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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_SRTPTRANSPORT_H_
#define PC_SRTPTRANSPORT_H_
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "p2p/base/icetransportinternal.h"
#include "pc/rtptransportinternaladapter.h"
#include "pc/srtpfilter.h"
#include "pc/srtpsession.h"
#include "rtc_base/checks.h"
namespace webrtc {
// This class will eventually be a wrapper around RtpTransportInternal
// that protects and unprotects sent and received RTP packets.
class SrtpTransport : public RtpTransportInternalAdapter {
public:
SrtpTransport(bool rtcp_mux_enabled, const std::string& content_name);
SrtpTransport(std::unique_ptr<RtpTransportInternal> rtp_transport,
const std::string& content_name);
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 IsActive() const;
// TODO(zstein): Remove this when we remove RtpTransportAdapter.
RtpTransportAdapter* GetInternal() override { return nullptr; }
// 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);
// Helper method to get RTP Absoulute SendTime extension header id if
// present in remote supported extensions list.
void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
}
private:
void CreateSrtpSessions();
void ConnectToRtpTransport();
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags);
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
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);
const std::string content_name_;
std::unique_ptr<RtpTransportInternal> rtp_transport_;
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_;
bool external_auth_enabled_ = false;
int rtp_abs_sendtime_extn_id_ = -1;
};
} // namespace webrtc
#endif // PC_SRTPTRANSPORT_H_