庖丁解牛-----Live555源码彻底解密(根据MediaServer讲解Rtsp的建立过程)
live555MediaServer.cpp服務端源碼講解(testRelay.cpp,http://blog.csdn.net/smilestone_322/article/details/18923139)
int main(int argc, char** argv) {
???? // Begin by setting up our usage environment:
???? TaskScheduler* scheduler = BasicTaskScheduler::createNew();
???? UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
?
???? UserAuthenticationDatabase* authDB = NULL;
????
???? // Create the RTSP server.? Try first with the default port number (554),
???? // and then with the alternative port number (8554):
???? RTSPServer* rtspServer;
???? portNumBits rtspServerPortNum = 554;
???? //先使用554默認端口建立Rtsp Server
???? rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
???? //如果建立不成功，使用8554建立rtsp server
???? if (rtspServer == NULL) {
???????? rtspServerPortNum = 8554;
???????? rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
???? }
???? if (rtspServer == NULL) {
???????? *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
???????? // exit(1);
???????? return -1;
???? }
????
???? env->taskScheduler().doEventLoop(); // does not return
?
???? return 0; // only to prevent compiler warning
}
?
跟蹤進入CreateNew函數;
DynamicRTSPServer*
DynamicRTSPServer::createNew(UsageEnvironment&env,PortourPort,
????????????? ???? UserAuthenticationDatabase*authDatabase,
????????????? ???? unsigned reclamationTestSeconds) {
? int ourSocket = setUpOurSocket(env,ourPort);?//建立tcp socket
? if (ourSocket == -1)returnNULL;
?
? return new DynamicRTSPServer(env,ourSocket,ourPort,authDatabase,reclamationTestSeconds);
}
?
?
DynamicRTSPServer::DynamicRTSPServer(UsageEnvironment&env,intourSocket,
?????????????????? ???? Port ourPort,
?????????????????? ???? UserAuthenticationDatabase*authDatabase,unsignedreclamationTestSeconds)
? : RTSPServerSupportingHTTPStreaming(env,ourSocket,ourPort,authDatabase,reclamationTestSeconds) {
}
?
首先建立socket，然后在調用DynamicRtspServer的構造函數，DynamicRtspServer繼承RTSPServerSupportingHTTPStreaming類; RTSPServerSupportingHTTPStreaming類又繼承RTSPServer類；
RTSPServerSupportingHTTPStreaming類的主要作用是支持Http；
?
接著看setUpOurSocket函數在前面已經講過；就是建立socket；最后我們跟蹤進入RTSPServer類的構造函數：
?
RTSPServer::RTSPServer(UsageEnvironment& env,
???????? ?????? int ourSocket, Port ourPort,
???????? ?????? UserAuthenticationDatabase* authDatabase,
???????? ?????? unsigned reclamationTestSeconds)
? : Medium(env),
??? fRTSPServerPort(ourPort), fRTSPServerSocket(ourSocket), fHTTPServerSocket(-1), fHTTPServerPort(0),
??? fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),
??? fClientConnections(HashTable::create(ONE_WORD_HASH_KEYS)),
??? fClientConnectionsForHTTPTunneling(NULL), // will get created if needed
??? fClientSessions(HashTable::create(STRING_HASH_KEYS)),
??? fPendingRegisterRequests(HashTable::create(ONE_WORD_HASH_KEYS)),
??? fAuthDB(authDatabase), fReclamationTestSeconds(reclamationTestSeconds) {
? ignoreSigPipeOnSocket(ourSocket); // so that clients on the same host that are killed don't also kill us
?
? // Arrange to handle connections from others:
? env.taskScheduler().turnOnBackgroundReadHandling(fRTSPServerSocket,
??????????????????????????? ?? (TaskScheduler::BackgroundHandlerProc*)&incomingConnectionHandlerRTSP,this);
}
?
當fRTSPServerSocket收到數據時，調用incomingConnectionHandlerRTSP回調函數，繼續跟進到incomingConnectionHandlerRTSP函數，源碼如下：
?
void RTSPServer::incomingConnectionHandlerRTSP(void* instance,int/*mask*/) {
? RTSPServer* server = (RTSPServer*)instance;
? server->incomingConnectionHandlerRTSP1();
}
?
