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) << "
";
#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] = '';
        if (urlPreSuffix[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
"
                    "CSeq: %s
"
                    "%s
", 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
CSeq: %s
"
                "%s"
                "Content-Base: %s/
"
                "Content-Type: application/sdp
"
                "Content-Length: %d

"
                "%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"
        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