Add PeerConnection IsClosed check.

webrtc/api/webrtcsession.h

 /*
  *  Copyright 2012 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 303 matching lines @@
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
 
+  // Set the transport controller and subscribe the Signals from the
+  // TransportController.
+  void SetTransportController(
+      cricket::TransportController* transport_controller);
+
   // Called when voice_channel_, video_channel_ and data_channel_ are created
   // and destroyed. As a result of, for example, setting a new description.
   sigslot::signal0<> SignalVoiceChannelCreated;
   sigslot::signal0<> SignalVoiceChannelDestroyed;
   sigslot::signal0<> SignalVideoChannelCreated;
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
   // Called when the whole session is destroyed.
   sigslot::signal0<> SignalDestroyed;
@@ skipping 192 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_