Created the DtlsSrtpTransport.

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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.
 
 import("../webrtc.gni")
 if (is_android) {
@@ skipping 20 matching lines @@
     "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
     "channel.cc",
     "channel.h",
     "channelmanager.cc",
     "channelmanager.h",
     "currentspeakermonitor.cc",
     "currentspeakermonitor.h",
+    "dtlssrtptransport.cc",
+    "dtlssrtptransport.h",
     "externalhmac.cc",
     "externalhmac.h",
     "mediamonitor.cc",
     "mediamonitor.h",
     "mediasession.cc",
     "mediasession.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "rtptransport.cc",
     "rtptransport.h",
@@ skipping 200 matching lines @@
   }
 
   rtc_test("rtc_pc_unittests") {
     testonly = true
 
     sources = [
       "bundlefilter_unittest.cc",
       "channel_unittest.cc",
       "channelmanager_unittest.cc",
       "currentspeakermonitor_unittest.cc",
+      "dtlssrtptransport_unittest.cc",
       "mediasession_unittest.cc",
       "rtcpmuxfilter_unittest.cc",
       "rtptransport_unittest.cc",
       "rtptransporttestutil.h",
       "srtpfilter_unittest.cc",
       "srtpsession_unittest.cc",
       "srtptestutil.h",
       "srtptransport_unittest.cc",
     ]
 
@@ skipping 186 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }