It's not simply about compression, Skype has a decade-long head start on tweaked echo/feedback suppression, gain control, dejitter, adaptive quality adjustment and so on. The call quality is marvelous, even with Macbook mic and lossy hotel wifi (I spend at least an hour every day using it right now).
I'm sure the browser vendors will catch up, but haven't even seen mention of some of these features yet with regard to WebRTC.
My experience is that Mumble blows Skype out of the water for background noise suppression. I can't have a call in a windy area without riding the mute button on Skype, but Mumble lets me adjust the signal/noise threshold to the point where it only picks up my voice.
The fact that it doesn't crash every other day is a nice bonus too.
No offense, and this isn't likely to be popular given the downvote above... but look harder. These are things that at they very least ARE being considered already for WebRTC.
I promise I'm not trying to be abrasive. I'll update with some links when I get a chance to get to a proper computer.
I'm sure they didn't, but Skype is just that good. Every single VOIP system my group used had issues. The microphone picking up audio from speakers and keyboard typing being two of the most problematic. Skype was the only one where that wasn't an issue. Their system just automagically filters out the cruft.
Meanwhile the voice chat built into gmail was one of the absolute worst and was completely unusable. It's a much more complicated problem than people realize.
Skype's codec (SILK) is superior to the ones listed for WebRTC.
Mumble's current codec (old version of CELT) is competitive and possibly better.
SILK was open sourced a while ago, and was combined with CELT to make the Opus voice codec, which is best-of-class and is even competitive with codecs like AAC for music compression. I suppose any client who wants to could adopt it, but none have yet.
Thanks for this info. In a another side project I'm working on video transcoding and (at this point) WebM. I'm still learning and it appears I need to familiarize myself with the lower level details of both video codecs and codecs for this sort of live calling, I'll have to check out Mumble/SILK. thanks.
