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)
Convenient (Practical)
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.
WAV is the universal file format for call recordings, IVR prompts, voicemail exports, and PSTN debug captures. Every telephony platform (Twilio, Asterisk, FreeSWITCH) reads and writes WAV-wrapped G.711.
Codec mismatch debugging starts here — SDP says `PCMU/8000` which is PT 0 per RFC 3551. Static vs dynamic payload types explain most SIP/WebRTC negotiation confusion.
IVR navigation, call routing, and fax detection depend on RFC 4733 DTMF — not literal G.711 tone audio. Misconfigured `telephone-event` in SDP breaks keypad input on SIP calls.