Fix WebRTC transport to use single offer-answer exchange instead of two.

remoting/protocol/connection_to_client.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef REMOTING_PROTOCOL_CONNECTION_TO_CLIENT_H_
 #define REMOTING_PROTOCOL_CONNECTION_TO_CLIENT_H_
 
 #include <stdint.h>
 
 #include <string>
@@ skipping 23 matching lines @@
 class ConnectionToClient {
  public:
   class EventHandler {
    public:
     // Called when the network connection is authenticating
     virtual void OnConnectionAuthenticating(ConnectionToClient* connection) = 0;
 
     // Called when the network connection is authenticated.
     virtual void OnConnectionAuthenticated(ConnectionToClient* connection) = 0;
 
+    // Called to request creation of video streams. May be called before or
+    // after OnConnectionChannelsConnected().
+    virtual void CreateVideoStreams(ConnectionToClient* connection) = 0;
+
     // Called when the network connection is authenticated and all
     // channels are connected.
     virtual void OnConnectionChannelsConnected(
         ConnectionToClient* connection) = 0;
 
     // Called when the network connection is closed or failed.
     virtual void OnConnectionClosed(ConnectionToClient* connection,
                                     ErrorCode error) = 0;
 
     // Called when a new input event is received.
@@ skipping 44 matching lines @@
   // must be called in EventHandler::OnConnectionAuthenticated().
   virtual void set_clipboard_stub(ClipboardStub* clipboard_stub) = 0;
   virtual void set_host_stub(HostStub* host_stub) = 0;
   virtual void set_input_stub(InputStub* input_stub) = 0;
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_CONNECTION_TO_CLIENT_H_