Written by 
Anyone that uses an in game voice chat program like Vent, Mumble, Team Speak or even if you are using Skype, you should know by now the codec is everything. Any spike in latency can be frustrating and annoying and a good audio codec can make a huge difference. Mumble prides itself in a codec developed specifically for speech and promises low bandwidth with high quality!
Skype has been knocking this new Opus codec around since March 2009 intending to create a low bandwidth codec “designed for the internet.” This should mean full CD quality audio in full band stereo over Skype, regardless of internet connection. They are also bringing in new technologies to help with packet loss so your friends and family wont sound like some dubstep audio sample. Personally this makes me VERY hopeful to see Skype on the BlackBerry PlayBook.
There is no specific ETA but they expect it to become standard across all Skype platforms. Below is a video that likely has way more information than anyone would be interested in…
This is very exciting. The very low latency of Opus could allow musicians to jam together over Skype … Does anyone know what the effective latency of a “jam” session over Skype might be with opus?
Skype currently uses SILK which has about 25ms of latency. Opus has a default codec latency of 22.5 ms, configurable down to 5 ms. The VoIP client Mumble uses this and the difference is noticeable. If another person is in the same room as you you don’t hear an echo at all. This all depends on the enviroment of course and different internet connections may result in different results.