Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 9c253894719278a48b354763b4aef0a3235d0817..8d04f978d7d62fd19fc9f58f158cff9f2070cdf5 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -150,7 +150,8 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return false;
+  return channel_proxy_->ReceivedRTCPPacket(static_cast<const uint8_t*>(packet),
+                                            length) == 0;
 }
 
 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
@@ -177,7 +178,8 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
-  return true;
+
+  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {