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六　建立RTP會話首先更正一個概念：ServerMediasession原先說代表一個流，其實(shí)是不準(zhǔn)確的。它代表的是server端的一個媒體的名字，而說ServerMediaSubsession代表一個Track是準(zhǔn)確的。以后流指的是那些有數(shù)據(jù)流動的組合。RTP的建立：RTP的建立過程無非是這樣：client告訴server自己的rtp/rtcp端口號，server建立自己的rtp/rtcp socket，然后在收到PLAY請求時向客戶端發(fā)數(shù)據(jù)。看起來非常簡單。在收到SETUP請求時才建立連接，讓我們看一下處理這個命令的函數(shù)：[cpp] view plain copyvoid RTSPServer::RTSPClientSession::handleCmd_SETUP( char const* cseq, char const* url[cpp] view plain copyvoid OnDemandServerMediaSubsession::getStreamParameters( unsigned clientSessionId, netAddressBits clientAddress, Port const& clientRTPPort, Port const& clientRTCPPort, int tcpSocketNum, unsigned char rtpChannelId, unsigned char rtcpChannelId, netAddressBits& destinationAddress, u_int8_t& /*destinationTTL*/, Boolean& isMulticast, Port& serverRTPPort, Port& serverRTCPPort, void*& streamToken) { if (destinationAddress == 0) destinationAddress = clientAddress; struct in_addr destinationAddr; destinationAddr.s_addr = destinationAddress; isMulticast = False; //ServerMediaSubsession并沒有保存所有基于自己的數(shù)據(jù)流，而是只記錄了最后一次建立的數(shù)據(jù)流。 //利用這個變量和fReuseFirstSource可以實(shí)現(xiàn)多client連接到一個流的形式。 if (fLastStreamToken != NULL && fReuseFirstSource) { //如果已經(jīng)基于這個ServerMediaSubsession創(chuàng)建了一個連接，并且希望使用這個連接 //則直接返回這個連接。 // Special case: Rather than creating a new 'StreamState', // we reuse the one that we've already created: serverRTPPort = ((StreamState*) fLastStreamToken)->serverRTPPort(); serverRTCPPort = ((StreamState*) fLastStreamToken)->serverRTCPPort(); ++((StreamState*) fLastStreamToken)->referenceCount(); streamToken = fLastStreamToken; } else { // Normal case: Create a new media source: unsigned streamBitrate; FramedSource* mediaSource = createNewStreamSource(clientSessionId,streamBitrate); // Create 'groupsock' and 'sink' objects for the destination, // using previously unused server port numbers: RTPSink* rtpSink; BasicUDPSink* udpSink; Groupsock* rtpGroupsock; Groupsock* rtcpGroupsock; portNumBits serverPortNum; if (clientRTCPPort.num() == 0) { // We're streaming raw UDP (not RTP). Create a single groupsock: NoReuse dummy; // ensures that we sk [cpp] view plain copyvoid RTSPServer::RTSPClientSession::handleCmd_PLAY( ServerMediaSubsession* subsession, char const* cseq, char const* fullRequestStr) { char* rtspURL = fOurServer.rtspURL(fOurServerMediaSession,fClientInputSocket); unsigned rtspURLSize = strlen(rtspURL); // Parse the client's "Scale:" header, if any: float scale; Boolean sawScaleHeader = parseScaleHeader(fullRequestStr, scale); // Try to set the stream's scale factor to this value: if (subsession == NULL /*aggregate op*/) { fOurServerMediaSession->testScaleFactor(scale); } else { subsession->testScaleFactor(scale); } char buf[100]; char* scaleHeader; if (!sawScaleHeader) { buf[0] = '/0'; // Because we didn't see a Scale: header, don't send one back } else { sprintf(buf, "Scale: %f/r/n", scale); } scaleHeader = strDup(buf); //分析客戶端對于播放范圍的要求 // Parse the client's "Range:" header, if any: double rangeStart = 0.0, rangeEnd = 0.0; Boolean sawRangeHeader = parseRangeHeader(fullRequestStr, rangeStart,rangeEnd); // Use this information, plus the stream's duration (if known), to create // our own "Range:" header, for the response: float duration = subsession == NULL /*aggregate op*/ ? fOurServerMediaSession->duration() : subsession->duration(); if (duration < 0.0) { // We're an aggregate PLAY, but the subsessions have different durations. // Use the largest of these durations in our header duration = -duration; } // Make sure that "rangeStart" and "rangeEnd" (from the client's "Range:" header) have sane values // before we send back our own "Range:" header in our response: if (rangeStart < 0.0) rangeStart = 0.0; else if (rangeStart > duration) rangeStart = duration; if (rangeEnd < 0.0) rangeEnd = 0.0; else if (rangeEnd > duration) rangeEnd = duration; if ((scale > 0.0 && rangeStart > rangeEnd && rangeEnd > 0.0) || (scale < 0.0 && rangeStart < rangeEnd)) { // "rangeStart" and "rangeEnd" were the wrong way around; swap them: double tmp = rangeStart; rangeStart = rangeEnd; rangeEnd = tmp; } char* rangeHeader; if (!sawRangeHeader) { buf[0] = '/0'; // Because we didn't see a Range: header, don't send one back } else if (rangeEnd == 0.0 && scale >= 0.0) { sprintf(buf, "Range: npt=%.3f-/r/n", rangeStart); } else { sprintf(buf, "Range: npt=%.3f-%.3f/r/n", rangeStart, rangeEnd); } rangeHeader = strDup(buf); // Create a "RTP-Info:" line. It will get filled in from each subsession's state: char const* rtpInfoFmt = "%s" // "RTP-Info:", plus any preceding rtpInfo items "%s"// comma separator, if needed "url=%s/%s" ";seq=%d" ";rtptime=%u"; unsigned