live555学习笔记5－RTSP服务运作

五　RTSP服务运作


基础基本搞明白了，那么RTSP,RTP等这些协议又是如何利用这些基础机制运作的呢？
首先来看RTSP.


RTSP首先需建立TCP侦听socket。可见于此函数：

DynamicRTSPServer* DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,UserAuthenticationDatabase* authDatabase,unsigned reclamationTestSeconds) {int ourSocket = setUpOurSocket(env, ourPort); //建立TCP socketif (ourSocket == -1)return NULL;return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase,reclamationTestSeconds);}

要帧听客户端的连接，就需要利用任务调度机制了，所以需添加一个socket handler。可见于此函数：

RTSPServer::RTSPServer(UsageEnvironment& env, int ourSocket, Port ourPort,UserAuthenticationDatabase* authDatabase,unsigned reclamationTestSeconds) :Medium(env), fRTSPServerSocket(ourSocket),fRTSPServerPort(ourPort),fHTTPServerSocket(-1),fHTTPServerPort(0),fClientSessionsForHTTPTunneling(NULL), fAuthDB(authDatabase),fReclamationTestSeconds(reclamationTestSeconds),fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)) {#ifdef USE_SIGNALS// Ignore the SIGPIPE signal, so that clients on the same host that are killed// don't also kill us:signal(SIGPIPE, SIG_IGN);#endif// Arrange to handle connections from others:env.taskScheduler().turnOnBackgroundReadHandling(fRTSPServerSocket,(TaskScheduler::BackgroundHandlerProc*) &incomingConnectionHandlerRTSP,this);}

当收到客户的连接时需保存下代表客户端的新socket，以后用这个socket与这个客户通讯。每个客户将来会对应一个rtp会话，而且各客户的RTSP请求只控制自己的rtp会话，那么最好建立一个会话类，代表各客户的rtsp会话。于是类RTSPServer::RTSPClientSession产生，它保存的代表客户的socket。下为RTSPClientSession的创建过程

void RTSPServer::incomingConnectionHandler(int serverSocket) {struct sockaddr_in clientAddr;SOCKLEN_T clientAddrLen = sizeof clientAddr;//接受连接int clientSocket = accept(serverSocket,(struct sockaddr*) &clientAddr,&clientAddrLen);if (clientSocket < 0) {int err = envir().getErrno();if (err != EWOULDBLOCK) {envir().setResultErrMsg("accept() failed: ");}return;}//设置socket的参数makeSocketNonBlocking(clientSocket);increaseSendBufferTo(envir(), clientSocket, 50 * 1024);#ifdef DEBUGenvir() << "accept()ed connection from " << our_inet_ntoa(clientAddr.sin_addr) << "\n";#endif//产生一个sesson id// Create a new object for this RTSP session.// (Choose a random 32-bit integer for the session id (it will be encoded as a 8-digit hex number).  We don't bother checking for//  a collision; the probability of two concurrent sessions getting the same session id is very low.)// (We do, however, avoid choosing session id 0, because that has a special use (by "OnDemandServerMediaSubsession").)unsigned sessionId;do {sessionId = (unsigned) our_random();} while (sessionId == 0);//创建RTSPClientSession，注意传入的参数(void) createNewClientSession(sessionId, clientSocket, clientAddr);}

 

RTSPClientSession要提供什么功能呢？可以想象：需要监听客户端的rtsp请求并回应它，需要在DESCRIBE请求中返回所请求的流的信息，需要在SETUP请求中建立起RTP会话，需要在TEARDOWN请求中关闭RTP会话，等等...

RTSPClientSession要侦听客户端的请求，就需把自己的socket handler加入计划任务。证据如下：

RTSPServer::RTSPClientSession::RTSPClientSession(RTSPServer& ourServer,unsigned sessionId,int clientSocket,struct sockaddr_in clientAddr) :fOurServer(ourServer),fOurSessionId(sessionId),fOurServerMediaSession(NULL),fClientInputSocket(clientSocket),fClientOutputSocket(clientSocket),fClientAddr(clientAddr),fSessionCookie(NULL),fLivenessCheckTask(NULL),fIsMulticast(False),fSessionIsActive(True),fStreamAfterSETUP(False),fTCPStreamIdCount(0),fNumStreamStates(0),fStreamStates(NULL),fRecursionCount(0){// Arrange to handle incoming requests:resetRequestBuffer();envir().taskScheduler().turnOnBackgroundReadHandling(fClientInputSocket,(TaskScheduler::BackgroundHandlerProc*) &incomingRequestHandler,this);noteLiveness();}

