Live555學習之(三)------建立RTSP連接的過程（RTSP服務器端）

本文轉載自查看原文 2015-04-05 12:26 13302 Live555學習

　　上一篇我們簡單分析了testOnDemandRTSPServer.cpp的main函數，主要步驟是創建RTSPServer，創建ServerMediaSession對象，然后等待RTSP客戶端的連接。接下來我們分析一下Live555中建立RTSP連接的詳細過程，首先我們需要簡單了解一下RTSP協議建立連接的過程：

　　1.（可選）

RTSP客戶端 —> RTSP服務器端 OPTIONS命令詢問服務器端有哪些方法可使用

RTSP服務器端 —> RTSP客戶端回復OPTIONS命令回復客戶端服務器支持的方法

　　2. （可選）

　　 RTSP客戶端 —> RTSP服務器端 DESCRIBE命令　　　請求對某個媒體資源（Live555中用ServerMediaSession表示）的描述信息

RTSP服務器端 —> RTSP客戶端　　回復DESCRIBE命令　回復客戶端某個媒體資源的描述信息（即SDP）

3. （必選）

RTSP客戶端 —> RTSP服務器端　 SETUP命令　　　　　請求建立對某個媒體資源的連接

　　 RTSP服務器端 —> RTSP客戶端　　回復SETUP命令　　　回復建立連接的結果

4. （必選）

RTSP客戶端 —> RTSP服務器端　 PLAY命令　　　　　　請求播放媒體資源

　　 RTSP服務器端 —> RTSP客戶端　　回復PLAY命令　　　　回復播放的結果

--------------------RTSP服務器端發送RTP包（封裝了數據）給RTSP客戶端-------------------------------

　　下面我們從RTSPServer::incomingConnectionHandlerRTSP函數開始，在incomingConnectionHandlerRTSP函數中又調用了RTSPServer::incomingConnectionHandler函數，在這個函數中accept客戶端的TCP連接，然后調用RTSPServer::createNewClientConnection函數創建一個RTSPClientConnection實例，該實例表示一個與客戶端的RTSP連接。

 1 RTSPServer::RTSPClientConnection
 2 ::RTSPClientConnection(RTSPServer& ourServer, int clientSocket, struct sockaddr_in clientAddr)
 3   : fOurServer(ourServer), fIsActive(True),
 4     fClientInputSocket(clientSocket), fClientOutputSocket(clientSocket), fClientAddr(clientAddr),
 5     fRecursionCount(0), fOurSessionCookie(NULL) {
 6   // Add ourself to our 'client connections' table:  把這個RTSPClientConnection實例添加到RTSPServer的列表中
 7   fOurServer.fClientConnections->Add((char const*)this, this);
 8   
 9   // Arrange to handle incoming requests:
10   resetRequestBuffer();
11   envir().taskScheduler().setBackgroundHandling(fClientInputSocket, SOCKET_READABLE|SOCKET_EXCEPTION,
12                         (TaskScheduler::BackgroundHandlerProc*)&incomingRequestHandler, this);
13 }

　　RTSPClientConnection的構造函數中，將自己添加到RTSPServer的連接列表中，然后將客戶端socket添加到SOCKET SET中，並且設置相應的回調處理函數incomingRequestHandler，然后就開始等待客戶端發送命令。服務器端收到客戶端的命令即回調RTSPClientConnection::incomingRequestHandler來處理。

　　在RTSPClientConnection::incomingRequestHandler函數中又調用RTSPClientConnection::incomingRequestHandler1函數，在這個函數中，從客戶端socket中讀取數據，讀取的數據存儲在RTSPClientConnection::fRequestBuffer這個數組中，然后調RTSPClientConnection::handleRequestBytes函數處理剛才讀到的數據。handleRequestBytes函數的內容（比較多）主要是分析讀取的數據，提取出命令名等數據，然后根據不同的命令調用不同的函數去處理，將處理后的結果保存在fResponseBuffer這個數組中，然后發送給客戶端。在此，我們假設客戶端跳過OPTINS命令，直接發送DESCRIBE命令請求建立連接，則在handleRequestBytes函數中會調用RTSPClientConnection::handleCmd_DESCRIBE函數來處理，下面來看一下handleCmd_DESCRIBE函數。先說一下urlPreSuffix和urlSuffix吧，假設客戶端請求媒體資源的RTSP地址是rtsp://127.0.0.1:8554/test1/test2/test.264，urlPreSuffix表示的是ip:port之后（不含緊跟的“/”）到最后一個“/”之前的部分，即test1/test2，urlSuffix表示的是最后一個“/”之后（不含緊跟的“/”）的內容，即test.264。

 1 void RTSPServer::RTSPClientConnection
 2 ::handleCmd_DESCRIBE(char const* urlPreSuffix, char const* urlSuffix, char const* fullRequestStr) {
 3   char* sdpDescription = NULL;
 4   char* rtspURL = NULL;
 5   do {
 6     char urlTotalSuffix[RTSP_PARAM_STRING_MAX];
 7     if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2 > sizeof urlTotalSuffix) {
 8       handleCmd_bad();
 9       break;
10     }
11     urlTotalSuffix[0] = '\0';                              // 拼接urlPreSuffix和urlSuffix，保存在urlTotalSuffix中 12     if (urlPreSuffix[0] != '\0') {
13       strcat(urlTotalSuffix, urlPreSuffix);
14       strcat(urlTotalSuffix, "/");
15     }
16     strcat(urlTotalSuffix, urlSuffix);
17     
18     if (!authenticationOK("DESCRIBE", urlTotalSuffix, fullRequestStr)) break;
19     
20     // We should really check that the request contains an "Accept:" #####
21     // for "application/sdp", because that's what we're sending back #####
22     
23     // Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix":
24     ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);                 // 在RTSPServer中查找對應的ServerMediaSession 25     if (session == NULL) {
26       handleCmd_notFound();
27       break;
28     }
29     
30     // Then, assemble a SDP description for this session:
31     sdpDescription = session->generateSDPDescription();                                                // 產生SDP描述信息字符串 32     if (sdpDescription == NULL) {
33       // This usually means that a file name that was specified for a
34       // "ServerMediaSubsession" does not exist.
35       setRTSPResponse("404 File Not Found, Or In Incorrect Format");
36       break;
37     }
38     unsigned sdpDescriptionSize = strlen(sdpDescription);
39     
40     // Also, generate our RTSP URL, for the "Content-Base:" header
41     // (which is necessary to ensure that the correct URL gets used in subsequent "SETUP" requests).
42     rtspURL = fOurServer.rtspURL(session, fClientInputSocket);
43     
44     snprintf((char*)fResponseBuffer, sizeof fResponseBuffer,                              // 構造回復信息 45          "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
46          "%s"
47          "Content-Base: %s/\r\n"
48          "Content-Type: application/sdp\r\n"
49          "Content-Length: %d\r\n\r\n"
50          "%s",
51          fCurrentCSeq,
52          dateHeader(),
53          rtspURL,
54          sdpDescriptionSize,
55          sdpDescription);
56   } while (0);
57   
58  59 
60   delete[] sdpDescription;
61   delete[] rtspURL;
62 }

　　在handleCmd_DESCRIBE函數中，主要調用了ServerMediaSession::generateSDPDescription函數產生SDP信息，ServerMediaSession的SDP信息由每個ServerMediaSubsession的SDP信息構成，然后將產生的SDP回復給客戶端。我們就來看一下generateSDPDescription函數。

  1 char* ServerMediaSession::generateSDPDescription() {
  2   AddressString ipAddressStr(ourIPAddress(envir()));
  3   unsigned ipAddressStrSize = strlen(ipAddressStr.val());
  4 
  5   // For a SSM sessions, we need a "a=source-filter: incl ..." line also:
  6   char* sourceFilterLine;
  7   if (fIsSSM) {
  8     char const* const sourceFilterFmt =
  9       "a=source-filter: incl IN IP4 * %s\r\n"
 10       "a=rtcp-unicast: reflection\r\n";
 11     unsigned const sourceFilterFmtSize = strlen(sourceFilterFmt) + ipAddressStrSize + 1;
 12 
 13     sourceFilterLine = new char[sourceFilterFmtSize];
 14     sprintf(sourceFilterLine, sourceFilterFmt, ipAddressStr.val());
 15   } else {
 16     sourceFilterLine = strDup("");
 17   }
 18 
 19   char* rangeLine = NULL; // for now
 20   char* sdp = NULL; // for now
 21 
 22   do {
 23     // Count the lengths of each subsession's media-level SDP lines.
 24     // (We do this first, because the call to "subsession->sdpLines()"
 25     // causes correct subsession 'duration()'s to be calculated later.)
  
