Ozeki VoIP SDK - Product Guide
Did you know?This SDK was used to build:
Ozeki Phone System XE - VoIP PBX Software for Developers Which is a high performance PBX system supporting Mobile and Desktop phones.
It was also used to create Ozeki 3D VoIP softphone. A cool SIP client that allows 3D Video calls.
Introducing G711 aLaw
In communication it is vital to use the best possible solutions that provide the quality and excellence. For this reason, the solution of G711 codec, also known as Pulse Code Modulation (PCM) is a frequently used waveform codec. The G711 uses a sampling rate of 8000 samples/sec, with a tolerance of 50 parts per million (ppm). The G711 codec comes in two different compression algorithms: μ-law and A-law. A-law is the standard compression for international circuits.
The G711 A-law compression algorithm is used in Europe, and almost all over the world. The A-law is logarithmic, and lighter for the computer to process. The G711 A-law encodes a 13 bit signed linear PCM sample into logarithmic 8-bit sample. As a result, the G711 encoder will be able to produce a 64 kbit/s bitstream for a signal that is sampled at 8 KHz. The A-law compression enables more quantization levels at lower signal levels. A 13-bit signed linear audio sample as input is converted to an 8 bit value as follows:
|Linear input code||Compressed code|
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 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 μ-Law.
G.711 aLaw algorithm is used in all over the world for telephony to optimize, for example, modify the dynamic range of an analog signal for digitizing. This is a logarithmic algorithm and it has been designed to be simpler for computer processing than uLaw algorithm. It also provides a more dynamic range resulted in a better sound quality because sampling artifacts are better suppressed. It has a very low processor requirements and needs at least 128 kbps for two-way.
- 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 A-law, PCM, Lossy||8 kHz||64 kbit/s||13 bit||125ms||Yes||No||No||No|
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