285 lines
14 KiB
C++
285 lines
14 KiB
C++
/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like 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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#ifndef SESSION_RTSPSESSION_H_
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#define SESSION_RTSPSESSION_H_
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#include <set>
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#include <vector>
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#include <unordered_set>
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#include "Network/Session.h"
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#include "RtspSplitter.h"
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#include "RtpReceiver.h"
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#include "Rtcp/RtcpContext.h"
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#include "RtspMediaSource.h"
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#include "RtspMediaSourceImp.h"
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#include "RtpMultiCaster.h"
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namespace mediakit {
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using BufferRtp = toolkit::BufferOffset<toolkit::Buffer::Ptr>;
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class RtspSession : public toolkit::Session, public RtspSplitter, public RtpReceiver, public MediaSourceEvent {
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public:
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using Ptr = std::shared_ptr<RtspSession>;
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using onGetRealm = std::function<void(const std::string &realm)>;
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// encrypted为true是则表明是md5加密的密码，否则是明文密码 [AUTO-TRANSLATED:cad96e51]
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// `encrypted` being `true` indicates an MD5 encrypted password, otherwise it is a plain text password
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// 在请求明文密码时如果提供md5密码者则会导致认证失败 [AUTO-TRANSLATED:8a38bff8]
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// When requesting a plain text password, providing an MD5 password will result in authentication failure
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using onAuth = std::function<void(bool encrypted, const std::string &pwd_or_md5)>;
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RtspSession(const toolkit::Socket::Ptr &sock);
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////Session override////
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void onRecv(const toolkit::Buffer::Ptr &buf) override;
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void onError(const toolkit::SockException &err) override;
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void onManager() override;
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protected:
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/////RtspSplitter override/////
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// 收到完整的rtsp包回调，包括sdp等content数据 [AUTO-TRANSLATED:efbe20df]
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// Callback for receiving a complete RTSP packet, including SDP and other content data
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void onWholeRtspPacket(Parser &parser) override;
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// 收到rtp包回调 [AUTO-TRANSLATED:119f1cca]
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// Callback for receiving an RTP packet
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void onRtpPacket(const char *data, size_t len) override;
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// 从rtsp头中获取Content长度 [AUTO-TRANSLATED:0e6f033e]
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// Get the Content length from the RTSP header
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ssize_t getContentLength(Parser &parser) override;
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////RtpReceiver override////
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void onRtpSorted(RtpPacket::Ptr rtp, int track_idx) override;
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void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override;
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///////MediaSourceEvent override///////
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// 关闭 [AUTO-TRANSLATED:92392f02]
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// Close
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bool close(MediaSource &sender) override;
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// 播放总人数 [AUTO-TRANSLATED:c42a3161]
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// Total number of players
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int totalReaderCount(MediaSource &sender) override;
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// 获取媒体源类型 [AUTO-TRANSLATED:34290a69]
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// Get the media source type
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MediaOriginType getOriginType(MediaSource &sender) const override;
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// 获取媒体源url或者文件路径 [AUTO-TRANSLATED:fa34d795]
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// Get the media source URL or file path
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std::string getOriginUrl(MediaSource &sender) const override;
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// 获取媒体源客户端相关信息 [AUTO-TRANSLATED:037ef910]
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// Get the media source client related information
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std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
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// 由于支持断连续推，存在OwnerPoller变更的可能 [AUTO-TRANSLATED:1c863b40]
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// Due to support for continuous pushing, there is a possibility of OwnerPoller changes
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toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
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/////Session override////
