mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-27 13:49:01 +08:00
4783ac0808
最后一个连通的候选地址会被赋值并锁定为_selected_session，如果之前的候选地址再发送数据，将通过_selected_session回复，导致无法切换为旧的候选地址。
346 lines
12 KiB
C++
346 lines
12 KiB
C++
/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#pragma once
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#include <memory>
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#include <string>
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#include "DtlsTransport.hpp"
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#include "IceServer.hpp"
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#include "SrtpSession.hpp"
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#include "StunPacket.hpp"
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#include "Sdp.h"
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#include "Util/mini.h"
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#include "Poller/EventPoller.h"
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#include "Network/Socket.h"
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#include "Network/Session.h"
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#include "Nack.h"
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#include "TwccContext.h"
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#include "SctpAssociation.hpp"
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#include "Rtcp/RtcpContext.h"
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namespace mediakit {
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//RTC配置项目
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namespace Rtc {
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extern const std::string kPort;
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extern const std::string kTcpPort;
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extern const std::string kTimeOutSec;
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}//namespace RTC
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class WebRtcInterface {
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public:
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WebRtcInterface() = default;
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virtual ~WebRtcInterface() = default;
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virtual std::string getAnswerSdp(const std::string &offer) = 0;
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virtual const std::string &getIdentifier() const = 0;
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};
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class WebRtcException : public WebRtcInterface {
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public:
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WebRtcException(const SockException &ex) : _ex(ex) {};
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~WebRtcException() override = default;
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std::string getAnswerSdp(const std::string &offer) override {
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throw _ex;
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}
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const std::string &getIdentifier() const override {
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static std::string s_null;
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return s_null;
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}
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private:
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SockException _ex;
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};
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class WebRtcTransport : public WebRtcInterface, public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener, public std::enable_shared_from_this<WebRtcTransport>
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#ifdef ENABLE_SCTP
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, public RTC::SctpAssociation::Listener
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#endif
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{
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public:
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using Ptr = std::shared_ptr<WebRtcTransport>;
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WebRtcTransport(const EventPoller::Ptr &poller);
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~WebRtcTransport() override = default;
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/**
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* 创建对象
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*/
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virtual void onCreate();
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/**
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* 销毁对象
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*/
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virtual void onDestory();
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/**
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* 创建webrtc answer sdp
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* @param offer offer sdp
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* @return answer sdp
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*/
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std::string getAnswerSdp(const std::string &offer) override;
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/**
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* 获取对象唯一id
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*/
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const std::string& getIdentifier() const override;
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/**
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* socket收到udp数据
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* @param buf 数据指针
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* @param len 数据长度
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* @param tuple 数据来源
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*/
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void inputSockData(char *buf, int len, RTC::TransportTuple *tuple);
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/**
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* 发送rtp
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* @param buf rtcp内容
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* @param len rtcp长度
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* @param flush 是否flush socket
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* @param ctx 用户指针
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*/
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void sendRtpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
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void sendRtcpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
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const EventPoller::Ptr& getPoller() const;
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Session::Ptr getSession() const;
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protected:
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//// dtls相关的回调 ////
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void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override;
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void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport,
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RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
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uint8_t *srtpLocalKey,
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size_t srtpLocalKeyLen,
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uint8_t *srtpRemoteKey,
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size_t srtpRemoteKeyLen,
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std::string &remoteCert) override;
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void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override;
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void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
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void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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protected:
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//// ice相关的回调 ///
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void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override;
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void OnIceServerConnected(const RTC::IceServer *iceServer) override;
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void OnIceServerCompleted(const RTC::IceServer *iceServer) override;
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void OnIceServerDisconnected(const RTC::IceServer *iceServer) override;
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#ifdef ENABLE_SCTP
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void OnSctpAssociationConnecting(RTC::SctpAssociation* sctpAssociation) override;
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void OnSctpAssociationConnected(RTC::SctpAssociation* sctpAssociation) override;
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void OnSctpAssociationFailed(RTC::SctpAssociation* sctpAssociation) override;
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void OnSctpAssociationClosed(RTC::SctpAssociation* sctpAssociation) override;
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void OnSctpAssociationSendData(RTC::SctpAssociation* sctpAssociation, const uint8_t* data, size_t len) override;
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void OnSctpAssociationMessageReceived(RTC::SctpAssociation *sctpAssociation, uint16_t streamId, uint32_t ppid,
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const uint8_t *msg, size_t len) override;
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#endif
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protected:
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virtual void onStartWebRTC() = 0;
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virtual void onRtcConfigure(RtcConfigure &configure) const;
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virtual void onCheckSdp(SdpType type, RtcSession &sdp) = 0;
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virtual void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) = 0;
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virtual void onRtp(const char *buf, size_t len, uint64_t stamp_ms) = 0;
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virtual void onRtcp(const char *buf, size_t len) = 0;
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virtual void onShutdown(const SockException &ex) = 0;
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virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0;
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virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0;
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virtual void onRtcpBye() = 0;
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protected:
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void sendRtcpRemb(uint32_t ssrc, size_t bit_rate);
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void sendRtcpPli(uint32_t ssrc);
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private:
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void sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple);
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void setRemoteDtlsFingerprint(const RtcSession &remote);
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protected:
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RtcSession::Ptr _offer_sdp;
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RtcSession::Ptr _answer_sdp;
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std::shared_ptr<RTC::IceServer> _ice_server;
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private:
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std::string _identifier;
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EventPoller::Ptr _poller;
