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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/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_