        //首先調用每個ServerMediaSubsession的sdpLines函數，用來計算sdp的長度
 26     unsigned sdpLength = 0;
 27     ServerMediaSubsession* subsession;
 28     for (subsession = fSubsessionsHead; subsession != NULL;
 29      subsession = subsession->fNext) {
 30       char const* sdpLines = subsession->sdpLines();
 31       if (sdpLines == NULL) continue; // the media's not available
 32       sdpLength += strlen(sdpLines);
 33     }
 34     if (sdpLength == 0) break; // the session has no usable subsessions
 35 
 36     // Unless subsessions have differing durations, we also have a "a=range:" line:
        // 計算ServerMediaSession的持續時間，該返回值影響a=range字段，該字段決定了該媒體資源是否可以執行快進、快退、任意進度點播，ServerMediaSession的duration由各個ServerMediaSubsession的duration決定。
 // ServerMediaSubsession的duration默認實現是返回0，Live555只對部分格式的媒體文件實現了duration函數，如MKV文件的MatroskaFileServerMediaSubsession分析了mkv文件的播放時長
 37     float dur = duration();　 38     if (dur == 0.0) {
 39       rangeLine = strDup("a=range:npt=0-\r\n");
 40     } else if (dur > 0.0) {
 41       char buf[100];
 42       sprintf(buf, "a=range:npt=0-%.3f\r\n", dur);
 43       rangeLine = strDup(buf);
 44     } else { // subsessions have differing durations, so "a=range:" lines go there
 45       rangeLine = strDup("");
 46     }
 47 
 48     char const* const sdpPrefixFmt =
 49       "v=0\r\n"
 50       "o=- %ld%06ld %d IN IP4 %s\r\n"
 51       "s=%s\r\n"
 52       "i=%s\r\n"
 53       "t=0 0\r\n"
 54       "a=tool:%s%s\r\n"
 55       "a=type:broadcast\r\n"
 56       "a=control:*\r\n"
 57       "%s"
 58       "%s"
 59       "a=x-qt-text-nam:%s\r\n"
 60       "a=x-qt-text-inf:%s\r\n"
 61       "%s";
 62     sdpLength += strlen(sdpPrefixFmt)
 63       + 20 + 6 + 20 + ipAddressStrSize
 64       + strlen(fDescriptionSDPString)
 65       + strlen(fInfoSDPString)
 66       + strlen(libNameStr) + strlen(libVersionStr)
 67       + strlen(sourceFilterLine)
 68       + strlen(rangeLine)
 69       + strlen(fDescriptionSDPString)
 70       + strlen(fInfoSDPString)
 71       + strlen(fMiscSDPLines);
 72     sdpLength += 1000; // in case the length of the "subsession->sdpLines()" calls below change
 73     sdp = new char[sdpLength];
 74     if (sdp == NULL) break;
 75 
 76     // Generate the SDP prefix (session-level lines):
 77     snprintf(sdp, sdpLength, sdpPrefixFmt,
 78          fCreationTime.tv_sec, fCreationTime.tv_usec, // o= <session id>
 79          1, // o= <version> // (needs to change if params are modified)
 80          ipAddressStr.val(), // o= <address>
 81          fDescriptionSDPString, // s= <description>
 82          fInfoSDPString, // i= <info>
 83          libNameStr, libVersionStr, // a=tool:
 84          sourceFilterLine, // a=source-filter: incl (if a SSM session)
 85          rangeLine, // a=range: line
 86          fDescriptionSDPString, // a=x-qt-text-nam: line
 87          fInfoSDPString, // a=x-qt-text-inf: line
 88          fMiscSDPLines); // miscellaneous session SDP lines (if any)
 89 
 90     // Then, add the (media-level) lines for each subsession:  
        // 再次調用每個ServerMediaSubsession的sdpLines函數，這次真正將每個ServerMediaSubsession的sdp信息添加到ServerMediaSession的SDP信息中
 91     char* mediaSDP = sdp;
 92     for (subsession = fSubsessionsHead; subsession != NULL;
 93      subsession = subsession->fNext) {
 94       unsigned mediaSDPLength = strlen(mediaSDP);
 95       mediaSDP += mediaSDPLength;　　　　　　　　　　　　　　　　　　　　　　// 指針后移  96       sdpLength -= mediaSDPLength;
 97       if (sdpLength <= 1) break; // the SDP has somehow become too long
 98 
 99       char const* sdpLines = subsession->sdpLines();
100       if (sdpLines != NULL) snprintf(mediaSDP, sdpLength, "%s", sdpLines);
101     }
102   } while (0);
103 
104   delete[] rangeLine; delete[] sourceFilterLine;
105   return sdp;
106 }

　　到此，服務器端將客戶端請求的SDP信息發送給客戶端，然后等着客戶端發送下一個命令（SETUP命令），在分析服務器端如何處理SETUP命令之前，我們繼續深入看一下服務器端是如何獲得SDP信息的。從generateSDPDescription函數中可以看到，主要是調用了每個ServerMediaSubsession的sdpLines函數，默認實現在OnDemandServerMediaSubsession這個類中，下面我們就來看看OnDemandServerMediaSubsession::sdpLines函數。

 1 char const* OnDemandServerMediaSubsession::sdpLines() {
 2   if (fSDPLines == NULL) {
 3     // We need to construct a set of SDP lines that describe this
 4     // subsession (as a unicast stream).  To do so, we first create
 5     // dummy (unused) source and "RTPSink" objects,
 6     // whose parameters we use for the SDP lines:
       // 這幾句話的意思是說，為了獲得這個ServerMediaSubsession的sdp信息，我們先創建“虛設的”FramedSource和RTPSink來分析出sdp信息，並非正式的開始播放
 7     unsigned estBitrate;
       // 創建FramedSource對象，用來獲取數據
 //（這里實際調用的是子類H264VideoFileServerMediaSubsession的createNewStreamSource函數，創建的是ByteStreamFileSource，ByteStreamFileSource是FramedSource的子類）  8     FramedSource* inputSource = createNewStreamSource(0, estBitrate);              // 創建FramedSource對象，獲取視頻幀數據  9     if (inputSource == NULL) return NULL; // file not found
10 
11     struct in_addr dummyAddr;
12     dummyAddr.s_addr = 0;
13     Groupsock dummyGroupsock(envir(), dummyAddr, 0, 0);
14     unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
 // 創建RTPSink對象，用來保存RTP數據包（這里實際調用的是子類H264VideoFileServerMediaSubsession的createNewRTPSink函數，創建的是H264VideoRTPSink對象，H264VideoRTPSink是RTPSink的子類）
15     RTPSink* dummyRTPSink
16       = createNewRTPSink(&dummyGroupsock, rtpPayloadType, inputSource);   
17     if (dummyRTPSink != NULL && dummyRTPSink->estimatedBitrate() > 0) estBitrate = dummyRTPSink->estimatedBitrate();
18 
19     setSDPLinesFromRTPSink(dummyRTPSink, inputSource, estBitrate);           // 通過RTPSink對象獲得ServerMediaSubsession的sdp信息
20     Medium::close(dummyRTPSink);
21     closeStreamSource(inputSource);
22   }
23 
24   return fSDPLines;
25 }

　　我們再轉到OnDemandServerMediaSubsession::setSDPLinesFromRTPSink函數，在這個函數中，我們通過創建的FramedSource對象和RTPSink對象將文件播放一段以便產生出sdp信息。在此，我要插一下Live555 RTSPServer播放媒體資源的一個大體流程：RTSPServer使用RTPSink獲得和保存RTP包，RTPSink不斷地向FramedSource請求幀數據，FramedSource取得幀數據后就調用回調函數把數據給RTPSink處理，RTPSink在回調函數中將數據發送給客戶端（也可以保存在本地存成文件，即錄像的功能）

 1 void OnDemandServerMediaSubsession
 2 ::setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource, unsigned estBitrate) {
 3   if (rtpSink == NULL) return;
 4 
     //通過RTPSink獲取各種關於該ServerMediaSubsession的信息，最主要的是獲取auxSDPLine
 5   char const* mediaType = rtpSink->sdpMediaType();
 6   unsigned char rtpPayloadType = rtpSink->rtpPayloadType();
 7   AddressString ipAddressStr(fServerAddressForSDP);
 8   char* rtpmapLine = rtpSink->rtpmapLine();
 9   char const* rtcpmuxLine = fMultiplexRTCPWithRTP ? "a=rtcp-mux\r\n" : "";
10   char const* rangeLine = rangeSDPLine();
11   char const* auxSDPLine = getAuxSDPLine(rtpSink, inputSource);
12   if (auxSDPLine == NULL) auxSDPLine = "";
13 
14   char const* const sdpFmt =
15     "m=%s %u RTP/AVP %d\r\n"
16     "c=IN IP4 %s\r\n"
17     "b=AS:%u\r\n"
18     "%s"
19     "%s"
20     "%s"
21     "%s"
22     "a=control:%s\r\n";
23   unsigned sdpFmtSize = strlen(sdpFmt)
24     + strlen(mediaType) + 5 /* max short len */ + 3 /* max char len */
25     + strlen(ipAddressStr.val())
26     + 20 /* max int len */
27     + strlen(rtpmapLine)
28     + strlen(rtcpmuxLine)
29     + strlen(rangeLine)
30     + strlen(auxSDPLine)
31     + strlen(trackId());
32   char* sdpLines = new char[sdpFmtSize];
33   sprintf(sdpLines, sdpFmt,
34       mediaType, // m= <media>
35       fPortNumForSDP, // m= <port>
36       rtpPayloadType, // m= <fmt list>
37       ipAddressStr.val(), // c= address
38       estBitrate, // b=AS:<bandwidth>
39       rtpmapLine, // a=rtpmap:... (if present)
40       rtcpmuxLine, // a=rtcp-mux:... (if present)
41       rangeLine, // a=range:... (if present)
42       auxSDPLine, // optional extra SDP line
43       trackId()); // a=control:<track-id>
44   delete[] (char*)rangeLine; delete[] rtpmapLine;
45 
46   fSDPLines = strDup(sdpLines);
47   delete[] sdpLines;
48 }

