#pragma once #include #include #include "DtlsTransport.hpp" #include "IceServer.hpp" #include "SrtpSession.hpp" #include "StunPacket.hpp" #include "Sdp.h" #include "Poller/EventPoller.h" #include "Network/Socket.h" #include "Rtsp/RtspMediaSourceImp.h" #include "Rtcp/RtcpContext.h" using namespace toolkit; using namespace mediakit; class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener { public: using Ptr = std::shared_ptr; WebRtcTransport(const EventPoller::Ptr &poller); ~WebRtcTransport() override = default; /** * 消费对象 */ virtual void onDestory(); /** * 创建webrtc answer sdp * @param offer offer sdp * @return answer sdp */ std::string getAnswerSdp(const string &offer); /** * socket收到udp数据 * @param buf 数据指针 * @param len 数据长度 * @param tuple 数据来源 */ void inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple); /** * 发送rtp * @param buf rtcp内容 * @param len rtcp长度 * @param flush 是否flush socket * @param pt rtp payload type */ void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt); void sendRtcpPacket(char *buf, size_t len, bool flush); const EventPoller::Ptr& getPoller() const; protected: //// dtls相关的回调 //// void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override {}; void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport, RTC::SrtpSession::CryptoSuite srtpCryptoSuite, uint8_t *srtpLocalKey, size_t srtpLocalKeyLen, uint8_t *srtpRemoteKey, size_t srtpRemoteKeyLen, std::string &remoteCert) override; void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override {}; void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override {}; void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override; void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override {}; protected: //// ice相关的回调 /// void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override; void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override; void OnIceServerConnected(const RTC::IceServer *iceServer) override; void OnIceServerCompleted(const RTC::IceServer *iceServer) override; void OnIceServerDisconnected(const RTC::IceServer *iceServer) override; protected: virtual void onStartWebRTC() = 0; virtual void onRtcConfigure(RtcConfigure &configure) const {} virtual void onCheckSdp(SdpType type, RtcSession &sdp); virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0; virtual void onRtp(const char *buf, size_t len) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; protected: const RtcSession& getSdp(SdpType type) const; private: void onSendSockData(const char *buf, size_t len, bool flush = true); void setRemoteDtlsFingerprint(const RtcSession &remote); private: EventPoller::Ptr _poller; std::shared_ptr _ice_server; std::shared_ptr _dtls_transport; std::shared_ptr _srtp_session_send; std::shared_ptr _srtp_session_recv; RtcSession::Ptr _offer_sdp; RtcSession::Ptr _answer_sdp; }; class RtpReceiverImp; class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this{ public: using Ptr = std::shared_ptr; ~WebRtcTransportImp() override = default; /** * 创建WebRTC对象 * @param poller 改对象需要绑定的线程 * @return 对象 */ static Ptr create(const EventPoller::Ptr &poller); /** * 绑定rtsp媒体源 * @param src 媒体源 * @param is_play 是播放还是推流 */ void attach(const RtspMediaSource::Ptr &src, bool is_play = true); protected: void onStartWebRTC() override; void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override; void onCheckSdp(SdpType type, RtcSession &sdp) override; void onRtcConfigure(RtcConfigure &configure) const override; void onRtp(const char *buf, size_t len) override; void onRtcp(const char *buf, size_t len) override; private: WebRtcTransportImp(const EventPoller::Ptr &poller); void onDestory() override; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush); SdpAttrCandidate::Ptr getIceCandidate() const; bool canSendRtp() const; bool canRecvRtp() const; class RtpPayloadInfo { public: bool is_common_rtp; const RtcCodecPlan *plan; const RtcMedia *media; std::shared_ptr receiver; RtcpContext::Ptr rtcp_context_recv; RtcpContext::Ptr rtcp_context_send; }; void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp); void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp); private: //pli rtcp计时器 Ticker _pli_ticker; //rtc rtp推流的视频ssrc uint32_t _recv_video_ssrc; //记录协商的rtp的pt类型 uint8_t _send_rtp_pt[2] = {0, 0}; //复合udp端口,接收一切rtp与rtcp Socket::Ptr _socket; //推流的rtsp源 RtspMediaSource::Ptr _push_src; //播放的rtsp源 RtspMediaSource::Ptr _play_src; //播放rtsp源的reader对象 RtspMediaSource::RingType::RingReader::Ptr _reader; //根据rtp的pt获取相关信息 unordered_map _rtp_info_pt; //根据推流端rtp的ssrc获取相关信息 unordered_map _rtp_info_ssrc; };