Add support for GCM cipher suites from RFC 7714.

webrtc/p2p/base/transport.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 232 matching lines @@
   virtual bool VerifyCandidate(const Candidate& candidate,
                                std::string* error);
 
   virtual bool GetSslRole(rtc::SSLRole* ssl_role) const { return false; }
 
   // Must be called before channel is starting to connect.
   virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
     return false;
   }
 
+  // Must be called before channel is starting to connect.
+  virtual bool SetEnableGcmCiphers(bool enable) {
+    return false;
+  }
+
  protected:
   // These are called by Create/DestroyChannel above in order to create or
   // destroy the appropriate type of channel.
   virtual TransportChannelImpl* CreateTransportChannel(int component) = 0;
   virtual void DestroyTransportChannel(TransportChannelImpl* channel) = 0;
 
   // The current local transport description, for use by derived classes
   // when performing transport description negotiation.
   const TransportDescription* local_description() const {
     return local_description_.get();
@@ skipping 54 matching lines @@
 
   ChannelMap channels_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Transport);
 };
 
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_TRANSPORT_H_