Pick the DTLS handshake timeout based on the ICE RTT estimate

webrtc/p2p/base/p2ptransportchannel.h

 /*
  *  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 89 matching lines @@
 
   // From TransportChannel:
   int SendPacket(const char* data,
                  size_t len,
                  const rtc::PacketOptions& options,
                  int flags) override;
   int SetOption(rtc::Socket::Option opt, int value) override;
   bool GetOption(rtc::Socket::Option opt, int* value) override;
   int GetError() override { return error_; }
   bool GetStats(std::vector<ConnectionInfo>* stats) override;
+  rtc::Optional<int> GetRttEstimate() override;
 
   // TODO(honghaiz): Remove this method once the reference of it in
   // Chromoting is removed.
   const Connection* best_connection() const { return selected_connection_; }
 
   const Connection* selected_connection() const { return selected_connection_; }
   void set_incoming_only(bool value) { incoming_only_ = value; }
 
   // Note: These are only for testing purpose.
   // |ports_| and |pruned_ports| should not be changed from outside.
@@ skipping 247 matching lines @@
   bool writable_ = false;
 
   webrtc::MetricsObserverInterface* metrics_observer_ = nullptr;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_