Repo created

TMessagesProj/jni/voip/webrtc/pc/srtp_transport.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_TRANSPORT_H_
+#define PC_SRTP_TRANSPORT_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_error.h"
+#include "p2p/base/packet_transport_internal.h"
+#include "pc/rtp_transport.h"
+#include "pc/srtp_session.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network_route.h"
+
+namespace webrtc {
+
+// This subclass of the RtpTransport is used for SRTP which is reponsible for
+// protecting/unprotecting the packets. It provides interfaces to set the crypto
+// parameters for the SrtpSession underneath.
+class SrtpTransport : public RtpTransport {
+ public:
+  SrtpTransport(bool rtcp_mux_enabled, const FieldTrialsView& field_trials);
+
+  virtual ~SrtpTransport() = default;
+
+  bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                     const rtc::PacketOptions& options,
+                     int flags) override;
+
+  bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                      const rtc::PacketOptions& options,
+                      int flags) override;
+
+  // The transport becomes active if the send_session_ and recv_session_ are
+  // created.
+  bool IsSrtpActive() const override;
+
+  bool IsWritable(bool rtcp) const override;
+
+  // Create new send/recv sessions and set the negotiated crypto keys for RTP
+  // packet encryption. The keys can either come from SDES negotiation or DTLS
+  // handshake.
+  bool SetRtpParams(int send_crypto_suite,
+                    const uint8_t* send_key,
+                    int send_key_len,
+                    const std::vector<int>& send_extension_ids,
+                    int recv_crypto_suite,
+                    const uint8_t* recv_key,
+                    int recv_key_len,
+                    const std::vector<int>& recv_extension_ids);
+
+  // Create new send/recv sessions and set the negotiated crypto keys for RTCP
+  // packet encryption. The keys can either come from SDES negotiation or DTLS
+  // handshake.
+  bool SetRtcpParams(int send_crypto_suite,
+                     const uint8_t* send_key,
+                     int send_key_len,
+                     const std::vector<int>& send_extension_ids,
+                     int recv_crypto_suite,
+                     const uint8_t* recv_key,
+                     int recv_key_len,
+                     const std::vector<int>& recv_extension_ids);
+
+  void ResetParams();
+
+  // 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 crypto 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 SrtpTransport 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;
+
+  // Returns srtp overhead for rtp packets.
+  bool GetSrtpOverhead(int* srtp_overhead) const;
+
+  // Returns rtp auth params from srtp context.
+  bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
+
+  // Cache RTP Absoulute SendTime extension header ID. This is only used when
+  // external authentication is enabled.
+  void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
+    rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
+  }
+
+  // In addition to unregistering the sink, the SRTP transport
+  // disassociates all SSRCs of the sink from libSRTP.
+  bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
+
+ protected:
+  // If the writable state changed, fire the SignalWritableState.
+  void MaybeUpdateWritableState();
+
+ private:
+  void ConnectToRtpTransport();
+  void CreateSrtpSessions();
+
+  void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
+                           int64_t packet_time_us) override;
+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
+                            int64_t packet_time_us) override;
+  void OnNetworkRouteChanged(
+      absl::optional<rtc::NetworkRoute> network_route) override;
+
+  // Override the RtpTransport::OnWritableState.
+  void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
+
+  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);
+
+  bool MaybeSetKeyParams();
+  bool ParseKeyParams(const std::string& key_params, uint8_t* key, size_t len);
+
+  const std::string content_name_;
+
+  std::unique_ptr<cricket::SrtpSession> send_session_;
+  std::unique_ptr<cricket::SrtpSession> recv_session_;
+  std::unique_ptr<cricket::SrtpSession> send_rtcp_session_;
+  std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_;
+
+  absl::optional<int> send_crypto_suite_;
+  absl::optional<int> recv_crypto_suite_;
+  rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
+  rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
+
+  bool writable_ = false;
+
+  bool external_auth_enabled_ = false;
+
+  int rtp_abs_sendtime_extn_id_ = -1;
+
+  int decryption_failure_count_ = 0;
+
+  const FieldTrialsView& field_trials_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_SRTP_TRANSPORT_H_