mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-01 17:11:31 +08:00
250 lines
8.6 KiB
C++
250 lines
8.6 KiB
C++
/*
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* MIT License
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*
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* Copyright (c) 2016-2019 xiongziliang <771730766@qq.com>
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*
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* This file is part of ZLMediaKit(https://github.com/xiongziliang/ZLMediaKit).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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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 "RtspToRtmpMediaSource.h"
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using namespace std;
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using namespace toolkit;
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namespace mediakit {
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class RtspSession;
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class BufferRtp : public Buffer{
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public:
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typedef std::shared_ptr<BufferRtp> Ptr;
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BufferRtp(const RtpPacket::Ptr & pkt,uint32_t offset = 0 ):_rtp(pkt),_offset(offset){}
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virtual ~BufferRtp(){}
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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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uint32_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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RtpPacket::Ptr _rtp;
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uint32_t _offset;
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};
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class RtspSession: public TcpSession, public RtspSplitter, public RtpReceiver , public MediaSourceEvent{
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public:
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typedef std::shared_ptr<RtspSession> Ptr;
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typedef std::function<void(const string &realm)> onGetRealm;
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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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typedef std::function<void(bool encrypted,const string &pwd_or_md5)> onAuth;
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RtspSession(const Socket::Ptr &pSock);
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virtual ~RtspSession();
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////TcpSession override////
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void onRecv(const Buffer::Ptr &pBuf) override;
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void onError(const 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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/**
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* 收到完整的rtsp包回调，包括sdp等content数据
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* @param parser rtsp包
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*/
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void onWholeRtspPacket(Parser &parser) override;
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/**
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* 收到rtp包回调
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* @param data
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* @param len
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*/
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void onRtpPacket(const char *data,uint64_t len) override;
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/**
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* 从rtsp头中获取Content长度
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* @param parser
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* @return
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*/
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int64_t getContentLength(Parser &parser) override;
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//RtpReceiver override
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void onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx) override;
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//MediaSourceEvent override
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bool close(MediaSource &sender,bool force) override ;
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void onNoneReader(MediaSource &sender) override;
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//TcpSession override
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int send(const Buffer::Ptr &pkt) override;
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/**
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* 收到RTCP包回调
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* @param iTrackidx
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* @param track
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* @param pucData
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* @param uiLen
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*/
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virtual void onRtcpPacket(int iTrackidx, SdpTrack::Ptr &track, unsigned char *pucData, unsigned int uiLen);
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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 inline send_StreamNotFound();
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//不支持的传输模式
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void inline send_UnsupportedTransport();
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//会话id错误
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void inline 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 inline send_NotAcceptable();
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//ssrc转字符串
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inline string printSSRC(uint32_t ui32Ssrc);
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//获取track下标
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inline int getTrackIndexByTrackType(TrackType type);
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inline int getTrackIndexByControlSuffix(const string &controlSuffix);
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inline 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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inline void onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
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//配合onRcvPeerUdpData使用
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inline void startListenPeerUdpData(int iTrackIdx);
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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 string &realm,const 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 string &realm,const string &authorization);
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//校验base64方式的认证加密
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void onAuthBasic(const string &realm,const string &strBase64);
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//校验md5方式的认证加密
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void onAuthDigest(const string &realm,const string &strMd5);
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//发送rtp给客户端
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void sendRtpPacket(const RtpPacket::Ptr &pkt);
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//回复客户端
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bool sendRtspResponse(const string &res_code,const std::initializer_list<string> &header, const string &sdp = "" , const char *protocol = "RTSP/1.0");
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bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0");
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//服务器发送rtcp
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void sendSenderReport(bool overTcp,int iTrackIndex);
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//设置socket标志
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void setSocketFlags();
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private:
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//用于判断客户端是否超时
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Ticker _ticker;
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//收到的seq，回复时一致
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int _iCseq = 0;
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//ContentBase
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string _strContentBase;
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//Session号
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string _strSession;
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//是否第一次播放，第一次播放需要鉴权，第二次播放属于暂停恢复
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bool _bFirstPlay = true;
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//url解析后保存的相关信息
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MediaInfo _mediaInfo;
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//rtsp播放器绑定的直播源
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std::weak_ptr<RtspMediaSource> _pMediaSrc;
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//直播源读取器
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RingBuffer<RtpPacket::Ptr>::RingReader::Ptr _pRtpReader;
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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 _rtpType = Rtsp::RTP_Invalid;
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//sdp里面有效的track,包含音频或视频
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vector<SdpTrack::Ptr> _aTrackInfo;
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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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Socket::Ptr _apRtpSock[2];
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//RTCP端口,trackid idx 为数组下标
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Socket::Ptr _apRtcpSock[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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unordered_set<int> _udpSockConnected;
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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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//登录认证
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string _strNonce;
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//消耗的总流量
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uint64_t _ui64TotalBytes = 0;
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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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string _http_x_sessioncookie;
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function<void(const Buffer::Ptr &pBuf)> _onRecv;
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//是否开始发送rtp
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bool _enableSendRtp;
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//rtsp推流相关
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RtspToRtmpMediaSource::Ptr _pushSrc;
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//rtcp统计,trackid idx 为数组下标
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RtcpCounter _aRtcpCnt[2];
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//rtcp发送时间,trackid idx 为数组下标
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Ticker _aRtcpTicker[2];
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//时间戳修整器
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Stamp _stamp[2];
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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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typedef TcpSessionWithSSL<RtspSession> RtspSessionWithSSL;
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} /* namespace mediakit */
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#endif /* SESSION_RTSPSESSION_H_ */
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