| /* |
| * 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_SRTP_SESSION_H_ |
| #define PC_SRTP_SESSION_H_ |
| |
| #include <vector> |
| |
| #include "api/scoped_refptr.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/synchronization/mutex.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 { |
| |
| // Prohibits webrtc from initializing libsrtp. This can be used if libsrtp is |
| // initialized by another library or explicitly. Note that this must be called |
| // before creating an SRTP session with WebRTC. |
| void ProhibitLibsrtpInitialization(); |
| |
| // 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; |
| |
| 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); |
| |
| // Writes unencrypted packets in text2pcap format to the log file |
| // for debugging. |
| void DumpPacket(const void* buf, int len, bool outbound); |
| |
| // 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 webrtc::GlobalMutex lock_; |
| int last_send_seq_num_ = -1; |
| bool external_auth_active_ = false; |
| bool external_auth_enabled_ = false; |
| int decryption_failure_count_ = 0; |
| bool dump_plain_rtp_ = false; |
| RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession); |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_SRTP_SESSION_H_ |