Create the SrtpTransportInterface.

webrtc/BUILD.gn

 # Copyright (c) 2014 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 240 matching lines @@
       "api:transport_api",
       "audio",
       "base",
       "call",
       "common_audio",
       "common_video",
       "logging",
       "media",
       "modules",
       "modules/video_capture:video_capture_internal_impl",
+      "ortc",
       "p2p",
       "pc",
       "sdk",
       "stats",
       "system_wrappers",
       "video",
       "voice_engine",
     ]
 
     if (rtc_enable_protobuf) {
@@ skipping 239 matching lines @@
       ]
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/examples:AppRTCMobile_javalib",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
   }
 }