srtp_transport.h - mozsearch

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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(const rtc::ReceivedPacket& packet) override;

  void OnRtcpPacketReceived(const rtc::ReceivedPacket& packet) 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_