下面重点讲一下下RTSPClientSession响应DESCRIBE请求的过程：

void RTSPServer::RTSPClientSession::handleCmd_DESCRIBE(char const* cseq,char const* urlPreSuffix,char const* urlSuffix,char const* fullRequestStr){char* sdpDescription = NULL;char* rtspURL = NULL;do {//整理一下下RTSP地址char urlTotalSuffix[RTSP_PARAM_STRING_MAX];if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2> sizeof urlTotalSuffix) {handleCmd_bad(cseq);break;}urlTotalSuffix[0] = '\0';if (urlPreSuffix[0] != '\0') {strcat(urlTotalSuffix, urlPreSuffix);strcat(urlTotalSuffix, "/");}strcat(urlTotalSuffix, urlSuffix);//验证帐户和密码if (!authenticationOK("DESCRIBE", cseq, urlTotalSuffix, fullRequestStr))break;// We should really check that the request contains an "Accept:" #####// for "application/sdp", because that's what we're sending back #####// Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix"://跟据流的名字查找ServerMediaSession，如果找不到，会创建一个。每个ServerMediaSession中至少要包含一个//ServerMediaSubsession。一个ServerMediaSession对应一个媒体，可以认为是Server上的一个文件，或一个实时获取设备。其包含的每个ServerMediaSubSession代表媒体中的一个Track。所以一个ServerMediaSession对应一个媒体，如果客户请求的媒体名相同，就使用已存在的ServerMediaSession，如果不同，就创建一个新的。一个流对应一个StreamState，StreamState与ServerMediaSubsession相关，但代表的是动态的，而ServerMediaSubsession代表静态的。ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);if (session == NULL) {handleCmd_notFound(cseq);break;}// Then, assemble a SDP description for this session://获取SDP字符串，在函数内会依次获取每个ServerMediaSubSession的字符串然连接起来。sdpDescription = session->generateSDPDescription();if (sdpDescription == NULL) {// This usually means that a file name that was specified for a// "ServerMediaSubsession" does not exist.snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,"RTSP/1.0 404 File Not Found, Or In Incorrect Format\r\n""CSeq: %s\r\n""%s\r\n", cseq, dateHeader());break;}unsigned sdpDescriptionSize = strlen(sdpDescription);// Also, generate our RTSP URL, for the "Content-Base:" header// (which is necessary to ensure that the correct URL gets used in// subsequent "SETUP" requests).rtspURL = fOurServer.rtspURL(session, fClientInputSocket);//形成响应DESCRIBE请求的RTSP字符串。snprintf((char*) fResponseBuffer, sizeof fResponseBuffer,"RTSP/1.0 200 OK\r\nCSeq: %s\r\n""%s""Content-Base: %s/\r\n""Content-Type: application/sdp\r\n""Content-Length: %d\r\n\r\n""%s", cseq, dateHeader(), rtspURL, sdpDescriptionSize,sdpDescription);} while (0);delete[] sdpDescription;delete[] rtspURL;//返回后会被立即发送（没有把socket write操作放入计划任务中）。}

fOurServer.lookupServerMediaSession(urlTotalSuffix)中会在找不到同名ServerMediaSession时新建一个，代表一个RTP流的ServerMediaSession们是被RTSPServer管理的，而不是被RTSPClientSession拥有。为什么呢？因为ServerMediaSession代表的是一个静态的流，也就是可以从它里面获取一个流的各种信息，但不能获取传输状态。不同客户可能连接到同一个流，所以ServerMediaSession应被RTSPServer所拥有。创建一个ServerMediaSession过程值得一观：

