/* * MIT License * * Copyright (c) 2016-2019 xiongziliang <771730766@qq.com> * * This file is part of ZLMediaKit(https://github.com/xiongziliang/ZLMediaKit). * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in all * copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE * SOFTWARE. */ #ifndef SESSION_RTSPSESSION_H_ #define SESSION_RTSPSESSION_H_ #include #include #include #include #include "Util/util.h" #include "Util/logger.h" #include "Common/config.h" #include "Network/TcpSession.h" #include "Player/PlayerBase.h" #include "RtpBroadCaster.h" #include "RtspMediaSource.h" #include "RtspSplitter.h" #include "RtpReceiver.h" #include "RtspToRtmpMediaSource.h" using namespace std; using namespace toolkit; namespace mediakit { class RtspSession; class BufferRtp : public Buffer{ public: typedef std::shared_ptr Ptr; BufferRtp(const RtpPacket::Ptr & pkt,uint32_t offset = 0 ):_rtp(pkt),_offset(offset){} virtual ~BufferRtp(){} char *data() const override { return (char *)_rtp->data() + _offset; } uint32_t size() const override { return _rtp->size() - _offset; } private: RtpPacket::Ptr _rtp; uint32_t _offset; }; class RtspSession: public TcpSession, public RtspSplitter, public RtpReceiver , public MediaSourceEvent{ public: typedef std::shared_ptr Ptr; typedef std::function onGetRealm; //encrypted为true是则表明是md5加密的密码，否则是明文密码 //在请求明文密码时如果提供md5密码者则会导致认证失败 typedef std::function onAuth; RtspSession(const Socket::Ptr &pSock); virtual ~RtspSession(); ////TcpSession override//// void onRecv(const Buffer::Ptr &pBuf) override; void onError(const SockException &err) override; void onManager() override; protected: //RtspSplitter override /** * 收到完整的rtsp包回调，包括sdp等content数据 * @param parser rtsp包 */ void onWholeRtspPacket(Parser &parser) override; /** * 收到rtp包回调 * @param data * @param len */ void onRtpPacket(const char *data,uint64_t len) override; /** * 从rtsp头中获取Content长度 * @param parser * @return */ int64_t getContentLength(Parser &parser) override; //RtpReceiver override void onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx) override; //MediaSourceEvent override bool close(MediaSource &sender,bool force) override ; void onNoneReader(MediaSource &sender) override; //TcpSession override int send(const Buffer::Ptr &pkt) override; /** * 收到RTCP包回调 * @param iTrackidx * @param track * @param pucData * @param uiLen */ virtual void onRtcpPacket(int iTrackidx, SdpTrack::Ptr &track, unsigned char *pucData, unsigned int uiLen); private: //处理options方法,获取服务器能力 void handleReq_Options(const Parser &parser); //处理describe方法，请求服务器rtsp sdp信息 void handleReq_Describe(const Parser &parser); //处理ANNOUNCE方法，请求推流，附带sdp void handleReq_ANNOUNCE(const Parser &parser); //处理record方法，开始推流 void handleReq_RECORD(const Parser &parser); //处理setup方法，播放和推流协商rtp传输方式用 void handleReq_Setup(const Parser &parser); //处理play方法，开始或恢复播放 void handleReq_Play(const Parser &parser); //处理pause方法，暂停播放 void handleReq_Pause(const Parser &parser); //处理teardown方法，结束播放 void handleReq_Teardown(const Parser &parser); //处理Get方法,rtp over http才用到 void handleReq_Get(const Parser &parser); //处理Post方法，rtp over http才用到 void handleReq_Post(const Parser &parser); //处理SET_PARAMETER、GET_PARAMETER方法，一般用于心跳 void handleReq_SET_PARAMETER(const Parser &parser); //rtsp资源未找到 void inline send_StreamNotFound(); //不支持的传输模式 void inline send_UnsupportedTransport(); //会话id错误 void inline send_SessionNotFound(); //一般rtsp服务器打开端口失败时触发 void inline send_NotAcceptable(); //ssrc转字符串 inline string printSSRC(uint32_t ui32Ssrc); //获取track下标 inline int getTrackIndexByTrackType(TrackType type); inline int getTrackIndexByControlSuffix(const string &controlSuffix); inline int getTrackIndexByInterleaved(int interleaved); //一般用于接收udp打洞包，也用于rtsp推流 inline void onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBuf, const struct sockaddr &addr); //配合onRcvPeerUdpData使用 inline void startListenPeerUdpData(int iTrackIdx); ////rtsp专有认证相关//// //认证成功 void onAuthSuccess(); //认证失败 void onAuthFailed(const string &realm,const string &why,bool close = true); //开始走rtsp专有认证流程 void onAuthUser(const string &realm,const string &authorization); //校验base64方式的认证加密 void onAuthBasic(const string &realm,const string &strBase64); //校验md5方式的认证加密 void onAuthDigest(const string &realm,const string &strMd5); //发送rtp给客户端 void sendRtpPacket(const RtpPacket::Ptr &pkt); //回复客户端 bool sendRtspResponse(const string &res_code,const std::initializer_list &header, const string &sdp = "" , const char *protocol = "RTSP/1.0"); bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0"); //服务器发送rtcp void sendSenderReport(bool overTcp,int iTrackIndex); private: //用于判断客户端是否超时 Ticker _ticker; //收到的seq，回复时一致 int _iCseq = 0; //ContentBase string _strContentBase; //Session号 string _strSession; //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 bool _bFirstPlay = true; //url解析后保存的相关信息 MediaInfo _mediaInfo; //rtsp播放器绑定的直播源 std::weak_ptr _pMediaSrc; //直播源读取器 RingBuffer::RingReader::Ptr _pRtpReader; //推流或拉流客户端采用的rtp传输方式 Rtsp::eRtpType _rtpType = Rtsp::RTP_Invalid; //sdp里面有效的track,包含音频或视频 vector _aTrackInfo; ////////RTP over udp//////// //RTP端口,trackid idx 为数组下标 Socket::Ptr _apRtpSock[2]; //RTCP端口,trackid idx 为数组下标 Socket::Ptr _apRtcpSock[2]; //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 unordered_set _udpSockConnected; ////////RTP over udp_multicast//////// //共享的rtp组播对象 RtpBroadCaster::Ptr _pBrdcaster; //登录认证 string _strNonce; //消耗的总流量 uint64_t _ui64TotalBytes = 0; //RTSP over HTTP //quicktime 请求rtsp会产生两次tcp连接， //一次发送 get 一次发送post，需要通过x-sessioncookie关联起来 string _http_x_sessioncookie; function _onRecv; //是否开始发送rtp bool _enableSendRtp; //rtsp推流相关 RtspToRtmpMediaSource::Ptr _pushSrc; //rtcp统计,trackid idx 为数组下标 RtcpCounter _aRtcpCnt[2]; //rtcp发送时间,trackid idx 为数组下标 Ticker _aRtcpTicker[2]; //时间戳修整器 Stamp _stamp[2]; }; /** * 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 */ typedef TcpSessionWithSSL RtspSessionWithSSL; } /* namespace mediakit */ #endif /* SESSION_RTSPSESSION_H_ */