Created the DtlsSrtpTransport.

pc/channel.h

Index: pc/channel.h
diff --git a/pc/channel.h b/pc/channel.h
index 613597b990e19990f4aa54490caad6fca5e4645e..b4b20417715981583e8305eea24a7dd79cebd029 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -30,10 +30,13 @@
 #include "p2p/base/transportcontroller.h"
 #include "p2p/client/socketmonitor.h"
 #include "pc/audiomonitor.h"
+#include "pc/dtlssrtptransport.h"
 #include "pc/mediamonitor.h"
 #include "pc/mediasession.h"
 #include "pc/rtcpmuxfilter.h"
+#include "pc/rtptransport.h"
 #include "pc/srtpfilter.h"
+#include "pc/srtptransport.h"
 #include "rtc_base/asyncinvoker.h"
 #include "rtc_base/asyncudpsocket.h"
 #include "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(); }
 
@@ -335,6 +339,13 @@ class BaseChannel
                  ContentSource src,
                  const std::vector<int>& encrypted_extension_ids,
                  std::string* error_desc);
+  bool SetSdesSrtp_n(const std::vector<CryptoParams>& params,
+                     ContentAction action,
+                     ContentSource src,
+                     const std::vector<int>& encrypted_extension_ids);
+  bool SetDtlsSrtp_n(ContentSource src,
+                     const std::vector<int>& encrypted_extension_ids);
+
   bool SetRtcpMux_n(bool enable,
                     ContentAction action,
                     ContentSource src,
@@ -369,7 +380,13 @@ class BaseChannel
   int GetTransportOverheadPerPacket() const;
   void UpdateTransportOverhead();
   // Wraps the existing RtpTransport in an SrtpTransport.
-  void EnableSrtpTransport_n();
+  void EnableSdes_n();
+  // Create an SrtpTransport and wrap it in an DtlsSrptTransport.

[Taylor Brandstetter, 2017/09/27 23:54:48]

DtlsSrtpTransport

+  void EnableDtlsSrtp_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_;
@@ -384,13 +401,18 @@ 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::SrtpTransport* sdes_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_;
@@ -398,7 +420,6 @@ 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