G.729 CS-ACELP Audio Codec
G.729 remains common on legacy SIP/PBX trunks where bandwidth is constrained. License/patent history affected adoption — know it when debugging codec negotiation on older enterprise systems.
What It Defines
ITU-T 8 kbps speech codec using Conjugate Structure Algebraic-Code Excited Linear Prediction (CS-ACELP). Dominant low-bitrate codec for SIP trunks and enterprise VoIP before Opus. Annexes add VAD (G.729B) and wide variants. Typically carried in RTP (payload type 18) rather than standalone files.
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.
Opus is the codec for real-time audio on the internet. WebRTC mandates it; Discord, Zoom, and most VoIP apps use it. Understanding bitrate, FEC, and frame sizing directly impacts audio quality and bandwidth.
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.
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.