diff --git a/.changeset/webrtc-single-peer-connection-option.md b/.changeset/webrtc-single-peer-connection-option.md
new file mode 100644
index 00000000..ff59e5be
--- /dev/null
+++ b/.changeset/webrtc-single-peer-connection-option.md
@@ -0,0 +1,5 @@
+---
+"@elevenlabs/client": minor
+---
+
+Add an optional `webRtc.singlePeerConnection` session option for WebRTC connections. Set it to `false` to force LiveKit's dual peer connection path, which restores the microphone request timing used before livekit-client began defaulting to the v1 join protocol. Leaving it unset keeps livekit-client's own default.
diff --git a/packages/client/src/utils/BaseConnection.ts b/packages/client/src/utils/BaseConnection.ts
index b9ecf9d4..63bf4b44 100644
--- a/packages/client/src/utils/BaseConnection.ts
+++ b/packages/client/src/utils/BaseConnection.ts
@@ -38,6 +38,14 @@ export type BaseSessionConfig = {
      * Defaults to "all".
      */
     iceTransportPolicy?: "all" | "relay";
+    /**
+     * Whether to negotiate over a single peer connection (LiveKit's v1 join
+     * protocol, which bundles the publisher offer in the JoinRequest). Set to
+     * false to force the dual peer connection path, e.g. on platforms that
+     * reject the microphone request at the point v1 issues it. Defaults to
+     * livekit-client's own default, currently true.
+     */
+    singlePeerConnection?: boolean;
   };
   overrides?: {
     agent?: {
diff --git a/packages/client/src/utils/WebRTCConnection.test.ts b/packages/client/src/utils/WebRTCConnection.test.ts
index 6d7005c1..96e18157 100644
--- a/packages/client/src/utils/WebRTCConnection.test.ts
+++ b/packages/client/src/utils/WebRTCConnection.test.ts
@@ -165,6 +165,97 @@ describe("WebRTCConnection", () => {
     connection.close();
   });
 
+  describe("webRtc.singlePeerConnection", () => {
+    // Options the Room constructor was last called with, if any.
+    function lastRoomOptions(): { singlePeerConnection?: boolean } | undefined {
+      return (Room as unknown as ReturnType<typeof vi.fn>).mock.calls.at(
+        -1
+      )?.[0];
+    }
+
+    // Returns a room mock whose connect sequence resolves, so create() runs to
+    // completion and we can inspect how the Room itself was constructed.
+    function mockConnectingRoom() {
+      const mockRoom = new Room() as any;
+      (mockRoom.on as ReturnType<typeof vi.fn>).mockImplementation(
+        (event: string, callback: () => void) => {
+          if (event === "connected") {
+            queueMicrotask(callback);
+          }
+        }
+      );
+      (mockRoom.once as ReturnType<typeof vi.fn>).mockImplementation(
+        (event: string, callback: () => void) => {
+          if (event === "signalConnected") {
+            queueMicrotask(callback);
+          }
+        }
+      );
+      return mockRoom;
+    }
+
+    it("forces the dual peer connection path when set to false", async () => {
+      mockConnectingRoom();
+
+      const connection = await WebRTCConnection.create({
+        conversationToken: "test-token",
+        connectionType: "webrtc",
+        webRtc: { singlePeerConnection: false },
+      });
+
+      expect(Room).toHaveBeenLastCalledWith({ singlePeerConnection: false });
+
+      connection.close();
+    });
+
+    it("passes the option through when set to true", async () => {
+      mockConnectingRoom();
+
+      const connection = await WebRTCConnection.create({
+        conversationToken: "test-token",
+        connectionType: "webrtc",
+        webRtc: { singlePeerConnection: true },
+      });
+
+      expect(Room).toHaveBeenLastCalledWith({ singlePeerConnection: true });
+
+      connection.close();
+    });
+
+    // The back-compat control: an unset option must leave livekit-client's own
+    // default in place rather than pinning it from here. Asserts that no mode
+    // was pinned rather than the exact call arity, since `new Room()` and
+    // `new Room(undefined)` are the same thing to LiveKit.
+    it("leaves the LiveKit default alone when the option is omitted", async () => {
+      mockConnectingRoom();
+
+      const connection = await WebRTCConnection.create({
+        conversationToken: "test-token",
+        connectionType: "webrtc",
+      });
+
+      expect(lastRoomOptions()?.singlePeerConnection).toBeUndefined();
+
+      connection.close();
+    });
+
+    // The two webRtc options are independent: setting the sibling must not
+    // start pinning the peer connection mode.
+    it("does not pin the peer connection mode when only iceTransportPolicy is set", async () => {
+      mockConnectingRoom();
+
+      const connection = await WebRTCConnection.create({
+        conversationToken: "test-token",
+        connectionType: "webrtc",
+        webRtc: { iceTransportPolicy: "relay" },
+      });
+
+      expect(lastRoomOptions()?.singlePeerConnection).toBeUndefined();
+
+      connection.close();
+    });
+  });
+
   it("reconnects input analyser after unmuting", async () => {
     const mockRoom = new Room() as any;
 
diff --git a/packages/client/src/utils/WebRTCConnection.ts b/packages/client/src/utils/WebRTCConnection.ts
index 82165db2..a74a6d87 100644
--- a/packages/client/src/utils/WebRTCConnection.ts
+++ b/packages/client/src/utils/WebRTCConnection.ts
@@ -289,7 +289,15 @@ export class WebRTCConnection extends BaseConnection {
       );
     }
 
-    const room = new Room();
+    // livekit-client defaults `singlePeerConnection` to true, which bundles the
+    // publisher offer in the JoinRequest and moves the microphone request
+    // earlier in the connection sequence. Callers that need the dual peer
+    // connection path can opt back into it. Left undefined so the LiveKit
+    // default applies unless the caller says otherwise.
+    const singlePeerConnection = config.webRtc?.singlePeerConnection;
+    const room = new Room(
+      singlePeerConnection === undefined ? undefined : { singlePeerConnection }
+    );
 
     try {
       // Create connection instance first to set up event listeners