?
void RTSPServer::incomingConnectionHandler(int serverSocket) {
? struct sockaddr_in clientAddr;
? SOCKLEN_T clientAddrLen = sizeof clientAddr;
? int clientSocket = accept(serverSocket, (struct sockaddr*)&clientAddr, &clientAddrLen);
? if (clientSocket < 0) {
??? int err = envir().getErrno();
??? if (err != EWOULDBLOCK) {
??????? envir().setResultErrMsg("accept() failed: ");
??? }
??? return;
? }
? makeSocketNonBlocking(clientSocket);
? increaseSendBufferTo(envir(), clientSocket, 50*1024);
?
#ifdef DEBUG
? envir() << "accept()ed connection from " << AddressString(clientAddr).val() << "\n";
#endif
?
? // Create a new object for handling this RTSP connection:
? (void)createNewClientConnection(clientSocket, clientAddr);
}
?
當收到客戶的連接時需保存下代表客戶端的新socket，以后用這個socket與這個客戶通訊。每個客戶將來會對應一個rtp會話，而且各客戶的RTSP請求只控制自己的rtp會話；
?
incomingConnectionHandler函數的作用是accept接受客戶端的socket連接,然后設置clientSocket的屬性,這里需要注意,我們在建立服務端socket時已經對服務端socket設置了非阻塞屬性,這個地方又要設置accept后的clientSecket的屬性;
?
incomingConnectionHandler函數最后調用createNewClientConnection函數，源碼如下：
RTSPServer::RTSPClientConnection*
RTSPServer::createNewClientConnection(int clientSocket,struct sockaddr_in clientAddr) {
? return new RTSPClientConnection(*this, clientSocket, clientAddr);
}
?
對于每個新建立的客戶端連接請求,new RTSPClientConnection的對象進行管理;
RTSPServer::RTSPClientConnection
::RTSPClientConnection(RTSPServer& ourServer, int clientSocket, struct sockaddr_in clientAddr)
? : fOurServer(ourServer), fIsActive(True),
??? fClientInputSocket(clientSocket), fClientOutputSocket(clientSocket), fClientAddr(clientAddr),
??? fRecursionCount(0), fOurSessionCookie(NULL) {
? // Add ourself to our 'client connections' table:
? fOurServer.fClientConnections->Add((charconst*)this,this);
?
? // Arrange to handle incoming requests:
? resetRequestBuffer();
? envir().taskScheduler().setBackgroundHandling(fClientInputSocket, SOCKET_READABLE|SOCKET_EXCEPTION,
??????????????????????????? (TaskScheduler::BackgroundHandlerProc*)&incomingRequestHandler,this);
}
?
在該函數中首先對RTSPServer的成員變量進行賦值：
fOurServer= ourServer;
fClientInputSocket= clientSocket;
fClientOutputSocket= clientSocket;
fClientAddr= clientAddr;
?
setBackgroundHandling函數用來處理fClientInputSocket socket上收到數據，或異常時，調用incomingRequestHandler回調函數;
?
下面在跟進到incomingRequestHandler函數:
void RTSPServer::RTSPClientConnection::incomingRequestHandler(void* instance,int/*mask*/) {
? RTSPClientConnection* session = (RTSPClientConnection*)instance;
? session->incomingRequestHandler1();
}
?
Session 為剛才new的RTSPClientConnection 對象，這個地方需要調試驗證下;調用成員函數incomingRequestHandler1;跟進到該成員函數的代碼:
?
void RTSPServer::RTSPClientConnection::incomingRequestHandler1() {
? struct sockaddr_in dummy; // 'from' address, meaningless in this case
?
? int bytesRead = readSocket(envir(), fClientInputSocket, &fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft, dummy);
? handleRequestBytes(bytesRead);
}
?
該函數調用ReadSocket從fClientInputSocket上讀取數據；讀到的數據保存在fRequestBuffer中，readSocket的返回值為實際讀到的數據的長度；源碼如下：
int readSocket(UsageEnvironment& env,
???? ?????? int socket, unsigned char* buffer, unsigned bufferSize,
???? ?????? struct sockaddr_in& fromAddress) {
? SOCKLEN_T addressSize = sizeof fromAddress;
? int bytesRead = recvfrom(socket, (char*)buffer, bufferSize, 0,
????????????? ?? (struct sockaddr*)&fromAddress,
????????????? ?? &addressSize);
? if (bytesRead < 0) {
??? //##### HACK to work around bugs in Linux and Windows:
??? int err = env.getErrno();
??? if (err == 111 /*ECONNREFUSED (Linux)*/
#if defined(__WIN32__) ||defined(_WIN32)
???? // What a piece of crap Windows is.?Sometimes
???? // recvfrom() returns -1, but with an 'errno' of 0.
