RTSP Server会不断用select查询是否有socket连接，如果有则在(*handler->handlerProc)(handler->clientData, resultConditionSet) 进行处理。
在RTSPServer::RTSPClientConnection::incomingRequestHandler1()函数中，从socket读取客户端请求信息，然后解析RTSP命令，在变量fRequestBuffer中保存了RTSP请求信息:
例如：
OPTIONS rtsp://192.168.2.100:8554/3.ts RTSP/1.0
CSeq: 2
User-Agent: LibVLC/2.0.7 (LIVE555 Streaming Media v2012.12.18)

void RTSPServer::RTSPClientConnection::incomingRequestHandler1() {
  struct sockaddr_in dummy;
  // 'from' address, meaningless in this case  
  int bytesRead = readSocket(envir(), fClientInputSocket, &fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft, dummy);
handleRequestBytes(bytesRead);
}

Live555首先预解析请求字符串，判断请求是否符合标准。目前支持解析以下几种命令：
OPTIONS
GET_PARAMETER
SET_PARAMETER
DESCRIBE
SETUP
TEARDOWN
PLAY
PAUSE

然后发送响应给客户端:
例如：

RTSP/1.0 200 OK
CSeq: 2
Date: Wed, Jun 19 2013 16:05:27 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, GET_PARAMETER, SET_PARAMETER

如果使用PLAY请求命令，Live555将调用
void RTSPServer::RTSPClientSession
::handleCmd_PLAY(
  RTSPServer::RTSPClientConnection* ourClientConnection,
  ServerMediaSubsession*  subsession,
  char const* fullRequestStr)函数进行处理。

请求数据如下:

PLAY rtsp://192.168.2.100:8554/3.ts/ RTSP/1.0
CSeq: 5
User-Agent: LibVLC/2.0.7 (LIVE555 Streaming Media v2012.12.18)
Session: E0406EAB
Range: npt=0.000-

首先定位流的位置，因为调试时使用的是ts流，
void MPEG2TransportFileServerMediaSubsession
::seekStream(
  unsigned clientSessionId,
  void* streamToken,
  double& seekNPT,
  double streamDuration,
  u_int64_t& numBytes);

之后就是开始播放流，

fStreamStates[i].subsession->startStream(fOurSessionId,
   fStreamStates[i].streamToken,    
   (TaskFunc*)noteClientLiveness, this,
   rtpSeqNum, rtpTimestamp,
    RTSPServer::RTSPClientConnection::handleAlternativeRequestByte,
    ourClientConnection);

该函数内会调用void OnDemandServerMediaSubsession::startStream()，继而调用StreamState::startPlaying()
void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,
   void* streamToken,
    TaskFunc* rtcpRRHandler,
    void* rtcpRRHandlerClientData,
        unsigned short& rtpSeqNum,
    unsigned& rtpTimestamp,
    ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
    void* serverRequestAlternativeByteHandlerClientData)；

在StreamState::startPlaying()中会创建一个RTCPInstance实例，并把客户端地址和端口添加到RTP和RTCPgroupsocks目的地中。再调用RTPSink::startPlaying()开始推送数据。