live555做流媒体服务器时解决rtp over udp模式下, 客户端没有发送teardown时直接关闭导致RTSPServer端没有即时关闭流的问题解决方法

char fClientSessionIdStr[16];	//for rtp over udp
		
		GenericMediaServer.hh  增加createNewClientSessionWithId的参数char *pSessionIdStr
		ClientSession* createNewClientSessionWithId(UsageEnvironment	*pEnv, char *pSessionIdStr);
		
		GenericMediaServer::ClientSession* GenericMediaServer::createNewClientSessionWithId(UsageEnvironment	*_pEnv, char *pSessionIdStr) {
		  u_int32_t sessionId;
		  char sessionIdStr[16] = {0};

		  // Choose a random (unused) 32-bit integer for the session id
		  // (it will be encoded as a 8-digit hex number).  (We avoid choosing session id 0,
		  // because that has a special use by some servers.)
		  do {
		    sessionId = (u_int32_t)our_random32();
		    snprintf(sessionIdStr, sizeof sessionIdStr, "%08X", sessionId);
		  } while (sessionId == 0 || lookupClientSession(sessionIdStr) != NULL);

		  ClientSession* clientSession = createNewClientSession(sessionId, _pEnv);
		  if (clientSession != NULL) fClientSessions->Add(sessionIdStr, clientSession);

		  if (NULL != pSessionIdStr)	strcpy(pSessionIdStr, sessionIdStr);		//此处返回生成的sessionId, 后续要根据该值找到对应的ClientSession

		  return clientSession;
		}		
		
		
		
		void RTSPServer::RTSPClientConnection::handleRequestBytes(int newBytesRead, UsageEnvironment *pEnv) {

			//找到以下代码		
		  if (authenticationOK("SETUP", urlTotalSuffix, (char const*)fRequestBuffer)) {
			  memset(fClientSessionIdStr, 0x00, sizeof(fClientSessionIdStr));
		    clientSession
		      = (RTSPServer::RTSPClientSession*)fOurRTSPServer.createNewClientSessionWithId(pEnv, fClientSessionIdStr);		//此处记录ClientSession的sessionId
		  }		
		
		}

void RTSPServer::stopTCPStreamingOnSocket(UsageEnvironment *pEnv, int socketNum, int *clientTrackNum, char *clientSessionIdStr){
		  // Close any stream that is streaming over "socketNum" (using RTP/RTCP-over-TCP streaming):

			RTSPClientSession	*pClientSession = NULL;

			LockClientConnection();
			do
			{
			  streamingOverTCPRecord* sotcp
				= (streamingOverTCPRecord*)fTCPStreamingDatabase->Lookup((char const*)socketNum);
			  if (sotcp != NULL) {		//rtp over tcp
				do {
				  RTSPClientSession* clientSession
				= (RTSPServer::RTSPClientSession*)lookupClientSession(sotcp->fSessionId);
				  if (clientSession != NULL) {
					  //clientSession->SetAssignSink(assignSink);
				clientSession->deleteStreamByTrack(pEnv, sotcp->fTrackNum, False, clientTrackNum);
				  }

				  streamingOverTCPRecord* sotcpNext = sotcp->fNext;
				  sotcp->fNext = NULL;
				  delete sotcp;
				  sotcp = sotcpNext;
				} while (sotcp != NULL);
				fTCPStreamingDatabase->Remove((char const*)socketNum);
			  }
			  else if ( (clientTrackNum) && (*clientTrackNum==0))		//rtp over udp
			  {
				  pClientSession = (RTSPServer::RTSPClientSession*)lookupClientSession(clientSessionIdStr);
			  }
			}while (0);

			UnlockClientConnection();

			if (pClientSession != NULL) 		//pClientSession不为空, 说明为rtp over udp
			{
				delete pClientSession;
			}

		}