Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.

talk/app/webrtc/peerconnectioninterface.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 215 matching lines @@
     kRtcpMuxPolicyRequire,
   };
 
   enum TcpCandidatePolicy {
     kTcpCandidatePolicyEnabled,
     kTcpCandidatePolicyDisabled
   };
 
   // TODO(hbos): Change into class with private data and public getters.
   struct RTCConfiguration {
+    static const int kUndefined = -1;
+    // Default maximum number of packets in the audio jitter buffer.
+    static const int kAudioJitterBufferMaxPackets = 50;
     // TODO(pthatcher): Rename this ice_transport_type, but update
     // Chromium at the same time.
     IceTransportsType type;
     // TODO(pthatcher): Rename this ice_servers, but update Chromium
     // at the same time.
     IceServers servers;
     // A localhost candidate is signaled whenever a candidate with the any
     // address is allocated.
     bool enable_localhost_ice_candidate;
     BundlePolicy bundle_policy;
     RtcpMuxPolicy rtcp_mux_policy;
     TcpCandidatePolicy tcp_candidate_policy;
     int audio_jitter_buffer_max_packets;
     bool audio_jitter_buffer_fast_accelerate;
+    int ice_connection_receiving_timeout;
     std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
 
     RTCConfiguration()
         : type(kAll),
           enable_localhost_ice_candidate(false),
           bundle_policy(kBundlePolicyBalanced),
           rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
           tcp_candidate_policy(kTcpCandidatePolicyEnabled),
-          audio_jitter_buffer_max_packets(50),
-          audio_jitter_buffer_fast_accelerate(false) {}
+          audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
+          audio_jitter_buffer_fast_accelerate(false),
+          ice_connection_receiving_timeout(kUndefined) {}
   };
 
   struct RTCOfferAnswerOptions {
     static const int kUndefined = -1;
     static const int kMaxOfferToReceiveMedia = 1;
 
     // The default value for constraint offerToReceiveX:true.
     static const int kOfferToReceiveMediaTrue = 1;
 
     int offer_to_receive_video;
@@ skipping 81 matching lines @@
   // Sets the local session description.
   // JsepInterface takes the ownership of |desc| even if it fails.
   // The |observer| callback will be called when done.
   virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
                                    SessionDescriptionInterface* desc) = 0;
   // Sets the remote session description.
   // JsepInterface takes the ownership of |desc| even if it fails.
   // The |observer| callback will be called when done.
   virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                                     SessionDescriptionInterface* desc) = 0;
-  // Sets the ICE connection receiving timeout value in milliseconds.
-  virtual void SetIceConnectionReceivingTimeout(int timeout_ms) {}
   // Restarts or updates the ICE Agent process of gathering local candidates
   // and pinging remote candidates.
   virtual bool UpdateIce(const IceServers& configuration,
                          const MediaConstraintsInterface* constraints) = 0;
   // Provides a remote candidate to the ICE Agent.
   // A copy of the |candidate| will be created and added to the remote
   // description. So the caller of this method still has the ownership of the
   // |candidate|.
   // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
   // take the ownership of the |candidate|.
@@ skipping 228 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_