live555源码分析----RTP的打包与发送

    这里主要分析一下，live555中关于RTP打包发送的部分。在处理完PLAY命令之后，就开始发送RTP数据包了(其实在发送PLAY命令的response包之前，就会发送一个RTP包，这里传输就已经开始了)
    RTP包的发送是从MediaSink::startPlaying函数调用开始的

Boolean MediaSink::startPlaying(MediaSource& source,
				afterPlayingFunc* afterFunc,
				void* afterClientData) {
  // Make sure we're not already being played:
  if (fSource != NULL) {
    envir().setResultMsg("This sink is already being played");
    return False;
  }


  // Make sure our source is compatible:
  if (!sourceIsCompatibleWithUs(source)) {
    envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");
    return False;
  }
  fSource = (FramedSource*)&source;


  fAfterFunc = afterFunc;
  fAfterClientData = afterClientData;
  return continuePlaying();     //重要的函数在这里
}

    这个函数只有最后一句最重要，即continuePlaying函数的调用。continuePlaying函数是定义在MediaSink类中的纯虚函数，需要到特定媒体的sink子类中实现，对于H264来讲是在H264VideoRTPSink中实现的。
    H264VideoRTPSink继承关系：H264VideoRTPSink->VideoRTPSink->MultiFramedRTPSink->RTPSink->MediaSink。

Boolean H264VideoRTPSink::continuePlaying() {
  // First, check whether we have a 'fragmenter' class set up yet.
  // If not, create it now:
  if (fOurFragmenter == NULL) {
    //创建一个辅助类H264FUAFragmenter，用于H264的RTP打包


    fOurFragmenter = new H264FUAFragmenter(envir(), fSource, OutPacketBuffer::maxSize,
					   ourMaxPacketSize() - 12/*RTP hdr size*/);
    fSource = fOurFragmenter;
  }


  // Then call the parent class's implementation:
  return MultiFramedRTPSink::continuePlaying();
}

    上面的代码中创建了一个辅助类H264FUAFragmenter，因为H264的RTP包，有些特殊需要进一步处理，可以参考RFC3986。接着调用MultiFramedRTPSink类的continuePlaying实现

Boolean MultiFramedRTPSink::continuePlaying() {
  // Send the first packet.
  // (This will also schedule any future sends.)
  buildAndSendPacket(True);
  return True;
}
    这时调用buildAndSendPacket函数时，看名字就知道其中将完成打包并发送工作。传递了一个True参数，表示这是第一个packet。继续看buildAndSendPacket函数定义
void MultiFramedRTPSink::buildAndSendPacket(Boolean isFirstPacket) {
  fIsFirstPacket = isFirstPacket;
    //
    //设置RTP头,注意，接收端需要根据RTP包的序号fSeqNo来重新排序
    //
  // Set up the RTP header:
  unsigned rtpHdr = 0x80000000; // RTP version 2; marker ('M') bit not set (by default; it can be set later)
  rtpHdr |= (fRTPPayloadType<<16);
  rtpHdr |= fSeqNo; // sequence number
  fOutBuf->enqueueWord(rtpHdr);


  //保留一个4 bytes空间，用于设置time stamp
  // Note where the RTP timestamp will go.
  // (We can't fill this in until we start packing payload frames.)
  fTimestampPosition = fOutBuf->curPacketSize();
  fOutBuf->skipBytes(4); // leave a hole for the timestamp


  fOutBuf->enqueueWord(SSRC());     //跟RTCP相关，作用暂不清楚
    
    //在RTP头后面，添加一个payload-format-specific头，
// Allow for a special, payload-format-specific header following the
  // RTP header:
  fSpecialHeaderPosition = fOutBuf->curPacketSize();
    //
    //specialHeaderSize在MultiFramedRTPSink中的默认实现返回0，对于H264的实现不需要处理这个字段
    //
  fSpecialHeaderSize = specialHeaderSize();
  fOutBuf->skipBytes(fSpecialHeaderSize);   //预留空间


   //填充尽可能多的frames到packet中
  // Begin packing as many (complete) frames into the packet as we can:
  fTotalFrameSpecificHeaderSizes = 0;
  fNoFramesLeft = False;
  fNumFramesUsedSoFar = 0;
  packFrame();
}

    buildAndSendPacket函数中，完成RTP头的准备工作。可以看到RTP头是非常简单的，RTP头中的序号非常重要，客户端需要据此进行RTP包的重排序操作。RTP包内容存放在一个OutPacketBuffer类型的fOutBuf成员变量中，OutPacketBuffer类的细节在文章的最后还会讨论。在RTP头中预留了一些空间没有进行实际