Telegram-Android/TMessagesProj/jni/voip/webrtc/pc/srtp_session.h

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/*
* 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"
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#include "api/sequence_checker.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/synchronization/mutex.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);
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// Writes unencrypted packets in text2pcap format to the log file
// for debugging.
void DumpPacket(const void* buf, int len, bool outbound);
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// 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);
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webrtc::SequenceChecker thread_checker_;
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srtp_ctx_t_* session_ = nullptr;
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// 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
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// of the kCsAesCm128HmacSha1_32 cipher suite. The additional four bytes
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// required for RTCP protection are not included.
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int rtp_auth_tag_len_ = 0;
int rtcp_auth_tag_len_ = 0;
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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;
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bool dump_plain_rtp_ = false;
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RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
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};
} // namespace cricket
#endif // PC_SRTP_SESSION_H_