Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/BUILD.gn

Index: webrtc/pc/BUILD.gn
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 783fd67a554afaa0bda7028c7f30908fc91b900d..c7558048231f519b7e93a4a8a354deab21ccbfff 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -48,10 +48,13 @@ rtc_static_library("rtc_pc_base") {
     "rtcpmuxfilter.h",
     "rtptransport.cc",
     "rtptransport.h",
+    "rtptransportinternal.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "srtpsession.cc",
     "srtpsession.h",
+    "srtptransport.cc",
+    "srtptransport.h",
     "voicechannel.h",
   ]
 
@@ -258,9 +261,11 @@ if (rtc_include_tests) {
       "mediasession_unittest.cc",
       "rtcpmuxfilter_unittest.cc",
       "rtptransport_unittest.cc",
+      "rtptransporttestutil.h",
       "srtpfilter_unittest.cc",
       "srtpsession_unittest.cc",
       "srtptestutil.h",
+      "srtptransport_unittest.cc",
     ]
 
     include_dirs = [ "//third_party/libsrtp/srtp" ]