live555学习笔记－RTSP服务运作

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 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。

原文地址：http://blog.csdn.net/niu_gao/article/details/6911130

live555源代码（VC6）：http://download.csdn.net/detail/leixiaohua1020/6374387