NetEQ is the adaptive jitter buffer algorithm implemented in libWebRTC. It is one of the most critical audio processing components in the WebRTC media engine.
What NetEQ does
NetEQ handles the challenging task of converting a jittery stream of audio packets arriving over the network into a smooth, continuous audio playback. It performs several functions:
- Adaptive buffering: Dynamically adjusts the jitter buffer size based on observed network jitter patterns, balancing between low latency and smooth playback
- Packet loss concealment: When packets are lost or arrive too late, NetEQ generates replacement audio using sophisticated algorithms that minimize audible artifacts
- Packet reordering: Handles out-of-order packet arrival gracefully
- Time-stretching and compression: Adjusts audio timing without changing pitch to smoothly adapt the buffer size
Why NetEQ matters
NetEQ is one of the key reasons why WebRTC audio quality is generally superior to many other VoIP implementations. It was developed by Global IP Solutions (GIPS), which Google acquired in 2011 and subsequently open-sourced as part of WebRTC. The algorithm has been refined over many years and represents significant intellectual property that would be very difficult to replicate.
