RtspSession.h

/*
 * 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 <set>
#include <vector>
#include <unordered_set>
#include <unordered_map>
#include "Util/util.h"
#include "Util/logger.h"
#include "Common/config.h"
#include "Network/TcpSession.h"
#include "Player/PlayerBase.h"
#include "RtpMultiCaster.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<BufferRtp> 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<RtspSession> Ptr;
	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;

	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<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");
	//服务器发送rtcp
	void sendSenderReport(bool overTcp,int iTrackIndex);
	//设置socket标志
	void setSocketFlags();
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<RtspMediaSource> _pMediaSrc;
	//直播源读取器
	RingBuffer<RtpPacket::Ptr>::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<SdpTrack::Ptr> _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<int> _udpSockConnected;
	////////RTP over udp_multicast////////
	//共享的rtp组播对象
	RtpMultiCaster::Ptr _multicaster;

	//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<void(const Buffer::Ptr &pBuf)> _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<RtspSession> RtspSessionWithSSL;

} /* namespace mediakit */

#endif /* SESSION_RTSPSESSION_H_ */