Allow applications to limit the ICE check rate through RTCConfiguration

webrtc/p2p/base/jseptransport.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 169 matching lines @@
 
   // The time period in which we will not switch the selected connection
   // when a new connection becomes receiving but the selected connection is not
   // in case that the selected connection may become receiving soon.
   rtc::Optional<int> receiving_switching_delay;
 
   // TODO(honghaiz): Change the default to regular nomination.
   // Default nomination mode if the remote does not support renomination.
   NominationMode default_nomination_mode = NominationMode::SEMI_AGGRESSIVE;
 
+  // ICE checks (STUN pings) will not be sent at higher rate (lower interval)
+  // than this, no matter what other settings there are.
+  // Measure in milliseconds.
+  rtc::Optional<int> ice_check_min_interval;
+
   IceConfig() {}
   IceConfig(int receiving_timeout_ms,
             int backup_connection_ping_interval,
             ContinualGatheringPolicy gathering_policy,
             bool prioritize_most_likely_candidate_pairs,
             int stable_writable_connection_ping_interval_ms,
             bool presume_writable_when_fully_relayed,
             int regather_on_failed_networks_interval_ms,
             int receiving_switching_delay_ms)
       : receiving_timeout(receiving_timeout_ms),
@@ skipping 158 matching lines @@
 
   // Candidate component => DTLS channel
   std::map<int, DtlsTransportInternal*> channels_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_JSEPTRANSPORT_H_