live555的RTCP

live555中默认支持RTCP，如果要监视网络状态就要了解RTCP。我们这里以openRTSP为例看看RTCP的过程。

在前面的openRTSP分析中分析了openRTSP的流程，其中在在continueAfterDESCRIBE中有subsession->initiate(simpleRTPoffsetArg)，在这里进行了RTP socket和RTCP socket的建立。

if (isSSM()) {
	  fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
	} else {
	  fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
	}

这里建立了RTCP 的socket，下面初始化fRTCPInstance

if (fRTPSource != NULL && fRTCPSocket != NULL) {// Finally, create our RTCP instance. (It starts running automatically)
      // If bandwidth is specified, use it and add 5% for RTCP overhead.
      // Otherwise make a guess at 500 kbps.
      unsigned totSessionBandwidth
	= fBandwidth ? fBandwidth + fBandwidth / 20 : 500;
      fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
					      totSessionBandwidth,
					      (unsigned char const*)
					      fParent.CNAME(),
					      NULL /* we're a client */,
					      fRTPSource);
      if (fRTCPInstance == NULL) {
	env().setResultMsg("Failed to create RTCP instance");
	break;
      }
    }

根据注释提示，fRTCPInstance会自动开始运行，如何运行的呢，我们再来看看。

RTCPInstance* RTCPInstance::createNew(UsageEnvironment& env, Groupsock* RTCPgs,
				      unsigned totSessionBW,
				      unsigned char const* cname,
				      RTPSink* sink, RTPSource const* source,
				      Boolean isSSMSource) {
  return new RTCPInstance(env, RTCPgs, totSessionBW, cname, sink, source,
			  isSSMSource);
}

为了过程的清晰，这里还是贴出所有代码

RTCPInstance::RTCPInstance(UsageEnvironment& env, Groupsock* RTCPgs,
			   unsigned totSessionBW,
			   unsigned char const* cname,
			   RTPSink* sink, RTPSource const* source,
			   Boolean isSSMSource)
  : Medium(env), fRTCPInterface(this, RTCPgs), fTotSessionBW(totSessionBW),
    fSink(sink), fSource(source), fIsSSMSource(isSSMSource),
    fCNAME(RTCP_SDES_CNAME, cname), fOutgoingReportCount(1),
    fAveRTCPSize(0), fIsInitial(1), fPrevNumMembers(0),
    fLastSentSize(0), fLastReceivedSize(0), fLastReceivedSSRC(0),
    fTypeOfEvent(EVENT_UNKNOWN), fTypeOfPacket(PACKET_UNKNOWN_TYPE),
    fHaveJustSentPacket(False), fLastPacketSentSize(0),
    fByeHandlerTask(NULL), fByeHandlerClientData(NULL),
    fSRHandlerTask(NULL), fSRHandlerClientData(NULL),
    fRRHandlerTask(NULL), fRRHandlerClientData(NULL),
    fSpecificRRHandlerTable(NULL) {
#ifdef DEBUG
  fprintf(stderr, "RTCPInstance[%p]::RTCPInstance()\n", this);
#endif
  if (fTotSessionBW == 0) { // not allowed!
    env << "RTCPInstance::RTCPInstance error: totSessionBW parameter should not be zero!\n";
    fTotSessionBW = 1;
  }

  if (isSSMSource) RTCPgs->multicastSendOnly(); // don't receive multicast

  double timeNow = dTimeNow();
  fPrevReportTime = fNextReportTime = timeNow;

  fKnownMembers = new RTCPMemberDatabase(*this);
  fInBuf = new unsigned char[maxPacketSize];
  if (fKnownMembers == NULL || fInBuf == NULL) return;
  fNumBytesAlreadyRead = 0;

  // A hack to save buffer space, because RTCP packets are always small:
  unsigned savedMaxSize = OutPacketBuffer::maxSize;
  OutPacketBuffer::maxSize = maxPacketSize;
  fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);
  OutPacketBuffer::maxSize = savedMaxSize;
  if (fOutBuf == NULL) return;

  // Arrange to handle incoming reports from others:
  TaskScheduler::BackgroundHandlerProc* handler
    = (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler;
  fRTCPInterface.startNetworkReading(handler);

  // Send our first report.
  fTypeOfEvent = EVENT_REPORT;
  onExpire(this);
}

可以看到这里加入了handler，handler为incomingReportHandler。看看这个startNetworkReading。

void RTPInterface
::startNetworkReading(TaskScheduler::BackgroundHandlerProc* handlerProc) {
  // Normal case: Arrange to read UDP packets:
  envir().taskScheduler().
    turnOnBackgroundReadHandling(fGS->socketNum(), handlerProc, fOwner);

  // Also, receive RTP over TCP, on each of our TCP connections:
  fReadHandlerProc = handlerProc;
  for (tcpStreamRecord* streams = fTCPStreams; streams != NULL;
       streams = streams->fNext) {
    // Get a socket descriptor for "streams->fStreamSocketNum":
    SocketDescriptor* socketDescriptor = lookupSocketDescriptor(envir(), streams->fStreamSocketNum);

    // Tell it about our subChannel:
    socketDescriptor->registerRTPInterface(streams->fStreamChannelId, this);
  }
}

第一句代码表示默认是通过udp来发送rtcp的包，如果rtp是tcp的就通过下面的代码处理。

这里的turnOnBackgroundReadHanding就是调用setBackgroundHandling(socketNum, SOCKET_READABLE, handlerProc, clientData);在前面分析过就是将socket加入到select set集中进行监视并传入handlerProc。这里不做重复讨论，接着看下面的注释是send our first report，接着调用了onExpire(this)，来看看

void RTCPInstance::onExpire(RTCPInstance* instance) {
  instance->onExpire1();
}

void RTCPInstance::onExpire1() {
  // Note: fTotSessionBW is kbits per second
  double rtcpBW = 0.05*fTotSessionBW*1024/8; // -> bytes per second

  OnExpire(this, // event
	   numMembers(), // members
	   (fSink != NULL) ? 1 : 0, // senders
	   rtcpBW, // rtcp_bw
	   (fSink != NULL) ? 1 : 0, // we_sent
	   &fAveRTCPSize, // ave_rtcp_size
	   &fIsInitial, // initial
	   dTimeNow(), // tc
	   &fPrevReportTime, // tp
	   &fPrevNumMembers // pmembers
	   );
}

再来看看这个OnExpire

void OnExpire(event e,
                 int    members,
                 int    senders,
                 double rtcp_bw,
                 int    we_sent,
                 double *avg_rtcp_size,
                 int    *initial,
                 time_tp   tc,
                 time_tp   *tp,
                 int    *pmembers)
   {
       /* This function is responsible for deciding whether to send
        * an RTCP report or BYE packet now, or to reschedule transmission.
        * It is also responsible for updating the pmembers, initial, tp,
        * and avg_rtcp_size state variables. This function should be called
        * upon expiration of the event timer used by Schedule(). */

       double t;     /* Interval */
       double tn;    /* Next transmit time */

       /* In the case of a BYE, we use "unconditional reconsideration" to
        * reschedule the transmission of the BYE if necessary */

       if (TypeOfEvent(e) == EVENT_BYE) {
           t = rtcp_interval(members,
                             senders,
                             rtcp_bw,
                             we_sent,
                             *avg_rtcp_size,
                             *initial);
           tn = *tp + t;
           if (tn <= tc) {
               SendBYEPacket(e);
               exit(1);
           } else {
               Schedule(tn, e);
           }

       } else if (TypeOfEvent(e) == EVENT_REPORT) {
           t = rtcp_interval(members,
                             senders,
                             rtcp_bw,
                             we_sent,
                             *avg_rtcp_size,
                             *initial);
           tn = *tp + t;

           if (tn <= tc) {
               SendRTCPReport(e);
               *avg_rtcp_size = (1./16.)*SentPacketSize(e) +
                   (15./16.)*(*avg_rtcp_size);
               *tp = tc;

               /* We must redraw the interval. Don't reuse the
                  one computed above, since its not actually
                  distributed the same, as we are conditioned
                  on it being small enough to cause a packet to
                  be sent */

               t = rtcp_interval(members,
                                 senders,
                                 rtcp_bw,
                                 we_sent,
                                 *avg_rtcp_size,
                                 *initial);

               Schedule(t+tc,e);
               *initial = 0;
           } else {
               Schedule(tn, e);
           }
           *pmembers = members;
       }
   }