Completed the functionalities of SrtpTransport.

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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.
  */
 
@@ skipping 15 matching lines @@
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/p2p/base/dtlstransportinternal.h"
 #include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
 #include "webrtc/pc/mediamonitor.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/rtcpmuxfilter.h"
-#include "webrtc/pc/rtptransportinternal.h"
 #include "webrtc/pc/srtpfilter.h"
 #include "webrtc/rtc_base/asyncinvoker.h"
 #include "webrtc/rtc_base/asyncudpsocket.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/network.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/window.h"
 
 namespace webrtc {
 class AudioSinkInterface;
+class RtpTransportInternal;
+class SrtpTransport;
 }  // namespace webrtc
 
 namespace cricket {
 
 struct CryptoParams;
 class MediaContentDescription;
 
 // BaseChannel contains logic common to voice and video, including enable,
 // marshaling calls to a worker and network threads, and connection and media
 // monitors.
@@ skipping 35 matching lines @@
   // done.
   void Deinit();
 
   rtc::Thread* worker_thread() const { return worker_thread_; }
   rtc::Thread* network_thread() const { return network_thread_; }
   const std::string& content_name() const { return content_name_; }
   // TODO(deadbeef): This is redundant; remove this.
   const std::string& transport_name() const { return transport_name_; }
   bool enabled() const { return enabled_; }
 
   // This function returns true if we are using SRTP.

[Taylor Brandstetter, 2017/08/26 02:40:39]

nit: Should update this comment.

[Zhi Huang, 2017/08/29 18:40:34]

Done.

-  bool secure() const { return srtp_filter_.IsActive(); }
+  bool secure_sdes() const { return srtp_filter_.IsActive(); }

[pthatcher, 2017/08/28 21:42:56]

While we're changing the method name, can we make it something more
understandable, like "sdes_active" (and the other "dtls_active")?

[Zhi Huang, 2017/08/29 18:40:34]

Done.

   // The following function returns true if we are using
   // DTLS-based keying. If you turned off SRTP later, however
   // you could have secure() == false and dtls_secure() == true.
-  bool secure_dtls() const { return dtls_keyed_; }
+  bool secure_dtls() const { return dtls_active_; }
+
+  bool secure() const { return secure_sdes() || secure_dtls(); }

[pthatcher, 2017/08/28 21:42:56]

Would "srtp_active" be a good name here, to along with the others?

[Zhi Huang, 2017/08/29 18:40:34]

It sounds good to me.

 
   bool writable() const { return writable_; }
 
   // Set the transport(s), and update writability and "ready-to-send" state.
   // |rtp_transport| must be non-null.
   // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning
   // RTCP muxing is not fully active yet).
   // |rtp_transport| and |rtcp_transport| must share the same transport name as
   // well.
   // Can not start with "rtc::PacketTransportInternal" and switch to
@@ skipping 273 matching lines @@
   std::string transport_name_;
 
   const bool rtcp_mux_required_;
 
   // 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;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;

[pthatcher, 2017/08/28 21:42:56]

It feels like this really should be renamed to SdesX sdes_x_.  What should the x
be?  Hmmm... maybe SdesSession or SdesNegotiation.  But it's not a filter any
more, and it's SDES-specific, not just any SRTP.

[Zhi Huang, 2017/08/29 18:40:34]

What about sdes_negotiator_? I plan to rename the class and files in a separate
CL since it would affect internal projects.

   RtcpMuxFilter rtcp_mux_filter_;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
-  bool dtls_keyed_ = false;
+  bool dtls_active_ = false;
   const bool srtp_required_ = true;
-  int rtp_abs_sendtime_extn_id_ = -1;
 
   // MediaChannel related members that should be accessed from the worker
   // thread.
   MediaChannel* const media_channel_;
   // Currently the |enabled_| flag is accessed from the signaling thread as
   // well, but it can be changed only when signaling thread does a synchronous
   // call to the worker thread, so it should be safe.
   bool enabled_ = false;
   std::vector<StreamParams> local_streams_;
   std::vector<StreamParams> remote_streams_;
@@ skipping 307 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_