|  | /* | 
|  | *  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 <stddef.h> | 
|  | #include <stdint.h> | 
|  |  | 
|  | #include <vector> | 
|  |  | 
|  | #include "api/field_trials_view.h" | 
|  | #include "api/sequence_checker.h" | 
|  | #include "rtc_base/buffer.h" | 
|  | #include "rtc_base/copy_on_write_buffer.h" | 
|  |  | 
|  | // Forward declaration to avoid pulling in libsrtp headers here | 
|  | struct srtp_event_data_t; | 
|  | struct srtp_ctx_t_;  // Trailing _ is required. | 
|  |  | 
|  | 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(); | 
|  | explicit SrtpSession(const webrtc::FieldTrialsView& field_trials); | 
|  | ~SrtpSession(); | 
|  |  | 
|  | SrtpSession(const SrtpSession&) = delete; | 
|  | SrtpSession& operator=(const SrtpSession&) = delete; | 
|  |  | 
|  | // Configures the session for sending data using the specified | 
|  | // crypto suite and key. Receiving must be done by a separate session. | 
|  | bool SetSend(int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  | bool UpdateSend(int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  |  | 
|  | // Configures the session for receiving data using the specified | 
|  | // crypto suite and key. Sending must be done by a separate session. | 
|  | bool SetReceive(int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  | bool UpdateReceive(int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | 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. | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtp(void* data, | 
|  | int in_len, | 
|  | int max_len, | 
|  | int* out_len); | 
|  | bool ProtectRtp(rtc::CopyOnWriteBuffer& buffer); | 
|  | // Overloaded version, outputs packet index. | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtp(void* data, | 
|  | int in_len, | 
|  | int max_len, | 
|  | int* out_len, | 
|  | int64_t* index); | 
|  | bool ProtectRtp(rtc::CopyOnWriteBuffer& buffer, int64_t* index); | 
|  |  | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] bool ProtectRtcp(void* data, | 
|  | int in_len, | 
|  | int max_len, | 
|  | int* out_len); | 
|  | bool ProtectRtcp(rtc::CopyOnWriteBuffer& buffer); | 
|  | // Decrypts/verifies an invidiual RTP/RTCP packet. | 
|  | // If an HMAC is used, this will decrease the packet size. | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] bool UnprotectRtp(void* data, | 
|  | int in_len, | 
|  | int* out_len); | 
|  | bool UnprotectRtp(rtc::CopyOnWriteBuffer& buffer); | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] bool UnprotectRtcp(void* data, | 
|  | int in_len, | 
|  | int* out_len); | 
|  | bool UnprotectRtcp(rtc::CopyOnWriteBuffer& buffer); | 
|  |  | 
|  | // 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; | 
|  |  | 
|  | // Removes a SSRC from the underlying libSRTP session. | 
|  | // Note: this should only be done for SSRCs that are received. | 
|  | // Removing SSRCs that were sent and then reusing them leads to | 
|  | // cryptographic weaknesses described in | 
|  | // https://www.rfc-editor.org/rfc/rfc3711#section-8 | 
|  | // https://www.rfc-editor.org/rfc/rfc7714#section-8.4 | 
|  | bool RemoveSsrcFromSession(uint32_t ssrc); | 
|  |  | 
|  | private: | 
|  | bool DoSetKey(int type, | 
|  | int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  | bool SetKey(int type, | 
|  | int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  | bool UpdateKey(int type, | 
|  | int crypto_suite, | 
|  | const rtc::ZeroOnFreeBuffer<uint8_t>& key, | 
|  | const std::vector<int>& extension_ids); | 
|  | // Returns send stream current packet index from srtp db. | 
|  | bool GetSendStreamPacketIndex(rtc::CopyOnWriteBuffer& buffer, int64_t* index); | 
|  |  | 
|  | // Writes unencrypted packets in text2pcap format to the log file | 
|  | // for debugging. | 
|  | void DumpPacket(const rtc::CopyOnWriteBuffer& buffer, bool outbound); | 
|  | [[deprecated("Pass CopyOnWriteBuffer")]] void DumpPacket(const void* buf, | 
|  | int len, | 
|  | bool outbound); | 
|  |  | 
|  | void HandleEvent(const srtp_event_data_t* ev); | 
|  | static void HandleEventThunk(srtp_event_data_t* ev); | 
|  |  | 
|  | webrtc::SequenceChecker thread_checker_; | 
|  | srtp_ctx_t_* session_ = nullptr; | 
|  |  | 
|  | // Overhead of the SRTP auth tag for RTP and RTCP in bytes. | 
|  | // Depends on the cipher suite used and is usually the same with the exception | 
|  | // of the kCsAesCm128HmacSha1_32 cipher suite. The additional four bytes | 
|  | // required for RTCP protection are not included. | 
|  | int rtp_auth_tag_len_ = 0; | 
|  | int rtcp_auth_tag_len_ = 0; | 
|  |  | 
|  | bool inited_ = false; | 
|  | 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; | 
|  | }; | 
|  |  | 
|  | }  // namespace cricket | 
|  |  | 
|  | #endif  // PC_SRTP_SESSION_H_ |