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 name
35 } else {
36 sprintf(streamName, "proxyStream-%d", i); // there's more than one stream; distinguish them by name
37 }
38 ServerMediaSession* sms
39 = 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 HTTP
56 // 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 return
66
67 return 0; // only to prevent compiler warning
68 }
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) stream
10 int verbosityLevel = 0,
11 int socketNumToServer = -1);
12 // Hack: "tunnelOverHTTPPortNum" == 0xFFFF (i.e., all-ones) means: Stream RTP/RTCP-over-TCP, but *not* using HTTP
13 // "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* ourCreateNewProxyRTSPClientFunc
34 = defaultCreateNewProxyRTSPClientFunc);
35
36 // If you subclass "ProxyRTSPClient", then you will also need to define your own function
37 // - with signature "createNewProxyRTSPClientFunc" (see above) - that creates a new object
38 // of this subclass. You should also subclass "ProxyServerMediaSession" and, in your
39 // 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 seconds
40 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 ProxyServerMediaSession
12 ::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 fProxyRTSPClient
26 = (*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 the
11 // 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 packets
14 // 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 that
26 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 previously
46 // indicated (in its response to our earlier "OPTIONS" command) that it supported "GET_PARAMETER". This is because
47 // "GET_PARAMETER" crashes some camera servers (even though they claimed to support "GET_PARAMETER").
48 #ifdef SEND_GET_PARAMETER_IF_SUPPORTED
49 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 #endif
55 rtspClient->sendOptionsCommand(::continueAfterOPTIONS, rtspClient->auth());
56 #ifdef SEND_GET_PARAMETER_IF_SUPPORTED
57 }
58 #endif
59 }
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] seconds
69 }
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 subsession
25 ProxyServerMediaSubsession* fNext; // used when we're part of a queue
26 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 #endif
112 } 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 a
168 // 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 default
186 Boolean doNormalMBitRule = True; // by default
187 // Some codecs change the above default parameters:
188 if (strcmp(codecName, "MP2T") == 0) {
189 doNormalMBitRule = False; // no RTP 'M' bit
190 }
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 source
14 //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 ready
17 }
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 ready
23 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 set
11
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: != NULL
14 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 done
30 // 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's
34 // 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);
38
39 }
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 check
20
21 delete fOutBuf;
22 fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);
23 fOurMaxPacketSize = maxPacketSize; // save value, in case subclasses need it
24 }
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。
作者：昨夜星辰
- 该日志由 求索阁 于5年前发表在工业·编程分类下，最后更新于 2020年11月21日.
- 转载请注明: Live555学习之(五):live555ProxyServer.cpp的学习 | 求索阁 +复制链接
- 关键字: live555