Ozeki VoIP SDK - Product Guide
Introducing G711 Ulaw
G.711 is an ITU-T standard algorithm for audio companding that is used for digital communication systems and supported by most of VoIP providers. G.711 encoder creates a 64 kbit/s bitstream for a signal sampled at 8 kHz. G.711 codec provides the best voice quality for VoIP.
Voice and audio signals are analogic, while data network is digital. The transformation of the
analogic signal to a digital one is made by Analog-to-Digital Converter (ADC).
This process of Analog-to-Digital Converter or Pulse Code Modulation (PCM) is done in three steps:
There are two types of quantization: uniform and not uniform. The not uniform quantization process follows a certain feature called encoding law. The two main encoding laws used nowadays are A law (a-law) and µ law (u-law), that are also known as G.711 codec. A Law (a-law) is used mainly in European PCM systems, and the µ law (u-law) is used in American PCM systems. The difference is in the method the analog signal being sampled. In both schemes, the signal is not sampled linearly, but in a logarithmic fashion.
G.711 µ-law tends to give more resolution to higher range signals while G.711 A-law provides more quantization levels at lower signal levels. When using µ-law G.711 in networks where suppression of the all 0 character signal is required, the character signal corresponding to negative input values between decision values numbers 127 and 128 should be 00000010 and the value at the decoder output is -7519. The corresponding decoder output value number is 125......
µ-law encoding takes a 14-bit signed linear audio sample as input, increases the magnitude by 32 (binary 100000), and converts it to an 8 bit value as follows:
|Linear input code||Compressed code|
Since G.711 was released in 1972 its patents have long since expired, so it is freely available.
G.711 is an ITU-T standard algorithm for audio companding that is used for digital communication systems and supported by most of VoIP providers. G.711 encoder creates a 64 kbit/s bitstream for a signal sampled at 8 kHz. G.711 codec provides the best voice quality for VoIP. Since it uses no compression it sounds like a regular or ISDN phone and it ensures a lowest latency. However it takes more bandwidth than other codecs. It defines two slightly different algorithms: A-Law and U-Law.
G.711 is a standard companding algorithm that is principally used in the digital telecommunication systems of North America and Japan. Companding algorithms reduce audio signal's dynamic range. In practice it means that in case of an analog system, the signal-to-noise ratio (SNR) can be increased while in case of digital systems the quantization error can be reduced.
This algorithm has two forms: an analog version, and a quantized digital version. It can be implemented in three ways: analog, non-linear ADC and digital.
Use an amplifier to achieve companding in the analog domain.
Use an Analog to Digital Converter with quantization levels that are unequally spaced to match the U-law algorithm.
Use the quantized digital version of the U-law algorithm in order to convert data in the digital domain.
- Encoded bandwidth: ~ 200-3400 Hz
- Standardized: ITU-T 1972
- Coding type: Companded PCM
- Bit rate: 64 kbps
- Delay (ms):
- Frame size: 0.125
- Lookahead: 0
- Quality: Toll
- MIPS: << 1
- RAM (words): 1
Features of G.711
- Sampling frequency 8 kHz
- 64 kbit/s bitrate (8 kHz sampling frequency x 8 bits per sample)
- Typical algorithmic delay is 0.125 ms, with no look-ahead delay
- G.711 is a waveform speech coder
- Digital telephony – PSTN, VoIP, wireless (G.711 is the mandatory minimum standard for ISDN terminal equipment)
- Videoconferencing (G.711 support is required for H.320/H.323 videoconferencing)
- Multimedia devices
Summary for G711 codec
|Algorithm||Sample Rate||Bit rate||Bits per sample||Latency||CBR||VBR||Stereo||Multi - |
|companding U-law, PCM, Lossy||8 kHz||64 kbit/s||13 bit||125ms||Yes||No||No||No|