Set actual transport overhead in rtp_rtcp

webrtc/pc/channel.cc

Index: webrtc/pc/channel.cc
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index f22bdc4ca41029155f2fe4fe511bb9c2765ee679..477c2deb19e297a8a71ae9643b83de28bc0bcb3e 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -585,7 +585,8 @@ void BaseChannel::OnSelectedCandidatePairChanged(
     network_route = rtc::NetworkRoute(
         ready_to_send, selected_candidate_pair->local_candidate().network_id(),
         selected_candidate_pair->remote_candidate().network_id(),
-        last_sent_packet_id);
+        last_sent_packet_id,
+        GetTransportOverheadPerPacket(*selected_candidate_pair));
   }
   invoker_.AsyncInvoke<void>(
       RTC_FROM_HERE, worker_thread_,
@@ -1661,6 +1662,29 @@ void BaseChannel::UpdateMediaSendRecvState() {
       Bind(&BaseChannel::UpdateMediaSendRecvState_w, this));
 }
 
+int BaseChannel::GetTransportOverheadPerPacket(
+    const CandidatePairInterface& selected_candidate_pair) {
+  constexpr int kIpv4Overhaed = 20;
+  constexpr int kIpv6Overhaed = 40;
+  constexpr int kUdpOverhaed = 8;
+  constexpr int kTcpOverhaed = 20;
+  constexpr int kSrtpAuthOverhaed = 4;
+
+  int ransport_overhead_per_packet = 0;
+  ransport_overhead_per_packet +=
+      selected_candidate_pair.local_candidate().protocol() == TCP_PROTOCOL_NAME
+          ? kTcpOverhaed
+          : kUdpOverhaed;
+  ransport_overhead_per_packet +=
+      selected_candidate_pair.local_candidate().address().family() == AF_INET
+          ? kIpv4Overhaed
+          : kIpv6Overhaed;
+  if (secure()) {
+    ransport_overhead_per_packet += kSrtpAuthOverhaed;
+  }
+  return ransport_overhead_per_packet;
+}
+
 void VoiceChannel::UpdateMediaSendRecvState_w() {
   // Render incoming data if we're the active call, and we have the local
   // content. We receive data on the default channel and multiplexed streams.