Support encrypted RTP extensions (RFC 6904)

webrtc/pc/srtpfilter.h

 /*
  *  Copyright 2009 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 61 matching lines @@
                             ContentSource source);
   // Indicates which crypto algorithms and keys were contained in the answer.
   // answer_params should contain the negotiated parameters, which may be none,
   // if crypto was not desired or could not be negotiated (and not required).
   // This must be called after SetOffer. If crypto negotiation completes
   // successfully, this will advance the filter to the active state.
   bool SetAnswer(const std::vector<CryptoParams>& answer_params,
                  ContentSource source);
 
   // Just set up both sets of keys directly.
   // Used with DTLS-SRTP.

[Taylor Brandstetter, 2017/03/22 18:00:11]

nit: Update the comments in this file to describe the new parameters.

[joachim, 2017/03/23 00:04:33]

Functionality has moved to a new method, I added a comment there.

   bool SetRtpParams(int send_cs,
                     const uint8_t* send_key,
                     int send_key_len,
+                    const std::vector<int>& send_encrypted_headers,
                     int recv_cs,
                     const uint8_t* recv_key,
-                    int recv_key_len);
+                    int recv_key_len,
+                    const std::vector<int>& recv_encrypted_headers);
   bool SetRtcpParams(int send_cs,
                      const uint8_t* send_key,
                      int send_key_len,
                      int recv_cs,
                      const uint8_t* recv_key,
                      int recv_key_len);
 
   // Encrypts/signs an individual RTP/RTCP packet, in-place.
   // If an HMAC is used, this will increase the packet size.
   bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
@@ skipping 89 matching lines @@
 };
 
 // Class that wraps a libSRTP session.
 class SrtpSession {
  public:
   SrtpSession();
   ~SrtpSession();
 
   // Configures the session for sending data using the specified
   // cipher-suite and key. Receiving must be done by a separate session.
-  bool SetSend(int cs, const uint8_t* key, size_t len);
+  bool SetSend(int cs, const uint8_t* key, size_t len,
+      const std::vector<int>& encrypted_headers);
   // Configures the session for receiving data using the specified
   // cipher-suite and key. Sending must be done by a separate session.
-  bool SetRecv(int cs, const uint8_t* key, size_t len);
+  bool SetRecv(int cs, const uint8_t* key, size_t len,
+      const std::vector<int>& encrypted_headers);
 
   // Encrypts/signs an individual RTP/RTCP packet, in-place.
   // If an HMAC is used, this will increase the packet size.
   bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
   // Overloaded version, outputs packet index.
   bool ProtectRtp(void* data,
                   int in_len,
                   int max_len,
                   int* out_len,
                   int64_t* index);
@@ skipping 24 matching lines @@
   // Update the silent threshold (in ms) for signaling errors.
   void set_signal_silent_time(int signal_silent_time_in_ms);
 
   // Calls srtp_shutdown if it's initialized.
   static void Terminate();
 
   sigslot::repeater3<uint32_t, SrtpFilter::Mode, SrtpFilter::Error>
       SignalSrtpError;
 
  private:
-  bool SetKey(int type, int cs, const uint8_t* key, size_t len);
+  bool SetKey(int type, int cs, const uint8_t* key, size_t len,

[Taylor Brandstetter, 2017/03/22 18:00:11]

SetKeyAndEncryptedHeaders?

[joachim, 2017/03/23 00:04:34]

Done.

+      const std::vector<int>& encrypted_headers);
     // Returns send stream current packet index from srtp db.
   bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index);
 
   static bool Init();
   void HandleEvent(const srtp_event_data_t* ev);
   static void HandleEventThunk(srtp_event_data_t* ev);
 
   rtc::ThreadChecker thread_checker_;
   srtp_ctx_t_* session_ = nullptr;
   int rtp_auth_tag_len_ = 0;
@@ skipping 75 matching lines @@
   std::map<FailureKey, FailureStat> failures_;
   // Threshold in ms to silent the signaling errors.
   int signal_silent_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_SRTPFILTER_H_