Negotiate the same SRTP crypto suites for every DTLS association formed.

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 185 matching lines @@
 
   // Only public for unit tests.  Otherwise, consider protected.
   int SetOption(SocketType type, rtc::Socket::Option o, int val)
       override;
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
   virtual cricket::MediaType media_type() = 0;
 
-  bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
-
   // This function returns true if we require SRTP for call setup.
   bool srtp_required_for_testing() const { return srtp_required_; }
 
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
 
   void SetTransports_n(DtlsTransportInternal* rtp_dtls_transport,
                        DtlsTransportInternal* rtcp_dtls_transport,
                        rtc::PacketTransportInternal* rtp_packet_transport,
                        rtc::PacketTransportInternal* rtcp_packet_transport);
@@ skipping 81 matching lines @@
 
   bool AddRecvStream_w(const StreamParams& sp);
   bool RemoveRecvStream_w(uint32_t ssrc);
   bool AddSendStream_w(const StreamParams& sp);
   bool RemoveSendStream_w(uint32_t ssrc);
   bool ShouldSetupDtlsSrtp_n() const;
   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
   bool SetupDtlsSrtp_n(bool rtcp);
   void MaybeSetupDtlsSrtp_n();
-  // Set the DTLS-SRTP cipher policy on this channel as appropriate.
-  bool SetDtlsSrtpCryptoSuites_n(DtlsTransportInternal* transport, bool rtcp);
 
   // Should be called whenever the conditions for
   // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
   // Updates the send/recv state of the media channel.
   void UpdateMediaSendRecvState();
   virtual void UpdateMediaSendRecvState_w() = 0;
 
   // Gets the content info appropriate to the channel (audio or video).
   virtual const ContentInfo* GetFirstContent(
       const SessionDescription* sdesc) = 0;
@@ skipping 31 matching lines @@
                  ContentSource src,
                  std::string* error_desc);
   bool SetRtcpMux_n(bool enable,
                     ContentAction action,
                     ContentSource src,
                     std::string* error_desc);
 
   // From MessageHandler
   void OnMessage(rtc::Message* pmsg) override;
 
-  const rtc::CryptoOptions& crypto_options() const {
-    return crypto_options_;
-  }
-
   // Handled in derived classes
-  // Get the SRTP crypto suites to use for RTP media
-  virtual void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const = 0;
   virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) = 0;
 
   // Helper function for invoking bool-returning methods on the worker thread.
   template <class FunctorT>
   bool InvokeOnWorker(const rtc::Location& posted_from,
                       const FunctorT& functor) {
     return worker_thread_->Invoke<bool>(posted_from, functor);
   }
 
@@ skipping 33 matching lines @@
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
   bool dtls_keyed_ = false;
   const bool srtp_required_ = true;
-  rtc::CryptoOptions crypto_options_;
   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_;
@@ skipping 94 matching lines @@
                          ContentAction action,
                          std::string* error_desc) override;
   bool SetRemoteContent_w(const MediaContentDescription* content,
                           ContentAction action,
                           std::string* error_desc) override;
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
 
   void OnMessage(rtc::Message* pmsg) override;
-  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
   void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) override;
   void OnMediaMonitorUpdate(VoiceMediaChannel* media_channel,
                             const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
 
   static const int kEarlyMediaTimeout = 1000;
   MediaEngineInterface* media_engine_;
   bool received_media_;
@@ skipping 62 matching lines @@
                           ContentAction action,
                           std::string* error_desc) override;
   bool GetStats_w(VideoMediaInfo* stats);
   webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
   bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
   webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
   bool SetRtpReceiveParameters_w(uint32_t ssrc,
                                  webrtc::RtpParameters parameters);
 
   void OnMessage(rtc::Message* pmsg) override;
-  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
   void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) override;
   void OnMediaMonitorUpdate(VideoMediaChannel* media_channel,
                             const VideoMediaInfo& info);
 
   std::unique_ptr<VideoMediaMonitor> media_monitor_;
 
   // Last VideoSendParameters sent down to the media_channel() via
   // SetSendParameters.
@@ skipping 88 matching lines @@
                                        std::string* error_desc);
   bool SetLocalContent_w(const MediaContentDescription* content,
                          ContentAction action,
                          std::string* error_desc) override;
   bool SetRemoteContent_w(const MediaContentDescription* content,
                           ContentAction action,
                           std::string* error_desc) override;
   void UpdateMediaSendRecvState_w() override;
 
   void OnMessage(rtc::Message* pmsg) override;
-  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
   void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) override;
   void OnMediaMonitorUpdate(DataMediaChannel* media_channel,
                             const DataMediaInfo& info);
   void OnDataReceived(
       const ReceiveDataParams& params, const char* data, size_t len);
   void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
   void OnDataChannelReadyToSend(bool writable);
 
   std::unique_ptr<DataMediaMonitor> media_monitor_;
   bool ready_to_send_data_ = false;
 
   // Last DataSendParameters sent down to the media_channel() via
   // 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_