　　在setSDPLinesFromRTPSink函數中通過RTPSink對象獲得各種信息，最復雜的是獲取auxSDPLine的過程，這個函數在H264VideoFileServerMediaSubsession類中被重寫了，由於我們現在分析的媒體資源是.264文件，所以我們來看一下H264VideoFileServerMediaSubsession::getAuxSDPLine函數:

 1 char const* H264VideoFileServerMediaSubsession::getAuxSDPLine(RTPSink* rtpSink, FramedSource* inputSource) {
 2   if (fAuxSDPLine != NULL) return fAuxSDPLine; // it's already been set up (for a previous client)
 3 
 4   if (fDummyRTPSink == NULL) { // we're not already setting it up for another, concurrent stream
 5     // Note: For H264 video files, the 'config' information ("profile-level-id" and "sprop-parameter-sets") isn't known
 6     // until we start reading the file.  This means that "rtpSink"s "auxSDPLine()" will be NULL initially,
 7     // and we need to start reading data from our file until this changes.
 8     fDummyRTPSink = rtpSink;
 9 
10     // Start reading the file:   //調用RTPSink的startPlaying函數來播放，對於文件型的ServerMediaSubsession，Live555的做法是播放一段文件來獲取sdp信息
11     fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
12 
13     // Check whether the sink's 'auxSDPLine()' is ready:
14     checkForAuxSDPLine(this);
15   }
16 
17   envir().taskScheduler().doEventLoop(&fDoneFlag);  // fDoneFlag初始值為NULL，讓程序在此循環等待，直到成功分析出sdp信息 18 
19   return fAuxSDPLine;
20 }

　　在這個函數中調用RTPSink的startPlaying函數開始讀取數據，調用H264VideoFileServerMediaSubsession::checkForAuxSDPLine函數來檢查是否已經從讀取的數據中分析出sdp信息，看一下checkForAuxSDPLine函數:

 1 static void checkForAuxSDPLine(void* clientData) {
 2   H264VideoFileServerMediaSubsession* subsess = (H264VideoFileServerMediaSubsession*)clientData;
 3   subsess->checkForAuxSDPLine1();
 4 }
 5 
 6 void H264VideoFileServerMediaSubsession::checkForAuxSDPLine1() {
 7   char const* dasl;
 8 
 9   if (fAuxSDPLine != NULL) {                       //說明已經分析出了sdp信息 10     // Signal the event loop that we're done:
11     setDoneFlag();                                 // 使程序退出循環等待         
12   }  // 還沒分析出sdp信息，調用RTPSink的auxSDPLine函數分析sdp信息
     else if (fDummyRTPSink != NULL && (dasl = fDummyRTPSink->auxSDPLine()) != NULL) {
13     fAuxSDPLine = strDup(dasl);
14     fDummyRTPSink = NULL;
15 
16     // Signal the event loop that we're done:
17     setDoneFlag();　　　　　　　　　　　　　　　　　　　　// 分析出了sdp信息，使程序退出循環等待 18   } else if (!fDoneFlag)                       // 仍然沒有分析出sdp信息，則稍后一會兒再執行checkForAuxSDPLine函數 19     // try again after a brief delay:
20     int uSecsToDelay = 100000; // 100 ms
21     nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecsToDelay,
22                   (TaskFunc*)checkForAuxSDPLine, this);
23   }
24 }

　　上面檢查發現沒有分析出sdp信息后，調用H264VideoRTPSink::auxSDPLine函數再次試圖分析出sdp信息，看看auxSDPLine函數:

 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:
       //  fOurFragmenter在調用startPlaying函數后被創建
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 
15     framerSource->getVPSandSPSandPPS(vpsDummy, vpsDummySize, sps, spsSize, pps, ppsSize);
16     if (sps == NULL || pps == NULL) return NULL; // our source isn't ready
17   }
18 
     // 已經從文件里面成功讀出了數據，接下來就分析sdp信息
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 }

　　至此，RTSPServer成功地處理了客戶端發送來的DESCRIBE命令，將SDP信息回復客戶端。然后，客戶端對應每個ServerMediaSubsession發送一個SETUP命令請求建立與該ServerMediaSubsession的連接，服務器端收到后會調用RTSPClientSession::handleCmd_SETUP函數來處理SETUP命令。RTSPClientSession類是服務器端用來維護和客戶端的一個會話，SETUP命令、PLAY命令、PAUSE命令、TEARDOWN命令等都是在RTSPClientSession中處理的，RTSPClientSession是RTSPClientConnection的內部類，來看一下這個類：

  1  // The state of an individual client session (using one or more sequential TCP connections) handled by a RTSP server:
  2    class RTSPClientSession {
  3    protected:
  4      RTSPClientSession(RTSPServer& ourServer, u_int32_t sessionId);
  5      virtual ~RTSPClientSession();
  6  
  7       friend class RTSPServer;
  8       friend class RTSPClientConnection;
  9       // Make the handler functions for each command virtual, to allow subclasses to redefine them:
 10      virtual void handleCmd_SETUP(RTSPClientConnection* ourClientConnection,　　　　　　　　　　　　　　　　　　// 處理SETUP命令
 11                   char const* urlPreSuffix, char const* urlSuffix, char const* fullRequestStr);
 12      virtual void handleCmd_withinSession(RTSPClientConnection* ourClientConnection,
 13                       char const* cmdName,
 14                       char const* urlPreSuffix, char const* urlSuffix,
 15                       char const* fullRequestStr);
 16      virtual void handleCmd_TEARDOWN(RTSPClientConnection* ourClientConnection,　　　　　　　　　　　　　　　　// 處理TEARDOWN命令（結束會話）
 17                      ServerMediaSubsession* subsession);
 18      virtual void handleCmd_PLAY(RTSPClientConnection* ourClientConnection,　　　　　　　　　　　　　　　　　　 // 處理PLAY命令
 19                  ServerMediaSubsession* subsession, char const* fullRequestStr);
 20      virtual void handleCmd_PAUSE(RTSPClientConnection* ourClientConnection,　　　　　　　　　　　　　　　　　　// 處理PAUSE命令
 21                   ServerMediaSubsession* subsession);
 22      virtual void handleCmd_GET_PARAMETER(RTSPClientConnection* ourClientConnection,
 23                       ServerMediaSubsession* subsession, char const* fullRequestStr);
 24      virtual void handleCmd_SET_PARAMETER(RTSPClientConnection* ourClientConnection,
 25                       ServerMediaSubsession* subsession, char const* fullRequestStr);
 26    protected:
 27      UsageEnvironment& envir() { return fOurServer.envir(); }
 28      void reclaimStreamStates();
 29      Boolean isMulticast() const { return fIsMulticast; }
 30      void noteLiveness();
 31      static void noteClientLiveness(RTSPClientSession* clientSession); 　　　　　　　　　　// 與客戶端的心跳
 32      static void livenessTimeoutTask(RTSPClientSession* clientSession);
 33  
 34      // Shortcuts for setting up a RTSP response (prior to sending it):
 35      void setRTSPResponse(RTSPClientConnection* ourClientConnection, char const* responseStr) { ourClientConnection->setRTSPResponse(responseStr); }
 36      void setRTSPResponse(RTSPClientConnection* ourClientConnection, char const* responseStr, u_int32_t sessionId) { ourClientConnection->setRTSPResponse(responseStr, sessionId); }
 37      void setRTSPResponse(RTSPClientConnection* ourClientConnection, char const* responseStr, char const* contentStr) { ourClientConnection->setRTSPResponse(responseStr, contentStr); }
 38      void setRTSPResponse(RTSPClientConnection* ourClientConnection, char const* responseStr, u_int32_t sessionId, char const* contentStr) { ourClientConnection->setRTSPResponse(responseStr, sessionId, contentStr); }
 39  
 40    protected:
 41      RTSPServer& fOurServer;
 42      u_int32_t fOurSessionId;
 43      ServerMediaSession* fOurServerMediaSession;
 44      Boolean fIsMulticast, fStreamAfterSETUP;
 45      unsigned char fTCPStreamIdCount; // used for (optional) RTP/TCP
 46      Boolean usesTCPTransport() const { return fTCPStreamIdCount > 0; }
 47      TaskToken fLivenessCheckTask;
 48      unsigned fNumStreamStates;　　　　　　　　// streamState對象的數目
 49      struct streamState {　　　　　　　　　　　　　　　　　　　　　　　　// streamState結構體，保存一個請求的ServerMediaSubsession以及對應的StreamState對象
 50        ServerMediaSubsession* subsession;
 51        void* streamToken;　　　　　　　　　　　// streamToken指向一個StreamState對象
 52      } * fStreamStates;                     // fStreamStates是streamState數組
 53    };
 54  
      /* StreamState類從名字上就可以看出是服務器端用來保存對某個ServerMediaSubsession的流化的狀態（包括serverRTPPort、serverRTCPPort、rtpSink、mediaSource等）
         當某個ServerMediaSubsession被客戶端請求SETUP時，服務器端會創建一個StreamState對象，並創建相關的服務器端socket、RTPSink、FramedSource為后面的播放做好准備，
　　　　　在創建一個ServerMediaSubsession對象時（詳情見testOnDemandRTSPServer.cpp的main函數），會傳入reuseFirstSource這個參數。如果reuseFirstSource為true，
　　　　　則表示對於請求該ServerMediaSubsession的所有客戶端都使用同一個StreamState對象，即服務器端使用同一個RTP端口、RTCP端口、RTPSink、FramedSource來為請求該
　　　　　ServerMediaSubsession的多個客戶端服務（一對多，節省服務器端資源）；而如果reuseFirstSource為false，則服務器端為每個對ServerMediaSubsession的請求創建一個StreamState對象（多對多，需要占用服務器端較多資源） */
 55 class StreamState {                                        // StreamState類，表示服務器端對一個ServerMediaSubsession的一次流化，並保存相關狀態  56 public:
 57   StreamState(OnDemandServerMediaSubsession& master,
 58               Port const& serverRTPPort, Port const& serverRTCPPort,
 59           RTPSink* rtpSink, BasicUDPSink* udpSink,
 60           unsigned totalBW, FramedSource* mediaSource,
 61           Groupsock* rtpGS, Groupsock* rtcpGS);
 62   virtual ~StreamState();
 63 
 64   void startPlaying(Destinations* destinations,　　　　　　　　　　　// 開始播放，服務器端在收到PLAY命令后，就是調用各個StreamState的startPlaying函數來開始播放一個ServerMediaSubsession  65             TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
 66             ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
 67                     void* serverRequestAlternativeByteHandlerClientData);
 68   void pause();
 69   void endPlaying(Destinations* destinations);　　　　　　　　　　　// 結束播放  70   void reclaim();
 71 
 72   unsigned& referenceCount() { return fReferenceCount; }         // 引用該路流的客戶端數目  73 
 74   Port const& serverRTPPort() const { return fServerRTPPort; }
 75   Port const& serverRTCPPort() const { return fServerRTCPPort; }
 76 
 77   RTPSink* rtpSink() const { return fRTPSink; }
 78 
 79   float streamDuration() const { return fStreamDuration; }
 80 
 81   FramedSource* mediaSource() const { return fMediaSource; }
 82   float& startNPT() { return fStartNPT; }
 83 
 84 private:
 85   OnDemandServerMediaSubsession& fMaster;
 86   Boolean fAreCurrentlyPlaying;
 87   unsigned fReferenceCount;
 88 
 89   Port fServerRTPPort, fServerRTCPPort;
 90 
 91   RTPSink* fRTPSink;
 92   BasicUDPSink* fUDPSink;
 93 
 94   float fStreamDuration;
 95   unsigned fTotalBW;
 96   RTCPInstance* fRTCPInstance;
 97 
 98   FramedSource* fMediaSource;
 99   float fStartNPT; // initial 'normal play time'; reset after each seek
100 
101   Groupsock* fRTPgs;
102   Groupsock* fRTCPgs;
103 };

　　接下來，在handleCmd_SETUP函數中，服務器首先找到客戶端請求的ServerMediaSession，再找到客戶端請求的ServerMediaSubsession，然后從客戶端的請求中獲取一些客戶端參數（如：客戶端的RTP端口、RTCP端口），最后調用OnDemandServerMediaSubsession::getStreamParameters函數創建RTP連接和RTCP連接。看一下getStreamParameters函數：

  1 void OnDemandServerMediaSubsession
  2 ::getStreamParameters(unsigned clientSessionId,netAddressBits clientAddress,Port const& clientRTPPort,
  3               Port const& clientRTCPPort,int tcpSocketNum,unsigned char rtpChannelId,
  4               unsigned char rtcpChannelId,netAddressBits& destinationAddress,u_int8_t& /*destinationTTL*/,
  5               Boolean& isMulticast,Port& serverRTPPort,Port& serverRTCPPort,void*& streamToken) {
  6   if (destinationAddress == 0) destinationAddress = clientAddress;
  7   struct in_addr destinationAddr; destinationAddr.s_addr = destinationAddress;
  8   isMulticast = False;
  9 
 10   if (fLastStreamToken != NULL && fReuseFirstSource) {                       // 如果fReuseFirstSource為true，則使用之前已經創建的StreamState對象  11     // Special case: Rather than creating a new 'StreamState',
 12     // we reuse the one that we've already created:
 13     serverRTPPort = ((StreamState*)fLastStreamToken)->serverRTPPort();
 14     serverRTCPPort = ((StreamState*)fLastStreamToken)->serverRTCPPort();
 15     ++((StreamState*)fLastStreamToken)->referenceCount();
 16     streamToken = fLastStreamToken;
 17   } else {　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　// 對於該ServerMediaSubsession尚未創建StreamState對象，或者fReuseFirstSource為false  18     // Normal case: Create a new media source:
 19     unsigned streamBitrate;
 20     FramedSource* mediaSource　　　　　　// 創建FramedSource對象，實際調用的是子類的createNewStreamSource函數，對應H264VideoFileServerMediaSubsession創建的是ByteStreamFileSource  21       = createNewStreamSource(clientSessionId, streamBitrate);
 22 
 23     // Create 'groupsock' and 'sink' objects for the destination,
 24     // using previously unused server port numbers:
 25     RTPSink* rtpSink = NULL;
 26     BasicUDPSink* udpSink = NULL;
 27     Groupsock* rtpGroupsock = NULL;
 28     Groupsock* rtcpGroupsock = NULL;
 29 
 30     if (clientRTPPort.num() != 0 || tcpSocketNum >= 0) { // Normal case: Create destinations
 31       portNumBits serverPortNum;
 32       if (clientRTCPPort.num() == 0) {
 33     // We're streaming raw UDP (not RTP). Create a single groupsock:
 34     NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
 35     for (serverPortNum = fInitialPortNum; ; ++serverPortNum) {
 36       struct in_addr dummyAddr; dummyAddr.s_addr = 0;
 37       
 38       serverRTPPort = serverPortNum;
 39       rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
 40       if (rtpGroupsock->socketNum() >= 0) break; // success
 41     }
 42 
 43     udpSink = BasicUDPSink::createNew(envir(), rtpGroupsock);
 44       } else {　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　// 創建兩個服務器端socket用來傳輸RTP包和RTCP包，對應rtpGroupsock和rtcpGroupsock  45     // Normal case: We're streaming RTP (over UDP or TCP).  Create a pair of
 46     // groupsocks (RTP and RTCP), with adjacent port numbers (RTP port number even).
 47     // (If we're multiplexing RTCP and RTP over the same port number, it can be odd or even.)
 48     NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
 49     for (portNumBits serverPortNum = fInitialPortNum; ; ++serverPortNum) {
 50       struct in_addr dummyAddr; dummyAddr.s_addr = 0;
 51 
 52       serverRTPPort = serverPortNum;
 53       rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
 54       if (rtpGroupsock->socketNum() < 0) {
 55         delete rtpGroupsock;
 56         continue; // try again
 57       }
 58 
 59       if (fMultiplexRTCPWithRTP) {
 60         // Use the RTP 'groupsock' object for RTCP as well:
 61         serverRTCPPort = serverRTPPort;
 62         rtcpGroupsock = rtpGroupsock;
 63       } else {
 64         // Create a separate 'groupsock' object (with the next (odd) port number) for RTCP:
 65         serverRTCPPort = ++serverPortNum;
 66         rtcpGroupsock = new Groupsock(envir(), dummyAddr, serverRTCPPort, 255);
 67         if (rtcpGroupsock->socketNum() < 0) {
 68           delete rtpGroupsock;
 69           delete rtcpGroupsock;
 70           continue; // try again
 71         }
 72       }
 73 
 74       break; // success
 75     }
 76 
 77     unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
         //創建RTPSink，實際調用的是子類的createNewRTPSink函數，對應H264VideoFileServerMediaSubsession創建的是H264VideoRTPSink
 78     rtpSink = createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);
 79     if (rtpSink != NULL && rtpSink->estimatedBitrate() > 0) streamBitrate = rtpSink->estimatedBitrate();    80       }
 81 
 82       // Turn off the destinations for each groupsock.  They'll get set later
 83       // (unless TCP is used instead):
 84 
 85       if (rtpGroupsock != NULL) rtpGroupsock->removeAllDestinations();
 86       if (rtcpGroupsock != NULL) rtcpGroupsock->removeAllDestinations();
 87 
 88       if (rtpGroupsock != NULL) {
 89     // Try to use a big send buffer for RTP -  at least 0.1 second of
 90     // specified bandwidth and at least 50 KB
 91     
 92     unsigned rtpBufSize = streamBitrate * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
 93     if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
 94     increaseSendBufferTo(envir(), rtpGroupsock->socketNum(), rtpBufSize);
 95       }
 96     }
 97 
 98     // Set up the state of the stream.  The stream will get started later:
 99     streamToken = fLastStreamToken
100       = new StreamState(*this, serverRTPPort, serverRTCPPort, rtpSink, udpSink,
101             streamBitrate, mediaSource,
102             rtpGroupsock, rtcpGroupsock);
103   }
104 
105   // Record these destinations as being for this client session id:
106   Destinations* destinations;
107   if (tcpSocketNum < 0) { // UDP
108     destinations = new Destinations(destinationAddr, clientRTPPort, clientRTCPPort);
109   } else { // TCP
110     destinations = new Destinations(tcpSocketNum, rtpChannelId, rtcpChannelId);
111   }
112   fDestinationsHashTable->Add((char const*)clientSessionId, destinations);
113 }

