mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-26 20:47:08 +08:00
229 lines
8.6 KiB
C++
229 lines
8.6 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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#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 <unordered_map>
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#include "Util/util.h"
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#include "Util/logger.h"
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#include "Common/config.h"
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#include "Network/TcpSession.h"
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#include "Player/PlayerBase.h"
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#include "RtpMultiCaster.h"
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#include "RtspMediaSource.h"
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#include "RtspSplitter.h"
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#include "RtpReceiver.h"
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#include "RtspMediaSourceImp.h"
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#include "Common/Stamp.h"
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#include "Rtcp/RtcpContext.h"
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namespace mediakit {
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class RtspSession;
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class BufferRtp : public toolkit::Buffer{
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public:
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using Ptr = std::shared_ptr<BufferRtp>;
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BufferRtp(Buffer::Ptr pkt, size_t offset = 0) : _offset(offset), _rtp(std::move(pkt)) {}
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~BufferRtp() override {}
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char *data() const override {
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return (char *)_rtp->data() + _offset;
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}
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size_t size() const override {
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return _rtp->size() - _offset;
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}
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private:
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size_t _offset;
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Buffer::Ptr _rtp;
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};
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class RtspSession : public toolkit::TcpSession, 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
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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
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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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virtual ~RtspSession();
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////TcpSession 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
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void onWholeRtspPacket(Parser &parser) override;
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//收到rtp包回调
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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
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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
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bool close(MediaSource &sender, bool force) override;
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// 播放总人数
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int totalReaderCount(MediaSource &sender) override;
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// 获取媒体源类型
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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
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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
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std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
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/////TcpSession override////
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ssize_t send(toolkit::Buffer::Ptr pkt) override;
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//收到RTCP包回调
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virtual void onRtcpPacket(int track_idx, SdpTrack::Ptr &track, const char *data, size_t len);
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private:
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//处理options方法,获取服务器能力
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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
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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
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void handleReq_ANNOUNCE(const Parser &parser);
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//处理record方法，开始推流
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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
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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
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void handleReq_Play(const Parser &parser);
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//处理pause方法，暂停播放
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void handleReq_Pause(const Parser &parser);
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//处理teardown方法，结束播放
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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
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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
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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
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void handleReq_SET_PARAMETER(const Parser &parser);
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//rtsp资源未找到
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void send_StreamNotFound();
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//不支持的传输模式
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void send_UnsupportedTransport();
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//会话id错误
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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
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void send_NotAcceptable();
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//获取track下标
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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
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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
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void startListenPeerUdpData(int track_idx);
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////rtsp专有认证相关////
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//认证成功
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void onAuthSuccess();
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//认证失败
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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
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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
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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
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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
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void emitOnPlay();
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//发送rtp给客户端
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void sendRtpPacket(const RtspMediaSource::RingDataType &pkt);
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//触发rtcp发送
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void updateRtcpContext(const RtpPacket::Ptr &rtp);
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//回复客户端
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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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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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//设置socket标志
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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
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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
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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
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Rtsp::eRtpType _rtp_type = Rtsp::RTP_Invalid;
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//收到的seq，回复时一致
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int _cseq = 0;
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//消耗的总流量
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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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//Session号
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std::string _sessionid;
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//记录是否需要rtsp专属鉴权，防止重复触发事件
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std::string _rtsp_realm;
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//登录认证
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std::string _auth_nonce;
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//用于判断客户端是否超时
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toolkit::Ticker _alive_ticker;
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//url解析后保存的相关信息
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MediaInfo _media_info;
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//rtsp推流相关绑定的源
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RtspMediaSourceImp::Ptr _push_src;
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//推流器所有权
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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
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std::weak_ptr<RtspMediaSource> _play_src;
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//直播源读取器
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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
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std::vector<SdpTrack::Ptr> _sdp_track;
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////////RTP over udp////////
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//RTP端口,trackid idx 为数组下标
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toolkit::Socket::Ptr _rtp_socks[2];
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//RTCP端口,trackid idx 为数组下标
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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
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std::unordered_set<int> _udp_connected_flags;
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////////RTP over udp_multicast////////
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//共享的rtp组播对象
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RtpMultiCaster::Ptr _multicaster;
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////////RTSP over HTTP ////////
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//quicktime 请求rtsp会产生两次tcp连接，
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//一次发送 get 一次发送post，需要通过x-sessioncookie关联起来
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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
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toolkit::Ticker _rtcp_send_tickers[2];
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//统计rtp并发送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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*/
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using RtspSessionWithSSL = toolkit::TcpSessionWithSSL<RtspSession>;
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} /* namespace mediakit */
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#endif /* SESSION_RTSPSESSION_H_ */
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