What does all this mean exactly? Well, in short, Opus is extremely flexible, and because of that, it can be used for low bit rate voice over IP and outperform existing codecs such as speex and g729.
Opus also supports a wide range of bitrates from 6-510kbps and variable frame rates from 2.5-20ms.
This is effectively the range of a PSTN phone call in G711 at 8khz to CD quality audio at 48khz. Opus is literally a hybrid codec that joins two separate codecs it spans the range of narrow band to wide band sample rates 8-48khz. In March 2011, the Opus implementation was beating AAC, and Vorbis in human listening tests. By July of 2010, they had created the first working prototype of a SILK+CELT hybrid codec. The working group resulted in collaboration between several organizations including, Broadcom, and Skype (Microsoft). After significant debate and push back from many organizations who hold patents relating to current codec technologies, the IETF created a working group in February of 2010. The first version of CELT became available in 2009, and shortly thereafter Skype joined the IETF and asked to create a working group to develop a standardized “Internet Wideband Audio Codec”. Opus originally comes from two independent efforts: The SILK codec that Skype started developing in 2007, and the CELT codec from which was also under development in 2007.
Beyond the new standard, Opus is interesting because of its technical claims, capability to provide high-quality real time audio encoding and decoding for a wide range of bit rates and sampling rates, and the fact that Opus is not only free, it's open sourced. Opus has seen a lot of press lately due to its receiving a newly IETF approved standard in RFC 6716. Opus is a relatively new audio codec that was created through a joint effort between several organizations based on two previously available codecs: SILK from Skype, and CELT from.