Completed the functionalities of SrtpTransport.

webrtc/pc/rtptransportinternal.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 36 matching lines @@
   sigslot::signal1<bool> SignalReadyToSend;
 
   // TODO(zstein): Consider having two signals - RtpPacketReceived and
   // RtcpPacketReceived.
   // The first argument is true for RTCP packets and false for RTP packets.
   sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
       SignalPacketReceived;
 
   virtual bool IsWritable(bool rtcp) const = 0;
 
+  virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                             const rtc::PacketOptions& options) = 0;
+
+  virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                              const rtc::PacketOptions& options) = 0;
+
   virtual bool SendPacket(bool rtcp,
                           rtc::CopyOnWriteBuffer* packet,
                           const rtc::PacketOptions& options,
                           int flags) = 0;
 
   virtual bool HandlesPayloadType(int payload_type) const = 0;
 
   virtual void AddHandledPayloadType(int payload_type) = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTPTRANSPORTINTERNAL_H_