Live555学习之(五)------live555ProxyServer.cpp的学习

live555ProxyServer.cpp在live/proxyServer目录下，这个程序展示了如何利用live555来做一个代理服务器转发rtsp视频（例如，IPCamera的视频）。

　　首先来看一下main函数

 1 int main(int argc, char** argv)  2 { 3   // Increase the maximum size of video frames that we can 'proxy' without truncation. 4   // (Such frames are unreasonably large; the back-end servers should really not be sending frames this large!) 5   OutPacketBuffer::maxSize = 300000; // bytes 6  7   // Begin by setting up our usage environment: 8   TaskScheduler* scheduler = BasicTaskScheduler::createNew(); 9   env = BasicUsageEnvironment::createNew(*scheduler);10 11   /*12    .... 对各种输入参数的处理，在此略去13   */14   15 // Create the RTSP server.  Try first with the default port number (554),16   // and then with the alternative port number (8554):17   RTSPServer* rtspServer;18   portNumBits rtspServerPortNum = 554;19   rtspServer = createRTSPServer(rtspServerPortNum);20   if (rtspServer == NULL) {21     rtspServerPortNum = 8554;22     rtspServer = createRTSPServer(rtspServerPortNum);23   }24   if (rtspServer == NULL) {25     *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";26     exit(1);27   }28 29   // Create a proxy for each "rtsp://" URL specified on the command line:30   for (i = 1; i < argc; ++i) {31     char const* proxiedStreamURL = argv[i];32     char streamName[30];33     if (argc == 2) {34       sprintf(streamName, "%s", "proxyStream"); // there's just one stream; give it this name35     } else {36       sprintf(streamName, "proxyStream-%d", i); // there's more than one stream; distinguish them by name37     }38     ServerMediaSession* sms39       = ProxyServerMediaSession::createNew(*env, rtspServer,40                        proxiedStreamURL, streamName,41                        username, password, tunnelOverHTTPPortNum, verbosityLevel);42     rtspServer->addServerMediaSession(sms);43     // proxiedStreamURL是代理的源rtsp地址字符串，streamName表示代理后的ServerMediaSession的名字44     char* proxyStreamURL = rtspServer->rtspURL(sms);45     *env << "RTSP stream, proxying the stream \"" << proxiedStreamURL << "\"\n";46     *env << "\tPlay this stream using the URL: " << proxyStreamURL << "\n";47     delete[] proxyStreamURL;48   }49 50   if (proxyREGISTERRequests) {51     *env << "(We handle incoming \"REGISTER\" requests on port " << rtspServerPortNum << ")\n";52   }53 54   // Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.55   // Try first with the default HTTP port (80), and then with the alternative HTTP56   // port numbers (8000 and 8080).57 58   if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {59     *env << "\n(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling.)\n";60   } else {61     *env << "\n(RTSP-over-HTTP tunneling is not available.)\n";62   }63 64   // Now, enter the event loop:65   env->taskScheduler().doEventLoop(); // does not return66 67   return 0; // only to prevent compiler warning68 }

　　main函数还是很简单，第一行是设置OutPacketBuffer::maxSize的值，经过测试，我设置成300000个字节时就可以传送1080p的视频了。

　　然后还是创建TaskShcheduler和UsageEnvironment对象，中间是对各种输入参数的处理，在此我就省略不作分析了。

　　然后创建RTSPServer，根据输入的rtsp地址串创建ProxyServerMediaSession并添加到RTSPServer，然后开始程序的无限循环。

　　看一下ProxyServerMediaSession这个类

 1 class ProxyServerMediaSession: public ServerMediaSession { 2 public: 3   static ProxyServerMediaSession* createNew(UsageEnvironment& env, 4                         RTSPServer* ourRTSPServer, // Note: We can be used by just one "RTSPServer" 5                         char const* inputStreamURL, // the "rtsp://" URL of the stream we'll be proxying 6                         char const* streamName = NULL, 7                         char const* username = NULL, char const* password = NULL, 8                         portNumBits tunnelOverHTTPPortNum = 0, 9                             // for streaming the *proxied* (i.e., back-end) stream10                         int verbosityLevel = 0,11                         int socketNumToServer = -1);12  // Hack: "tunnelOverHTTPPortNum" == 0xFFFF (i.e., all-ones) means: Stream RTP/RTCP-over-TCP, but *not* using HTTP13    // "verbosityLevel" == 1 means display basic proxy setup info; "verbosityLevel" == 2 means display RTSP client protocol also.14 // If "socketNumToServer" >= 0,then it is the socket number of an already-existing TCP connection to the server.15 //(In this case, "inputStreamURL" must point to the socket's endpoint, so that it can be accessed via the socket.)16 17   virtual ~ProxyServerMediaSession();18 19   char const* url() const;20 21   char describeCompletedFlag;22     // initialized to 0; set to 1 when the back-end "DESCRIBE" completes.23     // (This can be used as a 'watch variable' in "doEventLoop()".)24   Boolean describeCompletedSuccessfully() const { return fClientMediaSession != NULL; }25     // This can be used - along with "describeCompletdFlag" - to check whether the back-end "DESCRIBE" completed *successfully*.26 27 protected:28   ProxyServerMediaSession(UsageEnvironment& env, RTSPServer* ourRTSPServer,29               char const* inputStreamURL, char const* streamName,30               char const* username, char const* password,31               portNumBits tunnelOverHTTPPortNum, int verbosityLevel,32               int socketNumToServer,33               createNewProxyRTSPClientFunc* ourCreateNewProxyRTSPClientFunc34               = defaultCreateNewProxyRTSPClientFunc);35 36   // If you subclass "ProxyRTSPClient", then you will also need to define your own function37   // - with signature "createNewProxyRTSPClientFunc" (see above) - that creates a new object38   // of this subclass.  You should also subclass "ProxyServerMediaSession" and, in your39   // subclass's constructor, initialize the parent class (i.e., "ProxyServerMediaSession")40   // constructor by passing your new function as the "ourCreateNewProxyRTSPClientFunc"41   // parameter.42 43 protected:44   RTSPServer* fOurRTSPServer;                  // 添加该ProxyServerMediaSession的RTSPServer对象45   ProxyRTSPClient* fProxyRTSPClient;　　　　    // 通过一个ProxyRTSPClient对象与给定rtsp服务器进行沟通46   MediaSession* fClientMediaSession;           // 通过一个MediaSession对象去请求给定rtsp地址表示的媒体资源47 48 private:49   friend class ProxyRTSPClient;50   friend class ProxyServerMediaSubsession;51   void continueAfterDESCRIBE(char const* sdpDescription);52   void resetDESCRIBEState(); // undoes what was done by "contineAfterDESCRIBE()"53 54 private:55   int fVerbosityLevel;56   class PresentationTimeSessionNormalizer* fPresentationTimeSessionNormalizer;57   createNewProxyRTSPClientFunc* fCreateNewProxyRTSPClientFunc;58 };

