G.711 PCM Audio Codec
G.711 is the common denominator of telephony. Every PSTN gateway and legacy SIP trunk speaks G.711. When codec negotiation fails and you're left with 64 kbps PCM, this is why.
What It Defines
G.711 defines the Pulse Code Modulation (PCM) standard for telephony: two companding variants — A-law (Europe/international) and μ-law (North America/Japan) — encoding 8000 samples/second at 8 bits per sample, yielding 64 kbps uncompressed audio. The universal codec for PSTN and VoIP fallback.
Canonical (Normative)
The international standards body for telecommunications under the UN umbrella. Publishes Recommendations in series (E, G, H, Q, T, X…): E.164 phone numbering, G.711/G.722 audio codecs, H.323 VoIP framework, and G.9959 (Z-Wave). Free access to most Recommendations.
Related Specs
RTP carries all real-time media: VoIP, WebRTC, live video, and video conferencing. Understanding SSRC demultiplexing, jitter buffers, and RTCP feedback loops is essential for any media application.
SIP is the PSTN of the internet. Every VoIP platform (Twilio, Vonage, AWS Chime, Asterisk, FreeSWITCH) speaks SIP. Building any communications product requires understanding INVITE flows, registration, and codec negotiation via SDP.
G.722 is the standard for HD voice in SIP and enterprise telephony. When you enable 'HD Voice' on a SIP trunk, G.722 is typically what's negotiated. Essential for enterprise UC deployments.