???? // This appears not to be a real error; just treat
???? // it as if it were a read of zero bytes, and hope
???? // we don't have to do anything else to 'reset'
???? // this alleged error:
???? || err == 0 || err == EWOULDBLOCK
#else
???? || err == EAGAIN
#endif
???? || err == 113 /*EHOSTUNREACH (Linux)*/) {// Why does Linux return this for datagram sock?
????? fromAddress.sin_addr.s_addr = 0;
????? return 0;
??? }
??? //##### END HACK
??? socketErr(env, "recvfrom() error: ");
? } else if (bytesRead == 0) {
??? // "recvfrom()" on a stream socket can return 0 if the remote end has closed the connection.?Treat this as an error:
??? return -1;
? }
?
? return bytesRead;
}
?
從socket中讀到數據后必須對數據進行解析，解析的源碼如下：
void RTSPServer::RTSPClientConnection::handleRequestBytes(int newBytesRead) {
? int numBytesRemaining = 0;
? ++fRecursionCount;
?
? do {
??? RTSPServer::RTSPClientSession* clientSession = NULL;
?
??? if (newBytesRead < 0 || (unsigned)newBytesRead >= fRequestBufferBytesLeft) {
????? // Either the client socket has died, or the request was too big for us.
????? // Terminate this connection:
#ifdef DEBUG
????? fprintf(stderr, "RTSPClientConnection[%p]::handleRequestBytes() read %d new bytes (of %d); terminating connection!\n", this, newBytesRead, fRequestBufferBytesLeft);
#endif
????? fIsActive = False;
????? break;
??? }
???
??? Boolean endOfMsg = False;
??? unsigned char* ptr = &fRequestBuffer[fRequestBytesAlreadySeen];
#ifdef DEBUG
??? ptr[newBytesRead] = '\0';
??? fprintf(stderr, "RTSPClientConnection[%p]::handleRequestBytes() %s %d new bytes:%s\n",
???? ??? this, numBytesRemaining > 0 ? "processing" : "read", newBytesRead, ptr);
#endif
???
??? if (fClientOutputSocket != fClientInputSocket) {
????? // We're doing RTSP-over-HTTP tunneling, and input commands are assumed to have been Base64-encoded.
????? // We therefore Base64-decode as much of this new data as we can (i.e., up to a multiple of 4 bytes).
?
????? // But first, we remove any whitespace that may be in the input data:
????? unsigned toIndex = 0;
????? for (int fromIndex = 0; fromIndex < newBytesRead; ++fromIndex) {
???? char c = ptr[fromIndex];
???? if (!(c == ' ' || c == '\t' || c == '\r' || c == '\n')) { // not 'whitespace': space,tab,CR,NL
???? ? ptr[toIndex++] = c;
???? }
????? }
????? newBytesRead = toIndex;
?
????? unsigned numBytesToDecode = fBase64RemainderCount + newBytesRead;
????? unsigned newBase64RemainderCount = numBytesToDecode%4;
????? numBytesToDecode -= newBase64RemainderCount;
????? if (numBytesToDecode > 0) {
???? ptr[newBytesRead] = '\0';
???? unsigned decodedSize;
???? unsigned char* decodedBytes = base64Decode((char const*)(ptr-fBase64RemainderCount), numBytesToDecode, decodedSize);
#ifdef DEBUG
???? fprintf(stderr, "Base64-decoded %d input bytes into %d new bytes:", numBytesToDecode, decodedSize);
???? for (unsigned k = 0; k < decodedSize; ++k) fprintf(stderr, "%c", decodedBytes[k]);
???? fprintf(stderr, "\n");
#endif
????
???? // Copy the new decoded bytes in place of the old ones (we can do this because there are fewer decoded bytes than original):
???? unsigned char* to = ptr-fBase64RemainderCount;
???? for (unsigned i = 0; i < decodedSize; ++i) *to++ = decodedBytes[i];
????
???? // Then copy any remaining (undecoded) bytes to the end:
???? for (unsigned j = 0; j < newBase64RemainderCount; ++j) *to++ = (ptr-fBase64RemainderCount+numBytesToDecode)[j];
????
???? newBytesRead = decodedSize + newBase64RemainderCount; // adjust to allow for the size of the new decoded data (+ remainder)
???? delete[] decodedBytes;
????? }
????? fBase64RemainderCount = newBase64RemainderCount;
????? if (fBase64RemainderCount > 0)break;// because we know that we have more input bytes still to receive
??? }
???
??? // Look for the end of the message: <CR><LF><CR><LF>
??? unsigned char *tmpPtr = fLastCRLF + 2;
??? if (tmpPtr < fRequestBuffer) tmpPtr = fRequestBuffer;
??? while (tmpPtr < &ptr[newBytesRead-1]) {
????? if (*tmpPtr == '\r' && *(tmpPtr+1) == '\n') {
???? if (tmpPtr - fLastCRLF == 2) {// This is it:
???? ? endOfMsg = True;
???? ? break;
???? }
???? fLastCRLF = tmpPtr;
????? }
????? ++tmpPtr;
??? }
???
??? fRequestBufferBytesLeft -= newBytesRead;
??? fRequestBytesAlreadySeen += newBytesRead;
???
??? if (!endOfMsg) break; // subsequent reads will be needed to complete the request
???
??? // Parse the request string into command name and 'CSeq', then handle the command:
??? fRequestBuffer[fRequestBytesAlreadySeen] = '\0';
??? char cmdName[RTSP_PARAM_STRING_MAX];
??? char urlPreSuffix[RTSP_PARAM_STRING_MAX];
??? char urlSuffix[RTSP_PARAM_STRING_MAX];
??? char cseq[RTSP_PARAM_STRING_MAX];
??? char sessionIdStr[RTSP_PARAM_STRING_MAX];
??? unsigned contentLength = 0;
fLastCRLF[2] = '\0'; // temporarily, for parsing
?
//解析Rtsp請求字符串
??? Boolean parseSucceeded = parseRTSPRequestString((char*)fRequestBuffer, fLastCRLF+2 - fRequestBuffer,
??????????????????????????? ??? cmdName, sizeof cmdName,
??????????????????????????? ??? urlPreSuffix, sizeof urlPreSuffix,
??????????????????????????? ??? urlSuffix, sizeof urlSuffix,
??????????????????????????? ??? cseq, sizeof cseq,
??????????????????????????? ??? sessionIdStr, sizeof sessionIdStr,
??????????????????????????? ??? contentLength);
??? fLastCRLF[2] = '\r'; // restore its value
??? if (parseSucceeded) {
#ifdef DEBUG
????? fprintf(stderr, "parseRTSPRequestString() succeeded, returning cmdName \"%s\", urlPreSuffix \"%s\", urlSuffix \"%s\", CSeq \"%s\", Content-Length %u, with %d bytes following the message.\n", cmdName, urlPreSuffix, urlSuffix, cseq, contentLength, ptr + newBytesRead - (tmpPtr + 2));
#endif
????? // If there was a "Content-Length:" header, then make sure we've received all of the data that it specified:
????? if (ptr + newBytesRead < tmpPtr + 2 + contentLength)break;// we still need more data; subsequent reads will give it to us
?????
????? // We now have a complete RTSP request.
????? // Handle the specified command (beginning by checking those that don't require session ids):
????? fCurrentCSeq = cseq;
???? //收到客戶端的OPTIONS請求
????? if (strcmp(cmdName, "OPTIONS") == 0) {
???? // If the request included a "Session:" id, and it refers to a client session that's current ongoing, then use this
???? // command to indicate 'liveness' on that client session:
???? if (sessionIdStr[0] != '\0') {
???? ? clientSession = (RTSPServer::RTSPClientSession*)(fOurServer.fClientSessions->Lookup(sessionIdStr));
???? //根據sessionIdStr查表，看該客戶端的會話是否存在,存在會話，調用noteLiveness函數
???? ? if (clientSession != NULL) clientSession->noteLiveness();
???? }
???? //處理Opinion請求,構建應答包
???? handleCmd_OPTIONS();
????? } else if (urlPreSuffix[0] == '\0' && urlSuffix[0] =='*' && urlSuffix[1] =='\0') {
???? // The special "*" URL means: an operation on the entire server.?This works only for GET_PARAMETER and SET_PARAMETER:
???? if (strcmp(cmdName, "GET_PARAMETER") == 0) {
???? ? handleCmd_GET_PARAMETER((charconst*)fRequestBuffer);
???? } else if (strcmp(cmdName, "SET_PARAMETER") == 0) {
???? ? handleCmd_SET_PARAMETER((charconst*)fRequestBuffer);
???? } else {
???? ? handleCmd_notSupported();
???? }
????? } else if (strcmp(cmdName, "DESCRIBE") == 0) {
???? //收到客戶端的Describe請求,處理該請求，構建應答包
???? handleCmd_DESCRIBE(urlPreSuffix, urlSuffix, (charconst*)fRequestBuffer);
????? } else if (strcmp(cmdName, "SETUP") == 0) {
???? //收到客戶端的Setup請求,如果是第一次Setup，那么就需要調用createNewClientSession函數進行會話，然后將sessionIdStr和clientSession關聯起來
???? if (sessionIdStr[0] == '\0') {
???? ? // No session id was present in the request.?So create a new "RTSPClientSession" object for this request.
???? ? // Choose a random (unused) 32-bit integer for the session id (it will be encoded as a 8-digit hex number).
???? ? // (We avoid choosing session id 0, because that has a special use (by "OnDemandServerMediaSubsession").)
???? ? u_int32_t sessionId;
???? ? do {
???? ??? sessionId = (u_int32_t)our_random32();
???? ??? sprintf(sessionIdStr, "%08X", sessionId);
???? ? } while (sessionId == 0 || fOurServer.fClientSessions->Lookup(sessionIdStr) != NULL);
???? ? clientSession = fOurServer.createNewClientSession(sessionId);
???? ? fOurServer.fClientSessions->Add(sessionIdStr, clientSession);
???? } else {
???? ? // The request included a session id.?Make sure it's one that we have already set up:
???? ? //如果存在會話，直接查找原來的會話；
???? ? clientSession = (RTSPServer::RTSPClientSession*)(fOurServer.fClientSessions->Lookup(sessionIdStr));
?
???? ? if (clientSession == NULL) {
???? ??? handleCmd_sessionNotFound();
???? ? }
???? }
???? //構建Setup應答包
???? if (clientSession != NULL) clientSession->handleCmd_SETUP(this, urlPreSuffix, urlSuffix, (charconst*)fRequestBuffer);
????? } else if (strcmp(cmdName, "TEARDOWN") == 0
???????? ?|| strcmp(cmdName, "PLAY") == 0
???????? ?|| strcmp(cmdName, "PAUSE") == 0
???????? ?|| strcmp(cmdName, "GET_PARAMETER") == 0
???????? ?|| strcmp(cmdName, "SET_PARAMETER") == 0) {
???? RTSPServer::RTSPClientSession* clientSession
???? ? = sessionIdStr[0] == '\0' ? NULL : (RTSPServer::RTSPClientSession*)(fOurServer.fClientSessions->Lookup(sessionIdStr));
???? if (clientSession == NULL) {
???? ? handleCmd_sessionNotFound();
???? } else {
???? ? clientSession->handleCmd_withinSession(this, cmdName, urlPreSuffix, urlSuffix, (charconst*)fRequestBuffer);
???? }
????? } else if (strcmp(cmdName, "REGISTER") == 0 || strcmp(cmdName,"REGISTER_REMOTE") == 0) {
???? // Because - unlike other commands - an implementation of these commands needs the entire URL, we re-parse the
???? // command to get it:
???? char* url = strDupSize((char*)fRequestBuffer);
???? if (sscanf((char*)fRequestBuffer,"%*s %s", url) == 1) {
