Add a continual_gathering_policy in PeerConnection RTCConfiguration

webrtc/p2p/base/transportcontroller.h

 /*
  *  Copyright 2015 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 27 matching lines @@
   rtc::Thread* signaling_thread() const { return signaling_thread_; }
   rtc::Thread* worker_thread() const { return worker_thread_; }
 
   PortAllocator* port_allocator() const { return port_allocator_; }
 
   // Can only be set before transports are created.
   // TODO(deadbeef): Make this an argument to the constructor once BaseSession
   // and WebRtcSession are combined
   bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
 
-  void SetIceConnectionReceivingTimeout(int timeout_ms);
+  void SetIceConfig(const IceConfig& config);
   void SetIceRole(IceRole ice_role);
 
   // TODO(deadbeef) - Return role of each transport, as role may differ from
   // one another.
   // In current implementaion we just return the role of the first transport
   // alphabetically.
   bool GetSslRole(rtc::SSLRole* role);
 
   // Specifies the identity to use in this session.
   // Can only be called once.
@@ skipping 60 matching lines @@
   Transport* GetTransport_w(const std::string& transport_name);
 
  private:
   void OnMessage(rtc::Message* pmsg) override;
 
   Transport* GetOrCreateTransport_w(const std::string& transport_name);
   void DestroyTransport_w(const std::string& transport_name);
   void DestroyAllTransports_w();
 
   bool SetSslMaxProtocolVersion_w(rtc::SSLProtocolVersion version);
-  void SetIceConnectionReceivingTimeout_w(int timeout_ms);
+  void SetIceConfig_w(const IceConfig& config);
   void SetIceRole_w(IceRole ice_role);
   bool GetSslRole_w(rtc::SSLRole* role);
   bool SetLocalCertificate_w(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
   bool GetLocalCertificate_w(
       const std::string& transport_name,
       rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
   bool GetRemoteSSLCertificate_w(const std::string& transport_name,
                                  rtc::SSLCertificate** cert);
   bool SetLocalTransportDescription_w(const std::string& transport_name,
@@ skipping 33 matching lines @@
 
   PortAllocator* const port_allocator_ = nullptr;
   rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_10;
 
   // Aggregate state for Transports
   IceConnectionState connection_state_ = kIceConnectionConnecting;
   bool receiving_ = false;
   IceGatheringState gathering_state_ = kIceGatheringNew;
 
   // TODO(deadbeef): Move the fields below down to the transports themselves
-
-  // Timeout value in milliseconds for which no ICE connection receives
-  // any packets
-  int ice_receiving_timeout_ms_ = -1;
+  IceConfig ice_config_;
   IceRole ice_role_ = ICEROLE_CONTROLLING;
   // Flag which will be set to true after the first role switch
   bool ice_role_switch_ = false;
   uint64 ice_tiebreaker_ = rtc::CreateRandomId64();
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_TRANSPORTCONTROLLER_H_