Update the call when the network route changes

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 2317 matching lines @@
   // channels. So all RTCP packets must be forwarded all send channels. VoE
   // will filter out RR internally.
   for (const auto& ch : send_streams_) {
     engine()->voe()->network()->ReceivedRTCPPacket(
         ch.second->channel(), packet->cdata(), packet->size());
   }
 }
 
 void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
     const std::string& transport_name,
-    const NetworkRoute& network_route) {
-  // TODO(honghaiz): uncomment this once the function in call is implemented.
-  // call_->OnNetworkRouteChanged(transport_name, network_route);
+    const rtc::NetworkRoute& network_route) {
+  call_->OnNetworkRouteChanged(transport_name, network_route);
 }
 
 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   int channel = GetSendChannelId(ssrc);
   if (channel == -1) {
     LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
     return false;
   }
   if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
@@ skipping 213 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE