RtspSession.h 7.24 KB
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/*
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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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 *
 * 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.
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 */

#ifndef SESSION_RTSPSESSION_H_
#define SESSION_RTSPSESSION_H_

#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"
#include "Util/logger.h"
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#include "Common/config.h"
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#include "Network/TcpSession.h"
#include "Player/PlayerBase.h"
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#include "RtpBroadCaster.h"
#include "RtspMediaSource.h"
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#include "RtspSplitter.h"
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#include "RtpReceiver.h"
#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:
    typedef std::shared_ptr<BufferRtp> Ptr;
    BufferRtp(const RtpPacket::Ptr & pkt,uint32_t offset = 0 ):_rtp(pkt),_offset(offset){}
    virtual ~BufferRtp(){}

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    char *data() const override {
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        return (char *)_rtp->data() + _offset;
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    }
    uint32_t size() const override {
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        return _rtp->size() - _offset;
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    }
private:
    RtpPacket::Ptr _rtp;
    uint32_t _offset;
};

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class RtspSession: public TcpSession, public RtspSplitter, public RtpReceiver , public MediaSourceEvent{
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public:
	typedef std::shared_ptr<RtspSession> Ptr;
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	typedef std::function<void(const string &realm)> 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<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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	void onRecv(const Buffer::Ptr &pBuf) override;
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	void onError(const SockException &err) override;
	void onManager() override;
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protected:
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	//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;

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	//RtpReceiver override
	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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    /**
     * 收到RTCP包回调
     * @param iTrackidx
     * @param track
     * @param pucData
     * @param uiLen
     */
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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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    void handleReq_Options(const Parser &parser); //处理options方法
    void handleReq_Describe(const Parser &parser); //处理describe方法
    void handleReq_ANNOUNCE(const Parser &parser); //处理options方法
    void handleReq_RECORD(const Parser &parser); //处理options方法
    void handleReq_Setup(const Parser &parser); //处理setup方法
    void handleReq_Play(const Parser &parser); //处理play方法
    void handleReq_Pause(const Parser &parser); //处理pause方法
    void handleReq_Teardown(const Parser &parser); //处理teardown方法
    void handleReq_Get(const Parser &parser); //处理Get方法
    void handleReq_Post(const Parser &parser); //处理Post方法
    void handleReq_SET_PARAMETER(const Parser &parser); //处理SET_PARAMETER方法
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	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
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	inline string printSSRC(uint32_t ui32Ssrc);
	inline int getTrackIndexByTrackType(TrackType type);
    inline int getTrackIndexByControlSuffix(const string &controlSuffix);
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	inline int getTrackIndexByInterleaved(int interleaved);
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	inline void onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
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	inline void startListenPeerUdpData(int iTrackIdx);
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    //认证相关
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    void onAuthSuccess();
    void onAuthFailed(const string &realm,const string &why,bool close = true);
    void onAuthUser(const string &realm,const string &authorization);
    void onAuthBasic(const string &realm,const string &strBase64);
    void onAuthDigest(const string &realm,const string &strMd5);
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    void sendRtpPacket(const RtpPacket::Ptr &pkt);
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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");
	bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0");
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	void sendSenderReport(bool overTcp,int iTrackIndex);
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private:
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	Ticker _ticker;
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	int _iCseq = 0;
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	string _strContentBase;
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	string _strSdp;
	string _strSession;
	bool _bFirstPlay = true;
    MediaInfo _mediaInfo;
	std::weak_ptr<RtspMediaSource> _pMediaSrc;
	RingBuffer<RtpPacket::Ptr>::RingReader::Ptr _pRtpReader;
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	Rtsp::eRtpType _rtpType = Rtsp::RTP_Invalid;
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	vector<SdpTrack::Ptr> _aTrackInfo;

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	//RTP over udp
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	Socket::Ptr _apRtpSock[2]; //RTP端口,trackid idx 为数组下标
	Socket::Ptr _apRtcpSock[2];//RTCP端口,trackid idx 为数组下标
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    unordered_set<int> _udpSockConnected;
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	//RTP over udp_multicast
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	RtpBroadCaster::Ptr _pBrdcaster;
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	//登录认证
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    string _strNonce;
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    //消耗的总流量
    uint64_t _ui64TotalBytes = 0;
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	//RTSP over HTTP
	//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
	string _http_x_sessioncookie;
	function<void(const Buffer::Ptr &pBuf)> _onRecv;
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    bool _enableSendRtp;
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    //rtsp推流相关
	RtspToRtmpMediaSource::Ptr _pushSrc;

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	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
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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
 */
typedef TcpSessionWithSSL<RtspSession> RtspSessionWithSSL;

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} /* namespace mediakit */
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#endif /* SESSION_RTSPSESSION_H_ */