static ServerMediaSession* createNewSMS(UsageEnvironment& env,char const* fileName, FILE* /*fid*/) {// Use the file name extension to determine the type of "ServerMediaSession":char const* extension = strrchr(fileName, '.');if (extension == NULL)return NULL;ServerMediaSession* sms = NULL;Boolean const reuseSource = False;if (strcmp(extension, ".aac") == 0) {// Assumed to be an AAC Audio (ADTS format) file:NEW_SMS("AAC Audio");sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".amr") == 0) {// Assumed to be an AMR Audio file:NEW_SMS("AMR Audio");sms->addSubsession(AMRAudioFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".ac3") == 0) {// Assumed to be an AC-3 Audio file:NEW_SMS("AC-3 Audio");sms->addSubsession(AC3AudioFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".m4e") == 0) {// Assumed to be a MPEG-4 Video Elementary Stream file:NEW_SMS("MPEG-4 Video");sms->addSubsession(MPEG4VideoFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".264") == 0) {// Assumed to be a H.264 Video Elementary Stream file:NEW_SMS("H.264 Video");OutPacketBuffer::maxSize = 100000; // allow for some possibly large H.264 framessms->addSubsession(H264VideoFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".mp3") == 0) {// Assumed to be a MPEG-1 or 2 Audio file:NEW_SMS("MPEG-1 or 2 Audio");// To stream using 'ADUs' rather than raw MP3 frames, uncomment the following://#define STREAM_USING_ADUS 1// To also reorder ADUs before streaming, uncomment the following://#define INTERLEAVE_ADUS 1// (For more information about ADUs and interleaving,//  see <http://www.live555.com/rtp-mp3/>)Boolean useADUs = False;Interleaving* interleaving = NULL;#ifdef STREAM_USING_ADUSuseADUs = True;#ifdef INTERLEAVE_ADUSunsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...unsigned const interleaveCycleSize= (sizeof interleaveCycle)/(sizeof (unsigned char));interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);#endif#endifsms->addSubsession(MP3AudioFileServerMediaSubsession::createNew(env, fileName,reuseSource, useADUs, interleaving));} else if (strcmp(extension, ".mpg") == 0) {// Assumed to be a MPEG-1 or 2 Program Stream (audio+video) file:NEW_SMS("MPEG-1 or 2 Program Stream");MPEG1or2FileServerDemux* demux = MPEG1or2FileServerDemux::createNew(env,fileName, reuseSource);sms->addSubsession(demux->newVideoServerMediaSubsession());sms->addSubsession(demux->newAudioServerMediaSubsession());} else if (strcmp(extension, ".ts") == 0) {// Assumed to be a MPEG Transport Stream file:// Use an index file name that's the same as the TS file name, except with ".tsx":unsigned indexFileNameLen = strlen(fileName) + 2; // allow for trailing "x\0"char* indexFileName = new char[indexFileNameLen];sprintf(indexFileName, "%sx", fileName);NEW_SMS("MPEG Transport Stream");sms->addSubsession(MPEG2TransportFileServerMediaSubsession::createNew(env,fileName, indexFileName, reuseSource));delete[] indexFileName;} else if (strcmp(extension, ".wav") == 0) {// Assumed to be a WAV Audio file:NEW_SMS("WAV Audio Stream");// To convert 16-bit PCM data to 8-bit u-law, prior to streaming,// change the following to True:Boolean convertToULaw = False;sms->addSubsession(WAVAudioFileServerMediaSubsession::createNew(env, fileName,reuseSource, convertToULaw));} else if (strcmp(extension, ".dv") == 0) {// Assumed to be a DV Video file// First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).OutPacketBuffer::maxSize = 300000;NEW_SMS("DV Video");sms->addSubsession(DVVideoFileServerMediaSubsession::createNew(env, fileName,reuseSource));} else if (strcmp(extension, ".mkv") == 0) {// Assumed to be a Matroska fileNEW_SMS("Matroska video+audio+(optional)subtitles");// Create a Matroska file server demultiplexor for the specified file.  (We enter the event loop to wait for this to complete.)newMatroskaDemuxWatchVariable = 0;MatroskaFileServerDemux::createNew(env, fileName,onMatroskaDemuxCreation, NULL);env.taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);ServerMediaSubsession* smss;while ((smss = demux->newServerMediaSubsession()) != NULL) {sms->addSubsession(smss);}}return sms;}

可以看到NEW_SMS("AMR Audio")会创建新的ServerMediaSession，之后马上调用sms->addSubsession（）为这个ServerMediaSession添加一个 ServerMediaSubSession 。看起来ServerMediaSession应该可以添加多个ServerMediaSubSession，但这里并没有这样做。如果可以添加多个 ServerMediaSubsession 那么ServerMediaSession与流名字所指定与文件是没有关系的，也就是说它不会操作文件，而文件的操作是放在 ServerMediaSubsession中的。具体应改是在ServerMediaSubsession的sdpLines()函数中打开。

 

转自： http://blog.csdn.net/niu_gao/article/details/6911130