rtp over udp改成独占式端口，提高性能

src/Common/config.cpp

@@ -58,7 +58,7 @@ const char kBroadcastHttpRequest[] = "kBroadcastHttpRequest";
 const char kBroadcastOnGetRtspRealm[] = "kBroadcastOnGetRtspRealm";
 const char kBroadcastOnRtspAuth[] = "kBroadcastOnRtspAuth";
 const char kBroadcastMediaPlayed[] = "kBroadcastMediaPlayed";
-const char kBroadcastRtmpPublish[] = "kBroadcastRtmpPublish";
+const char kBroadcastMediaPublish[] = "kBroadcastMediaPublish";
 const char kBroadcastFlowReport[] = "kBroadcastFlowReport";
 const char kBroadcastReloadConfig[] = "kBroadcastReloadConfig";
 const char kBroadcastShellLogin[] = "kBroadcastShellLogin";

src/Common/config.h

@@ -92,9 +92,13 @@ extern const char kBroadcastOnRtspAuth[];
 //如果errMessage为空则代表鉴权成功
 typedef std::function<void(const string &errMessage)> AuthInvoker;
 
-//收到rtmp推流事件广播，通过该事件控制推流鉴权
+//收到rtsp/rtmp推流事件广播，通过该事件控制推流鉴权
-extern const char kBroadcastRtmpPublish[];
+extern const char kBroadcastMediaPublish[];
-#define BroadcastRtmpPublishArgs const MediaInfo &args,const Broadcast::AuthInvoker &invoker,TcpSession &sender
+#define BroadcastMediaPublishArgs const MediaInfo &args,const Broadcast::AuthInvoker &invoker,TcpSession &sender
+
+//兼容旧代码的宏
+#define BroadcastRtmpPublishArgs BroadcastMediaPublishArgs
+#define kBroadcastRtmpPublish kBroadcastMediaPublish
 
 //播放rtsp/rtmp/http-flv事件广播，通过该事件控制播放鉴权
 extern const char kBroadcastMediaPlayed[];

src/Rtmp/RtmpSession.cpp

@@ -190,7 +190,7 @@ void RtmpSession::onCmd_publish(AMFDecoder &dec) {
             onRes(err);
         });
     };
-    auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastRtmpPublish,
+    auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish,
                                                    _mediaInfo,
                                                    invoker,
                                                    *this);

src/Rtsp/RtspSession.cpp

@@ -89,10 +89,9 @@ void RtspSession::shutdown_l(bool close){
 
 void RtspSession::onError(const SockException& err) {
 	TraceL << err.getErrCode() << " " << err.what();
-	if (_bListenPeerUdpData) {
+	if (_rtpType == PlayerBase::RTP_MULTICAST) {
 		//取消UDP端口监听
 		UDPServer::Instance().stopListenPeer(get_peer_ip().data(), this);
-		_bListenPeerUdpData = false;
 	}
 	if (!_bBase64need && _strSessionCookie.size() != 0) {
 		//quickTime http getter
@@ -131,7 +130,9 @@ void RtspSession::onManager() {
 			return;
 		}
 	}
-	if (_rtpType != PlayerBase::RTP_TCP && _ticker.elapsedTime() > 15 * 1000) {
+
+	//组播不检查心跳是否超时
+	if (_rtpType != PlayerBase::RTP_MULTICAST && _ticker.elapsedTime() > 15 * 1000) {
 		WarnL << "RTSP会话超时:" << get_peer_ip();
 		shutdown();
 		return;
@@ -157,6 +158,7 @@ int64_t RtspSession::onRecvHeader(const char *header,uint64_t len) {
 		s_handler_map.emplace("OPTIONS",&RtspSession::handleReq_Options);
 		s_handler_map.emplace("DESCRIBE",&RtspSession::handleReq_Describe);
 		s_handler_map.emplace("ANNOUNCE",&RtspSession::handleReq_ANNOUNCE);
+        s_handler_map.emplace("RECORD",&RtspSession::handleReq_RECORD);
 		s_handler_map.emplace("SETUP",&RtspSession::handleReq_Setup);
 		s_handler_map.emplace("PLAY",&RtspSession::handleReq_Play);
 		s_handler_map.emplace("PAUSE",&RtspSession::handleReq_Pause);
@@ -207,7 +209,7 @@ void RtspSession::inputRtspOrRtcp(const char *data,uint64_t len) {
 
 int RtspSession::handleReq_Options() {
 	//支持这些命令
-	sendRtspResponse("200 OK",{"Public" , "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, ANNOUNCE, SET_PARAMETER, GET_PARAMETER"});
+	sendRtspResponse("200 OK",{"Public" , "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, ANNOUNCE, RECORD, SET_PARAMETER, GET_PARAMETER"});
 	return 0;
 }
 
