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g711 ulaw codec - VoIP SIP SDK
https://voip-sip-sdk.com/p_7220-g711-ulaw-codec.html
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.
Audio Codec -G.711 Support - UCM6100 series IP PBX Appliance
http://forums5.grandstream.com/t/audio-codec-g-711-support/13485
“Supports a wide range of popular voice codes (including G.711 A-law/U-law, G.722, G.723.1, G.726, G.729A/B, iLBC (30ms only), GSM), video codec (including H.264, H.263, H.263+), and Fax (T.38).” However, G.711 isn’t listed in the available codecs in the SIP Trunks or the extensions. Where do I set the default codec to G.711?
Web audio codec guide - Web media technologies | MDN
https://developer.mozilla.org/en-US/docs/Web/Media/Formats/Audio_codecs
The G.711 specification, published by the International Telecommunications Union (ITU), was issued in 1972 to define standard audio encoding for telephone applications. It supports voice-grade audio covering frequencies from 300 to 3400 Hz.
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