rtpInfoFmtSize = strlen(rtpInfoFmt); char* rtpInfo = strDup("RTP-Info: "); unsigned i, numRTPInfoItems = 0; // Do any required seeking/scaling on each subsession, before starting streaming: for (i = 0; i < fNumStreamStates; ++i) { if (subsession == NULL /* means: aggregated Operation */ || subsession == fStreamStates[i].subsession) { if (sawScaleHeader) { fStreamStates[i].subsession->setStreamScale(fOurSessionId, fStreamStates[i].streamToken, scale); } if (sawRangeHeader) { double streamDuration = 0.0; // by default; means: stream until the end of the media if (rangeEnd > 0.0 && (rangeEnd + 0.001) < duration) { // the 0.001 is because we limited the values to 3 decimal places // We want the stream to end early. Set the duration we want: streamDuration = rangeEnd - rangeStart; if (streamDuration < 0.0) streamDuration = -streamDuration; // should happen only if scale < 0.0 } u_int64_t numBytes; fStreamStates[i].subsession->seekStream(fOurSessionId, fStreamStates[i].streamToken, rangeStart, streamDuration, numBytes); } } } // Now, start streaming: for (i = 0; i < fNumStreamStates; ++i) { if (subsession == NULL /* means: aggregated operation */ || subsession == fStreamStates[i].subsession) { unsigned short rtpSeqNum = 0; unsigned rtpTimestamp = 0; //啟動流 fStreamStates[i].subsession->startStream(fOurSessionId, fStreamStates[i].streamToken, (TaskFunc*) noteClientLiveness, this, rtpSeqNum, rtpTimestamp, handleAlternativeRequestByte, this); const char *urlSuffix = fStreamStates[i].subsession->trackId(); char* prevRTPInfo = rtpInfo; unsigned rtpInfoSize = rtpInfoFmtSize + strlen(prevRTPInfo) + 1 + rtspURLSize + strlen(urlSuffix) + 5 /*max unsigned short len*/ + 10 /*max unsigned (32-bit) len*/ + 2 /*allows for trailing /r/n at final end of string*/; rtpInfo = new char[rtpInfoSize]; sprintf(rtpInfo, rtpInfoFmt, prevRTPInfo, numRTPInfoItems++ == 0 ? "" : ",", rtspURL, urlSuffix, rtpSeqNum, rtpTimestamp); delete[] prevRTPInfo; } } if (numRTPInfoItems == 0) { rtpInfo[0] = '/0'; } else { unsigned rtpInfoLen = strlen(rtpInfo); rtpInfo[rtpInfoLen] = '/r'; rtpInfo[rtpInfoLen + 1] = '/n'; rtpInfo[rtpInfoLen + 2] = '/0'; } // Fill in the response: snprintf((char*) fResponseBuffer, sizeof fResponseBuffer, "RTSP/1.0 200 OK/r/n" "CSeq: %s/r/n" "%s" "%s" "%s" "Session: %08X/r/n" "%s/r/n", cseq, dateHeader(), scaleHeader, rangeHeader, fOurSessionId, rtpInfo); delete[] rtpInfo; delete[] rangeHeader; delete[] scaleHeader; delete[] rtspURL; } 有個問題，如果這個streamToken使用的是已存在的（還記得ReuseFirstSource嗎），為它再次調(diào)用startStream()時，究竟會做什么呢？你猜啊！呵呵我猜吧：應(yīng)該是把這個客戶端的地址和rtp/rtcp端口傳給rtp server的groupSocket，rtp server自然就開始向這個客戶端發(fā)送數(shù)據(jù)了。是不是這樣尼？看代碼吧：[cpp] view plain copyvoid StreamState::startPlaying( Destinations* dests, TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData, ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler, void* serverRequestAlternativeByteHandlerClientData) { //目標(biāo)ip address rtp && rtcp port if (dests == NULL) return; //創(chuàng)建RTCPInstance對象，以計算和收發(fā)RTCP包 if (fRTCPInstance == NULL && fRTPSink != NULL) { // Create (and start) a 'RTCP instance' for this RTP sink: fRTCPInstance = RTCPInstance::createNew(fRTPSink->envir(), fRTCPgs, fTotalBW, (unsigned char*) fMaster.fCNAME, fRTPSink, NULL /* we're a server */); // Note: This starts RTCP running automatically } if (dests->isTCP) { //如果RTP over TCP // Change RTP and RTCP to use the TCP socket instead of UDP: if (fRTPSink != NULL) { fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId); fRTPSink->setServerRequestAlternativeByteHandler( dests->tcpSocketNum, serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData); } if (fRTCPInstance != NULL) { fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId); fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId, rtcpRRHandler, rtcpRRHandlerClientData); } } else { //向RTP和RTCP的groupsocket增加這個目標(biāo) // Tell the RTP and RTCP 'groupsocks' about this destination // (in case they don't already have it): if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort); if (fRTCPgs != NULL) fRTCPgs->addDestination(dests->addr, dests->rtcpPort); if (fRTCPInstance != NULL) { fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort, rtcpRRHandler, rtcpRRHandlerClientData); } } if (!fAreCurrentlyPlaying && fMediaSource != NULL) { //如果還沒有啟動傳輸，現(xiàn)在啟動之。 if (fRTPSink != NULL) { fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this); fAreCurrentlyPlaying = True; } else if (fUDPSink != NULL) { fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState,this); fAreCurrentlyPlaying = True; } } } 嘿嘿，還真的這樣嘀！

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live555學(xué)習(xí)筆記6－建立RTP會話