G.711.0 - Compressed G.711
Posted: Tue Oct 13, 2009 8:40 pm
Folks,
The ITU formally approved a new Recommendation for audio coding that is sure to grab some attention. It's a lossless compression algorithm for G.711, supporting both a-law and mu-law.
As most people know, using codecs like G.729 results in reduced audio quality. Compressing, decompressing, mixing in a conference bridge, re-compressing, and then decompressing results in an even more degraded audio performance. An these steps are not uncommon when calls pass through different carrier networks.
G.711.0 suffers absolutely no loss whatsoever and the compression is very good. It can be compressed and decompressed any number of times and no reduction of quality occurs, because the compression algorithm is a lossless algorithm. When there is silence, the codec consumes virtually no bandwidth at all. When there is speech activity, then the bandwidth it consumes varies, with as much as 50% reduction in bandwidth.
This is significant. G.711.0 has been the standard for telephone systems for years and virtually everything is designed for its use, including conference bridges, modems, fax devices, etc. While it is unlikely that any real compression would be seen if a modem is employed, it can significantly reduce the bandwidth required for most voice calls. Rather than consuming 80Kbps (G.711 with IP packet overhead using 20ms per packet), it will be possible to see bandwidth consumption below 64Kbps for most calls. The reason that is significant, of course, is that existing T1/E1 infrastructure can more easily be replaced with IP without sacrificing any audio quality whatsoever and without reducing the number of simultaneous calls that can be placed over that existing infrastructure.
The document has not yet been posted, but when it has undergone final edits, it will be posted here:
http://www.itu.int/rec/T-REC-G.711.0/en
Paul
The ITU formally approved a new Recommendation for audio coding that is sure to grab some attention. It's a lossless compression algorithm for G.711, supporting both a-law and mu-law.
As most people know, using codecs like G.729 results in reduced audio quality. Compressing, decompressing, mixing in a conference bridge, re-compressing, and then decompressing results in an even more degraded audio performance. An these steps are not uncommon when calls pass through different carrier networks.
G.711.0 suffers absolutely no loss whatsoever and the compression is very good. It can be compressed and decompressed any number of times and no reduction of quality occurs, because the compression algorithm is a lossless algorithm. When there is silence, the codec consumes virtually no bandwidth at all. When there is speech activity, then the bandwidth it consumes varies, with as much as 50% reduction in bandwidth.
This is significant. G.711.0 has been the standard for telephone systems for years and virtually everything is designed for its use, including conference bridges, modems, fax devices, etc. While it is unlikely that any real compression would be seen if a modem is employed, it can significantly reduce the bandwidth required for most voice calls. Rather than consuming 80Kbps (G.711 with IP packet overhead using 20ms per packet), it will be possible to see bandwidth consumption below 64Kbps for most calls. The reason that is significant, of course, is that existing T1/E1 infrastructure can more easily be replaced with IP without sacrificing any audio quality whatsoever and without reducing the number of simultaneous calls that can be placed over that existing infrastructure.
The document has not yet been posted, but when it has undergone final edits, it will be posted here:
http://www.itu.int/rec/T-REC-G.711.0/en
Paul