By Sagar Shankaran, Founder of CallSphere
Call in Studio runs on WebRTC. AI now screens callers, generates topic summaries, and drafts host prep notes in real time. Here is the 2026 build for AI-augmented radio.
Key takeaways
Talk radio in 2026 is browser-based. Call in Studio, BroadcastCaller, and a wave of new AI-augmented platforms run entirely on WebRTC — no SIP trunks, no PCs in the studio. The AI layer screens callers, summarizes the topic, drafts a host-prep card, and even flags policy risks before the caller goes on air.
Live talk radio still produces 1-in-5 minutes of US AM/FM listening, but every station shed its hardware call-in console between 2020 and 2025. The 2026 reference setup is a host browser, a producer browser, and N caller browsers — all peering through a WebRTC SFU. The producer adds AI: a real-time transcript of every caller, an LLM that classifies topic and tone, and a "would this caller drop an FCC indecency violation in the next 30 seconds?" risk model.
This is not theoretical. Call in Studio runs WebRTC for radio, podcasts, and public-meeting facilitation. KUDO and similar platforms add live translation. Smaller stations are gluing OpenAI Realtime + Pion to roll their own.
```mermaid flowchart LR Caller[Caller Browser] -- WebRTC --> SFU[Pion Go gateway 1.23] Producer[Producer Browser] -- WebRTC --> SFU Host[Host Browser] -- WebRTC --> SFU SFU -- audio --> AI[ASR + Topic + Risk LLM] AI -- screening card --> Producer Host -- "go on air" --> SFU SFU -- broadcast feed --> ICY[Icecast / Triton] ```
CallSphere's primary verticals are not broadcast, but the same WebRTC + AI pipeline powers two adjacent use cases:
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```typescript // 1. Caller joins as a constrained transceiver (mic-only, no video) const pc = new RTCPeerConnection({ iceServers }); const stream = await navigator.mediaDevices.getUserMedia({ audio: { echoCancellation: true, noiseSuppression: true }, }); stream.getTracks().forEach(t => pc.addTrack(t, stream));
// 2. Producer side: subscribe to caller's transcript stream const subs = new WebSocket("wss://callsphere.ai/producer/transcripts"); subs.onmessage = (e) => { const { callerId, text, topic, riskScore } = JSON.parse(e.data); renderScreeningCard({ callerId, text, topic, riskScore }); if (riskScore > 0.8) flagForHost(callerId); };
// 3. Risk + topic LLM call (server-side) import { openai } from "./openai"; async function classify(transcript: string) { const r = await openai.chat.completions.create({ model: "gpt-5", messages: [ { role: "system", content: "Classify topic + FCC risk for talk radio caller." }, { role: "user", content: transcript }, ], response_format: { type: "json_object" }, }); return JSON.parse(r.choices[0].message.content!); }
// 4. "Take to air" — promote caller's audio track to the broadcast mix async function takeToAir(callerId: string) { await sfu.subscribe(callerId, { destinations: ["broadcast", "host"] }); } ```
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Can I run this in a browser only? Yes — caller, producer, and host can all be browser-based; the broadcast feed exits via Icecast or Triton.
How do I integrate with my existing audio console? Pion + libsoundio or JACK gives you a virtual device that plugs into Pro Tools, Adobe Audition, or Wheatstone.
What about phone callers (PSTN)? Bridge SIP to WebRTC at the gateway; FreeSWITCH or Asterisk in front of Pion is the common pattern.
Does AI screening replace the producer? No — it augments. The producer still picks the order and tone.
Latency target? Under 200 ms host-to-caller for natural turn-taking; under 7 s caller-to-broadcast for FCC delay.
Written by
Sagar Shankaran· Founder, CallSphere
LinkedInSagar 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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