/* * MIT License * * Copyright (c) 2016 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 "Common/config.h" #include "Rtsp.h" #include "RtpBroadCaster.h" #include "RtspMediaSource.h" #include "Player/PlayerBase.h" #include "Util/util.h" #include "Util/logger.h" #include "Network/TcpSession.h" #include "Http/HttpRequestSplitter.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->payload + _offset; } uint32_t size() const override { return _rtp->length - _offset; } private: RtpPacket::Ptr _rtp; uint32_t _offset; }; class RtspSession: public TcpSession, public HttpRequestSplitter { 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 std::shared_ptr &pTh, const Socket::Ptr &pSock); virtual ~RtspSession(); void onRecv(const Buffer::Ptr &pBuf) override; void onError(const SockException &err) override; void onManager() override; protected: //HttpRequestSplitter override int64_t onRecvHeader(const char *data,uint64_t len) override ; private: void inputRtspOrRtcp(const char *data,uint64_t len); int send(const Buffer::Ptr &pkt) override{ _ui64TotalBytes += pkt->size(); return _pSender->send(pkt,_flags); } void shutdown() override ; void shutdown_l(bool close); bool handleReq_Options(); //处理options方法 bool handleReq_Describe(); //处理describe方法 bool handleReq_Setup(); //处理setup方法 bool handleReq_Play(); //处理play方法 bool handleReq_Pause(); //处理pause方法 bool handleReq_Teardown(); //处理teardown方法 bool handleReq_Get(); //处理Get方法 bool handleReq_Post(); //处理Post方法 bool handleReq_SET_PARAMETER(); //处理SET_PARAMETER方法 void inline send_StreamNotFound(); //rtsp资源未找到 void inline send_UnsupportedTransport(); //不支持的传输模式 void inline send_SessionNotFound(); //会话id错误 void inline send_NotAcceptable(); //rtsp同时播放数限制 inline bool findStream(); //根据rtsp url查找 MediaSource实例 inline void initSender(const std::shared_ptr &pSession); //处理rtsp over http,quicktime使用的 inline void sendRtpPacket(const RtpPacket::Ptr &pkt); inline string printSSRC(uint32_t ui32Ssrc) { char tmp[9] = { 0 }; ui32Ssrc = htonl(ui32Ssrc); uint8_t *pSsrc = (uint8_t *) &ui32Ssrc; for (int i = 0; i < 4; i++) { sprintf(tmp + 2 * i, "%02X", pSsrc[i]); } return tmp; } inline int getTrackIndexByTrackType(TrackType type) { for (unsigned int i = 0; i < _aTrackInfo.size(); i++) { if (type == _aTrackInfo[i]->type) { return i; } } return -1; } inline int getTrackIndexByControlSuffix(const string &controlSuffix) { for (unsigned int i = 0; i < _aTrackInfo.size(); i++) { if (controlSuffix == _aTrackInfo[i]->_control_surffix) { return i; } } return -1; } inline void onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr &addr); inline void startListenPeerUdpData(); //认证相关 static void onAuthSuccess(const weak_ptr &weakSelf); static void onAuthFailed(const weak_ptr &weakSelf,const string &realm); static void onAuthUser(const weak_ptr &weakSelf,const string &realm,const string &authorization); static void onAuthBasic(const weak_ptr &weakSelf,const string &realm,const string &strBase64); static void onAuthDigest(const weak_ptr &weakSelf,const string &realm,const string &strMd5); private: char *_pcBuf = nullptr; Ticker _ticker; Parser _parser; //rtsp解析类 string _strUrl; string _strSdp; string _strSession; bool _bFirstPlay = true; MediaInfo _mediaInfo; std::weak_ptr _pMediaSrc; //RTP缓冲 weak_ptr > _pWeakRing; RingBuffer::RingReader::Ptr _pRtpReader; PlayerBase::eRtpType _rtpType = PlayerBase::RTP_UDP; bool _bSetUped = false; int _iCseq = 0; SdpAttr _sdpAttr; vector _aTrackInfo; bool _bGotAllPeerUdp = false; #ifdef RTSP_SEND_RTCP RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标 Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标 inline void sendRTCP(); #endif //RTP over UDP bool _abGotPeerUdp[2] = { false, false }; //获取客户端udp端口计数 weak_ptr _apUdpSock[2]; //发送RTP的UDP端口,trackid idx 为数组下标 std::shared_ptr _apPeerUdpAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标 bool _bListenPeerUdpData = false; RtpBroadCaster::Ptr _pBrdcaster; //登录认证 string _strNonce; //RTSP over HTTP function _onDestory; bool _bBase64need = false; //是否需要base64解码 Socket::Ptr _pSender; //回复rtsp时走的tcp通道,供quicktime用 //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sessioncookie关联起来 string _strSessionCookie; //消耗的总流量 uint64_t _ui64TotalBytes = 0; static recursive_mutex g_mtxGetter; //对quicktime上锁保护 static recursive_mutex g_mtxPostter; //对quicktime上锁保护 static unordered_map > g_mapGetter; static unordered_map > g_mapPostter; }; } /* namespace mediakit */ #endif /* SESSION_RTSPSESSION_H_ */