　　ProxyServerMediaSession是ServerMediaSession的子类，它与普通的ServerMediaSession相比多了三个重要的成员变量：RTSPServer* fOurRTSPServer，ProxyRTSPClient* fProxyRTSPClient，MediaSession* fClientMediaSession。fOurRTSPServer保存添加该ProxyServerMediaSession的RTSPServer对象，fProxyRTSPClient保存该ProxyServerMediaSession对应的ProxyRTSPClient对象，fClientMediaSession保存该ProxyServerMediaSession对应的MediaSession对象。每个ProxyServerMediaSession对应一个ProxyRTSPClient对象和MediaSession对象，从这个地方可以看出，live555代理服务器同时作为RTSP服务器端和RTSP客户端，作为RTSP客户端去获取给定rtsp地址（比如IPCamera的rtsp地址）的媒体资源，然后作为RTSP服务器端转发给其他的RTSP客户端（比如VLC）。

　　ProxyRTSPClient是RTSPClient的子类，我们来看一下它的定义

 1 // A subclass of "RTSPClient", used to refer to the particular "ProxyServerMediaSession" object being used. 2 // It is used only within the implementation of "ProxyServerMediaSession", but is defined here, in case developers wish to 3 // subclass it. 4  5 class ProxyRTSPClient: public RTSPClient { 6 public: 7   ProxyRTSPClient(class ProxyServerMediaSession& ourServerMediaSession, char const* rtspURL, 8                   char const* username, char const* password, 9                   portNumBits tunnelOverHTTPPortNum, int verbosityLevel, int socketNumToServer);10   virtual ~ProxyRTSPClient();11 12   void continueAfterDESCRIBE(char const* sdpDescription);   //包含了continueAfterDESCRIBE回调函数13   void continueAfterLivenessCommand(int resultCode, Boolean serverSupportsGetParameter); //发送心跳命令后的回调函数 14   void continueAfterSETUP();                                //包含了continueAfterSETUP回调函数15 16 private:17   void reset();18 19   Authenticator* auth() { return fOurAuthenticator; }20 21   void scheduleLivenessCommand();                      // 设置何时执行发送心跳命令的任务22   static void sendLivenessCommand(void* clientData);   // 发送心跳命令23 24   void scheduleDESCRIBECommand();　　　　　　　　　　　　　// 设置何时执行发送DESCRIBE命令的任务25   static void sendDESCRIBE(void* clientData);　　　　　　// 发送DESCRIBE命令26 27   static void subsessionTimeout(void* clientData);28   void handleSubsessionTimeout();29 30 private:31   friend class ProxyServerMediaSession;32   friend class ProxyServerMediaSubsession;33   ProxyServerMediaSession& fOurServerMediaSession;34   char* fOurURL;35   Authenticator* fOurAuthenticator;36   Boolean fStreamRTPOverTCP;37   class ProxyServerMediaSubsession *fSetupQueueHead, *fSetupQueueTail;38   unsigned fNumSetupsDone;39   unsigned fNextDESCRIBEDelay; // in seconds40   Boolean fServerSupportsGetParameter, fLastCommandWasPLAY;41   TaskToken fLivenessCommandTask, fDESCRIBECommandTask, fSubsessionTimerTask;42 };

　　我们接下来看一下创建ProxyServerMediaSession对象的过程

 1 ProxyServerMediaSession* ProxyServerMediaSession 2 ::createNew(UsageEnvironment& env, RTSPServer* ourRTSPServer, 3         char const* inputStreamURL, char const* streamName, 4         char const* username, char const* password, 5         portNumBits tunnelOverHTTPPortNum, int verbosityLevel, int socketNumToServer) { 6   return new ProxyServerMediaSession(env, ourRTSPServer, inputStreamURL, streamName, username, password, 7                      tunnelOverHTTPPortNum, verbosityLevel, socketNumToServer); 8 } 9 10 11 ProxyServerMediaSession12 ::ProxyServerMediaSession(UsageEnvironment& env, RTSPServer* ourRTSPServer,13               char const* inputStreamURL, char const* streamName,14               char const* username, char const* password,15               portNumBits tunnelOverHTTPPortNum, int verbosityLevel,16               int socketNumToServer,17               createNewProxyRTSPClientFunc* ourCreateNewProxyRTSPClientFunc)18   : ServerMediaSession(env, streamName, NULL, NULL, False, NULL),19     describeCompletedFlag(0), fOurRTSPServer(ourRTSPServer), fClientMediaSession(NULL),20     fVerbosityLevel(verbosityLevel),21     fPresentationTimeSessionNormalizer(new PresentationTimeSessionNormalizer(envir())),22     fCreateNewProxyRTSPClientFunc(ourCreateNewProxyRTSPClientFunc) {23   // Open a RTSP connection to the input stream, and send a "DESCRIBE" command.24   // We'll use the SDP description in the response to set ourselves up.25   fProxyRTSPClient26     = (*fCreateNewProxyRTSPClientFunc)(*this, inputStreamURL, username, password,27                        tunnelOverHTTPPortNum,28                        verbosityLevel > 0 ? verbosityLevel-1 : verbosityLevel,29                        socketNumToServer);30   ProxyRTSPClient::sendDESCRIBE(fProxyRTSPClient);31 }

　　在ProxyServerMediaSession中创建了ProxyRTSPClient对象，是通过fCreateNewProxyRTSPClientFunc函数来创建的，该函数默认是defaultCreateNewProxyRTSPClientFunc函数。

1 ProxyRTSPClient*2 defaultCreateNewProxyRTSPClientFunc(ProxyServerMediaSession& ourServerMediaSession,3                     char const* rtspURL,4                     char const* username, char const* password,5                     portNumBits tunnelOverHTTPPortNum, int verbosityLevel,6                     int socketNumToServer) {7   return new ProxyRTSPClient(ourServerMediaSession, rtspURL, username, password,8                  tunnelOverHTTPPortNum, verbosityLevel, socketNumToServer);9 }