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ssize_t send(toolkit::Buffer::Ptr pkt) override;
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// 收到RTCP包回调 [AUTO-TRANSLATED:249f4807]
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// Callback for receiving an RTCP packet
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virtual void onRtcpPacket(int track_idx, SdpTrack::Ptr &track, const char *data, size_t len);
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// 回复客户端 [AUTO-TRANSLATED:8108ebea]
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// Reply to the client
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virtual bool sendRtspResponse(const std::string &res_code, const StrCaseMap &header = StrCaseMap(), const std::string &sdp = "", const char *protocol = "RTSP/1.0");
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protected:
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// url解析后保存的相关信息 [AUTO-TRANSLATED:1c26e4e3]
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// Information related to the URL after parsing
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MediaInfo _media_info;
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////////RTP over udp_multicast////////
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// 共享的rtp组播对象 [AUTO-TRANSLATED:d4a5cfdd]
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// Shared RTP multicast object
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RtpMultiCaster::Ptr _multicaster;
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// Session号 [AUTO-TRANSLATED:4552ec74]
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// Session number
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std::string _sessionid;
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uint32_t _multicast_ip = 0;
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uint16_t _multicast_video_port = 0;
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uint16_t _multicast_audio_port = 0;
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private:
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// 处理options方法,获取服务器能力 [AUTO-TRANSLATED:a51f6d7c]
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// Handle the OPTIONS method, get server capabilities
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void handleReq_Options(const Parser &parser);
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// 处理describe方法，请求服务器rtsp sdp信息 [AUTO-TRANSLATED:ed2c8fcb]
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// Handle the DESCRIBE method, request server RTSP SDP information
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void handleReq_Describe(const Parser &parser);
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// 处理ANNOUNCE方法，请求推流，附带sdp [AUTO-TRANSLATED:aa4b4517]
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// Handle the ANNOUNCE method, request streaming, with SDP attached
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void handleReq_ANNOUNCE(const Parser &parser);
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// 处理record方法，开始推流 [AUTO-TRANSLATED:885cf8a9]
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// Handle the RECORD method, start streaming
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void handleReq_RECORD(const Parser &parser);
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// 处理setup方法，播放和推流协商rtp传输方式用 [AUTO-TRANSLATED:cbe5dcfc]
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// Handle the SETUP method, used for negotiating RTP transport methods for playback and streaming
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void handleReq_Setup(const Parser &parser);
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// 处理play方法，开始或恢复播放 [AUTO-TRANSLATED:f15d151d]
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// Handle the PLAY method, start or resume playback
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void handleReq_Play(const Parser &parser);
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// 处理pause方法，暂停播放 [AUTO-TRANSLATED:0c3b8f79]
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// Handle the PAUSE method, pause playback
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void handleReq_Pause(const Parser &parser);
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// 处理teardown方法，结束播放 [AUTO-TRANSLATED:64d82572]
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// Handle the TEARDOWN method, end playback
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void handleReq_Teardown(const Parser &parser);
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// 处理Get方法,rtp over http才用到 [AUTO-TRANSLATED:c7c51eb6]
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// Handle the GET method, only used for RTP over HTTP
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void handleReq_Get(const Parser &parser);
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// 处理Post方法，rtp over http才用到 [AUTO-TRANSLATED:228bdbbe]
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// Handle the POST method, only used for RTP over HTTP
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void handleReq_Post(const Parser &parser);
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// 处理SET_PARAMETER、GET_PARAMETER方法，一般用于心跳 [AUTO-TRANSLATED:b9e333e1]
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// Handle the SET_PARAMETER, GET_PARAMETER methods, generally used for heartbeats
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void handleReq_SET_PARAMETER(const Parser &parser);
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// rtsp资源未找到 [AUTO-TRANSLATED:9b779890]
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// RTSP resource not found
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void send_StreamNotFound();
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// 不支持的传输模式 [AUTO-TRANSLATED:ef90414c]
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// Unsupported transport mode
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void send_UnsupportedTransport();
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// 会话id错误 [AUTO-TRANSLATED:7cf632d3]
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// Session ID error
