feat: Add TURN server support

e2e/helpers/seed-test-endpoint.ts

@@ -26,7 +26,7 @@ interface SeededEndpointStorageState {
 }
 
 function buildSeededEndpointStorageState(
-  endpointsOrPort: ReadonlyArray<SeededEndpointInput> | number = Number(process.env.TEST_SERVER_PORT) || 3099
+  endpointsOrPort: readonly SeededEndpointInput[] | number = Number(process.env.TEST_SERVER_PORT) || 3099
 ): SeededEndpointStorageState {
   const endpoints = Array.isArray(endpointsOrPort)
     ? endpointsOrPort.map((endpoint) => ({
@@ -81,7 +81,7 @@ export async function installTestServerEndpoint(
 
 export async function installTestServerEndpoints(
   context: BrowserContext,
-  endpoints: ReadonlyArray<SeededEndpointInput>
+  endpoints: readonly SeededEndpointInput[]
 ): Promise<void> {
   const storageState = buildSeededEndpointStorageState(endpoints);
 
@@ -111,7 +111,7 @@ export async function seedTestServerEndpoint(
 
 export async function seedTestServerEndpoints(
   page: Page,
-  endpoints: ReadonlyArray<SeededEndpointInput>
+  endpoints: readonly SeededEndpointInput[]
 ): Promise<void> {
   const storageState = buildSeededEndpointStorageState(endpoints);
 

e2e/helpers/webrtc-helpers.ts

@@ -129,6 +129,48 @@ export async function installWebRTCTracking(page: Page): Promise<void> {
 /**
  * Wait until at least one RTCPeerConnection reaches the 'connected' state.
  */
+
+/**
+ * Ensure every `AudioContext` created by the page auto-resumes so that
+ * the input-gain Web Audio pipeline (`source -> gain -> destination`) never
+ * stalls in the "suspended" state.
+ *
+ * On Linux with multiple headless Chromium instances, `new AudioContext()`
+ * can start suspended without a user-gesture gate, causing the media
+ * pipeline to emit only a single RTP packet.
+ *
+ * Call once per page, BEFORE navigating, alongside `installWebRTCTracking`.
+ */
+export async function installAutoResumeAudioContext(page: Page): Promise<void> {
+  await page.addInitScript(() => {
+    const OrigAudioContext = window.AudioContext;
+
+    (window as any).AudioContext = function(this: AudioContext, ...args: any[]) {
+      const ctx: AudioContext = new OrigAudioContext(...args);
+      // Track all created AudioContexts for test diagnostics
+      const tracked = ((window as any).__trackedAudioContexts ??= []) as AudioContext[];
+
+      tracked.push(ctx);
+
+      if (ctx.state === 'suspended') {
+        ctx.resume().catch(() => { /* noop */ });
+      }
+
+      // Also catch transitions to suspended after creation
+      ctx.addEventListener('statechange', () => {
+        if (ctx.state === 'suspended') {
+          ctx.resume().catch(() => { /* noop */ });
+        }
+      });
+
+      return ctx;
+    } as any;
+
+    (window as any).AudioContext.prototype = OrigAudioContext.prototype;
+    Object.setPrototypeOf((window as any).AudioContext, OrigAudioContext);
+  });
+}
+
 export async function waitForPeerConnected(page: Page, timeout = 30_000): Promise<void> {
   await page.waitForFunction(
     () => (window as any).__rtcConnections?.some(
@@ -172,7 +214,7 @@ export async function waitForConnectedPeerCount(page: Page, expectedCount: numbe
 /**
  * Resume all suspended AudioContext instances created by the synthetic
  * media patch.  Uses CDP `Runtime.evaluate` with `userGesture: true` so
- * Chrome treats the call as a user-gesture — this satisfies the autoplay
+ * Chrome treats the call as a user-gesture - this satisfies the autoplay
  * policy that otherwise blocks `AudioContext.resume()`.
  */
 export async function resumeSyntheticAudioContexts(page: Page): Promise<number> {