　　然后就通过刚创建的ProxyRTSPClient对象发送DESCRIBE命令，请求获得媒体资源的SDP信息。

 1 void ProxyRTSPClient::sendDESCRIBE(void* clientData) { 2   ProxyRTSPClient* rtspClient = (ProxyRTSPClient*)clientData; 3   if (rtspClient != NULL) rtspClient->sendDescribeCommand(::continueAfterDESCRIBE, rtspClient->auth()); 4 } 5  6 void ProxyRTSPClient::continueAfterDESCRIBE(char const* sdpDescription) { 7   if (sdpDescription != NULL) { 8     fOurServerMediaSession.continueAfterDESCRIBE(sdpDescription); 9 10     // Unlike most RTSP streams, there might be a long delay between this "DESCRIBE" command (to the downstream server) and the11     // subsequent "SETUP"/"PLAY" - which doesn't occur until the first time that a client requests the stream.12     // To prevent the proxied connection (between us and the downstream server) from timing out, we send periodic 'liveness'13     // ("OPTIONS" or "GET_PARAMETER") commands.  (The usual RTCP liveness mechanism wouldn't work here, because RTCP packets14     // don't get sent until after the "PLAY" command.)15     scheduleLivenessCommand();16   } else {17     // The "DESCRIBE" command failed, most likely because the server or the stream is not yet running.18     // Reschedule another "DESCRIBE" command to take place later:19     scheduleDESCRIBECommand();20   }21 }22 23 void ProxyRTSPClient::scheduleLivenessCommand() {24   // Delay a random time before sending another 'liveness' command.25   unsigned delayMax = sessionTimeoutParameter(); // if the server specified a maximum time between 'liveness' probes, then use that26   if (delayMax == 0) {27     delayMax = 60;28   }29 30   // Choose a random time from [delayMax/2,delayMax-1) seconds:31   unsigned const us_1stPart = delayMax*500000;32   unsigned uSecondsToDelay;33   if (us_1stPart <= 1000000) {34     uSecondsToDelay = us_1stPart;35   } else {36     unsigned const us_2ndPart = us_1stPart-1000000;37     uSecondsToDelay = us_1stPart + (us_2ndPart*our_random())%us_2ndPart;38   }39   fLivenessCommandTask = envir().taskScheduler().scheduleDelayedTask(uSecondsToDelay, sendLivenessCommand, this);40 }41 42 void ProxyRTSPClient::sendLivenessCommand(void* clientData) {43   ProxyRTSPClient* rtspClient = (ProxyRTSPClient*)clientData;44 45   // Note.  By default, we do not send "GET_PARAMETER" as our 'liveness notification' command, even if the server previously46   // indicated (in its response to our earlier "OPTIONS" command) that it supported "GET_PARAMETER".  This is because47   // "GET_PARAMETER" crashes some camera servers (even though they claimed to support "GET_PARAMETER").48 #ifdef SEND_GET_PARAMETER_IF_SUPPORTED49   MediaSession* sess = rtspClient->fOurServerMediaSession.fClientMediaSession;50 51   if (rtspClient->fServerSupportsGetParameter && rtspClient->fNumSetupsDone > 0 && sess != NULL) {52     rtspClient->sendGetParameterCommand(*sess, ::continueAfterGET_PARAMETER, "", rtspClient->auth());53   } else {54 #endif55     rtspClient->sendOptionsCommand(::continueAfterOPTIONS, rtspClient->auth());56 #ifdef SEND_GET_PARAMETER_IF_SUPPORTED57   }58 #endif59 }60 61 void ProxyRTSPClient::scheduleDESCRIBECommand() {62   // Delay 1s, 2s, 4s, 8s ... 256s until sending the next "DESCRIBE".  Then, keep delaying a random time from [256..511] seconds:63   unsigned secondsToDelay;64   if (fNextDESCRIBEDelay <= 256) {65     secondsToDelay = fNextDESCRIBEDelay;66     fNextDESCRIBEDelay *= 2;67   } else {68     secondsToDelay = 256 + (our_random()&0xFF); // [256..511] seconds69   }70 71   if (fVerbosityLevel > 0) {72     envir() << *this << ": RTSP \"DESCRIBE\" command failed; trying again in " << secondsToDelay << " seconds\n";73   }74   fDESCRIBECommandTask = envir().taskScheduler().scheduleDelayedTask(secondsToDelay*MILLION, sendDESCRIBE, this);75 }76 77 void ProxyRTSPClient::sendDESCRIBE(void* clientData) {78   ProxyRTSPClient* rtspClient = (ProxyRTSPClient*)clientData;79   if (rtspClient != NULL) rtspClient->sendDescribeCommand(::continueAfterDESCRIBE, rtspClient->auth());80 }

 　　发送DESCRIBE命令后，回调::continueAfterDESCRIBE函数(static void continueAfterDESCRIBE函数)，在该函数中再调用ProxyServerMediaSession::continueAfterDESCRIBE函数，在ProxyServerMediaSession::continueAfterDESCRIBE函数中判断是否成功获取了SDP信息。若成功获取了，则调用ProxyServerMediaSession::continueAfterDESCRIBE，然后调用scheduleLivenessCommand函数设置发送心跳命令的任务；若没有成功获取则调用scheduleDESCRIBECommand函数设置重新发送DESCRIBE命令的任务。

　　ProxyRTSPClient使用GET_PARAMETER命令或者OPTIONS命令作为心跳命令，scheduleLivenessCommand函数中，从[delayMax / 2,delayMax - 1)中随机选取一个值作为发送下一个心跳命令的延时。scheduleDESCRIBECommand函数中，根据上次发送DESCRIBE命令的延时来计算下一次发送DESCRIBE命令的延时，若上次发送DESCRIBE命令的延时小于256s，则按照1,2,4,8,.....256这样一个等比数列来选择一个值作为发送下一个DESCRIBE命令的延时，否则就从[256,511]中随机选择一个值作为下次发送DESCRIBE命令的延时。

　　成功获取SDP信息后，调用ProxyServerMediaSession::continueAfterDESCRIBE函数：

 1 void ProxyServerMediaSession::continueAfterDESCRIBE(char const* sdpDescription) { 2   describeCompletedFlag = 1; 3  4   // Create a (client) "MediaSession" object from the stream's SDP description ("resultString"), then iterate through its 5   // "MediaSubsession" objects, to set up corresponding "ServerMediaSubsession" objects that we'll use to serve the stream's tracks. 6   do { 7     fClientMediaSession = MediaSession::createNew(envir(), sdpDescription); 8     if (fClientMediaSession == NULL) break; 9 10     MediaSubsessionIterator iter(*fClientMediaSession);11     for (MediaSubsession* mss = iter.next(); mss != NULL; mss = iter.next()) {12       ServerMediaSubsession* smss = new ProxyServerMediaSubsession(*mss);13       addSubsession(smss);14       if (fVerbosityLevel > 0) {15     envir() << *this << " added new \"ProxyServerMediaSubsession\" for "16         << mss->protocolName() << "/" << mss->mediumName() << "/" << mss->codecName() << " track\n";17       }18     }19   } while (0);20 }

　　在continueAfterDESCRIBE函数中，首先创建了MediaSession对象，然后创建ProxyServerMediaSubsession对象并添加到ProxyServerMediaSession。ProxyServerMediaSubsession继承自OnDemandServerMediaSubsession类

 1 class ProxyServerMediaSubsession: public OnDemandServerMediaSubsession { 2 public: 3   ProxyServerMediaSubsession(MediaSubsession& mediaSubsession); 4   virtual ~ProxyServerMediaSubsession(); 5  6   char const* codecName() const { return fClientMediaSubsession.codecName(); } 7  8 private: // redefined virtual functions 9   virtual FramedSource* createNewStreamSource(unsigned clientSessionId,10                                               unsigned& estBitrate);11   virtual void closeStreamSource(FramedSource *inputSource);12   virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,13                                     unsigned char rtpPayloadTypeIfDynamic,14                                     FramedSource* inputSource);15 16 private:17   static void subsessionByeHandler(void* clientData);18   void subsessionByeHandler();19 20   int verbosityLevel() const { return ((ProxyServerMediaSession*)fParentSession)->fVerbosityLevel; }21 22 private:23   friend class ProxyRTSPClient;24   MediaSubsession& fClientMediaSubsession; // the 'client' media subsession object that corresponds to this 'server' media subsession25   ProxyServerMediaSubsession* fNext; // used when we're part of a queue26   Boolean fHaveSetupStream;27 };