@@ -218,14 +220,71 @@ void RtspSession::onRecvContent(const char *data, uint64_t len) {
 		_onContent = nullptr;
 	}
 }
+
 int RtspSession::handleReq_ANNOUNCE() {
 	sendRtspResponse("200 OK");
 	_onContent = [this](const char *data, uint64_t len){
-
+	    _strSdp.assign(data,len);
+	    SdpAttr attr(_strSdp);
+        _aTrackInfo = attr.getAvailableTrack();
 	};
 	return atoi(_parser["Content-Length"].data());
 }
 
+int RtspSession::handleReq_RECORD(){
+	if (_aTrackInfo.empty() || _parser["Session"] != _strSession) {
+		send_SessionNotFound();
+		return -1;
+	}
+	auto onRes = [this](const string &err){
+		bool authSuccess = err.empty();
+		if(!authSuccess){
+			//第一次play是播放，否则是恢复播放。只对播放鉴权
+			sendRtspResponse("401 Unauthorized", {"Content-Type", "text/plain"}, err);
+			shutdown();
+			return;
+		}
+
+		_StrPrinter rtp_info;
+		for(auto &track : _aTrackInfo){
+			if (track->_inited == false) {
+				//还有track没有setup
+				shutdown();
+				return;
+			}
+			rtp_info << "url=" << _strUrl << "/" << track->_control_surffix << ",";
+		}
+
+		rtp_info.pop_back();
+		sendRtspResponse("200 OK", {"RTP-Info",rtp_info});
+		SockUtil::setNoDelay(_pSender->rawFD(),false);
+	};
+
+	weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
+	Broadcast::AuthInvoker invoker = [weakSelf,onRes](const string &err){
+		auto strongSelf = weakSelf.lock();
+		if(!strongSelf){
+			return;
+		}
+		strongSelf->async([weakSelf,onRes,err](){
+			auto strongSelf = weakSelf.lock();
+			if(!strongSelf){
+				return;
+			}
+			onRes(err);
+		});
+	};
+
+	//rtsp推流需要鉴权
+	auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish,_mediaInfo,invoker,*this);
+	if(!flag){
+		//该事件无人监听,默认不鉴权
+		onRes("");
+	}
+	return 0;
+}
+
+
 int RtspSession::handleReq_Describe() {
     {
         //解析url获取媒体名称
@@ -510,21 +569,22 @@ int RtspSession::handleReq_Setup() {
 		break;
 	case PlayerBase::RTP_UDP: {
 		//我们用trackIdx区分rtp和rtcp包
-		auto pSockRtp = UDPServer::Instance().getSock(get_local_ip().data(),2*trackIdx);
+		auto pSockRtp = std::make_shared<Socket>(_sock->getPoller());
-		if (!pSockRtp) {
+		if (!pSockRtp->bindUdpSock(0,get_local_ip().data())) {
 			//分配端口失败
 			WarnL << "分配rtp端口失败";
 			send_NotAcceptable();
 			return -1;
 		}
-		auto pSockRtcp = UDPServer::Instance().getSock(get_local_ip().data(),2*trackIdx + 1 ,pSockRtp->get_local_port() + 1);
+		auto pSockRtcp = std::make_shared<Socket>(_sock->getPoller());
-		if (!pSockRtcp) {
+		if (!pSockRtcp->bindUdpSock(pSockRtp->get_local_port() + 1,get_local_ip().data())) {
 			//分配端口失败
 			WarnL << "分配rtcp端口失败";
 			send_NotAcceptable();
 			return -1;
 		}
-		_apUdpSock[trackIdx] = pSockRtp;
+		_apRtpSock[trackIdx] = pSockRtp;
+		_apRtcpSock[trackIdx] = pSockRtcp;
 		//设置客户端内网端口信息
 		string strClientPort = FindField(_parser["Transport"].data(), "client_port=", NULL);
 		uint16_t ui16PeerPort = atoi( FindField(strClientPort.data(), NULL, "-").data());
@@ -533,9 +593,9 @@ int RtspSession::handleReq_Setup() {
 		peerAddr.sin_port = htons(ui16PeerPort);
 		peerAddr.sin_addr.s_addr = inet_addr(get_peer_ip().data());
 		bzero(&(peerAddr.sin_zero), sizeof peerAddr.sin_zero);
-		_apPeerUdpAddr[trackIdx].reset((struct sockaddr *) (new struct sockaddr_in(peerAddr)));
+		_apPeerRtpPortAddr[trackIdx].reset((struct sockaddr *) (new struct sockaddr_in(peerAddr)));
 		//尝试获取客户端nat映射地址
-		startListenPeerUdpData();
+		startListenPeerUdpData(trackIdx);
 		//InfoL << "分配端口:" << srv_port;
 
