Completed the functionalities of SrtpTransport.

webrtc/p2p/base/fakepackettransport.h

 /*
  *  Copyright 2017 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 68 matching lines @@
       SendPacketInternal(packet);
     }
     SentPacket sent_packet(options.packet_id, TimeMillis());
     SignalSentPacket(this, sent_packet);
     return static_cast<int>(len);
   }
   int SetOption(Socket::Option opt, int value) override { return true; }
   bool GetOption(Socket::Option opt, int* value) override { return true; }
   int GetError() override { return 0; }
 
+  const CopyOnWriteBuffer* last_sent_packet() { return &last_sent_packet_; }
+
  private:
   void set_writable(bool writable) {
     if (writable_ == writable) {
       return;
     }
     writable_ = writable;
     if (writable_) {
       SignalReadyToSend(this);
     }
     SignalWritableState(this);
   }
 
   void set_receiving(bool receiving) {
     if (receiving_ == receiving) {
       return;
     }
     receiving_ = receiving;
     SignalReceivingState(this);
   }
 
   void SendPacketInternal(const CopyOnWriteBuffer& packet) {
+    last_sent_packet_ = packet;
     if (dest_) {
       dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
                               CreatePacketTime(0), 0);
     }
   }
 
+  CopyOnWriteBuffer last_sent_packet_;
   AsyncInvoker invoker_;
   std::string debug_name_;
   FakePacketTransport* dest_ = nullptr;
   bool async_ = false;
   int async_delay_ms_ = 0;
   bool writable_ = false;
   bool receiving_ = false;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_P2P_BASE_FAKEPACKETTRANSPORT_H_