Allow applications to limit the ICE check rate through RTCConfiguration

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 42 matching lines @@
   if (pings_since_last_response.size() == 0)
     return false;
 
   auto first = pings_since_last_response[0];
   return now > (first.sent_time + maximum_time);
 }
 
 // We will restrict RTT estimates (when used for determining state) to be
 // within a reasonable range.
 const int MINIMUM_RTT = 100;   // 0.1 seconds
-const int MAXIMUM_RTT = 3000;  // 3 seconds
+const int MAXIMUM_RTT = 60000;  // 60 seconds

[Taylor Brandstetter, 2017/02/02 07:03:55]

See other comment

 
 // When we don't have any RTT data, we have to pick something reasonable.  We
 // use a large value just in case the connection is really slow.
-const int DEFAULT_RTT = MAXIMUM_RTT;
+const int DEFAULT_RTT = 3000;  // 3 seconds
 
 // Computes our estimate of the RTT given the current estimate.
 inline int ConservativeRTTEstimate(int rtt) {
   return std::max(MINIMUM_RTT, std::min(MAXIMUM_RTT, 2 * rtt));
 }
 
 // Weighting of the old rtt value to new data.
 const int RTT_RATIO = 3;  // 3 : 1
 
 // The delay before we begin checking if this port is useless.
@@ skipping 1505 matching lines @@
     RTC_DCHECK(sent < 0);
     error_ = port_->GetError();
     stats_.sent_discarded_packets++;
   } else {
     send_rate_tracker_.AddSamples(sent);
   }
   return sent;
 }
 
 }  // namespace cricket