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175 lines
6.1 KiB
175 lines
6.1 KiB
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_SRTP_TRANSPORT_H_
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#define PC_SRTP_TRANSPORT_H_
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#include <stddef.h>
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#include <cstdint>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/crypto_params.h"
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#include "api/rtc_error.h"
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#include "p2p/base/packet_transport_internal.h"
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#include "pc/rtp_transport.h"
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#include "pc/srtp_session.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network_route.h"
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namespace webrtc {
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// This subclass of the RtpTransport is used for SRTP which is reponsible for
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// protecting/unprotecting the packets. It provides interfaces to set the crypto
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// parameters for the SrtpSession underneath.
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class SrtpTransport : public RtpTransport {
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public:
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explicit SrtpTransport(bool rtcp_mux_enabled);
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virtual ~SrtpTransport() = default;
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// SrtpTransportInterface specific implementation.
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virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params);
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virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params);
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bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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// The transport becomes active if the send_session_ and recv_session_ are
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// created.
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bool IsSrtpActive() const override;
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bool IsWritable(bool rtcp) const override;
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// Create new send/recv sessions and set the negotiated crypto keys for RTP
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// packet encryption. The keys can either come from SDES negotiation or DTLS
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// handshake.
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bool SetRtpParams(int send_cs,
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const uint8_t* send_key,
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int send_key_len,
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const std::vector<int>& send_extension_ids,
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int recv_cs,
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const uint8_t* recv_key,
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int recv_key_len,
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const std::vector<int>& recv_extension_ids);
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// Create new send/recv sessions and set the negotiated crypto keys for RTCP
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// packet encryption. The keys can either come from SDES negotiation or DTLS
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// handshake.
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bool SetRtcpParams(int send_cs,
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const uint8_t* send_key,
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int send_key_len,
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const std::vector<int>& send_extension_ids,
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int recv_cs,
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const uint8_t* recv_key,
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int recv_key_len,
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const std::vector<int>& recv_extension_ids);
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void ResetParams();
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// If external auth is enabled, SRTP will write a dummy auth tag that then
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// later must get replaced before the packet is sent out. Only supported for
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// non-GCM cipher suites and can be checked through "IsExternalAuthActive"
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// if it is actually used. This method is only valid before the RTP params
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// have been set.
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void EnableExternalAuth();
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bool IsExternalAuthEnabled() const;
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// A SrtpTransport supports external creation of the auth tag if a non-GCM
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// cipher is used. This method is only valid after the RTP params have
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// been set.
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bool IsExternalAuthActive() const;
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// Returns srtp overhead for rtp packets.
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bool GetSrtpOverhead(int* srtp_overhead) const;
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// Returns rtp auth params from srtp context.
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bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
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// Cache RTP Absoulute SendTime extension header ID. This is only used when
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// external authentication is enabled.
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void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
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rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
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}
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protected:
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// If the writable state changed, fire the SignalWritableState.
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void MaybeUpdateWritableState();
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private:
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void ConnectToRtpTransport();
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void CreateSrtpSessions();
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void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) override;
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) override;
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void OnNetworkRouteChanged(
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absl::optional<rtc::NetworkRoute> network_route) override;
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// Override the RtpTransport::OnWritableState.
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void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
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bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
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// Overloaded version, outputs packet index.
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bool ProtectRtp(void* data,
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int in_len,
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int max_len,
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int* out_len,
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int64_t* index);
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bool ProtectRtcp(void* data, int in_len, int max_len, int* out_len);
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// Decrypts/verifies an invidiual RTP/RTCP packet.
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// If an HMAC is used, this will decrease the packet size.
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bool UnprotectRtp(void* data, int in_len, int* out_len);
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bool UnprotectRtcp(void* data, int in_len, int* out_len);
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bool MaybeSetKeyParams();
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bool ParseKeyParams(const std::string& key_params, uint8_t* key, size_t len);
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const std::string content_name_;
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std::unique_ptr<cricket::SrtpSession> send_session_;
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std::unique_ptr<cricket::SrtpSession> recv_session_;
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std::unique_ptr<cricket::SrtpSession> send_rtcp_session_;
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std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_;
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absl::optional<cricket::CryptoParams> send_params_;
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absl::optional<cricket::CryptoParams> recv_params_;
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absl::optional<int> send_cipher_suite_;
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absl::optional<int> recv_cipher_suite_;
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rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
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rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
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bool writable_ = false;
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bool external_auth_enabled_ = false;
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int rtp_abs_sendtime_extn_id_ = -1;
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int decryption_failure_count_ = 0;
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};
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} // namespace webrtc
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#endif // PC_SRTP_TRANSPORT_H_
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