G.726 ADPCM Audio Codec
G.726 appears in older VoIP gateways and recorded WAV files. Often confused with G.711 — verify fmt chunk bit rate and RTP payload type mapping.
What It Defines
ITU-T Adaptive Differential PCM at 16, 24, 32, or 40 kbps — commonly 32 kbps (same bandwidth as G.722 sub-band packing). Used in legacy VoIP, international trunks, and WAV format tag 0x0064. RTP payload types 2/3 historically map to G.726 at 32/16 kbps in RFC 3551.
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
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.
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.