Pick the DTLS handshake timeout based on the ICE RTT estimate

webrtc/p2p/base/port.cc

Index: webrtc/p2p/base/port.cc
diff --git a/webrtc/p2p/base/port.cc b/webrtc/p2p/base/port.cc
index 565ddb67a60e24fe28f3eb838c88d517a5641512..6fea346841f569df3bbc39517ce086aaceff2aba 100644
--- a/webrtc/p2p/base/port.cc
+++ b/webrtc/p2p/base/port.cc
@@ -1252,8 +1252,12 @@ void Connection::ReceivedPingResponse(int rtt, const std::string& request_id) {
   UpdateReceiving(last_ping_response_received_);
   set_write_state(STATE_WRITABLE);
   set_state(IceCandidatePairState::SUCCEEDED);
+  if (rtt_samples_ > 0) {
+    rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1);
+  } else {
+    rtt_ = rtt;

[Taylor Brandstetter, 2017/02/02 18:07:41]

May make more sense in a separate CL, but it would be great if this was renamed
to "smoothed_rtt_" and changed to an rtc::Optional (as opposed to using 3
seconds as an initial value).

[pthatcher2, 2017/02/02 20:09:44]

I agree that we should split it into "smoothed_rtt_ and "last_rtt_".  And I
think the DTLS code should used the last_rtt_, not the smoothed_rtt_.

[skvlad, 2017/02/02 22:19:33]

Let's make it rtc::Optional in a separate CL. There are too many places that
depend on there being some value (3000 or anything) that will break - such as
GetStats() or the connection sorting logic.

+  }
   rtt_samples_++;
-  rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1);
 }
 
 bool Connection::dead(int64_t now) const {