Use rtppacketutil.h in socket_host*.cc

content/content_tests.gypi

 # Copyright (c) 2013 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 {
   'variables': {
     'layouttest_support_content_sources': [
       'public/test/layouttest_support.h',
       'public/test/nested_message_pump_android.cc',
       'public/test/nested_message_pump_android.h',
@@ skipping 746 matching lines @@
     ],
     # WebRTC-specific sources. Put WebRTC plugin-related stuff further below.
     'content_unittests_webrtc_sources': [
       'browser/media/webrtc_internals_unittest.cc',
       'browser/renderer_host/media/webrtc_identity_service_host_unittest.cc',
       'browser/renderer_host/p2p/socket_host_tcp_server_unittest.cc',
       'browser/renderer_host/p2p/socket_host_tcp_unittest.cc',
       'browser/renderer_host/p2p/socket_host_test_utils.cc',
       'browser/renderer_host/p2p/socket_host_test_utils.h',
       'browser/renderer_host/p2p/socket_host_udp_unittest.cc',
-      'browser/renderer_host/p2p/socket_host_unittest.cc',
       'renderer/media/audio_repetition_detector_unittest.cc',
       'renderer/media/audio_track_recorder_unittest.cc',
       'renderer/media/canvas_capture_handler_unittest.cc',
       'renderer/media/html_video_element_capturer_source_unittest.cc',
       'renderer/media/media_recorder_handler_unittest.cc',
       'renderer/media/media_stream_audio_processor_unittest.cc',
       'renderer/media/media_stream_constraints_util_unittest.cc',
       'renderer/media/media_stream_dispatcher_unittest.cc',
       'renderer/media/media_stream_video_capturer_source_unittest.cc',
       'renderer/media/media_stream_video_renderer_sink_unittest.cc',
@@ skipping 1435 matching lines @@
                   'content_unittests_apk.isolate',
                 ],
               },
             ],
           },
         ],
       ],
     }],
   ],
 }