Question

我想要从文件中进行RTSP流式传输，所以我使用实时555库。在实时555测试目录“testMpeg2TransportStreamer”程序中流式传输TS格式文件，我做的是，我把整个代码放在线程中所以每当客户端请求流线程开始工作时，当客户端说DONT STREAM然后线程关闭线程我也写了代码Medium :: close（rtsp服务器的指针），所以要关闭rtsp服务器，这是有效的对于第一个STREAM和DONT STREAM请求很好，但是当客户端对STREAM说，我调试代码并发现它无法创建rtsp服务器时，在DONT STREAM请求之后。然后我使用另一种方法，在第一次STREAM请求时只创建一次rtsp服务器，而不是关闭它直到我的整个程序退出但这也失败了。任何人都可以建议任何其他方式我的代码如下： -

#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7

#define IMPLEMENT_RTSP_SERVER 
// To stream using "source-specific multicast" (SSM), uncomment the following:
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif

/********************Global variable***************************/
UsageEnvironment* env=NULL;
FramedSource* videoSource;
RTPSink* videoSink;
DeviceSourceFICard* fileSource;
FICardDeviceParameters fi_params;
HANDLE          g_hRtpComThread;
DWORD           g_dwRtpComThreadID;
char            g_ExitEventLoop;
void play(); // forward
RTSPServer* rtspServer=NULL;
ServerMediaSession* sms;
int initLm555Settings(void)
{
scheduler = BasicTaskScheduler::createNew();
  env = BasicUsageEnvironment::createNew(*scheduler);
 destinationAddressStr//make it global
#ifdef USE_SSM
    = "232.255.42.42";
#else
    = "239.255.42.42";
const unsigned short rtpPortNum = 18888;//make it global
  rtpPortNum1=rtpPortNum;
 const unsigned short rtcpPortNum = rtpPortNum+1;
  const unsigned char ttl = 7;
 struct in_addr destinationAddress;
  destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
  const Port rtpPort(rtpPortNum);
  const Port rtcpPort(rtcpPortNum);

  Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
  rtpGroupsock.multicastSendOnly();

  Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
  rtcpGroupsock.multicastSendOnly();
#ifdef USE_SSM
  rtpGroupsock.multicastSendOnly();
  rtcpGroupsock.multicastSendOnly();
#endif
 g_ExitEventLoop = 0;
videoSink =
    SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video", "MP2T",
                 1, True, False /*no 'M' bit*/);
setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);
  setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);

  // Create (and start) a 'RTCP instance' for this RTP sink:
  const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w share
  const unsigned maxCNAMElen = 100;
  unsigned char CNAME[maxCNAMElen+1];
  gethostname((char*)CNAME, maxCNAMElen);
  CNAME[maxCNAMElen] = '\0'; // just in case

  RTCPInstance* rtcp =
    RTCPInstance::createNew(*env, &rtcpGroupsock,
                estimatedSessionBandwidth, CNAME,
                videoSink, NULL /* we're a server */, isSSM);
 UserAuthenticationDatabase* authDB = NULL; 
      portNumBits rtspServerPortNum = 554;
      unsigned reclamationTestSeconds=65U;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum, authDB, reclamationTestSeconds); 
    if (rtspServer == NULL)
    {
        *env << "Failed to create RTSP server: " <<env->getResultMsg()<<"\n";

        rtspServerPortNum = 8554;
        rtspServer = RTSPServer::createNew(*env,rtspServerPortNum);
        if (rtspServer == NULL)
        {
return 0;
        }
Boolean const inputStreamIsRawUDP = False; 
char const* descriptionString={"Session streamed by \"testOnDemandRT\""};
sms=ServerMediaSession::createNew(*env, streamName, streamName,descriptionString);
sms->addSubsession(MPEG2TransportUDPServerMediaSubsession::createNew(*env,destinationAddressStr,rtpPortNum1,inputStreamIsRawUDP));
    rtspServer->addServerMediaSession(sms);

    char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
    delete[] url;
if (rtspServer->setUpTunnelingOverHTTP(sport) || rtspServer->setUpTunnelingOverHTTP(sport) || rtspServer->setUpTunnelingOverHTTP(sport))
        {
out<<"\n\n\n(We use port "<<rtspServer->httpServerPortNum()<<" for optional RTSP-over-HTTP tunneling.)\n";

        } 
        else
        {
             pDailyLogger->LogInfoString("(RTSP-over-HTTP tunneling is not available.)");
            cout<<"\n\n\n(RTSP-over-HTTP tunneling is not available.)";
        }
  play();
  env->taskScheduler().doEventLoop(&g_ExitEventLoop);
Medium::close(rtspServer);
 Medium::close(rtcp);
  Medium::close(videoSink);
 rtpGroupsock.removeAllDestinations();
  rtcpGroupsock.removeAllDestinations();

  env->reclaim();


  delete scheduler;



    pDailyLogger->LogDebugString("OUT::initLm555Settings Thread");

    return 0; // only to prevent compiler warning
}
void afterPlaying(void* /*clientData*/) {
  *env << "...done reading from file\n";

  videoSink->stopPlaying();
 Medium::close(videoSource);
  play();
}
void play() {
  // Open the input file as a 'byte-stream file source':

  fi_params.nFICardFrameSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET * TRANSPORT_PACKET_SIZE;
  fi_params.pfnGetRTPPayload = GetRTPPayload;
  fi_params.socketNum = videoSink->groupsockBeingUsed().socketNum();

  DeviceParameters temp;

    fileSource = DeviceSourceFICard::createNew(*env, fi_params, temp);
  if (fileSource == NULL) {
    *env << "Unable to open Foresight card as a byte-stream file source\n";
    exit(1);
  }
  FramedSource* videoES = fileSource;
 videoSource = MPEG1or2VideoStreamDiscreteFramer::createNew(*env, videoES);
*env << "Beginning to read from file...\n";
  videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
void StartRTPProcess(void)
{

    g_hRtpComThread = CreateThread((LPSECURITY_ATTRIBUTES) NULL, 0,
        (LPTHREAD_START_ROUTINE)initLm555Settings, 0, 0, &g_dwRtpComThreadID);

    if(g_hRtpComThread) SetThreadPriority(g_hRtpComThread, THREAD_PRIORITY_LOWEST/*THREAD_PRIORITY_NORMAL*/);

}
int StopRTProcess(void) 
{
  g_ExitEventLoop = 1;

     g_ExitEventLoop = 0;
  g_hRtpComThread = 0;
  g_dwRtpComThreadID = 0;
}

当我收到DONT STREAM消息时，调用StopRTProcess（）Sir，我缺少什么请告诉

Answer 1

可能您的RTSP服务器套接字未关闭。关闭媒体后删除媒体对象和RTSPServer对象。

如何一次又一次地创建和销毁rtsp服务器Live 555

1 个答案: