internet Speech Audio Codec

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internet Speech Audio Codec
Internet media type
Developed byGlobal IP Solutions, now Google Inc
Type of formatAudio compression format
Developer(s)Global IP Solutions, now Google Inc
Written inC
Operating systemCross-platform
TypeAudio codec, reference implementation
Licenseformerly proprietary, now 3-clause BSD

internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011).[2][3] It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, e.g. RTP.

It is one of the codecs used by AIM Triton, the Gizmo5, QQ, and Google Talk. It was formerly a proprietary codec licensed by Global IP Solutions. As of June 2011, it is part of open source WebRTC project,[4] which includes a royalty-free license for iSAC when using the WebRTC codebase.[5]

Parameters and features[edit]

  • Sampling frequency of 16 kHz (wideband) or 32 kHz (superwideband)[1][6][7]
  • Adaptive and variable bit rate of 10 kbit/s to 32 kbit/s (wideband) or 10 kbit/s to 52 kbit/s (superwideband)[1][6][7]
  • Adaptive packet size 30 to 60 ms
  • Complexity comparable to G.722.2 at comparable bit-rates
  • Algorithmic delay of frame size plus 3 ms

See also[edit]


  1. ^ a b c Grand, Tina le; Jones, Paul; Huart, Pascal; Shabestary, Turaj Zakizadeh; Alvestrand, Harald T. (2013). "RTP Payload Format for the iSAC Codec". Retrieved 2016-04-30.
  2. ^ Dana Blankenhorn (2010-05-18). "Why Google bought Global IP Solutions". ZDNet. Retrieved 2011-06-23.
  3. ^ "iLBC Freeware". Archived from the original on 2011-07-05. Retrieved 2011-06-23.
  4. ^ Archived 2011-06-07 at the Wayback Machine.
  5. ^
  6. ^ a b "WebRTC FAQ - What are the parameters of iSAC?". Archived from the original on 2016-11-17. Retrieved 2011-06-23.
  7. ^ a b "WebRTC components". Archived from the original on 2011-06-28. Retrieved 2011-06-23.

External links[edit]