/* * 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 ORTC_RTPTRANSPORTADAPTER_H_ #define ORTC_RTPTRANSPORTADAPTER_H_ #include #include #include "api/ortc/srtptransportinterface.h" #include "api/rtcerror.h" #include "media/base/streamparams.h" #include "ortc/rtptransportcontrolleradapter.h" #include "pc/channel.h" #include "rtc_base/constructormagic.h" #include "rtc_base/sigslot.h" namespace webrtc { // Implementation of SrtpTransportInterface to be used with RtpSenderAdapter, // RtpReceiverAdapter, and RtpTransportControllerAdapter classes. This class // is used to implement both a secure and insecure RTP transport. // // TODO(deadbeef): When BaseChannel is split apart into separate // "RtpTransport"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this // adapter object can be removed. class RtpTransportAdapter : public SrtpTransportInterface { public: // |rtp| can't be null. |rtcp| can if RTCP muxing is used immediately (meaning // |rtcp_parameters.mux| is also true). static RTCErrorOr> CreateProxied( const RtpTransportParameters& rtcp_parameters, PacketTransportInterface* rtp, PacketTransportInterface* rtcp, RtpTransportControllerAdapter* rtp_transport_controller); static RTCErrorOr> CreateSrtpProxied( const RtpTransportParameters& rtcp_parameters, PacketTransportInterface* rtp, PacketTransportInterface* rtcp, RtpTransportControllerAdapter* rtp_transport_controller); ~RtpTransportAdapter() override; // RtpTransportInterface implementation. PacketTransportInterface* GetRtpPacketTransport() const override; PacketTransportInterface* GetRtcpPacketTransport() const override; RTCError SetParameters(const RtpTransportParameters& parameters) override; RtpTransportParameters GetParameters() const override { return parameters_; } // SRTP specific implementation. RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override; RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override; // Methods used internally by OrtcFactory. RtpTransportControllerAdapter* rtp_transport_controller() { return rtp_transport_controller_; } void TakeOwnershipOfRtpTransportController( std::unique_ptr controller); // Used by RtpTransportControllerAdapter to tell when it should stop // returning this transport from GetTransports(). sigslot::signal1 SignalDestroyed; // Used by the RtpTransportControllerAdapter to tell if an rtp sender or // receiver can be created. bool is_srtp_transport() { return is_srtp_transport_; } // Used by the RtpTransportControllerAdapter to set keys for senders and // receivers. rtc::Optional send_key() { return send_key_; } rtc::Optional receive_key() { return receive_key_; } protected: RtpTransportAdapter* GetInternal() override { return this; } private: RtpTransportAdapter(const RtcpParameters& rtcp_params, PacketTransportInterface* rtp, PacketTransportInterface* rtcp, RtpTransportControllerAdapter* rtp_transport_controller, bool is_srtp_transport); PacketTransportInterface* rtp_packet_transport_; PacketTransportInterface* rtcp_packet_transport_; RtpTransportControllerAdapter* const rtp_transport_controller_; // Non-null if this class owns the transport controller. std::unique_ptr owned_rtp_transport_controller_; RtpTransportParameters parameters_; // SRTP specific members. rtc::Optional send_key_; rtc::Optional receive_key_; bool is_srtp_transport_; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportAdapter); }; } // namespace webrtc #endif // ORTC_RTPTRANSPORTADAPTER_H_