| /* | 
 |  *  Copyright 2017 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. | 
 |  */ | 
 |  | 
 | #ifndef PC_SRTPSESSION_H_ | 
 | #define PC_SRTPSESSION_H_ | 
 |  | 
 | #include <vector> | 
 |  | 
 | #include "api/umametrics.h" | 
 | #include "rtc_base/basictypes.h" | 
 | #include "rtc_base/scoped_ref_ptr.h" | 
 | #include "rtc_base/thread_checker.h" | 
 |  | 
 | // Forward declaration to avoid pulling in libsrtp headers here | 
 | struct srtp_event_data_t; | 
 | struct srtp_ctx_t_; | 
 |  | 
 | namespace cricket { | 
 |  | 
 | // 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, | 
 |                const std::vector<int>& extension_ids); | 
 |   bool UpdateSend(int cs, | 
 |                   const uint8_t* key, | 
 |                   size_t len, | 
 |                   const std::vector<int>& extension_ids); | 
 |  | 
 |   // 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, | 
 |                const std::vector<int>& extension_ids); | 
 |   bool UpdateRecv(int cs, | 
 |                   const uint8_t* key, | 
 |                   size_t len, | 
 |                   const std::vector<int>& extension_ids); | 
 |  | 
 |   // 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); | 
 |   bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len); | 
 |   // Decrypts/verifies an invidiual RTP/RTCP packet. | 
 |   // If an HMAC is used, this will decrease the packet size. | 
 |   bool UnprotectRtp(void* data, int in_len, int* out_len); | 
 |   bool UnprotectRtcp(void* data, int in_len, int* out_len); | 
 |  | 
 |   // Helper method to get authentication params. | 
 |   bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len); | 
 |  | 
 |   int GetSrtpOverhead() const; | 
 |  | 
 |   // If external auth is enabled, SRTP will write a dummy auth tag that then | 
 |   // later must get replaced before the packet is sent out. Only supported for | 
 |   // non-GCM cipher suites and can be checked through "IsExternalAuthActive" | 
 |   // if it is actually used. This method is only valid before the RTP params | 
 |   // have been set. | 
 |   void EnableExternalAuth(); | 
 |   bool IsExternalAuthEnabled() const; | 
 |  | 
 |   // A SRTP session supports external creation of the auth tag if a non-GCM | 
 |   // cipher is used. This method is only valid after the RTP params have | 
 |   // been set. | 
 |   bool IsExternalAuthActive() const; | 
 |  | 
 |   void SetMetricsObserver( | 
 |       rtc::scoped_refptr<webrtc::MetricsObserverInterface> metrics_observer); | 
 |  | 
 |  private: | 
 |   bool DoSetKey(int type, | 
 |                 int cs, | 
 |                 const uint8_t* key, | 
 |                 size_t len, | 
 |                 const std::vector<int>& extension_ids); | 
 |   bool SetKey(int type, | 
 |               int cs, | 
 |               const uint8_t* key, | 
 |               size_t len, | 
 |               const std::vector<int>& extension_ids); | 
 |   bool UpdateKey(int type, | 
 |                  int cs, | 
 |                  const uint8_t* key, | 
 |                  size_t len, | 
 |                  const std::vector<int>& extension_ids); | 
 |   // Returns send stream current packet index from srtp db. | 
 |   bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index); | 
 |  | 
 |   // These methods are responsible for initializing libsrtp (if the usage count | 
 |   // is incremented from 0 to 1) or deinitializing it (when decremented from 1 | 
 |   // to 0). | 
 |   // | 
 |   // Returns true if successful (will always be successful if already inited). | 
 |   static bool IncrementLibsrtpUsageCountAndMaybeInit(); | 
 |   static void DecrementLibsrtpUsageCountAndMaybeDeinit(); | 
 |  | 
 |   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; | 
 |   int rtcp_auth_tag_len_ = 0; | 
 |   bool inited_ = false; | 
 |   static rtc::GlobalLockPod lock_; | 
 |   int last_send_seq_num_ = -1; | 
 |   bool external_auth_active_ = false; | 
 |   bool external_auth_enabled_ = false; | 
 |   rtc::scoped_refptr<webrtc::MetricsObserverInterface> metrics_observer_; | 
 |   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession); | 
 | }; | 
 |  | 
 | }  // namespace cricket | 
 |  | 
 | #endif  // PC_SRTPSESSION_H_ |