　　ProxyServerMediaSubsession类中有一个MediaSubsession的引用，一个ProxyServerMediaSubsession对象对应一个MediaSubsession对象。ProxyServerMediaSubsession接下来并不会急着发送SETUP命令，而是等到有RTSP客户端（比如VLC）请求它时再发送SETUP命令去请求建立与IPCamera的连接。

　　然后，RTSPServer等待着RTSP客户端来请求，现在我们假设收到了来自VLC客户端的rtsp请求，然后流程就和前面《建立RTSP连接的过程（RTSP服务器端）》类似。下面我们简要来看一下这个流程，主要突出与之前不同的步骤，我们从RTSPServer::handleCmd_DESCRIBE函数看起：

　　hanleCmd_DESCRIBE函数处理来自客户端的DESCRIBE命令，调用ServerMediaSession::generateSDPDescription函数；

　　ServerMediaSession::generateSDPDescription函数中调用的是OnDemandServerMediaSubsession::sdpLines函数；

　　在sdpLines函数中，调用ProxyServerMediaSubsession::createNewStreamSource函数创建一个临时的FramedSource对象，调用ProxyServerMediaSubsession::createNewRTPSink创建临时的RTPSink对象，然后调用OnDemandServerMediaSubsession::setSDPLinesFromRTPSink函数。

  1 FramedSource* ProxyServerMediaSubsession::createNewStreamSource(unsigned clientSessionId, unsigned& estBitrate)
    {   
  2    ProxyServerMediaSession* const sms = (ProxyServerMediaSession*)fParentSession; 
  3   4   if (verbosityLevel() > 0) {  5     envir() << *this << "::createNewStreamSource(session id " << clientSessionId << ")\n";  6   }  7   8   // If we haven't yet created a data source from our 'media subsession' object, initiate() it to do so:  9   if (fClientMediaSubsession.readSource() == NULL) { 10     fClientMediaSubsession.receiveRawMP3ADUs(); // hack for MPA-ROBUST streams 11     fClientMediaSubsession.receiveRawJPEGFrames(); // hack for proxying JPEG/RTP streams. (Don't do this if we're transcoding.) 12     fClientMediaSubsession.initiate();        // 调用MediaSubsession的initiate函数，初始化MediaSubsession对象 13     if (verbosityLevel() > 0) { 14       envir() << "\tInitiated: " << *this << "\n"; 15     } 16      // 在fReadSource前面添加PresentationTimeSessionNormalizer作为Filter 17     if (fClientMediaSubsession.readSource() != NULL) { 18       // Add to the front of all data sources a filter that will 'normalize' their frames' presentation times, 19       // before the frames get re-transmitted by our server: 20       char const* const codecName = fClientMediaSubsession.codecName(); 21       FramedFilter* normalizerFilter = sms->fPresentationTimeSessionNormalizer 22     ->createNewPresentationTimeSubsessionNormalizer(fClientMediaSubsession.readSource(), fClientMediaSubsession.rtpSource(),codecName);
 23 24       fClientMediaSubsession.addFilter(normalizerFilter);   // ProxyServerMediaSubsession的FramedSource以MediaSubsession的FramedSource作为媒体源 25  26       // Some data sources require a 'framer' object to be added, before they can be fed into 27       // a "RTPSink".  Adjust for this now: 28       if (strcmp(codecName, "H264") == 0) {  // 再在fReadSource前面添加H264VideoStreamDiscreteFramer作为Filter 29     fClientMediaSubsession.addFilter(H264VideoStreamDiscreteFramer 30                      ::createNew(envir(), fClientMediaSubsession.readSource())); 31       } else if (strcmp(codecName, "H265") == 0) { 32     fClientMediaSubsession.addFilter(H265VideoStreamDiscreteFramer 33                      ::createNew(envir(), fClientMediaSubsession.readSource())); 34       } else if (strcmp(codecName, "MP4V-ES") == 0) { 35     fClientMediaSubsession.addFilter(MPEG4VideoStreamDiscreteFramer 36                      ::createNew(envir(), fClientMediaSubsession.readSource(), 37                              True/* leave PTs unmodified*/)); 38       } else if (strcmp(codecName, "MPV") == 0) { 39     fClientMediaSubsession.addFilter(MPEG1or2VideoStreamDiscreteFramer 40                      ::createNew(envir(), fClientMediaSubsession.readSource(), 41                              False, 5.0, True/* leave PTs unmodified*/)); 42       } else if (strcmp(codecName, "DV") == 0) { 43     fClientMediaSubsession.addFilter(DVVideoStreamFramer 44                      ::createNew(envir(), fClientMediaSubsession.readSource(), 45                              False, True/* leave PTs unmodified*/)); 46       } 47     } 48  49     if (fClientMediaSubsession.rtcpInstance() != NULL) { 50       fClientMediaSubsession.rtcpInstance()->setByeHandler(subsessionByeHandler, this); 51     } 52   } 53  54   ProxyRTSPClient* const proxyRTSPClient = sms->fProxyRTSPClient; 55   if (clientSessionId != 0) {　　　　//为了形成SDP信息而创建临时FramedSource时，传入的clientSessionID参数为0，就不会发送SETUP命令 56     // We're being called as a result of implementing a RTSP "SETUP". 57     if (!fHaveSetupStream) { 58       // This is our first "SETUP".  Send RTSP "SETUP" and later "PLAY" commands to the proxied server, to start streaming: 59       // (Before sending "SETUP", enqueue ourselves on the "RTSPClient"s 'SETUP queue', so we'll be able to get the correct 60       //  "ProxyServerMediaSubsession" to handle the response.  (Note that responses come back in the same order as requests.)) 61       Boolean queueWasEmpty = proxyRTSPClient->fSetupQueueHead == NULL; 62       if (queueWasEmpty) { 63     proxyRTSPClient->fSetupQueueHead = this; 64       } else { 65     proxyRTSPClient->fSetupQueueTail->fNext = this; 66       } 67       proxyRTSPClient->fSetupQueueTail = this; 68  69       // Hack: If there's already a pending "SETUP" request (for another track), don't send this track's "SETUP" right away, because 70       // the server might not properly handle 'pipelined' requests.  Instead, wait until after previous "SETUP" responses come back. 71       if (queueWasEmpty) {   // 发送SETUP命令 72     proxyRTSPClient->sendSetupCommand(fClientMediaSubsession, ::continueAfterSETUP, 73                       False, proxyRTSPClient->fStreamRTPOverTCP, False, proxyRTSPClient->auth()); 74     ++proxyRTSPClient->fNumSetupsDone; 75     fHaveSetupStream = True; 76       } 77     } else { 78       // This is a "SETUP" from a new client.  We know that there are no other currently active clients (otherwise we wouldn't 79       // have been called here), so we know that the substream was previously "PAUSE"d.  Send "PLAY" downstream once again, 80       // to resume the stream: 81       if (!proxyRTSPClient->fLastCommandWasPLAY) { // so that we send only one "PLAY"; not one for each subsession 82     proxyRTSPClient->sendPlayCommand(fClientMediaSubsession.parentSession(), NULL, -1.0f/*resume from previous point*/, 83                      -1.0f, 1.0f, proxyRTSPClient->auth()); 84     proxyRTSPClient->fLastCommandWasPLAY = True; 85       } 86     } 87   } 88  89   estBitrate = fClientMediaSubsession.bandwidth(); 90   if (estBitrate == 0) estBitrate = 50; // kbps, estimate 91   return fClientMediaSubsession.readSource(); 92 } 93  94 RTPSink* ProxyServerMediaSubsession 95 ::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* inputSource) { 96   if (verbosityLevel() > 0) { 97     envir() << *this << "::createNewRTPSink()\n"; 98   } 99 100   // Create (and return) the appropriate "RTPSink" object for our codec:101   RTPSink* newSink;102   char const* const codecName = fClientMediaSubsession.codecName();103   if (strcmp(codecName, "AC3") == 0 || strcmp(codecName, "EAC3") == 0) {104     newSink = AC3AudioRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,105                      fClientMediaSubsession.rtpTimestampFrequency()); 106 #if 0 // This code does not work; do *not* enable it:107   } else if (strcmp(codecName, "AMR") == 0 || strcmp(codecName, "AMR-WB") == 0) {108     Boolean isWideband = strcmp(codecName, "AMR-WB") == 0;109     newSink = AMRAudioRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,110                      isWideband, fClientMediaSubsession.numChannels());111 #endif112   } else if (strcmp(codecName, "DV") == 0) {113     newSink = DVVideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);114   } else if (strcmp(codecName, "GSM") == 0) {115     newSink = GSMAudioRTPSink::createNew(envir(), rtpGroupsock);116   } else if (strcmp(codecName, "H263-1998") == 0 || strcmp(codecName, "H263-2000") == 0) {117     newSink = H263plusVideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,118                           fClientMediaSubsession.rtpTimestampFrequency()); 119   } else if (strcmp(codecName, "H264") == 0) {  //创建H264VideoRTPSink对象120     newSink = H264VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,121                       fClientMediaSubsession.fmtp_spropparametersets());122   } else if (strcmp(codecName, "H265") == 0) {123     newSink = H265VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,124                       fClientMediaSubsession.fmtp_spropvps(),125                       fClientMediaSubsession.fmtp_spropsps(),126                       fClientMediaSubsession.fmtp_sproppps());127   } else if (strcmp(codecName, "JPEG") == 0) {128     newSink = SimpleRTPSink::createNew(envir(), rtpGroupsock, 26, 90000, "video", "JPEG",129                        1/*numChannels*/, False/*allowMultipleFramesPerPacket*/, False/*doNormalMBitRule*/);130   } else if (strcmp(codecName, "MP4A-LATM") == 0) {131     newSink = MPEG4LATMAudioRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,132                            fClientMediaSubsession.rtpTimestampFrequency(),133                            fClientMediaSubsession.fmtp_config(),134                            fClientMediaSubsession.numChannels());135   } else if (strcmp(codecName, "MP4V-ES") == 0) {136     newSink = MPEG4ESVideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,137                          fClientMediaSubsession.rtpTimestampFrequency(),138                          fClientMediaSubsession.attrVal_unsigned("profile-level-id"),139                          fClientMediaSubsession.fmtp_config()); 140   } else if (strcmp(codecName, "MPA") == 0) {141     newSink = MPEG1or2AudioRTPSink::createNew(envir(), rtpGroupsock);142   } else if (strcmp(codecName, "MPA-ROBUST") == 0) {143     newSink = MP3ADURTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);144   } else if (strcmp(codecName, "MPEG4-GENERIC") == 0) {145     newSink = MPEG4GenericRTPSink::createNew(envir(), rtpGroupsock,146                          rtpPayloadTypeIfDynamic, fClientMediaSubsession.rtpTimestampFrequency(),147                          fClientMediaSubsession.mediumName(),148                          fClientMediaSubsession.attrVal_strToLower("mode"),149                          fClientMediaSubsession.fmtp_config(), fClientMediaSubsession.numChannels());150   } else if (strcmp(codecName, "MPV") == 0) {151     newSink = MPEG1or2VideoRTPSink::createNew(envir(), rtpGroupsock);152   } else if (strcmp(codecName, "OPUS") == 0) {153     newSink = SimpleRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,154                        48000, "audio", "OPUS", 2, False/*only 1 Opus 'packet' in each RTP packet*/);155   } else if (strcmp(codecName, "T140") == 0) {156     newSink = T140TextRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);157   } else if (strcmp(codecName, "THEORA") == 0) {158     newSink = TheoraVideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,159                         fClientMediaSubsession.fmtp_config()); 160   } else if (strcmp(codecName, "VORBIS") == 0) {161     newSink = VorbisAudioRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic,162                         fClientMediaSubsession.rtpTimestampFrequency(), fClientMediaSubsession.numChannels(),163                         fClientMediaSubsession.fmtp_config()); 164   } else if (strcmp(codecName, "VP8") == 0) {165     newSink = VP8VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);166   } else if (strcmp(codecName, "AMR") == 0 || strcmp(codecName, "AMR-WB") == 0) {167     // Proxying of these codecs is currently *not* supported, because the data received by the "RTPSource" object is not in a168     // form that can be fed directly into a corresponding "RTPSink" object.169     if (verbosityLevel() > 0) {170       envir() << "\treturns NULL (because we currently don't support the proxying of \""171           << fClientMediaSubsession.mediumName() << "/" << codecName << "\" streams)\n";172     }173     return NULL;174   } else if (strcmp(codecName, "QCELP") == 0 ||175          strcmp(codecName, "H261") == 0 ||176          strcmp(codecName, "H263-1998") == 0 || strcmp(codecName, "H263-2000") == 0 ||177          strcmp(codecName, "X-QT") == 0 || strcmp(codecName, "X-QUICKTIME") == 0) {178     // This codec requires a specialized RTP payload format; however, we don't yet have an appropriate "RTPSink" subclass for it:179     if (verbosityLevel() > 0) {180       envir() << "\treturns NULL (because we don't have a \"RTPSink\" subclass for this RTP payload format)\n";181     }182     return NULL;183   } else {184     // This codec is assumed to have a simple RTP payload format that can be implemented just with a "SimpleRTPSink":185     Boolean allowMultipleFramesPerPacket = True; // by default186     Boolean doNormalMBitRule = True; // by default187     // Some codecs change the above default parameters:188     if (strcmp(codecName, "MP2T") == 0) {189       doNormalMBitRule = False; // no RTP 'M' bit190     }191     newSink = SimpleRTPSink::createNew(envir(), rtpGroupsock,192                        rtpPayloadTypeIfDynamic, fClientMediaSubsession.rtpTimestampFrequency(),193                        fClientMediaSubsession.mediumName(), fClientMediaSubsession.codecName(),194                        fClientMediaSubsession.numChannels(), allowMultipleFramesPerPacket, doNormalMBitRule);195   }196 197   // Because our relayed frames' presentation times are inaccurate until the input frames have been RTCP-synchronized,198   // we temporarily disable RTCP "SR" reports for this "RTPSink" object:199   newSink->enableRTCPReports() = False;200 201   // Also tell our "PresentationTimeSubsessionNormalizer" object about the "RTPSink", so it can enable RTCP "SR" reports later:202   PresentationTimeSubsessionNormalizer* ssNormalizer;203   if (strcmp(codecName, "H264") == 0 ||204       strcmp(codecName, "H265") == 0 ||205       strcmp(codecName, "MP4V-ES") == 0 ||206       strcmp(codecName, "MPV") == 0 ||207       strcmp(codecName, "DV") == 0) {208     // There was a separate 'framer' object in front of the "PresentationTimeSubsessionNormalizer", so go back one object to get it:209     ssNormalizer = (PresentationTimeSubsessionNormalizer*)(((FramedFilter*)inputSource)->inputSource());210   } else {211     ssNormalizer = (PresentationTimeSubsessionNormalizer*)inputSource;212   }213   ssNormalizer->setRTPSink(newSink);214 215   return newSink;216 }

