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

remoting/host/client_session_unittest.cc

 // 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.
 
 #include "remoting/host/client_session.h"
 
 #include <stdint.h>
 
 #include <algorithm>
 #include <string>
@@ skipping 187 matching lines @@
 
   // Stubs should be set only after connection is authenticated.
   EXPECT_FALSE(connection_->clipboard_stub());
   EXPECT_FALSE(connection_->input_stub());
 
   client_session_->OnConnectionAuthenticated(client_session_->connection());
 
   EXPECT_TRUE(connection_->clipboard_stub());
   EXPECT_TRUE(connection_->input_stub());
 
+  client_session_->OnCreateVideoStreams(client_session_->connection());
   client_session_->OnConnectionChannelsConnected(client_session_->connection());
 }
 
 void ClientSessionTest::NotifyVideoSize() {
   connection_->last_video_stream()->size_callback().Run(
       webrtc::DesktopSize(protocol::FakeDesktopCapturer::kWidth,
                           protocol::FakeDesktopCapturer::kHeight),
       webrtc::DesktopVector(kDefaultDpi, kDefaultDpi));
 }
 
@@ skipping 204 matching lines @@
 
   // ext2 was instantiated but not sent a message, and wrapped video encoder.
   EXPECT_TRUE(extension2.was_instantiated());
   EXPECT_FALSE(extension2.has_handled_message());
 
   // ext3 was sent a message but not instantiated.
   EXPECT_FALSE(extension3.was_instantiated());
 }
 
 }  // namespace remoting