By Sagar Shankaran, Founder of CallSphere
Krisp BVC went paid May 1 2026, RNNoise is unmaintained, and WebRTC's built-in NS is not enough for production voice AI. Here is the 2026 architecture for browser noise suppression.
Key takeaways
Two things changed the noise-suppression calculus in 2026: Krisp BVC moved to paid usage on May 1, and RNNoise stopped getting maintenance updates from Mozilla. Production WebRTC apps now have to make a real decision — pay for Krisp/Telnyx VIVA, ship RNNoise WASM yourself, or rely on WebRTC's built-in NS3 and accept the quality hit.
WebRTC's built-in noise suppressor (NS3, the Speex/WebRTC AEC3 chain) is fine for two-way human conversation in moderate environments. It is not fine for AI voice agents, where ASR and turn-detection are unforgiving — even 50 ms of background TV chatter can trigger a false interruption, derail a tool call, or push your end-pointing model into a hallucination. In 2026, every serious voice-AI vendor (LiveKit, Daily, Telnyx, Vonage, Twilio) ships a deep-learning suppressor on top of WebRTC's classic chain.
The market has bifurcated. Krisp BVC v3 is the gold standard but introduced metered pricing for SDK consumers in May 2026. RNNoise, originally a Mozilla project that married DSP to a small RNN, still works and compiles cleanly to WebAssembly — but it has not been maintained since 2024 and its model is showing its age on modern noise types (mechanical keyboard chatter, baby cries, fan noise from data centers). Newer entrants — Telnyx VIVA, Picovoice Koala, NVIDIA Maxine — split the difference with mid-tier pricing and on-device execution.
```mermaid
flowchart LR
Mic[Browser Mic] --> AWN[AudioWorkletNode]
AWN --> WASM[Noise Suppressor WASM
Krisp / RNNoise / Koala]
WASM --> AEC[WebRTC AEC3]
AEC --> RTP[RTP Encode Opus]
RTP --> Gateway[Pion Go gateway 1.23]
Gateway --> Agent[Voice Agent Pod]
```
CallSphere runs noise suppression in two places — server-side for inbound PSTN calls hitting the SIP gateway, and client-side for browser WebRTC sessions:
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```javascript // 1. Load RNNoise WASM into an AudioWorklet const ctx = new AudioContext({ sampleRate: 48000 }); await ctx.audioWorklet.addModule("/rnnoise-processor.js");
// 2. Build the graph: mic -> RNNoise -> destination -> RTCPeerConnection const stream = await navigator.mediaDevices.getUserMedia({ audio: { echoCancellation: true, // keep WebRTC AEC3 noiseSuppression: false, // disable WebRTC NS3 (we use RNNoise) autoGainControl: true, }, }); const source = ctx.createMediaStreamSource(stream); const rnnoise = new AudioWorkletNode(ctx, "rnnoise-processor"); const dest = ctx.createMediaStreamDestination(); source.connect(rnnoise).connect(dest);
// 3. Add the cleaned track to the peer connection const pc = new RTCPeerConnection({ iceServers }); dest.stream.getAudioTracks().forEach(t => pc.addTrack(t, dest.stream));
// 4. (Optional, paid) swap RNNoise for Krisp SDK at runtime async function upgradeToKrisp(apiKey) { const krisp = await import("@krisp-ai/sdk"); await krisp.init({ apiKey }); // replace the AudioWorkletNode in the graph } ```
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Written by
Sagar Shankaran· Founder, CallSphere
Sagar Shankaran is the founder of CallSphere, where he builds production AI voice and chat agents deployed across healthcare, hospitality, real estate, and home services. He writes about agentic AI, LLM engineering, and shipping voice agents that handle real calls in production.
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