/* * 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_RTP_TRANSPORT_INTERNAL_H_ #define PC_RTP_TRANSPORT_INTERNAL_H_ #include #include "call/rtp_demuxer.h" #include "p2p/base/ice_transport_internal.h" #include "pc/session_description.h" #include "rtc_base/network_route.h" #include "rtc_base/ssl_stream_adapter.h" #include "rtc_base/third_party/sigslot/sigslot.h" namespace rtc { class CopyOnWriteBuffer; struct PacketOptions; } // namespace rtc namespace webrtc { // This represents the internal interface beneath SrtpTransportInterface; // it is not accessible to API consumers but is accessible to internal classes // in order to send and receive RTP and RTCP packets belonging to a single RTP // session. Additional convenience and configuration methods are also provided. class RtpTransportInternal : public sigslot::has_slots<> { public: virtual ~RtpTransportInternal() = default; virtual void SetRtcpMuxEnabled(bool enable) = 0; virtual const std::string& transport_name() const = 0; // Sets socket options on the underlying RTP or RTCP transports. virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0; virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0; virtual bool rtcp_mux_enabled() const = 0; virtual bool IsReadyToSend() const = 0; // Called whenever a transport's ready-to-send state changes. The argument // is true if all used transports are ready to send. This is more specific // than just "writable"; it means the last send didn't return ENOTCONN. sigslot::signal1 SignalReadyToSend; // Called whenever an RTCP packet is received. There is no equivalent signal // for RTP packets because they would be forwarded to the BaseChannel through // the RtpDemuxer callback. sigslot::signal2 SignalRtcpPacketReceived; // Called whenever the network route of the P2P layer transport changes. // The argument is an optional network route. sigslot::signal1> SignalNetworkRouteChanged; // Called whenever a transport's writable state might change. The argument is // true if the transport is writable, otherwise it is false. sigslot::signal1 SignalWritableState; sigslot::signal1 SignalSentPacket; virtual bool IsWritable(bool rtcp) const = 0; // TODO(zhihuang): Pass the |packet| by copy so that the original data // wouldn't be modified. virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options, int flags) = 0; virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options, int flags) = 0; // This method updates the RTP header extension map so that the RTP transport // can parse the received packets and identify the MID. This is called by the // BaseChannel when setting the content description. // // TODO(zhihuang): Merging and replacing following methods handling header // extensions with SetParameters: // UpdateRtpHeaderExtensionMap, // UpdateSendEncryptedHeaderExtensionIds, // UpdateRecvEncryptedHeaderExtensionIds, // CacheRtpAbsSendTimeHeaderExtension, virtual void UpdateRtpHeaderExtensionMap( const cricket::RtpHeaderExtensions& header_extensions) = 0; virtual bool IsSrtpActive() const = 0; virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria, RtpPacketSinkInterface* sink) = 0; virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0; }; } // namespace webrtc #endif // PC_RTP_TRANSPORT_INTERNAL_H_