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
G.711.0 - Compressed G.711
Re: G.711.0 - Compressed G.711
Hi,
is there any specification how to use it in the H.245 and SDP?
Tomas
is there any specification how to use it in the H.245 and SDP?
Tomas
- paulej
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Re: G.711.0 - Compressed G.711
Not yet. A payload specification must first be defined in the IETF. Once that is done, then we could use it in H.245 and SDP. This specification is being written right now.
Re: G.711.0 - Compressed G.711
Does this mean the new G.711.O CoDec will consume as little as 40Kbps? For that bit rate, wouldn't it be better to have a broadband CoDec, i.e. G.722?paulej wrote: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.
- paulej
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- Location: Research Triangle Park, NC, USA
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Re: G.711.0 - Compressed G.711
It can be even better than 50%. The 50% number comes from the typical bi-directional conversation. If folks are on a conference bridge where most of the participants are fairly quiet, then the compression is even better.
The advantage of G.711.0 is that it is possible to go between G.711.0 and G.711 without any loss whatsoever. Unlike a real codec, G.711.0 is a lossless compression scheme explicitly created for G.711. So, if an IP phone uses G.711 inside the enterprise, and SBC sitting at the edge could easily convert it to G.711.0, significantly reducing bandwidth over the WAN connection. On the other end, it could be converted back again, all without a single bit of information loss.
G.722 is an entirely different codec. Once you start talking a different codec, then the door is wide open to G.722, G.728, iLBC, Silk, etc. there are tons of codecs out there. It's just that G.711 is the lingua franca of VoIP codecs.
The advantage of G.711.0 is that it is possible to go between G.711.0 and G.711 without any loss whatsoever. Unlike a real codec, G.711.0 is a lossless compression scheme explicitly created for G.711. So, if an IP phone uses G.711 inside the enterprise, and SBC sitting at the edge could easily convert it to G.711.0, significantly reducing bandwidth over the WAN connection. On the other end, it could be converted back again, all without a single bit of information loss.
G.722 is an entirely different codec. Once you start talking a different codec, then the door is wide open to G.722, G.728, iLBC, Silk, etc. there are tons of codecs out there. It's just that G.711 is the lingua franca of VoIP codecs.
Re: G.711.0 - Compressed G.711
Thank you Paul. Now, I understand better the reason of having the G.711.O CoDec.