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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/srtp_transport.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/srtp_transport.h')
-rw-r--r-- | third_party/libwebrtc/pc/srtp_transport.h | 176 |
1 files changed, 176 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/srtp_transport.h b/third_party/libwebrtc/pc/srtp_transport.h new file mode 100644 index 0000000000..ae62d5b780 --- /dev/null +++ b/third_party/libwebrtc/pc/srtp_transport.h @@ -0,0 +1,176 @@ +/* + * 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/crypto_params.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; + + virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params); + virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params); + + 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_cs, + const uint8_t* send_key, + int send_key_len, + const std::vector<int>& send_extension_ids, + int recv_cs, + 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_cs, + const uint8_t* send_key, + int send_key_len, + const std::vector<int>& send_extension_ids, + int recv_cs, + 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 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 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; + } + + 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<cricket::CryptoParams> send_params_; + absl::optional<cricket::CryptoParams> recv_params_; + absl::optional<int> send_cipher_suite_; + absl::optional<int> recv_cipher_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_ |