 		sendRtspResponse("200 OK",
@@ -564,6 +624,7 @@ int RtspSession::handleReq_Setup() {
 		}
 		int iSrvPort = _pBrdcaster->getPort(trackRef->_type);
 		//我们用trackIdx区分rtp和rtcp包
+		//由于组播udp端口是共享的，而rtcp端口为组播udp端口+1，所以rtcp端口需要改成共享端口
 		auto pSockRtcp = UDPServer::Instance().getSock(get_local_ip().data(),2*trackIdx + 1,iSrvPort + 1);
 		if (!pSockRtcp) {
 			//分配端口失败
@@ -571,7 +632,7 @@ int RtspSession::handleReq_Setup() {
 			send_NotAcceptable();
 			return -1;
 		}
-		startListenPeerUdpData();
+		startListenPeerUdpData(trackIdx);
         GET_CONFIG_AND_REGISTER(uint32_t,udpTTL,MultiCast::kUdpTTL);
 
 		sendRtspResponse("200 OK",
@@ -914,12 +975,12 @@ inline void RtspSession::sendRtpPacket(const RtpPacket::Ptr & pkt) {
 		break;
 	case PlayerBase::RTP_UDP: {
 		int iTrackIndex = getTrackIndexByTrackType(pkt->type);
-		auto pSock = _apUdpSock[iTrackIndex].lock();
+		auto &pSock = _apRtpSock[iTrackIndex];
 		if (!pSock) {
 			shutdown();
 			return;
 		}
-		auto peerAddr = _apPeerUdpAddr[iTrackIndex];
+		auto &peerAddr = _apPeerRtpPortAddr[iTrackIndex];
 		if (!peerAddr) {
 			return;
 		}
@@ -934,7 +995,11 @@ inline void RtspSession::sendRtpPacket(const RtpPacket::Ptr & pkt) {
 }
 
 inline void RtspSession::onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr& addr) {
+	//这是rtcp心跳包，说明播放器还存活
+	_ticker.resetTime();
+
 	if(iTrackIdx % 2 == 0){
+//		DebugL << "rtp数据包:" << iTrackIdx / 2;
 		//这是rtp探测包
 		if(!_bGotAllPeerUdp){
 			//还没有获取完整的rtp探测包
@@ -944,7 +1009,7 @@ inline void RtspSession::onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf
 				return;
 			}
 			//设置真实的客户端nat映射端口号
-			_apPeerUdpAddr[iTrackIdx / 2].reset(new struct sockaddr(addr));
+			_apPeerRtpPortAddr[iTrackIdx / 2].reset(new struct sockaddr(addr));
 			_abGotPeerUdp[iTrackIdx / 2] = true;
 			_bGotAllPeerUdp = true;//先假设获取到完整的rtp探测包
 			for (unsigned int i = 0; i < _aTrackInfo.size(); i++) {
@@ -956,18 +1021,15 @@ inline void RtspSession::onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf
 			}
 		}
 	}else{
-		//这是rtcp心跳包，说明播放器还存活
+//		TraceL << "rtcp数据包" << (iTrackIdx-1)/2 ;
-		_ticker.resetTime();
-		//TraceL << "rtcp:" << (iTrackIdx-1)/2 ;
 	}
 }
 