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void send_SessionNotFound();
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// 一般rtsp服务器打开端口失败时触发 [AUTO-TRANSLATED:82ecb043]
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// Triggered when the general RTSP server fails to open the port
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void send_NotAcceptable();
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// 获取track下标 [AUTO-TRANSLATED:36d0b2c2]
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// Get the track index
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int getTrackIndexByTrackType(TrackType type);
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int getTrackIndexByControlUrl(const std::string &control_url);
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int getTrackIndexByInterleaved(int interleaved);
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// 一般用于接收udp打洞包，也用于rtsp推流 [AUTO-TRANSLATED:0b55c12f]
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// Generally used to receive UDP hole punching packets, also used for RTSP pushing
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void onRcvPeerUdpData(int interleaved, const toolkit::Buffer::Ptr &buf, const struct sockaddr_storage &addr);
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// 配合onRcvPeerUdpData使用 [AUTO-TRANSLATED:811d2d1a]
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// Used in conjunction with onRcvPeerUdpData
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void startListenPeerUdpData(int track_idx);
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// //rtsp专有认证相关//// [AUTO-TRANSLATED:0f021bb5]
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// // RTSP specific authentication related ////
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// 认证成功 [AUTO-TRANSLATED:e1bafff3]
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// Authentication successful
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void onAuthSuccess();
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// 认证失败 [AUTO-TRANSLATED:a188326a]
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// Authentication failed
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void onAuthFailed(const std::string &realm, const std::string &why, bool close = true);
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// 开始走rtsp专有认证流程 [AUTO-TRANSLATED:2d773497]
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// Start the RTSP specific authentication process
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void onAuthUser(const std::string &realm, const std::string &authorization);
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// 校验base64方式的认证加密 [AUTO-TRANSLATED:bde8662f]
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// Verify base64 authentication encryption
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void onAuthBasic(const std::string &realm, const std::string &auth_base64);
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// 校验md5方式的认证加密 [AUTO-TRANSLATED:0cc37fa7]
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// Verify MD5 authentication encryption
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void onAuthDigest(const std::string &realm, const std::string &auth_md5);
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// 触发url鉴权事件 [AUTO-TRANSLATED:776dc4b5]
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// Trigger URL authentication event
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void emitOnPlay();
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// 发送rtp给客户端 [AUTO-TRANSLATED:18602be0]
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// Send RTP to the client
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void sendRtpPacket(const RtspMediaSource::RingDataType &pkt);
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// 触发rtcp发送 [AUTO-TRANSLATED:4fbe7706]
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// Trigger RTCP sending
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void updateRtcpContext(const RtpPacket::Ptr &rtp);
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// 回复客户端 [AUTO-TRANSLATED:8108ebea]
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// Reply to the client
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bool sendRtspResponse(const std::string &res_code, const std::initializer_list<std::string> &header, const std::string &sdp = "", const char *protocol = "RTSP/1.0");
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// 设置socket标志 [AUTO-TRANSLATED:4086e686]
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// Set socket flag
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void setSocketFlags();
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private:
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// 是否已经触发on_play事件 [AUTO-TRANSLATED:49c937ce]
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// Whether the on_play event has been triggered
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bool _emit_on_play = false;
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bool _send_sr_rtcp[2] = {true, true};
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// 断连续推延时 [AUTO-TRANSLATED:13ad578a]
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// Delay in continuous pushing
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uint32_t _continue_push_ms = 0;
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// 推流或拉流客户端采用的rtp传输方式 [AUTO-TRANSLATED:27411079]
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// RTP transport method used by the pushing or pulling client
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Rtsp::eRtpType _rtp_type = Rtsp::RTP_Invalid;
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// 收到的seq，回复时一致 [AUTO-TRANSLATED:64544fb4]
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// Received seq, consistent when replying
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int _cseq = 0;
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// 消耗的总流量 [AUTO-TRANSLATED:45ad2785]
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// Total traffic consumed
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uint64_t _bytes_usage = 0;