　　在ProxyServerMediaSubsession::createNewStreamSource函数中，首先调用MediaSubsession::initiate函数进行初始化，然后添加两个Filter：PresentationTimeSessionNormalizer和H264VideoStreamDiscreteFramer。PresentationTimeSessionNormalizer我没有细致的去看，大概的作用应该是给帧打时间戳的，H264VideoStreamDiscreteFramer是用来从接收到的数据分离出每一帧数据。

　　在ProxyServerMediaSubsession::createNewRTPSink函数中，主要就是创建了一个H264VideoRTPSink对象。

　　执行完以上两个函数后，调用OnDemandServerMediaSubsession::setSDPLinesFromRTPSink函数；

　　在setSDPLinesFromRTPSink函数中，调用OnDemandServerMediaSubsession::getAuxSDPLine函数；

　　在getAuxSDPLine函数中，调用H264VideoRTPSink::auxSDPLine函数：

 1 char const* H264VideoRTPSink::auxSDPLine() { 2   // Generate a new "a=fmtp:" line each time, using our SPS and PPS (if we have them), 3   // otherwise parameters from our framer source (in case they've changed since the last time that 4   // we were called): 5   H264or5VideoStreamFramer* framerSource = NULL; 6   u_int8_t* vpsDummy = NULL; unsigned vpsDummySize = 0; 7   u_int8_t* sps = fSPS; unsigned spsSize = fSPSSize; 8   u_int8_t* pps = fPPS; unsigned ppsSize = fPPSSize; 9   if (sps == NULL || pps == NULL) {10     // We need to get SPS and PPS from our framer source:11     if (fOurFragmenter == NULL) return NULL; // we don't yet have a fragmenter (and therefore not a source)12     framerSource = (H264or5VideoStreamFramer*)(fOurFragmenter->inputSource());13     if (framerSource == NULL) return NULL; // we don't yet have a source14     //获取VPS、SPS以及PPS信息15     framerSource->getVPSandSPSandPPS(vpsDummy, vpsDummySize, sps, spsSize, pps, ppsSize);16     if (sps == NULL || pps == NULL) return NULL; // our source isn't ready17   }18 19   // Set up the "a=fmtp:" SDP line for this stream:20   u_int8_t* spsWEB = new u_int8_t[spsSize]; // "WEB" means "Without Emulation Bytes"21   unsigned spsWEBSize = removeH264or5EmulationBytes(spsWEB, spsSize, sps, spsSize);22   if (spsWEBSize < 4) { // Bad SPS size => assume our source isn't ready23     delete[] spsWEB;24     return NULL;25   }26   u_int32_t profileLevelId = (spsWEB[1]<<16) | (spsWEB[2]<<8) | spsWEB[3];27   delete[] spsWEB;28 29   char* sps_base64 = base64Encode((char*)sps, spsSize);30   char* pps_base64 = base64Encode((char*)pps, ppsSize);31 32   char const* fmtpFmt =33     "a=fmtp:%d packetization-mode=1"34     ";profile-level-id=%06X"35     ";sprop-parameter-sets=%s,%s\r\n";36   unsigned fmtpFmtSize = strlen(fmtpFmt)37     + 3 /* max char len */38     + 6 /* 3 bytes in hex */39     + strlen(sps_base64) + strlen(pps_base64);40   char* fmtp = new char[fmtpFmtSize];41   sprintf(fmtp, fmtpFmt,42           rtpPayloadType(),43       profileLevelId,44           sps_base64, pps_base64);45 46   delete[] sps_base64;47   delete[] pps_base64;48 49   delete[] fFmtpSDPLine; fFmtpSDPLine = fmtp;50   return fFmtpSDPLine;51 }

　　在H264VideoRTPSink::auxSDPLine函数中，调用getVPSandSPSandPPS函数获取VPS、SPS和PPS信息，此后将组成的SDP信息发送给RTSP客户端（VLC客户端）。

　　然后RTSPServer就等待RTSP客户端（VLC客户端）发送SETUP命令，收到SETUP命令后就调用RTSPServer::handleCmd_SETUP函数来处理；

　　在handleCmd_SETUP函数中，调用OnDemandServerMediaSubsession::getStreamParameters函数；

　　在getStreamParameters函数中又调用ProxyServerMediaSubsession::createNewStreamSource函数创建FramedSource，调用ProxyServerMediaSubsession::createNewRTPSink函数创建RTPSink。这次调用createNewStreamSource函数的时候传入的参数clientSessionId就是一个非0值，这样在createNewStreamSource函数里，就会发送SETUP命令给IPCamera请求建立连接。并且在收到回复后会回调::continueAfterSETUP（static void continueAfterSETUP），在其中又调用ProxyRTSPClient::continueAfterSETUP函数。

