Created the DtlsSrtpTransport.

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index b95bd529b005a71e5b749dc1333425467ece376d..10b0b8f30b3a096ccf7b1c8adc5522802061eb08 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -30,10 +30,13 @@
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
+#include "webrtc/pc/dtlssrtptransport.h"
 #include "webrtc/pc/mediamonitor.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/rtcpmuxfilter.h"
+#include "webrtc/pc/rtptransport.h"
 #include "webrtc/pc/srtpfilter.h"
+#include "webrtc/pc/srtptransport.h"
 #include "webrtc/rtc_base/asyncinvoker.h"
 #include "webrtc/rtc_base/asyncudpsocket.h"
 #include "webrtc/rtc_base/criticalsection.h"
@@ -44,7 +47,6 @@
 namespace webrtc {
 class AudioSinkInterface;
 class RtpTransportInternal;
-class SrtpTransport;
 }  // namespace webrtc
 
 namespace cricket {
@@ -103,7 +105,9 @@ class BaseChannel
   // This function returns true if we are using SDES.
   bool sdes_active() const { return sdes_negotiator_.IsActive(); }
   // The following function returns true if we are using DTLS-based keying.
-  bool dtls_active() const { return dtls_active_; }
+  bool dtls_active() const {
+    return dtls_srtp_transport_ && dtls_srtp_transport_->IsActive();
+  }
   // This function returns true if using SRTP (DTLS-based keying or SDES).
   bool srtp_active() const { return sdes_active() || dtls_active(); }
 
@@ -379,6 +383,12 @@ class BaseChannel
   void UpdateTransportOverhead();
   // Wraps the existing RtpTransport in an SrtpTransport.
   void EnableSrtpTransport_n();
+  // Create an SrtpTransport and wrap it in an DtlsSrptTransport.
+  void EnableDtlsSrtpTransport_n();
+  // Cache the send/recv encrypted header extension ids before the
+  // DtlsSrtpTransport is enabled.
+  void CacheEncryptedHeaderExtensionIds(cricket::ContentSource source,
+                                        const std::vector<int>& extension_ids);
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
@@ -393,13 +403,19 @@ class BaseChannel
 
   const bool rtcp_mux_required_;
 
+  std::vector<int> send_encrypted_header_extension_ids_;
+  std::vector<int> recv_encrypted_header_extension_ids_;
+
   // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
   // Temporary measure until more refactoring is done.
   // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
   DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
   DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
+
   std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
   webrtc::SrtpTransport* srtp_transport_ = nullptr;
+  webrtc::DtlsSrtpTransport* dtls_srtp_transport_ = nullptr;
+
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter sdes_negotiator_;
@@ -407,7 +423,6 @@ class BaseChannel
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
-  bool dtls_active_ = false;
   const bool srtp_required_ = true;
 
   // MediaChannel related members that should be accessed from the worker