WebRTC is the new battleground for peer-to-peer vs. server-based models for communications

It would be wrong to classify Google as being purely objective here either. Despite high-profile moves like Google Voice, Gmail and Chat, I think that its dirty secret is that it doesn’t actually want to control or monetise communications per se. I suspect it sees a trillion-dollar market in telecoms services such as phone calls and SMS’s that could – eventually – be dissipated to near-zero and those sums diverted into alternate businesses in cloud infrastructure, advertising and other services.

I suspect Google believes (as do I) that a lot of communications will eventually move “into” applications and contexts. You’ll speak to a taxi driver from the taxi app, send messages inside social networks, or conclude business deals inside a collaboration service. You’ll do interviews “inside” LinkedIn, message/speak to possible partners inside a dating app etc. If your friend wants to meet you at the pub, you’ll send the message inside a mapping widget showing where it is… and so on.

I think Google wants to monetise communications context rather than communications sessions, through advertising or other enabling/exploiting capabilities.

Feature or Product (aka Service)? Perhaps like cameras they will remain both, albeit the Product version being a little more niche.

Opus Interactive Audio Codec

Overview

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. It is standardized by the Internet Engineering Task Force (IETF) as RFC 6716 which incorporated technology from Skype’s SILK codec and Xiph.Org’s CELT codec.

Technology

Opus can handle a wide range of audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bit-rate narrowband speech to very high quality stereo music. Supported features are:

  • Bit-rates from 6 kb/s to 510 kb/s
  • Sampling rates from 8 kHz (narrowband) to 48 kHz (fullband)
  • Frame sizes from 2.5 ms to 60 ms
  • Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
  • Audio bandwidth from narrowband to full-band
  • Support for speech and music
  • Support for mono and stereo
  • Support for up to 255 channels (multistream frames)
  • Dynamically adjustable bitrate, audio bandwidth, and frame size
  • Good loss robustness and packet loss concealment (PLC)
  • Floating point and fixed-point implementation

You can read the full specification, including the reference implementation, in RFC 6716. An up-to-date implementation of the Opus standard is also available from the downloads page.