 1 void ProxyRTSPClient::continueAfterSETUP() { 2   if (fVerbosityLevel > 0) { 3     envir() << *this << "::continueAfterSETUP(): head codec: " << fSetupQueueHead->fClientMediaSubsession.codecName() 4         << "; numSubsessions " << fSetupQueueHead->fParentSession->numSubsessions() << "\n\tqueue:"; 5     for (ProxyServerMediaSubsession* p = fSetupQueueHead; p != NULL; p = p->fNext) { 6       envir() << "\t" << p->fClientMediaSubsession.codecName(); 7     } 8     envir() << "\n"; 9   }10   envir().taskScheduler().unscheduleDelayedTask(fSubsessionTimerTask); // in case it had been set11 12   // Dequeue the first "ProxyServerMediaSubsession" from our 'SETUP queue'.  It will be the one for which this "SETUP" was done:13   ProxyServerMediaSubsession* smss = fSetupQueueHead; // Assert: != NULL14   fSetupQueueHead = fSetupQueueHead->fNext;15   if (fSetupQueueHead == NULL) fSetupQueueTail = NULL;16 17   if (fSetupQueueHead != NULL) {18     // There are still entries in the queue, for tracks for which we have still to do a "SETUP".19     // "SETUP" the first of these now:20     sendSetupCommand(fSetupQueueHead->fClientMediaSubsession, ::continueAfterSETUP,21              False, fStreamRTPOverTCP, False, fOurAuthenticator);22     ++fNumSetupsDone;23     fSetupQueueHead->fHaveSetupStream = True;24   } else {25     if (fNumSetupsDone >= smss->fParentSession->numSubsessions()) {26       // We've now finished setting up each of our subsessions (i.e., 'tracks').27       // Continue by sending a "PLAY" command (an 'aggregate' "PLAY" command, on the whole session):28       sendPlayCommand(smss->fClientMediaSubsession.parentSession(), NULL, -1.0f, -1.0f, 1.0f, fOurAuthenticator);29           // the "-1.0f" "start" parameter causes the "PLAY" to be sent without a "Range:" header, in case we'd already done30           // a "PLAY" before (as a result of a 'subsession timeout' (note below))31       fLastCommandWasPLAY = True;32     } else {33       // Some of this session's subsessions (i.e., 'tracks') remain to be "SETUP".  They might get "SETUP" very soon, but it's34       // also possible - if the remote client chose to play only some of the session's tracks - that they might not.35       // To allow for this possibility, we set a timer.  If the timer expires without the remaining subsessions getting "SETUP",36       // then we send a "PLAY" command anyway:37       fSubsessionTimerTask = envir().taskScheduler().scheduleDelayedTask(SUBSESSION_TIMEOUT_SECONDS*MILLION, (TaskFunc*)subsessionTimeout, this);
3839     }40   }41 }

　　在ProxyRTSPClient::continueAfterSETUP函数中，为剩余未建立连接的MediaSubsession发送SETUP命令，当所有的MediaSubsession都建立连接后，向IPCamera发送PLAY命令，开始请求传输媒体流。

 　　然后RTSPServer等待RTSP客户端（VLC客户端）的PLAY命令，收到PLAY命令后，调用RTSPServer::RTSPClientSession::handleCmd_PLAY函数进行处理；

　　然后调用OnDemandServerMediaSubsession::startStream函数，在其中调用StreamState::startPlaying函数；

　　然后就是H264VideoRTPSink不断地从H264VideoStreamDiscreteFramer中获取数据然后传给RTSP客户端（VLC客户端），而H264VideoStreamDiscreteFramer从MediaSubsession的FramedSource获取数据，MediaSubsession的FramedSource从IPCamera获取数据。

　　

　　以上就是live555作为代理服务器转发RTSP实时视频的过程，实际上是综合了前面两篇介绍的流程，对于IPCamera作为RTSP客户端，对于VLC作为RTSP服务器端。

　　关于live555ProxyServer.cpp的几个修改建议：

　   我们可以使用live555ProxyServer.cpp这个程序很方便地构建一个转发RTSP实时视频的代理服务器，比如转发IPCamera的实时视频。但我经过试验发现这个程序还是存在一些问题，还需要作出一些修改才能更好地作为代理服务器运行。由于楼主理解能力有限，这些修改不一定是从根本上解决问题，仅供大家参考。

　　（1）main函数的开头有OutPacketBuffer::maxSize = 300000，原本的语句是OutPacketBuffer::maxSize=30000。但我发现转发高清实时视频的时候，VLC会有大面积马赛克，而live555服务器端也打印出"MultiFramedRTPSink::afterGettingFrame1(): The input frame data was too large for out buffer size .............."。

　　　　我们找到这个提示语句在MultiFramedRTPSink::afterGettingFram1函数中，明显从提示的意思来看是说我们RTPSink的缓冲区设置的太小了，而高清视频的一帧数据太大了。MultiFramedRTPSink将数据保存在fOutBuf中，fOutBuf是指向OutPacketBuffer实例的指针，看一下OutPacketBuffer::totalBytesAvailable函数：

1 unsigned totalBytesAvailable() const {2     return fLimit - (fPacketStart + fCurOffset);3   }

内容很简单，那么totalBytesAvaiable返回值太小的就说明fLimit太小了，fLimit的值在OutPacketBuffer的构造函数中设置了：

 1 OutPacketBuffer 2 ::OutPacketBuffer(unsigned preferredPacketSize, unsigned maxPacketSize, unsigned maxBufferSize) 3   : fPreferred(preferredPacketSize), fMax(maxPacketSize), 4     fOverflowDataSize(0) { 5   if (maxBufferSize == 0) maxBufferSize = maxSize;      // maxBufferSize的默认值是0 6   unsigned maxNumPackets = (maxBufferSize + (maxPacketSize-1))/maxPacketSize; 7   fLimit = maxNumPackets*maxPacketSize; 8   fBuf = new unsigned char[fLimit]; 9   resetPacketStart();10   resetOffset();11   resetOverflowData();12 }

可以看出，fLimit的大小取决于maxNumPackets和maxPacketSize，maxPacketSize的值是在MultiFramedRTPSink类的构造函数中设置：

 1 MultiFramedRTPSink::MultiFramedRTPSink(UsageEnvironment& env, 2                        Groupsock* rtpGS, 3                        unsigned char rtpPayloadType, 4                        unsigned rtpTimestampFrequency, 5                        char const* rtpPayloadFormatName, 6                        unsigned numChannels) 7   : RTPSink(env, rtpGS, rtpPayloadType, rtpTimestampFrequency, 8         rtpPayloadFormatName, numChannels), 9     fOutBuf(NULL), fCurFragmentationOffset(0), fPreviousFrameEndedFragmentation(False),10     fOnSendErrorFunc(NULL), fOnSendErrorData(NULL) {11   setPacketSizes(1000, 1448);12       // Default max packet size (1500, minus allowance for IP, UDP, UMTP headers)13       // (Also, make it a multiple of 4 bytes, just in case that matters.)14 }15 16 void MultiFramedRTPSink::setPacketSizes(unsigned preferredPacketSize,17                     unsigned maxPacketSize) {18   if (preferredPacketSize > maxPacketSize || preferredPacketSize == 0) return;19       // sanity check20 21   delete fOutBuf;22   fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);23   fOurMaxPacketSize = maxPacketSize; // save value, in case subclasses need it24 }

可以看出maxPacketSize的大小默认值是1448，则fLimit太小就说明了maxBufferSize太小，maxBufferSize = maxSize，因为在OutPacketBuffer类的构造函数声明中可以看到maxBufferSize默认值是0，然后就会被赋值maxSize。而maxSize是OutPacketBuffer类的一个static的成员，因此，只要把OutPacketBuffer::maxSize的值设大一些就可以了。经过测试，我发现设置成300000时就可以转发1080p的高清视频。