　　經過上面的步驟后，服務器端就已經准備好向客戶端傳送RTP包以及RTCP包了，等待客戶端發送PLAY命令后開始傳輸。服務器端收到PLAY命令后，調用RTSPClientSession::handleCmd_PLAY函數處理。在handleCmd_PLAY函數中，首先提取Scale，表示客戶端期望的播放速度（正常、快進、快退），然后提取Range，表示客戶端期望的播放起止范圍，根據這兩個參數分別調用ServerMediaSubsession::setStreamScale函數和ServerMediaSubsession::seekStream函數，最后調用ServerMediaSubsession::startStream函數開始傳輸數據。實際調用的是OnDemandServerMediaSubsession::startStream函數，看一下這個函數的內容：

 1 void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,void* streamToken,
 2                         TaskFunc* rtcpRRHandler,void* rtcpRRHandlerClientData,
 3                         unsigned short& rtpSeqNum,unsigned& rtpTimestamp,
 4                         ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
 5                         void* serverRequestAlternativeByteHandlerClientData) {
 6   StreamState* streamState = (StreamState*)streamToken;
 7   Destinations* destinations
 8     = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));　　　　　　// 查找目的客戶端的地址  9   if (streamState != NULL) {
10     streamState->startPlaying(destinations,
11                   rtcpRRHandler, rtcpRRHandlerClientData,
12                   serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);　　　//調用StreamState::startPlaying函數開始播放 13     
14     RTPSink* rtpSink = streamState->rtpSink(); // alias
15     if (rtpSink != NULL) {
16       rtpSeqNum = rtpSink->currentSeqNo();
17       rtpTimestamp = rtpSink->presetNextTimestamp();
18     }
19   }
20 }

　　在OnDemandServerMediaSubsessionstartStream函數中，主要是調用了StreamState::startPlaying函數，來看一下這個函數：

 1 void StreamState
 2 ::startPlaying(Destinations* dests,
 3            TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
 4            ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
 5            void* serverRequestAlternativeByteHandlerClientData) {
 6   if (dests == NULL) return;
 7 
 8   if (fRTCPInstance == NULL && fRTPSink != NULL) {
 9     // Create (and start) a 'RTCP instance' for this RTP sink:
10     fRTCPInstance
11       = RTCPInstance::createNew(fRTPSink->envir(), fRTCPgs,
12                 fTotalBW, (unsigned char*)fMaster.fCNAME,
13                 fRTPSink, NULL /* we're a server */);
14         // Note: This starts RTCP running automatically
15   }
16 
17   if (dests->isTCP) {　　　　　　// 使用TCP傳輸RTP包和RTCP包 18     19     // Change RTP and RTCP to use the TCP socket instead of UDP:
20     if (fRTPSink != NULL) {
21       fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
22       RTPInterface
23     ::setServerRequestAlternativeByteHandler(fRTPSink->envir(), dests->tcpSocketNum,
24                          serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
25         // So that we continue to handle RTSP commands from the client
26     }
27     if (fRTCPInstance != NULL) {
28       fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
29       fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
30                       rtcpRRHandler, rtcpRRHandlerClientData);
31     }
32   } else {　　　　　　　　　　　　　　// 使用UDP傳輸RTP包和RTCP包 33       34     // Tell the RTP and RTCP 'groupsocks' about this destination
35     // (in case they don't already have it):
36     if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort);
37     if (fRTCPgs != NULL) fRTCPgs->addDestination(dests->addr, dests->rtcpPort);
38     if (fRTCPInstance != NULL) {
39       fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
40                       rtcpRRHandler, rtcpRRHandlerClientData);
41     }
42   }
43 
44   if (fRTCPInstance != NULL) {
45     // Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to
46     // get RTCP-synchronized presentation times immediately:
47     fRTCPInstance->sendReport();
48   }
49 
50   if (!fAreCurrentlyPlaying && fMediaSource != NULL) {　　　　　　　　　　　　　　　　//調用RTPSink::startPlaying函數開始傳輸數據 51     if (fRTPSink != NULL) {
52       fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);　　　　　　
53       fAreCurrentlyPlaying = True;
54     } else if (fUDPSink != NULL) {
55       fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
56       fAreCurrentlyPlaying = True;
57     }
58   }
59 }

　　在StreamState::startPlaying函數中，將客戶端添加到目的客戶端列表中去，然后調用RTPSink::startPlaying函數，實際調用的是MediaSink::startPlaying函數:

 1 Boolean MediaSink::startPlaying(MediaSource& source,
 2                 afterPlayingFunc* afterFunc,
 3                 void* afterClientData) {
 4   // Make sure we're not already being played:
 5   if (fSource != NULL) {
 6     envir().setResultMsg("This sink is already being played");
 7     return False;
 8   }
 9 
10   // Make sure our source is compatible:
11   if (!sourceIsCompatibleWithUs(source)) {
12     envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");
13     return False;
14   }
15   fSource = (FramedSource*)&source;
16 
17   fAfterFunc = afterFunc;　　　　　　　　　　　　　// 設置好回調函數后，就調用continuePlaying函數開始播放 18   fAfterClientData = afterClientData;
19   return continuePlaying();
20 }
21   /* 這里我們的媒體資源是.264文件，對應的是H264VideoFileServerMediaSubsession，對應H264VideoFileServerMediaSubsession創建的是
　　　H264VideoRTPSink對象，H264VideoRTPSink是H264or5VideoRTPSink的子類，因此上面實際調用的是H264or5VideoRTPSink::continuePlaying函數 */
22 Boolean H264or5VideoRTPSink::continuePlaying() {
23   // First, check whether we have a 'fragmenter' class set up yet.
24   // If not, create it now:　　　　　創建一個H264or5Fragmenter對象，　　
25   if (fOurFragmenter == NULL) {
26     fOurFragmenter = new H264or5Fragmenter(fHNumber, envir(), fSource, OutPacketBuffer::maxSize,
27                        ourMaxPacketSize() - 12/*RTP hdr size*/);
28   } else {
29     fOurFragmenter->reassignInputSource(fSource);
30   }
31   fSource = fOurFragmenter;   // 注意，此處fSource變成了H264or5Fragmenter對象，H264or5Fragmenter是FramedFilter的子類 32   // FramedFilter是FramedSource的子類，從名字可以看出FramedFilter的作用是對FramedSource送來的數據做一些“過濾”，並且FramedFilter的
        結果數據還可以給另外一個FramedFilter做進一步的“過濾”，這里類似於Java中的IO裝飾流，使用了裝飾模式。
33   // Then call the parent class's implementation:
34   return MultiFramedRTPSink::continuePlaying();　　　　　　//調用了MultiFramedRTPSink的continuePlaying函數 35 }

