Set actual transport overhead in rtp_rtcp

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index 6ea54194fbb8bf908cfd9d2e52f6ed76f0a7819e..f564347f0a6aa7907b003a82d835095f9f46ecf8 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -30,6 +30,7 @@
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
+#include "webrtc/p2p/base/candidatepair.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
@@ -361,6 +362,7 @@ class BaseChannel
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
   void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
+  int GetTransportOverheadPerPacket();
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
@@ -404,6 +406,7 @@ class BaseChannel
   std::vector<StreamParams> remote_streams_;
   MediaContentDirection local_content_direction_ = MD_INACTIVE;
   MediaContentDirection remote_content_direction_ = MD_INACTIVE;
+  rtc::Optional<CandidatePair> selected_candidate_pair_;
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,