　　（2）当向live555请求某一路视频资源的VLC客户端的数量减少到0时，live555会给出以下错误信息 RTCPInstance error: Hit limit when reading incoming packet over TCP. Increase "maxRTCPPacketSize"，我们找到此提示信息在RTCPInstance::incomingReportHandler1函数的最开头。提示信息让我们增大maxRTCPPacketSize的值，可是无论我怎么增大都还是会出现这个信息，无奈不知如何解决，然后觉得关于RTCP的一些包不去处理应该不会对转发数据有太大影响，但这样不停的提示总是很烦的，于是就采用了以下办法：

 1 void RTCPInstance::incomingReportHandler1()  2 { 3   do { 4     if (fNumBytesAlreadyRead >= maxRTCPPacketSize) { 5        memset(fInBuf,0,fNumBytesAlreadyRead); 6        fNumBytesAlreadyRead = 0; 7        break; 8     } 9     10    /*11     ......................     略去12      13    */14 }

　　（3）在live555ProxyServer.cpp的main函数中有一个输入参数是streamRTPOverTCP，streamRTPOverTCP默认是false。

　　首先，想要外网的客户端能访问流媒体服务器，则必须将streamRTPOverTCP设置为True；

      其次，想要转发外网的摄像机，也必须将streamRTPOverTCP设置为True。

　　（4）在ProxyServerMediaSession.cpp文件的ProxyServerMediaSubsession::closeStreamSource函数中，我们需要注释掉if(fHaveSetupStream)这个if语句，因为对于转发实时视频是不支持PAUSE命令的。如果不注释，当请求某一路实时视频的VLC客户端数目减少到0，再有VLC客户端重新请求该视频时就无法再播放了。

　　（5）对于同一路视频流，当请求的VLC客户端越来越多时，会发现后面请求的VLC客户端正在播放但没有图像。我们找到RTPInterface.cpp文件中RTPInterface的构造函数，注释掉其中调用makeSokcetNonBlocking函数的那一句即可。

 1 RTPInterface::RTPInterface(Medium* owner, Groupsock* gs) 2   : fOwner(owner), fGS(gs), 3     fTCPStreams(NULL), 4     fNextTCPReadSize(0), fNextTCPReadStreamSocketNum(-1), 5     fNextTCPReadStreamChannelId(0xFF), fReadHandlerProc(NULL), 6     fAuxReadHandlerFunc(NULL), fAuxReadHandlerClientData(NULL) { 7   // Make the socket non-blocking, even though it will be read from only asynchronously, when packets arrive. 8   // The reason for this is that, in some OSs, reads on a blocking socket can (allegedly) sometimes block, 9   // even if the socket was previously reported (e.g., by "select()") as having data available.10   // (This can supposedly happen if the UDP checksum fails, for example.)11         12   //makeSocketNonBlocking(fGS->socketNum());           //注释掉这一句13   increaseSendBufferTo(envir(), fGS->socketNum(), 50*1024);14 }

　　makeSocketNonBlocking这个函数顾名思义是使某个Socket成为非阻塞式的，在RTPInterface构造函数调用的这一句就是使发送RTP包给VLC客户端的Socket成为非阻塞。由于多个VLC客户端共享RTPInteface缓冲区中的RTP数据，那么当从IPCamera获取数据的速率要快于将缓冲区中的数据发送给所有VLC客户端的速率时（这种情况应该只可能发生在局域网内的测试环境，在生产环境中建议还是不要注释这一句了），缓冲区的数据就会被冲刷导致后面播放的VLC客户端播放不出图像。将makeSocketNonBlocking这一句注释掉后，就会等到给所有的VLC客户端都发送完数据后才会再从IPCamera获取数据。

　　（6）在OnDemandServerMediaSubsession.cpp文件中，找到OnDemandServerMediaSubsession::deleteStream函数

 1 void OnDemandServerMediaSubsession::deleteStream(unsigned clientSessionId, 2                          void*& streamToken) { 3   StreamState* streamState = (StreamState*)streamToken; 4  5   // Look up (and remove) the destinations for this client session: 6   Destinations* destinations 7     = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId)); 8   if (destinations != NULL) { 9     fDestinationsHashTable->Remove((char const*)clientSessionId);10 11     // Stop streaming to these destinations:12     if (streamState != NULL) streamState->endPlaying(destinations);13   }14 15   // Delete the "StreamState" structure if it's no longer being used:16   if (streamState != NULL) {17     if (streamState->referenceCount() > 0) --streamState->referenceCount();18     if (streamState->referenceCount() == 0) { //将这一句修改为if(streamState->referenceCount() == 0 && fParentSession->deleteWhenUnreferenced())19       delete streamState;20       streamToken = NULL;21     }22   }23 24   // Finally, delete the destinations themselves:25   delete destinations;26 }

　　将if(streamState->referenceCount() == 0)修改为 if(streamState->referenceCount() == 0 && fParentSession->deleteWhenUnreferenced())。修改之前，当请求某路视频资源的VLC客户端的数目减少到0时，就会delete streamState，即释放与该路视频流相关的资源，这样下次再有VLC客户端请求该路视频资源时，就需要重新申请资源，速度会比较慢。而且，在Windows下测试发现，执行delete streamState这一句时偶尔会发生异常崩溃。

　　ProxyServerMediaSubsession是OnDemandServerMediaSubsession的子类，但对于ProxyServerMediaSubsession而言，我们可以在请求该路视频流的VLC客户端数目减少到0时不释放相关资源，这样后面再有VLC客户端请求时速度就会加快。ServerMeiaSubsessio类中会保存有父会话ServerMediaSession的指针，ServerMediaSession类有一个属性fDeleteWhenUnreferenced，这个属性表示当不再被请求时是否删除会话并释放资源，默认是false。

　　（7）楼主本想使用此程序开发出一个Live555的代理服务器，结果发现在局域网内，Live555转发IPCamera的视频给VLC客户端，延时都将近3s（这个地方的延时是指VLC点击播放按钮后要等3s才能出来图像），楼主最终也找不到解决的办法（听说出不来图像是因为还没有I帧，但楼主水平有限，对视频编码什么的一窍不通）。哪位仁兄找到解决办法请联系我，谢谢。

     （8）问题（7）后来发现是VLC播放器的用法不正确，网上说是设置缓存时间的问题，在局域网内设置缓存时间是有些效果，但对于客户端在外网访问流媒体服务器，还是很慢，后来发现在"工具-首选项"中设置"live555流传输"方式为"RTP over RTSP"即可。如下图所示：

　　此问题刚解决后，又发现一个由此而来的新问题：对于同一路流，后面请求的VLC客户端会把前面请求的VLC客户端"挤掉"，具体表现是，后开启的客户端开始播放画面时，前面请求的客户端的画面就停止不动了，然后紧接着就和流媒体服务器断开了连接。并且，在之前将"Live555流传输"设置为"HTTP"时没有这个问题。

　　后来发现将RTPInterface::sendRTPorRTCPPacketOverTCP函数中的if(!sendDataOverTCP(socketNum,framingHeader,4,False))中的False修改为True即可。