Difference between revisions of "Information on Audio Codecs"

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codec."
 
codec."
  
Also it becomes clear that the X-Lite/X-Pro versions support 6 different audio compression codecs, G.711A and G.711U being the codecs with the best sound quality but with low voice compression.
 
  
 
=== G.711A & G.711U ===
 
=== G.711A & G.711U ===

Revision as of 22:17, 30 January 2007

X-lite has various audio codecs implemented. The implemented audio codecs of the X-lite 3.0 version are DVI4, G.711 uLaw/aLaw, GSM, iLBC as (narrow band audio codecs) and BV-32, BV FEC, DVI4, L16 PCM (wide band audio codecs). Eyebeam version 1.5 has a wider range of codecs available.

From the User’s Guide for X-Lite and X-PRO (p.34) the following can be said:

"When two VoIP systems are establishing a call, they negotiate an audio compression codec they are going to use. Which audio compression codec to choose depends on many factors: which audio compression codecs are installed on both systems, bandwidth limitations, desired sound quality, etc. During the negotiation, X-Lite/X-PRO offers to the remote system the first audio compression codec from the list. If the remote system rejects the audio compression codec, XLite/X-PRO offers the next one from top to bottom until they both accept the audio compression codec."


G.711A & G.711U

G.711 is a codec primarily used in telephony, released in 1972. It samples the analogue voice input signal using pulse-code modulation (PCM) at a sampling rate of 8000 samples/second.

The G.711 standard has two algorithms: the A-law (mostly used in Europe) and the U-law algorithm (used in North America and Japan).

The G.711 codec encodes the pulse-code modulation samples (sampling rate of 8000 samples/second) to logarithmic 8-bit samples. This will lead to a (8000 times 8-bit=) 64 kbit/s bitstream. Note that this is the datastream without information send in the headers of the packages, so the actual datastream will be higher. Because the sampling rate of 8000 samples/second and the encoding to 8-bit samples are intrinsic properties of the codec, the bitstream can not exceed 64 kbit/s. If one wants to increase the quality of the voice by increasing the used bandwidth (> 64 kbit/s) one has to use another codec.