diff --git a/ios/truapi-host/Sources/TrUAPIHost/Resources/truapi-container.js b/ios/truapi-host/Sources/TrUAPIHost/Resources/truapi-container.js
index 0a78c43a5..c9a54165e 100644
--- a/ios/truapi-host/Sources/TrUAPIHost/Resources/truapi-container.js
+++ b/ios/truapi-host/Sources/TrUAPIHost/Resources/truapi-container.js
@@ -94,5 +94,4 @@
     }
     return _createElement(tagName, options);
   });
-  freezeAndDelete(window, "RTCPeerConnection");
 })();
diff --git a/js/container/src/index.ts b/js/container/src/index.ts
index a914568a0..6147cf7ad 100644
--- a/js/container/src/index.ts
+++ b/js/container/src/index.ts
@@ -121,7 +121,4 @@ freezeValue(document, 'createElement', (tagName: string, options?: ElementCreati
   return _createElement(tagName, options);
 });
 
-// --- WebRTC: no permission path in TrUAPI mode ---
-freezeAndDelete(window, 'RTCPeerConnection');
-
 export {};
diff --git a/playground/src/lib/auto-test.ts b/playground/src/lib/auto-test.ts
index 1a1e90282..ffdfce8b6 100644
--- a/playground/src/lib/auto-test.ts
+++ b/playground/src/lib/auto-test.ts
@@ -1,6 +1,7 @@
 import { runExample, type LogEntry, type RunResult } from "./example-runner";
 import { getClientSync } from "@parity/truapi/sandbox";
 import type { MethodInfo, ServiceInfo } from "./services";
+import { WEBRTC_METHOD_NAME, WEBRTC_SERVICE_NAME } from "./webrtc-check";
 import type { DiagnosisStatus } from "@/shared/diagnosis";
 
 export const DIAGNOSIS_ID = "__diagnosis__";
@@ -47,6 +48,8 @@ const METHOD_TIMEOUT_MS = new Map<string, number>([
   ["Statement Store/create_proof_authorized", LIVE_ALLOCATION_TIMEOUT_MS],
   ["Statement Store/submit", LIVE_ALLOCATION_TIMEOUT_MS],
   ["Statement Store/subscribe", LIVE_ALLOCATION_TIMEOUT_MS],
+  // Two permission prompts (camera, microphone) may need the user.
+  [`${WEBRTC_SERVICE_NAME}/${WEBRTC_METHOD_NAME}`, SSO_TIMEOUT_MS],
 ]);
 
 type RunOneOpts = {
diff --git a/playground/src/lib/services.ts b/playground/src/lib/services.ts
index cfa3da55b..522ac7d8d 100644
--- a/playground/src/lib/services.ts
+++ b/playground/src/lib/services.ts
@@ -5,11 +5,14 @@ import type {
   ProductExecutionKind,
   ServiceInfo,
 } from "@parity/truapi/playground/services-types";
+import { WEBRTC_SERVICE } from "./webrtc-check";
 
 export type { MethodInfo, ProductExecutionKind, ServiceInfo };
 export { servicesForExecution };
 
-export const services: ServiceInfo[] = servicesForExecution(
-  generatedServices,
-  "Spa",
-);
+// Generated SPA-compatible services plus the synthetic WebRTC browser-capability
+// method, which is exercised the same way (an example) but is not a wire method.
+export const services: ServiceInfo[] = [
+  ...servicesForExecution(generatedServices, "Spa"),
+  WEBRTC_SERVICE,
+];
diff --git a/playground/src/lib/webrtc-check.ts b/playground/src/lib/webrtc-check.ts
new file mode 100644
index 000000000..b3cce222c
--- /dev/null
+++ b/playground/src/lib/webrtc-check.ts
@@ -0,0 +1,81 @@
+import type { ServiceInfo } from "@parity/truapi/playground/services-types";
+
+// A first-class playground method backed by an example that combines TrUAPI
+// permission calls with the browser WebRTC APIs. It runs through the same
+// example runner as every other method (`truapi` and `assert`/`console` are
+// injected; globals like `navigator`/`RTCPeerConnection` are reachable in the
+// runner's function scope), so it shows up in the method browser, ⌘K, and the
+// diagnosis alike.
+//
+// The example follows the product model — request every device permission via
+// TrUAPI first, then use the capability:
+//   1. requestDevicePermission("Camera") and ("Microphone") — device permissions;
+//   2. only once granted, getUserMedia + RTCPeerConnection.createOffer.
+
+export const WEBRTC_SERVICE_NAME = "WebRTC";
+export const WEBRTC_METHOD_NAME = "peer_connection";
+
+const WEBRTC_EXAMPLE_SOURCE = `// Fail fast: WebRTC media capture requires a secure context, so bail before
+// prompting for any permission if the origin isn't secure.
+console.log("secure context:", window.isSecureContext);
+assert(
+  window.isSecureContext,
+  "Not a secure context — WebRTC media capture requires HTTPS or a localhost/loopback origin.",
+);
+
+// Permission phase — request the camera and microphone through TrUAPI up front.
+const camera = await truapi.permissions.requestDevicePermission("Camera");
+assert(camera.isOk(), "camera permission request failed:", camera);
+assert(camera.value.granted, "camera permission denied");
+console.log("camera granted");
+
+const microphone = await truapi.permissions.requestDevicePermission("Microphone");
+assert(microphone.isOk(), "microphone permission request failed:", microphone);
+assert(microphone.value.granted, "microphone permission denied");
+console.log("microphone granted");
+
+// Capability phase — permissions granted, now access the capability.
+assert(
+  typeof RTCPeerConnection !== "undefined",
+  "RTCPeerConnection is unavailable — the host has not wired the WebRTC bridge (fail-closed).",
+);
+assert(
+  typeof navigator !== "undefined" && !!navigator.mediaDevices?.getUserMedia,
+  "navigator.mediaDevices.getUserMedia is unavailable in this host.",
+);
+
+const stream = await navigator.mediaDevices.getUserMedia({
+  video: true,
+  audio: true,
+});
+console.log(
+  "media captured:",
+  stream.getTracks().map((t) => t.kind).join(", "),
+);
+
+const pc = new RTCPeerConnection();
+try {
+  for (const track of stream.getTracks()) pc.addTrack(track, stream);
+  const offer = await pc.createOffer();
+  assert(!!offer.sdp, "createOffer returned an empty SDP");
+  console.log("offer created:", offer.type, offer.sdp.length + " bytes of SDP");
+} finally {
+  pc.close();
+  stream.getTracks().forEach((t) => t.stop());
+}`;
+
+/** Synthetic WebRTC method — browsable, runnable, and part of the diagnosis. */
+export const WEBRTC_SERVICE: ServiceInfo = {
+  name: WEBRTC_SERVICE_NAME,
+  methods: [
+    {
+      name: WEBRTC_METHOD_NAME,
+      type: "unary",
+      description:
+        "Requests camera + microphone (device permissions) through TrUAPI, " +
+        "then — once granted — opens an RTCPeerConnection and creates an " +
+        "offer with the captured media.",
+      exampleSource: WEBRTC_EXAMPLE_SOURCE,
+    },
+  ],
+};