Pick the DTLS handshake timeout based on the ICE RTT estimate

webrtc/p2p/base/port.cc

 /*
  *  Copyright 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 1234 matching lines @@
   if (iter != pings_since_last_response_.end() &&
       iter->nomination > acked_nomination_) {
     acked_nomination_ = iter->nomination;
   }
 
   pings_since_last_response_.clear();
   last_ping_response_received_ = rtc::TimeMillis();
   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;
+  }
   rtt_samples_++;
-  rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1);
 }
 
 bool Connection::dead(int64_t now) const {
   if (last_received() > 0) {
     // If it has ever received anything, we keep it alive until it hasn't
     // received anything for DEAD_CONNECTION_RECEIVE_TIMEOUT. This covers the
     // normal case of a successfully used connection that stops working. This
     // also allows a remote peer to continue pinging over a locally inactive
     // (pruned) connection.
     return (now > (last_received() + DEAD_CONNECTION_RECEIVE_TIMEOUT));
@@ skipping 316 matching lines @@
     RTC_DCHECK(sent < 0);
     error_ = port_->GetError();
     stats_.sent_discarded_packets++;
   } else {
     send_rate_tracker_.AddSamples(sent);
   }
   return sent;
 }
 
 }  // namespace cricket