Create the SrtpTransportInterface.

webrtc/api/ortc/ortcfactoryinterface.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.
  */
 
 #ifndef WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_
 #define WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_
 
 #include <memory>
 #include <string>
 #include <utility>  // For std::move.
 
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediatypes.h"
 #include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
 #include "webrtc/api/ortc/ortcrtpsenderinterface.h"
 #include "webrtc/api/ortc/packettransportinterface.h"
 #include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
 #include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/api/ortc/srtptransportinterface.h"
 #include "webrtc/api/ortc/udptransportinterface.h"
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/p2p/base/packetsocketfactory.h"
 
 namespace webrtc {
 
@@ skipping 83 matching lines @@
   // If |transport_controller| is null, one will automatically be created, and
   // its lifetime managed by the returned RtpTransport. This should only be
   // done if a single RtpTransport is being used to communicate with the remote
   // endpoint.
   virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
       const RtcpParameters& rtcp_parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp,
       RtpTransportControllerInterface* transport_controller) = 0;
 
+  // Creates an SrtpTransport which is an RTP transport that uses SRTP.
+  virtual RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
+  CreateSrtpTransport(
+      const RtcpParameters& rtcp_parameters,
+      PacketTransportInterface* rtp,
+      PacketTransportInterface* rtcp,
+      RtpTransportControllerInterface* transport_controller) = 0;
+
   // Returns the capabilities of an RTP sender of type |kind|. These
   // capabilities can be used to determine what RtpParameters to use to create
   // an RtpSender.
   //
   // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
   virtual RtpCapabilities GetRtpSenderCapabilities(
       cricket::MediaType kind) const = 0;
 
   // Creates an RTP sender with |track|. Will not start sending until Send is
   // called. This is provided as a convenience; it's equivalent to calling
@@ skipping 82 matching lines @@
 
   // Creates an new local audio track wrapping |source|.
   virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
       const std::string& id,
       AudioSourceInterface* source) = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_