Re: my future telephones [telecom]
- From: Thad Floryan <thad@xxxxxxxxxxxx>
- Date: Tue, 25 Aug 2009 17:25:06 -0400 (EDT)
[...]
****** Moderator's Note *****
Please explain what "G.711u" means,
Per <http://www.voip-info.org/wiki/view/ITU+G.711>:
ITU G.711
G.711 is a high bit rate (64 Kbps) ITU standard codec. It is the
native language of 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 sounds 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.
G.711 is supported by most VoIP providers.
.
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