 
-inline void RtspSession::startListenPeerUdpData() {
+inline void RtspSession::startListenPeerUdpData(int trackIdx) {
-	_bListenPeerUdpData = true;
 	weak_ptr<RtspSession> weakSelf = dynamic_pointer_cast<RtspSession>(shared_from_this());
-	UDPServer::Instance().listenPeer(get_peer_ip().data(), this,
+
-			[weakSelf](int iTrackIdx,const Buffer::Ptr &pBuf,struct sockaddr *pPeerAddr)->bool {
+	auto onUdpData = [weakSelf](const Buffer::Ptr &pBuf, struct sockaddr *pPeerAddr,int iTrackIdx){
 		auto strongSelf=weakSelf.lock();
 		if(!strongSelf) {
 			return false;
@@ -981,7 +1043,36 @@ inline void RtspSession::startListenPeerUdpData() {
 			strongSelf->onRcvPeerUdpData(iTrackIdx,pBuf,addr);
 		});
 		return true;
+	};
+
+	switch (_rtpType){
+		case PlayerBase::RTP_MULTICAST:{
+			//组播使用的共享rtcp端口
+			UDPServer::Instance().listenPeer(get_peer_ip().data(), this, [onUdpData](
+					int iTrackIdx, const Buffer::Ptr &pBuf, struct sockaddr *pPeerAddr) {
+				return onUdpData(pBuf,pPeerAddr,iTrackIdx);
 			});
+		}
+			break;
+		case PlayerBase::RTP_UDP:{
+			auto setEvent = [&](Socket::Ptr &sock,int iTrackIdx){
+				if(!sock){
+					WarnL << "udp端口为空:" << iTrackIdx;
+					return;
+				}
+				sock->setOnRead([onUdpData,iTrackIdx](const Buffer::Ptr &pBuf, struct sockaddr *pPeerAddr){
+					onUdpData(pBuf,pPeerAddr,iTrackIdx);
+				});
+			};
+			setEvent(_apRtpSock[trackIdx], 2*trackIdx );
+			setEvent(_apRtcpSock[trackIdx], 2*trackIdx + 1 );
+		}
+			break;
+
+		default:
+			break;
+	}
+
 }
 
 inline void RtspSession::initSender(const std::shared_ptr<RtspSession>& session) {

src/Rtsp/RtspSession.h

@@ -88,6 +88,7 @@ private:
 	int handleReq_Options(); //处理options方法
 	int handleReq_Describe(); //处理describe方法
 	int handleReq_ANNOUNCE(); //处理options方法
+    int handleReq_RECORD(); //处理options方法
 	int handleReq_Setup(); //处理setup方法
 	int handleReq_Play(); //处理play方法
 	int handleReq_Pause(); //处理pause方法
@@ -108,7 +109,7 @@ private:
     inline int getTrackIndexByControlSuffix(const string &controlSuffix);
 
 	inline void onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
-	inline void startListenPeerUdpData();
+	inline void startListenPeerUdpData(int iTrackIdx);
 
     //认证相关
     static void onAuthSuccess(const weak_ptr<RtspSession> &weakSelf);
@@ -142,9 +143,9 @@ private:
 	//RTP over udp
 	bool _bGotAllPeerUdp = false;
 	bool _abGotPeerUdp[2] = { false, false }; //获取客户端udp端口计数
-	weak_ptr<Socket> _apUdpSock[2]; //发送RTP的UDP端口,trackid idx 为数组下标
+	Socket::Ptr _apRtpSock[2]; //RTP端口,trackid idx 为数组下标
-	std::shared_ptr<struct sockaddr> _apPeerUdpAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标
+	Socket::Ptr _apRtcpSock[2];//RTCP端口,trackid idx 为数组下标
-	bool _bListenPeerUdpData = false;
+	std::shared_ptr<struct sockaddr> _apPeerRtpPortAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标
 
 	//RTP over udp_multicast
 	RtpBroadCaster::Ptr _pBrdcaster;