Wire up RTP keep-alive in ortc api.

webrtc/call/BUILD.gn

 # Copyright (c) 2015 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")
 
@@ skipping 58 matching lines @@
   ]
 }
 
 rtc_source_set("rtp_sender") {
   sources = [
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
   deps = [
     ":rtp_interfaces",
+    "..:webrtc_common",
     "../modules/congestion_controller",
     "../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "callfactory.cc",
@@ skipping 134 matching lines @@
     sources = [
       "test/mock_rtp_packet_sink_interface.h",
     ]
     deps = [
       ":rtp_interfaces",
       "../test:test_support",
       "//testing/gmock",
     ]
   }
 }