G.711 is a narrowband speech codec standardized by the ITU-T in 1972. It's one of the earliest and most widely used codecs for VoIP (Voice over IP) and telephony due to its simplicity, low computational complexity, and decent voice quality. It's a waveform codec, meaning it attempts to reproduce the actual waveform of the speech signal.
Here's a breakdown of its key aspects:
Coding Method: G.711 uses pulse-code modulation (PCM) with either μ-law (mu-law) or A-law companding. Companding, short for compressing-expanding, is used to reduce the dynamic range of the audio signal, improving the signal-to-noise ratio, particularly for low-amplitude signals.
Variants: There are two primary variations of G.711:
Bitrate: G.711 operates at a fixed bitrate of 64 kbps. This is derived from sampling the audio signal at 8 kHz and using 8 bits per sample. Thus Bitrate becomes an important property.
Sampling Rate: It uses an 8 kHz Sampling%20Rate.
Payload Size: For VoIP applications, G.711 audio is typically packetized into payloads. Common payload sizes are 20ms (160 bytes) and 30ms (240 bytes).
Advantages:
Disadvantages:
Applications: G.711 remains widely used in:
Standardization: Defined in ITU-T Recommendation G.711. You can find the standard description under Standardization.
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