 1  Boolean MultiFramedRTPSink::continuePlaying() {
 2    // Send the first packet.
 3    // (This will also schedule any future sends.)
 4    buildAndSendPacket(True);
 5    return True;
 6  }
 7  
 8  void MultiFramedRTPSink::buildAndSendPacket(Boolean isFirstPacket) {
 9    fIsFirstPacket = isFirstPacket;
　　　 // RTP version 2; marker ('M') bit not set (by default; it can be set later) 10    unsigned rtpHdr = 0x80000000;   Set up the RTP header:　　　/設置RTP頭部
11    rtpHdr |= (fRTPPayloadType<<16);
12    rtpHdr |= fSeqNo; // sequence number
13    fOutBuf->enqueueWord(rtpHdr);
14    // Note where the RTP timestamp will go.
15    // (We can't fill this in until we start packing payload frames.)
16    fTimestampPosition = fOutBuf->curPacketSize();
17    fOutBuf->skipBytes(4); // leave a hole for the timestamp
18    fOutBuf->enqueueWord(SSRC());
19    // Allow for a special, payload-format-specific header following the
20    // RTP header:
21    fSpecialHeaderPosition = fOutBuf->curPacketSize();
22    fSpecialHeaderSize = specialHeaderSize();
23    fOutBuf->skipBytes(fSpecialHeaderSize);
24  
25    // Begin packing as many (complete) frames into the packet as we can:
26    fTotalFrameSpecificHeaderSizes = 0;
27    fNoFramesLeft = False;
28    fNumFramesUsedSoFar = 0;
29    packFrame();                      // 調用packFrame函數
30  }
31  
32  void MultiFramedRTPSink::packFrame() {
33    // Get the next frame.
34  
35    // First, see if we have an overflow frame that was too big for the last pkt
36    if (fOutBuf->haveOverflowData()) {
37      // Use this frame before reading a new one from the source
38      unsigned frameSize = fOutBuf->overflowDataSize();
39      struct timeval presentationTime = fOutBuf->overflowPresentationTime();
40      unsigned durationInMicroseconds = fOutBuf->overflowDurationInMicroseconds();
41      fOutBuf->useOverflowData();
42  
43      afterGettingFrame1(frameSize, 0, presentationTime, durationInMicroseconds);
44    } else {
45      // Normal case: we need to read a new frame from the source
46      if (fSource == NULL) return;
47  
48      fCurFrameSpecificHeaderPosition = fOutBuf->curPacketSize();
49      fCurFrameSpecificHeaderSize = frameSpecificHeaderSize();
50      fOutBuf->skipBytes(fCurFrameSpecificHeaderSize);
51      fTotalFrameSpecificHeaderSizes += fCurFrameSpecificHeaderSize;
52      fSource->getNextFrame(fOutBuf->curPtr(), fOutBuf->totalBytesAvailable(),  // 調用FramedSource::getNextFrame函數獲取幀數據保存到fOutBuf中
53                afterGettingFrame, this, ourHandleClosure, this);       // 獲取后回調MultiFramedRTPSink::afterGettingFrame函數
        //對應H264VideoFileServerMediaSubsession，是在H264or5Fragmenter中回調MultiFramedRTPSink::afterGettingFrame函數
54    }
55  }
56 
57 void FramedSource::getNextFrame(unsigned char* to, unsigned maxSize,
58                 afterGettingFunc* afterGettingFunc,void* afterGettingClientData,
59                 onCloseFunc* onCloseFunc,void* onCloseClientData) {
60   // Make sure we're not already being read:
61   if (fIsCurrentlyAwaitingData) {
62     envir() << "FramedSource[" << this << "]::getNextFrame(): attempting to read more than once at the same time!\n";
63     envir().internalError();
64   }
65 
66   fTo = to;
67   fMaxSize = maxSize;
68   fNumTruncatedBytes = 0; // by default; could be changed by doGetNextFrame()
69   fDurationInMicroseconds = 0; // by default; could be changed by doGetNextFrame()
70   fAfterGettingFunc = afterGettingFunc;
71   fAfterGettingClientData = afterGettingClientData;
72   fOnCloseFunc = onCloseFunc;
73   fOnCloseClientData = onCloseClientData;
74   fIsCurrentlyAwaitingData = True;
75 
76   doGetNextFrame();　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　// 調用doGetNextFrame函數 77 }

　　對應H264VideoFileServerMediaSubsession之前創建的FramedSource是ByteStreamFileSource對象，然后又在ByteStreamFileSource外面套了一個H264or5Fragmenter，因此調用的是H264or5Fragmenter::doGetNextFrame函數：

 1 void H264or5Fragmenter::doGetNextFrame() {
 2   if (fNumValidDataBytes == 1) {
 3     // We have no NAL unit data currently in the buffer.  Read a new one:
        //在這里又調用了ByteStreamFileSource的doGetNextFrame函數，並且設置回調函數為H264or5Fragmenter::afterGettingFrame函數
 4     fInputSource->getNextFrame(&fInputBuffer[1], fInputBufferSize - 1,        
 5                    afterGettingFrame, this,
 6                    FramedSource::handleClosure, this);

 7   } else {
 8     // We have NAL unit data in the buffer.  There are three cases to consider:
 9     // 1. There is a new NAL unit in the buffer, and it's small enough to deliver
10     //    to the RTP sink (as is).
11     // 2. There is a new NAL unit in the buffer, but it's too large to deliver to
12     //    the RTP sink in its entirety.  Deliver the first fragment of this data,
13     //    as a FU packet, with one extra preceding header byte (for the "FU header").
14     // 3. There is a NAL unit in the buffer, and we've already delivered some
15     //    fragment(s) of this.  Deliver the next fragment of this data,
16     //    as a FU packet, with two (H.264) or three (H.265) extra preceding header bytes
17     //    (for the "NAL header" and the "FU header").
18 
19     if (fMaxSize < fMaxOutputPacketSize) { // shouldn't happen
20       envir() << "H264or5Fragmenter::doGetNextFrame(): fMaxSize ("
21           << fMaxSize << ") is smaller than expected\n";
22     } else {
23       fMaxSize = fMaxOutputPacketSize;
24     }
25 
26     fLastFragmentCompletedNALUnit = True; // by default
27     if (fCurDataOffset == 1) { // case 1 or 2
28       if (fNumValidDataBytes - 1 <= fMaxSize) { // case 1
29     memmove(fTo, &fInputBuffer[1], fNumValidDataBytes - 1);
30     fFrameSize = fNumValidDataBytes - 1;
31     fCurDataOffset = fNumValidDataBytes;
32       } else { // case 2
33     // We need to send the NAL unit data as FU packets.  Deliver the first
34     // packet now.  Note that we add "NAL header" and "FU header" bytes to the front
35     // of the packet (overwriting the existing "NAL header").
36     if (fHNumber == 264) {
37       fInputBuffer[0] = (fInputBuffer[1] & 0xE0) | 28; // FU indicator
38       fInputBuffer[1] = 0x80 | (fInputBuffer[1] & 0x1F); // FU header (with S bit)
39     } else { // 265
40       u_int8_t nal_unit_type = (fInputBuffer[1]&0x7E)>>1;
41       fInputBuffer[0] = (fInputBuffer[1] & 0x81) | (49<<1); // Payload header (1st byte)
42       fInputBuffer[1] = fInputBuffer[2]; // Payload header (2nd byte)
43       fInputBuffer[2] = 0x80 | nal_unit_type; // FU header (with S bit)
44     }
45     memmove(fTo, fInputBuffer, fMaxSize);
46     fFrameSize = fMaxSize;
47     fCurDataOffset += fMaxSize - 1;
48     fLastFragmentCompletedNALUnit = False;
49       }
50     } else { // case 3
51       // We are sending this NAL unit data as FU packets.  We've already sent the
52       // first packet (fragment).  Now, send the next fragment.  Note that we add
53       // "NAL header" and "FU header" bytes to the front.  (We reuse these bytes that
54       // we already sent for the first fragment, but clear the S bit, and add the E
55       // bit if this is the last fragment.)
56       unsigned numExtraHeaderBytes;
57       if (fHNumber == 264) {
58     fInputBuffer[fCurDataOffset-2] = fInputBuffer[0]; // FU indicator
59     fInputBuffer[fCurDataOffset-1] = fInputBuffer[1]&~0x80; // FU header (no S bit)
60     numExtraHeaderBytes = 2;
61       } else { // 265
62     fInputBuffer[fCurDataOffset-3] = fInputBuffer[0]; // Payload header (1st byte)
63     fInputBuffer[fCurDataOffset-2] = fInputBuffer[1]; // Payload header (2nd byte)
64     fInputBuffer[fCurDataOffset-1] = fInputBuffer[2]&~0x80; // FU header (no S bit)
65     numExtraHeaderBytes = 3;
66       }
67       unsigned numBytesToSend = numExtraHeaderBytes + (fNumValidDataBytes - fCurDataOffset);
68       if (numBytesToSend > fMaxSize) {
69     // We can't send all of the remaining data this time:
70     numBytesToSend = fMaxSize;
71     fLastFragmentCompletedNALUnit = False;
72       } else {
73     // This is the last fragment:
74     fInputBuffer[fCurDataOffset-1] |= 0x40; // set the E bit in the FU header
75     fNumTruncatedBytes = fSaveNumTruncatedBytes;
76       }
77       memmove(fTo, &fInputBuffer[fCurDataOffset-numExtraHeaderBytes], numBytesToSend);
78       fFrameSize = numBytesToSend;
79       fCurDataOffset += numBytesToSend - numExtraHeaderBytes;
80     }
81 
82     if (fCurDataOffset >= fNumValidDataBytes) {
83       // We're done with this data.  Reset the pointers for receiving new data:
84       fNumValidDataBytes = fCurDataOffset = 1;
85     }
86 
87     // Complete delivery to the client:
88     FramedSource::afterGetting(this);       // 回調MultiFramedRTPSink::afterGettingFrame函數   
89   }
90 }