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//ContentBase
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std::string _content_base;
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// 记录是否需要rtsp专属鉴权，防止重复触发事件 [AUTO-TRANSLATED:9cff90b9]
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// Record whether RTSP specific authentication is required to prevent duplicate event triggering
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std::string _rtsp_realm;
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// 登录认证 [AUTO-TRANSLATED:43fdb875]
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// Login authentication
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std::string _auth_nonce;
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// 用于判断客户端是否超时 [AUTO-TRANSLATED:86cb328a]
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// Used to determine if the client has timed out
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toolkit::Ticker _alive_ticker;
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// rtsp推流相关绑定的源 [AUTO-TRANSLATED:a25078d9]
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// Source bound to RTSP pushing
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RtspMediaSourceImp::Ptr _push_src;
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// 推流器所有权 [AUTO-TRANSLATED:e47b4bcb]
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// Pusher ownership
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std::shared_ptr<void> _push_src_ownership;
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// rtsp播放器绑定的直播源 [AUTO-TRANSLATED:a7e130b5]
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// Live source bound to the RTSP player
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std::weak_ptr<RtspMediaSource> _play_src;
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// 直播源读取器 [AUTO-TRANSLATED:e1edc193]
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// Live source reader
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RtspMediaSource::RingType::RingReader::Ptr _play_reader;
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// sdp里面有效的track,包含音频或视频 [AUTO-TRANSLATED:64e2fcdf]
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// Valid track in SDP, including audio or video
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std::vector<SdpTrack::Ptr> _sdp_track;
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// 播放器setup指定的播放track,默认为TrackInvalid表示不指定即音视频都推 [AUTO-TRANSLATED:c7a0df5e]
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// Track specified by the player setup, default is TrackInvalid, which means no specification, both audio and video are pushed
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TrackType _target_play_track = TrackInvalid;
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////////RTP over udp////////
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// RTP端口,trackid idx 为数组下标 [AUTO-TRANSLATED:77c186bb]
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// RTP port, trackid idx is the array index
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toolkit::Socket::Ptr _rtp_socks[2];
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// RTCP端口,trackid idx 为数组下标 [AUTO-TRANSLATED:446a7861]
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// RTCP port, trackid idx is the array index
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toolkit::Socket::Ptr _rtcp_socks[2];
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// 标记是否收到播放的udp打洞包,收到播放的udp打洞包后才能知道其外网udp端口号 [AUTO-TRANSLATED:ad039c25]
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// Flag whether the UDP hole punching packet for playback has been received. The external UDP port number can only be known after receiving the UDP hole punching packet for playback.
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std::unordered_set<int> _udp_connected_flags;
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////////RTSP over HTTP ////////
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// quicktime 请求rtsp会产生两次tcp连接， [AUTO-TRANSLATED:3f72e181]
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// QuickTime requests for RTSP will generate two TCP connections,
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// 一次发送 get 一次发送post，需要通过x-sessioncookie关联起来 [AUTO-TRANSLATED:f29a653f]
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// one for sending GET and one for sending POST. They need to be associated through x-sessioncookie.
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std::string _http_x_sessioncookie;
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std::function<void(const toolkit::Buffer::Ptr &)> _on_recv;
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////////// rtcp ////////////////
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// rtcp发送时间,trackid idx 为数组下标 [AUTO-TRANSLATED:bf3248b1]
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// RTCP send time, trackid idx is the array index
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toolkit::Ticker _rtcp_send_tickers[2];
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// 统计rtp并发送rtcp [AUTO-TRANSLATED:0ac2b665]
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// Count RTP and send RTCP
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std::vector<RtcpContext::Ptr> _rtcp_context;
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};
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/**
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* 支持ssl加密的rtsp服务器，可用于诸如亚马逊echo show这样的设备访问
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* RTSP server supporting SSL encryption, which can be used for devices such as Amazon Echo Show to access.
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* [AUTO-TRANSLATED:7d1eed83]
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*/
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using RtspSessionWithSSL = toolkit::SessionWithSSL<RtspSession>;
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} /* namespace mediakit */
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#endif /* SESSION_RTSPSESSION_H_ */
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