G.711 Codec

G.711 is a high bit rate audio codec from the ITU. It is the native format for voice on the modern digital telephone network.

Although formally standardised in 1988, the G.711 PCM codec is the granddaddy of digital telephony. Invented by Bell Systems and introduced in the early 70's, the T1 digital trunk employed an 8-bit uncompressed Pulse Code Modulation encoding scheme with a sample rate of 8000 samples per second. This allowed for a (theoretical) maximum voice bandwith of 4000 Hz. A T1 trunk carries 24 digital PCM channels multiplexed together. The improved European E1 standard carries 30 channels.

There are two versions: A-law and U-law. U-law is indigenous to the T1 standard used in North America and Japan. The A-law is indigenous to the E1 standard used in the rest of the world. The difference is in the method the analog signal being sampled. In both schemes, the signal is not sampled linearly, but in a logarithmic fashion. A-law provides more dynamic range as opposed to U-law. The result is a less 'fuzzy' sound as sampling artifacts are better supressed.

Using G.711 for VoIP will give the best voice quality; since it uses no compression and it is the same codec used by the PSTN network and ISDN lines, it sound just like using a regular or ISDN phone. It also has the lowest latency (lag) because there is no need for compression, which costs processing power. The downside is that it takes more bandwidth then other codecs, up to 84 Kbps including all TCP/IP overhead. However, with increasing broadband bandwith, this should not be a problem.

Vostron uses G.711 A-law for all VoIP applications where possible to ensure the highest voice quality and lowest latency.



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