/* * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit). * * Use of this source code is governed by MIT license that can be found in the * LICENSE file in the root of the source tree. All contributing project authors * may be found in the AUTHORS file in the root of the source tree. */ #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" #include "Rtcp/RtcpFCI.h" #include "Nack.h" #include "Network/Session.h" #include "TwccContext.h" using namespace toolkit; using namespace mediakit; //RTC配置项目 namespace RTC { extern const string kPort; extern const string kTimeOutSec; }//namespace RTC class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener, public std::enable_shared_from_this { public: using Ptr = std::shared_ptr; WebRtcTransport(const EventPoller::Ptr &poller); ~WebRtcTransport() override = default; /** * 创建对象 */ virtual void onCreate(); /** * 销毁对象 */ 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, int len, RTC::TransportTuple *tuple); /** * 发送rtp * @param buf rtcp内容 * @param len rtcp长度 * @param flush 是否flush socket * @param ctx 用户指针 */ void sendRtpPacket(const char *buf, int len, bool flush, void *ctx = nullptr); void sendRtcpPacket(const char *buf, int len, bool flush, void *ctx = nullptr); const EventPoller::Ptr& getPoller() const; const string& getIdentifier() 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) = 0; 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, uint64_t stamp_ms) = 0; virtual void onRtcp(const char *buf, size_t len) = 0; virtual void onShutdown(const SockException &ex) = 0; virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0; virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0; protected: RTC::TransportTuple* getSelectedTuple() const; void sendRtcpRemb(uint32_t ssrc, size_t bit_rate); void sendRtcpPli(uint32_t ssrc); private: void onSendSockData(const char *buf, size_t len, bool flush = true); void setRemoteDtlsFingerprint(const RtcSession &remote); protected: RtcSession::Ptr _offer_sdp; RtcSession::Ptr _answer_sdp; private: uint8_t _srtp_buf[2000]; string _identifier; 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; Ticker _ticker; }; class RtpChannel; class MediaTrack { public: using Ptr = std::shared_ptr; const RtcCodecPlan *plan_rtp; const RtcCodecPlan *plan_rtx; uint32_t offer_ssrc_rtp = 0; uint32_t offer_ssrc_rtx = 0; uint32_t answer_ssrc_rtp = 0; uint32_t answer_ssrc_rtx = 0; const RtcMedia *media; RtpExtContext::Ptr rtp_ext_ctx; //for send rtp NackList nack_list; RtcpContext::Ptr rtcp_context_send; //for recv rtp unordered_map > rtp_channel; std::shared_ptr getRtpChannel(uint32_t ssrc) const; }; class WebRtcTransportImp : public WebRtcTransport { public: using Ptr = std::shared_ptr; ~WebRtcTransportImp() override; void setSession(Session::Ptr session); const Session::Ptr& getSession() const; uint64_t getBytesUsage() const; uint64_t getDuration() const; bool canSendRtp() const; bool canRecvRtp() const; void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false); protected: WebRtcTransportImp(const EventPoller::Ptr &poller); 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, uint64_t stamp_ms) override; void onRtcp(const char *buf, size_t len) override; void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override; void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {}; void onCreate() override; void onDestory() override; void onShutdown(const SockException &ex) override; virtual void onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) = 0; private: SdpAttrCandidate::Ptr getIceCandidate() const; void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp); void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc); void onSendTwcc(uint32_t ssrc, const string &twcc_fci); void createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track); void registerSelf(); void unregisterSelf(); void unrefSelf(); void onCheckAnswer(RtcSession &sdp); private: uint16_t _rtx_seq[2] = {0, 0}; //用掉的总流量 uint64_t _bytes_usage = 0; //保持自我强引用 Ptr _self; //检测超时的定时器 Timer::Ptr _timer; //刷新计时器 Ticker _alive_ticker; //pli rtcp计时器 Ticker _pli_ticker; //udp session Session::Ptr _session; //twcc rtcp发送上下文对象 TwccContext _twcc_ctx; //根据发送rtp的track类型获取相关信息 MediaTrack::Ptr _type_to_track[2]; //根据rtcp的ssrc获取相关信息,收发rtp和rtx的ssrc都会记录 unordered_map _ssrc_to_track; //根据接收rtp的pt获取相关信息 unordered_map > _pt_to_track; }; class WebRtcTransportManager { mutable mutex _mtx; unordered_map > _map; WebRtcTransportManager() = default; public: static WebRtcTransportManager& instance(); void addItem(string key, const WebRtcTransportImp::Ptr &ptr); WebRtcTransportImp::Ptr getItem(const string &key); void removeItem(string key); };