live555学习笔记5

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 socket if (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 DEBUG envir() << "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代表一个rtp流，其包含的每个ServerMediaSubSession代表 //流中的一个Track。 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 frames sms->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_ADUS useADUs = True; #ifdef INTERLEAVE_ADUS unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own... unsigned const interleaveCycleSize = (sizeof interleaveCycle)/(sizeof (unsigned char)); interleaving = new Interleaving(interleaveCycleSize, interleaveCycle); #endif #endif sms->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 file NEW_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。