blob: c490c48f9d9164dfd974acbdc55cca60e2d6987b [file] [log] [blame]
/*
* 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 WEBRTC_PC_SRTPSESSION_H_
#define WEBRTC_PC_SRTPSESSION_H_
#include <vector>
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/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);
bool UpdateSend(int cs, const uint8_t* key, size_t len);
// 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 UpdateRecv(int cs, const uint8_t* key, size_t len);
void SetEncryptedHeaderExtensionIds(
const std::vector<int>& encrypted_header_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;
// Calls srtp_shutdown if it's initialized.
static void Terminate();
private:
bool DoSetKey(int type, int cs, const uint8_t* key, size_t len);
bool SetKey(int type, int cs, const uint8_t* key, size_t len);
bool UpdateKey(int type, int cs, const uint8_t* key, size_t len);
bool SetEncryptedHeaderExtensionIds(
int type,
const std::vector<int>& encrypted_header_extension_ids);
// 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;
int rtcp_auth_tag_len_ = 0;
static bool inited_;
static rtc::GlobalLockPod lock_;
int last_send_seq_num_ = -1;
bool external_auth_active_ = false;
bool external_auth_enabled_ = false;
std::vector<int> encrypted_header_extension_ids_;
RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
};
} // namespace cricket
#endif // WEBRTC_PC_SRTPSESSION_H_