internet Speech Audio Codec

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internet Speech Audio Codec
Internet media type audio/isac[1]
Developed by Global IP Solutions, now Google Inc
Type of format Audio compression format
Developer(s) Global IP Solutions, now Google Inc
Written in C
Operating system Cross-platform
Type Audio codec, reference implementation
License formerly 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 16 kHz[1] (or 32 kHz according to WebRTC)[6][7]
  • Adaptive and variable bit rate (10 kbit/s to 32 kbit/s) (or 10 kbit/s to 52 kbit/s according to WebRTC)[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 "RTP Payload Format for the iSAC Codec - Internet Draft - draft-legrand-rtp-isac-02.txt". 2009. Retrieved 2011-06-23. 
  2. ^ Dana Blankenhorn (2010-05-18). "Why Google bought Global IP Solutions". Retrieved 2011-06-23. 
  3. ^ "iLBC Freeware". Retrieved 2011-06-23. 
  4. ^
  5. ^
  6. ^ a b "WebRTC FAQ - What are the parameters of iSAC?". Retrieved 2011-06-23. 
  7. ^ a b "WebRTC components". Retrieved 2011-06-23. 

External links[edit]