Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.

talk/app/webrtc/peerconnection.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 82 matching lines @@
   virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
                            const MediaConstraintsInterface* constraints);
   virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
                            const RTCOfferAnswerOptions& options);
   virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
                             const MediaConstraintsInterface* constraints);
   virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
                                    SessionDescriptionInterface* desc);
   virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                                     SessionDescriptionInterface* desc);
-  virtual void SetIceConnectionReceivingTimeout(int timeout_ms);
   // TODO(mallinath) : Deprecated version, remove after all clients are updated.
   virtual bool UpdateIce(const IceServers& configuration,
                          const MediaConstraintsInterface* constraints);
   virtual bool UpdateIce(
       const PeerConnectionInterface::RTCConfiguration& config);
   virtual bool AddIceCandidate(const IceCandidateInterface* candidate);
 
   virtual void RegisterUMAObserver(UMAObserver* observer);
 
   virtual void Close();
@@ skipping 73 matching lines @@
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
   rtc::scoped_ptr<WebRtcSession> session_;
   rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_;
   rtc::scoped_ptr<MediaStreamHandlerContainer> stream_handler_container_;
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_