G.711 is the 1972 ITU telephone codec and the lowest common denominator of digital voice. It is companded PCM — 8 kHz sampling, 8 bits per sample, 64 kbps — using either the mu-law (North America, Japan) or A-law (rest of world) curve to give better low-level resolution than linear 8-bit would. The audio is narrowband and unmistakably 'telephone', but it has almost no delay, trivial CPU cost, and is understood by literally every VoIP and PSTN endpoint, so it remains the universal fallback in SIP and WebRTC when nothing better is negotiated. When you hear a call drop to thin, plain telephone quality, you're often hearing G.711.
The book · Volume 4 of 7
Audio for Video: A Complete Guide to Sound in Video Products: From PCM and Codecs to Loudness, Accessibility, and Real-Time Voice Processing
G.711 is still the codec every SIP trunk and WebRTC gateway falls back to, so you will meet it whenever calls cross into telephony. This volume covers G.711 in depth with u-law and A-law companding, G.722, AMR, EVS and Opus, so you know what quality and bandwidth to expect at each bridge. Written by Nikolay Sapunov, CEO at Fora Soft.
Connecting your video app to phone lines and stuck with G.711 audio quality?
Describe your SIP or PSTN setup and we will design the G.711 interop path, transcoding points and the quality you can expect on calls. 250+ video products shipped since 2005.