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std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
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std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
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std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
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Ticker _ticker;
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// 循环池
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ResourcePool<BufferRaw> _packet_pool;
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#ifdef ENABLE_SCTP
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RTC::SctpAssociationImp::Ptr _sctp;
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#endif
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};
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class RtpChannel;
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class MediaTrack {
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public:
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using Ptr = std::shared_ptr<MediaTrack>;
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const RtcCodecPlan *plan_rtp;
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const RtcCodecPlan *plan_rtx;
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uint32_t offer_ssrc_rtp = 0;
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uint32_t offer_ssrc_rtx = 0;
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uint32_t answer_ssrc_rtp = 0;
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uint32_t answer_ssrc_rtx = 0;
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const RtcMedia *media;
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RtpExtContext::Ptr rtp_ext_ctx;
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//for send rtp
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NackList nack_list;
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RtcpContext::Ptr rtcp_context_send;
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//for recv rtp
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std::unordered_map<std::string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
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std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
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};
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struct WrappedMediaTrack {
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MediaTrack::Ptr track;
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explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(ptr) {}
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virtual ~WrappedMediaTrack() {}
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virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) = 0;
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};
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struct WrappedRtxTrack: public WrappedMediaTrack {
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explicit WrappedRtxTrack(MediaTrack::Ptr ptr)
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: WrappedMediaTrack(std::move(ptr)) {}
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void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
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};
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class WebRtcTransportImp;
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struct WrappedRtpTrack : public WrappedMediaTrack {
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explicit WrappedRtpTrack(MediaTrack::Ptr ptr, TwccContext& twcc, WebRtcTransportImp& t)
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: WrappedMediaTrack(std::move(ptr))
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, _twcc_ctx(twcc)
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, _transport(t) {}
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TwccContext& _twcc_ctx;
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WebRtcTransportImp& _transport;
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void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
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};
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class WebRtcTransportImp : public WebRtcTransport {
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public:
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using Ptr = std::shared_ptr<WebRtcTransportImp>;
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~WebRtcTransportImp() override;
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uint64_t getBytesUsage() const;
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uint64_t getDuration() const;
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bool canSendRtp() const;
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bool canRecvRtp() const;
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void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
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void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
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void removeTuple(RTC::TransportTuple* tuple);
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protected:
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void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
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WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
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void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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void onStartWebRTC() override;
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void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
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void onCheckSdp(SdpType type, RtcSession &sdp) override;
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void onRtcConfigure(RtcConfigure &configure) const override;
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void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
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void onRtcp(const char *buf, size_t len) override;
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void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override;
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void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
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void onCreate() override;
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void onDestory() override;
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void onShutdown(const SockException &ex) override;
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virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
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void updateTicker();
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float getLossRate(TrackType type);
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void onRtcpBye() override;
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private:
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void onSortedRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp);
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void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
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void onSendTwcc(uint32_t ssrc, const std::string &twcc_fci);
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void registerSelf();
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void unregisterSelf();
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void unrefSelf();
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void onCheckAnswer(RtcSession &sdp);
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private:
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bool _preferred_tcp;
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uint16_t _rtx_seq[2] = {0, 0};
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//用掉的总流量
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uint64_t _bytes_usage = 0;
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//保持自我强引用
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Ptr _self;
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//检测超时的定时器
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Timer::Ptr _timer;
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//刷新计时器
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Ticker _alive_ticker;
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//pli rtcp计时器
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Ticker _pli_ticker;
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//twcc rtcp发送上下文对象
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TwccContext _twcc_ctx;
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//根据发送rtp的track类型获取相关信息
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MediaTrack::Ptr _type_to_track[2];
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//根据rtcp的ssrc获取相关信息，收发rtp和rtx的ssrc都会记录
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std::unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
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//根据接收rtp的pt获取相关信息
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std::unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
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};
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class WebRtcTransportManager {
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public:
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friend class WebRtcTransportImp;
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static WebRtcTransportManager &Instance();
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WebRtcTransportImp::Ptr getItem(const std::string &key);
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private:
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WebRtcTransportManager() = default;
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void addItem(const std::string &key, const WebRtcTransportImp::Ptr &ptr);
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void removeItem(const std::string &key);
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private:
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mutable std::mutex _mtx;
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std::unordered_map<std::string, std::weak_ptr<WebRtcTransportImp> > _map;
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};
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class WebRtcArgs {
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public:
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WebRtcArgs() = default;
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virtual ~WebRtcArgs() = default;
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virtual variant operator[](const std::string &key) const = 0;
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};
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class WebRtcPluginManager {
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public:
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using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
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using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
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static WebRtcPluginManager &Instance();
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void registerPlugin(const std::string &type, Plugin cb);
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void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
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private:
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WebRtcPluginManager() = default;
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private:
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mutable std::mutex _mtx_creator;
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std::unordered_map<std::string, Plugin> _map_creator;
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};
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}// namespace mediakit
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