 1 void ByteStreamFileSource::doGetNextFrame() {
 2   if (feof(fFid) || ferror(fFid) || (fLimitNumBytesToStream && fNumBytesToStream == 0)) {
 3     handleClosure();
 4     return;
 5   }
 6 
 7 #ifdef READ_FROM_FILES_SYNCHRONOUSLY
 8   doReadFromFile();　　　　　　　　　　　　　　　　　　　　　　　　　　　// 讀文件  9 #else
10   if (!fHaveStartedReading) {
11     // Await readable data from the file:
12     envir().taskScheduler().turnOnBackgroundReadHandling(fileno(fFid),
13            (TaskScheduler::BackgroundHandlerProc*)&fileReadableHandler, this);
14     fHaveStartedReading = True;
15   }
16 #endif
17 }
18 
19 void ByteStreamFileSource::doReadFromFile() {
20   // Try to read as many bytes as will fit in the buffer provided (or "fPreferredFrameSize" if less)
21   if (fLimitNumBytesToStream && fNumBytesToStream < (u_int64_t)fMaxSize) {
22     fMaxSize = (unsigned)fNumBytesToStream;
23   }
24   if (fPreferredFrameSize > 0 && fPreferredFrameSize < fMaxSize) {
25     fMaxSize = fPreferredFrameSize;
26   }
27 #ifdef READ_FROM_FILES_SYNCHRONOUSLY
28   fFrameSize = fread(fTo, 1, fMaxSize, fFid);        //調用fread函數讀取數據 29 #else
30   if (fFidIsSeekable) {
31     fFrameSize = fread(fTo, 1, fMaxSize, fFid);
32   } else {
33     // For non-seekable files (e.g., pipes), call "read()" rather than "fread()", to ensure that the read doesn't block:
34     fFrameSize = read(fileno(fFid), fTo, fMaxSize);
35   }
36 #endif
37   if (fFrameSize == 0) {
38     handleClosure();
39     return;
40   }
41   fNumBytesToStream -= fFrameSize;
42 
43   // Set the 'presentation time':
44   if (fPlayTimePerFrame > 0 && fPreferredFrameSize > 0) {
45     if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) {
46       // This is the first frame, so use the current time:
47       gettimeofday(&fPresentationTime, NULL);
48     } else {
49       // Increment by the play time of the previous data:
50       unsigned uSeconds    = fPresentationTime.tv_usec + fLastPlayTime;
51       fPresentationTime.tv_sec += uSeconds/1000000;
52       fPresentationTime.tv_usec = uSeconds%1000000;
53     }
54 
55     // Remember the play time of this data:
56     fLastPlayTime = (fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;
57     fDurationInMicroseconds = fLastPlayTime;
58   } else {
59     // We don't know a specific play time duration for this data,
60     // so just record the current time as being the 'presentation time':
61     gettimeofday(&fPresentationTime, NULL);
62   }
63  //讀取數據后，調用FramedSource::afterGetting函數
64   // Inform the reader that he has data:
65 #ifdef READ_FROM_FILES_SYNCHRONOUSLY
66   // To avoid possible infinite recursion, we need to return to the event loop to do this:
67   nextTask() = envir().taskScheduler().scheduleDelayedTask(0,
68                 (TaskFunc*)FramedSource::afterGetting, this);
69 #else
70   // Because the file read was done from the event loop, we can call the
71   // 'after getting' function directly, without risk of infinite recursion:
72   FramedSource::afterGetting(this);
73 #endif
74 }
75 
76 void FramedSource::afterGetting(FramedSource* source) {
77   source->fIsCurrentlyAwaitingData = False;
78       // indicates that we can be read again
79       // Note that this needs to be done here, in case the "fAfterFunc"
80       // called below tries to read another frame (which it usually will)
81    
82   if (source->fAfterGettingFunc != NULL) {　　　　　　　　　　　　　　　　　　　　　　　　　
83     (*(source->fAfterGettingFunc))(source->fAfterGettingClientData,
84                    source->fFrameSize, source->fNumTruncatedBytes,
85                    source->fPresentationTime,
86                    source->fDurationInMicroseconds);
87   }
88 }

　　在FramedSource::afterGetting函數中調用fAfterGettingFunc函數，對於ByteStreamFileSource對象，fAfterGettingFunc在之前被設置為H264or5Fragmenter::afterGettingFrame函數：

 1 void H264or5Fragmenter::afterGettingFrame(void* clientData, unsigned frameSize,
 2                       unsigned numTruncatedBytes,
 3                       struct timeval presentationTime,
 4                       unsigned durationInMicroseconds) {
 5   H264or5Fragmenter* fragmenter = (H264or5Fragmenter*)clientData;
 6   fragmenter->afterGettingFrame1(frameSize, numTruncatedBytes, presentationTime,
 7                  durationInMicroseconds);
 8 }
 9 
10 void H264or5Fragmenter::afterGettingFrame1(unsigned frameSize,
11                        unsigned numTruncatedBytes,
12                        struct timeval presentationTime,
13                        unsigned durationInMicroseconds) {
14   fNumValidDataBytes += frameSize;
15   fSaveNumTruncatedBytes = numTruncatedBytes;
16   fPresentationTime = presentationTime;
17   fDurationInMicroseconds = durationInMicroseconds;
18 
19   // Deliver data to the client:
20   doGetNextFrame();                    
21 }

　　在H264or5Fragmenter::afterGettingFrame1函數中又調用了H264or5Fragmenter::doGetNextFrame函數，在H264or5Fragmenter::doGetNextFrame函數中，當讀取的幀數據滿足條件時就又回調MultiFrameRTPSink的afterGettingFrame函數。

