Update the call when the network route changes

webrtc/media/engine/fakewebrtccall.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 221 matching lines @@
 
   DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const webrtc::PacketTime& packet_time) override;
 
   webrtc::Call::Stats GetStats() const override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  void OnNetworkRouteChanged(const std::string& transport_name,
+                             const rtc::NetworkRoute& network_route) override {}
   void SignalChannelNetworkState(webrtc::MediaType media,
                                  webrtc::NetworkState state) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState audio_network_state_;
   webrtc::NetworkState video_network_state_;
   rtc::SentPacket last_sent_packet_;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_