prediction (ACELP) speech coding to produce data streams of 6300 and 5300 bps.