Internet Low Bitrate Codec (iLBC)
iLBC remains in legacy WebRTC/SIP endpoints and some embedded VoIP devices. Frame independence (no inter-frame dependency) makes it resilient to packet loss — useful design comparison for Opus FEC.
What It Defines
Open royalty-free speech codec at 13.33 or 15.2 kbps designed for IP networks with packet loss. Fixed 20/30 ms frames. Widely deployed in WebRTC predecessors and early VoIP systems. RTP payload format in RFC 3952.
Canonical (Normative)
Convenient (Practical)
Related References
The canonical publication point for finalized RFCs. If a protocol is standardized as an RFC, the RFC Editor text is the normative final reference. Published by the IETF, IRTF, IAB, and independent stream.
Related Specs
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.
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.
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.
Video calls (Meet, Zoom web), voice calls, peer-to-peer file transfer, and collaborative tools. The browser API surface for real-time A/V communication.