For audio at least, things are looking up. Opus, a new royalty-free codec backed by both Mozilla Foundation and Skype (Microsoft), has been standardized as an RFC[1], and will likely be mandatory in WebRTC[2]. So we should be seeing that across all major browsers in a year or two.
Opus beats almost all other codecs (MP3, AAC and HE-AAC, Vorbis) in subjective quality[3], so it's a good standard to have.
The IETF is forming a new working group for an open video codec[1]. It is still a BoF and will be chartered in 3-6 months, which is when the real work will begin. More details on the charter can be found in this email [2].
As for the audio, WebRTC chose G.711 and Opus as mandatory to implement (MTI)[3]. The reason for G.711 is so that WebRTC can interoperate with legacy devices.
Not solely. Opus is a hybrid of the SILK codec, which is more for speech, and CELT, which is more aimed at music. It can seamlessly switch between the two methods and use them simultaneously: https://wiki.xiph.org/OpusFAQ#Why_not_keep_the_SILK_and_CELT...
The main goal was streaming (of both music and speech), and hence, low latency. Matching or bettering high-latency codecs (like Vorbis) on quality was just a bonus, and I believe somewhat of a surprise to the developers when listening test results came out.
> Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but also intended for storage and streaming applications
As for the audio, WebRTC chose G.711 and Opus as mandatory to implement (MTI)[3]. The reason for G.711 is so that WebRTC can interoperate with legacy devices.
[1] http://www.ietf.org/mail-archive/web/video-codec/current/mai...
[2] http://www.ietf.org/mail-archive/web/video-codec/current/msg...
[3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267...