  1 void MultiFramedRTPSink
  2 ::afterGettingFrame(void* clientData, unsigned numBytesRead,
  3             unsigned numTruncatedBytes,
  4             struct timeval presentationTime,
  5             unsigned durationInMicroseconds) {
  6   MultiFramedRTPSink* sink = (MultiFramedRTPSink*)clientData;
  7   sink->afterGettingFrame1(numBytesRead, numTruncatedBytes,
  8                presentationTime, durationInMicroseconds);
  9 }
 10 
 11 void MultiFramedRTPSink
 12 ::afterGettingFrame1(unsigned frameSize, unsigned numTruncatedBytes,
 13              struct timeval presentationTime,
 14              unsigned durationInMicroseconds) {
 15   if (fIsFirstPacket) {
 16     // Record the fact that we're starting to play now:
 17     gettimeofday(&fNextSendTime, NULL);
 18   }
 19 
 20   fMostRecentPresentationTime = presentationTime;
 21   if (fInitialPresentationTime.tv_sec == 0 && fInitialPresentationTime.tv_usec == 0) {
 22     fInitialPresentationTime = presentationTime;
 23   }    
 24 
 25   if (numTruncatedBytes > 0) {　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　　//超出緩沖區大小要被舍棄的數據  26     unsigned const bufferSize = fOutBuf->totalBytesAvailable();
 27     envir() << "MultiFramedRTPSink::afterGettingFrame1(): The input frame data was too large for our buffer size ("
 28         << bufferSize << ").  "
 29         << numTruncatedBytes << " bytes of trailing data was dropped!  Correct this by increasing \"OutPacketBuffer::maxSize\" to at least "
 30         << OutPacketBuffer::maxSize + numTruncatedBytes << ", *before* creating this 'RTPSink'.  (Current value is "
 31         << OutPacketBuffer::maxSize << ".)\n";
 32   }
 33   unsigned curFragmentationOffset = fCurFragmentationOffset;
 34   unsigned numFrameBytesToUse = frameSize;
 35   unsigned overflowBytes = 0;
 36 
 37   // If we have already packed one or more frames into this packet,
 38   // check whether this new frame is eligible to be packed after them.
 39   // (This is independent of whether the packet has enough room for this
 40   // new frame; that check comes later.)
 41   if (fNumFramesUsedSoFar > 0) {                              // 在這個RTP包中已經包含了若干幀數據  42     if ((fPreviousFrameEndedFragmentation
 43      && !allowOtherFramesAfterLastFragment())
 44     || !frameCanAppearAfterPacketStart(fOutBuf->curPtr(), frameSize)) {      //這個RTP包中不允許再添加多余的幀（比如：前面的幀作了結尾標記）  45       // Save away this frame for next time:
 46       numFrameBytesToUse = 0;
 47       fOutBuf->setOverflowData(fOutBuf->curPacketSize(), frameSize,
 48                    presentationTime, durationInMicroseconds);
 49     }
 50   }
 51   fPreviousFrameEndedFragmentation = False;
 52 
 53   if (numFrameBytesToUse > 0) {　　　　　　　　　　　　　　　　                  // 允許將這一幀添加到RTP包中，但要檢查大小是否超出了RTP包的剩余空間　　　　　　　　　　　　　　　  
 54     // Check whether this frame overflows the packet        
 55     if (fOutBuf->wouldOverflow(frameSize)) {　　　　　　　　　　　　　　　　　　　　　　　　 //這一幀數據超出了RTP包剩余空間  56       // Don't use this frame now; instead, save it as overflow data, and
 57       // send it in the next packet instead.  However, if the frame is too
 58       // big to fit in a packet by itself, then we need to fragment it (and
 59       // use some of it in this packet, if the payload format permits this.)
 60       if (isTooBigForAPacket(frameSize)
 61           && (fNumFramesUsedSoFar == 0 || allowFragmentationAfterStart())) {      //發送這一幀數據的一部分  62         // We need to fragment this frame, and use some of it now:
 63         overflowBytes = computeOverflowForNewFrame(frameSize);
 64         numFrameBytesToUse -= overflowBytes;
 65         fCurFragmentationOffset += numFrameBytesToUse;
 66       } else {
 67         // We don't use any of this frame now:                              // 不添加這一幀數據
 68         overflowBytes = frameSize;
 69         numFrameBytesToUse = 0;
 70       }
 71       fOutBuf->setOverflowData(fOutBuf->curPacketSize() + numFrameBytesToUse,         // 標記超出幀的位置和大小，以便后面調整packet  72                    overflowBytes, presentationTime, durationInMicroseconds);
 73     } else if (fCurFragmentationOffset > 0) {                                  
 74       // This is the last fragment of a frame that was fragmented over
 75       // more than one packet.  Do any special handling for this case:
 76       fCurFragmentationOffset = 0;
 77       fPreviousFrameEndedFragmentation = True;
 78     }
 79   }
 80 
 81   if (numFrameBytesToUse == 0 && frameSize > 0) {　　　　　　// 讀取適當的數據后，開始發送RTP包  82     // Send our packet now, because we have filled it up:
 83     sendPacketIfNecessary();　　　　　　　　　　　　　　　　　　　　　　　　 
 84   } else {
 85     // Use this frame in our outgoing packet:
 86     unsigned char* frameStart = fOutBuf->curPtr();
 87     fOutBuf->increment(numFrameBytesToUse);
 88         // do this now, in case "doSpecialFrameHandling()" calls "setFramePadding()" to append padding bytes
 89 
 90     // Here's where any payload format specific processing gets done:
 91     doSpecialFrameHandling(curFragmentationOffset, frameStart,
 92                numFrameBytesToUse, presentationTime,
 93                overflowBytes);
 94 
 95     ++fNumFramesUsedSoFar;
 96 
 97     // Update the time at which the next packet should be sent, based
 98     // on the duration of the frame that we just packed into it.
 99     // However, if this frame has overflow data remaining, then don't
100     // count its duration yet.
101     if (overflowBytes == 0) {
102       fNextSendTime.tv_usec += durationInMicroseconds;
103       fNextSendTime.tv_sec += fNextSendTime.tv_usec/1000000;
104       fNextSendTime.tv_usec %= 1000000;
105     }
106 
107     // Send our packet now if (i) it's already at our preferred size, or
108     // (ii) (heuristic) another frame of the same size as the one we just
109     //      read would overflow the packet, or
110     // (iii) it contains the last fragment of a fragmented frame, and we
111     //      don't allow anything else to follow this or
112     // (iv) one frame per packet is allowed:
113     if (fOutBuf->isPreferredSize()
114         || fOutBuf->wouldOverflow(numFrameBytesToUse)
115         || (fPreviousFrameEndedFragmentation &&
116             !allowOtherFramesAfterLastFragment())
117         || !frameCanAppearAfterPacketStart(fOutBuf->curPtr() - frameSize,
118                        frameSize) ) {
119       // The packet is ready to be sent now
120       sendPacketIfNecessary();
121     } else {
122       // There's room for more frames; try getting another:
123       packFrame();　　　　　　　　　　　　　　　　　　　　　　　　　　// 調用packFrame函數讀取更多的數據 124     }
125   }
126 }

　　在MultiFramedRTPSink中盡量讀取多的幀之后，調用sendPacketIfNecessary函數發送給客戶端:

 1 void MultiFramedRTPSink::sendPacketIfNecessary() {
 2   if (fNumFramesUsedSoFar > 0) {
 3     // Send the packet:
 4 #ifdef TEST_LOSS
 5     if ((our_random()%10) != 0) // simulate 10% packet loss #####
 6 #endif
 7       if (!fRTPInterface.sendPacket(fOutBuf->packet(), fOutBuf->curPacketSize())) {      // 通過RTPInterface發送RTP包  8     // if failure handler has been specified, call it
 9     if (fOnSendErrorFunc != NULL) (*fOnSendErrorFunc)(fOnSendErrorData);
10       }
11     ++fPacketCount;
12     fTotalOctetCount += fOutBuf->curPacketSize();
13     fOctetCount += fOutBuf->curPacketSize()
14       - rtpHeaderSize - fSpecialHeaderSize - fTotalFrameSpecificHeaderSizes;
15 
16     ++fSeqNo; // for next time
17   }
18 
19   if (fOutBuf->haveOverflowData()　　　　　　　　　　　　　　　　　　　　　　　　　　　　//未發送的幀數據，對RTP包作出調整 20       && fOutBuf->totalBytesAvailable() > fOutBuf->totalBufferSize()/2) {
21     // Efficiency hack: Reset the packet start pointer to just in front of
22     // the overflow data (allowing for the RTP header and special headers),
23     // so that we probably don't have to "memmove()" the overflow data
24     // into place when building the next packet:
25     unsigned newPacketStart = fOutBuf->curPacketSize()
26       - (rtpHeaderSize + fSpecialHeaderSize + frameSpecificHeaderSize());
27     fOutBuf->adjustPacketStart(newPacketStart);　　　　　　　　　　　　　　　　　　　　　　　　　　　　
28   } else {
29     // Normal case: Reset the packet start pointer back to the start:
30     fOutBuf->resetPacketStart();
31   }
32   fOutBuf->resetOffset();
33   fNumFramesUsedSoFar = 0;
34 
35   if (fNoFramesLeft) {
36     // We're done:
37     onSourceClosure();
38   } else {
39     // We have more frames left to send.  Figure out when the next frame
40     // is due to start playing, then make sure that we wait this long before
41     // sending the next packet.
42     struct timeval timeNow;
43     gettimeofday(&timeNow, NULL);
44     int secsDiff = fNextSendTime.tv_sec - timeNow.tv_sec;
45     int64_t uSecondsToGo = secsDiff*1000000 + (fNextSendTime.tv_usec - timeNow.tv_usec);
46     if (uSecondsToGo < 0 || secsDiff < 0) { // sanity check: Make sure that the time-to-delay is non-negative:
47       uSecondsToGo = 0;
48     }
49 
50     // Delay this amount of time:        // 准備下一次發送RTP包
51     nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo, (TaskFunc*)sendNext, this);
52   }
53 }
54 
55 // The following is called after each delay between packet sends:
56 void MultiFramedRTPSink::sendNext(void* firstArg) {
57   MultiFramedRTPSink* sink = (MultiFramedRTPSink*)firstArg;
58   sink->buildAndSendPacket(False);                                //循環調用buildAndSendPacket 59 }

　　以上的調用函數過程比較亂，特附一張圖以更清晰地展示以上的流程

　　服務器端通過RTPSink去讀數據，在RTPSink中又通過FramedSource讀數據，讀完數據后交給RTPSink處理，RTPSink處理完后繼續通過FramedSource讀取數據，如此在RTPSink和FramedSoruce之間形成一個循環，這是Live555讀取發送數據的總體流程。

　　以上便是從建立RTSP連接到發送RTP數據的流程（以H264文件為例），后面的停止發送數據到斷開連接不再關注和詳述。

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猜您在找 Live555學習之(四)------建立RTSP連接的過程（RTSP客戶端） live555學習之RTSP連接建立以及請求消息處理過程基於live555的一個簡單RTSP服務器視頻直播：libx264編碼一幀圖片+live555搭建RTSP流媒體服務器基於Live555,ffmpeg的RTSP播放器直播與點播視頻監控、直播——基於opencv，libx264，live555的RTSP流媒體服務器（zc301P攝像頭）By Chain_Gank live555_RTSP連接建立以及請求消息處理過程 RTSP服務端轉發服務（live555庫中的testH264VideoStreamer.cpp和testOnDemandRTSPServer.cpp實例） (轉)live555 RTSP Server RTP over TCP BUG 開源流媒體處理庫live555服務器端、客戶端源代碼分析總結