Revert change which removes GICE

webrtc/p2p/base/session.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 222 matching lines @@
   rtc::Thread* worker_thread() const { return worker_thread_; }
   PortAllocator* port_allocator() const { return port_allocator_; }
 
   // The ID of this session.
   const std::string& id() const { return sid_; }
 
   // TODO(juberti): This data is largely redundant, as it can now be obtained
   // from local/remote_description(). Remove these functions and members.
   // Returns the XML namespace identifying the type of this session.
   const std::string& content_type() const { return content_type_; }
+  // Returns the XML namespace identifying the transport used for this session.
+  const std::string& transport_type() const { return transport_type_; }
 
   // Indicates whether we initiated this session.
   bool initiator() const { return initiator_; }
 
   // Returns the application-level description given by our client.
   // If we are the recipient, this will be NULL until we send an accept.
   const SessionDescription* local_description() const;
 
   // Returns the application-level description given by the other client.
   // If we are the initiator, this will be NULL until we receive an accept.
@@ skipping 190 matching lines @@
   // from |description|. Returns false if it's not available.
   static bool GetTransportDescription(const SessionDescription* description,
                                       const std::string& content_name,
                                       TransportDescription* info);
 
   rtc::Thread* const signaling_thread_;
   rtc::Thread* const worker_thread_;
   PortAllocator* const port_allocator_;
   const std::string sid_;
   const std::string content_type_;
+  const std::string transport_type_;
   bool initiator_;
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
   rtc::SSLProtocolVersion ssl_max_version_;
   rtc::scoped_ptr<const SessionDescription> local_description_;
   rtc::scoped_ptr<SessionDescription> remote_description_;
   uint64 ice_tiebreaker_;
   // This flag will be set to true after the first role switch. This flag
   // will enable us to stop any role switch during the call.
   bool role_switch_;
   TransportMap transports_;
 
   // Timeout value in milliseconds for which no ICE connection receives
   // any packets.
   int ice_receiving_timeout_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_SESSION_H_