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

• 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

• iLBC
• Opus (audio codec)
• SILK
• G.722
• Speex
• List of codecs

References

External links

• WebRTC codecs - GIT- source code for iSAC and other codecs

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