???? ? handleCmd_REGISTER(url, urlSuffix, strcmp(cmdName, "REGISTER_REMOTE") == 0);
???? } else {
???? ? handleCmd_bad();
???? }
???? delete[] url;
????? } else {
???? // The command is one that we don't handle:
???? handleCmd_notSupported();
????? }
??? } else {
#ifdef DEBUG
????? fprintf(stderr, "parseRTSPRequestString() failed; checking now for HTTP commands (for RTSP-over-HTTP tunneling)...\n");
#endif
????? // The request was not (valid) RTSP, but check for a special case: HTTP commands (for setting up RTSP-over-HTTP tunneling):
????? char sessionCookie[RTSP_PARAM_STRING_MAX];
????? char acceptStr[RTSP_PARAM_STRING_MAX];
????? *fLastCRLF = '\0'; // temporarily, for parsing
????? parseSucceeded = parseHTTPRequestString(cmdName, sizeof cmdName,
?????????????????????? ????? urlSuffix, sizeof urlPreSuffix,
?????????????????????? ????? sessionCookie, sizeof sessionCookie,
?????????????????????? ????? acceptStr, sizeof acceptStr);
????? *fLastCRLF = '\r';
????? if (parseSucceeded) {
#ifdef DEBUG
???? fprintf(stderr, "parseHTTPRequestString() succeeded, returning cmdName \"%s\", urlSuffix \"%s\", sessionCookie \"%s\", acceptStr \"%s\"\n", cmdName, urlSuffix, sessionCookie, acceptStr);
#endif
???? // Check that the HTTP command is valid for RTSP-over-HTTP tunneling: There must be a 'session cookie'.
???? Boolean isValidHTTPCmd = True;
???? if (sessionCookie[0] == '\0') {
???? ? // There was no "x-sessioncookie:" header.?If there was an "Accept: application/x-rtsp-tunnelled" header,
???? ? // then this is a bad tunneling request.?Otherwise, assume that it's an attempt to access the stream via HTTP.
???? ? if (strcmp(acceptStr, "application/x-rtsp-tunnelled") == 0) {
???? ??? isValidHTTPCmd = False;
???? ? } else {
???? ??? handleHTTPCmd_StreamingGET(urlSuffix, (charconst*)fRequestBuffer);
???? ? }
???? } else if (strcmp(cmdName, "GET") == 0) {
???? ? handleHTTPCmd_TunnelingGET(sessionCookie);
???? } else if (strcmp(cmdName, "POST") == 0) {
???? ? // We might have received additional data following the HTTP "POST" command - i.e., the first Base64-encoded RTSP command.
???? ? // Check for this, and handle it if it exists:
???? ? unsigned char const* extraData = fLastCRLF+4;
???? ? unsigned extraDataSize = &fRequestBuffer[fRequestBytesAlreadySeen] - extraData;
???? ? if (handleHTTPCmd_TunnelingPOST(sessionCookie, extraData, extraDataSize)) {
???? ??? // We don't respond to the "POST" command, and we go away:
???? ??? fIsActive = False;
???? ??? break;
???? ? }
???? } else {
???? ? isValidHTTPCmd = False;
???? }
???? if (!isValidHTTPCmd) {
???? ? handleHTTPCmd_notSupported();
???? }
????? } else {
#ifdef DEBUG
???? fprintf(stderr, "parseHTTPRequestString() failed!\n");
#endif
???? handleCmd_bad();
????? }
??? }
???
#ifdef DEBUG
??? fprintf(stderr, "sending response: %s", fResponseBuffer);
#endif
???? //發送應答包
??? send(fClientOutputSocket, (charconst*)fResponseBuffer, strlen((char*)fResponseBuffer), 0);
???
??? if (clientSession != NULL && clientSession->fStreamAfterSETUP && strcmp(cmdName,"SETUP") == 0) {
????? // The client has asked for streaming to commence now, rather than after a
????? // subsequent "PLAY" command.? So, simulate the effect of a "PLAY" command:
????? clientSession->handleCmd_withinSession(this,"PLAY", urlPreSuffix, urlSuffix, (charconst*)fRequestBuffer);
??? }
???
??? // Check whether there are extra bytes remaining in the buffer, after the end of the request (a rare case).
??? // If so, move them to the front of our buffer, and keep processing it, because it might be a following, pipelined request.
??? unsigned requestSize = (fLastCRLF+4-fRequestBuffer) + contentLength;
??? numBytesRemaining = fRequestBytesAlreadySeen - requestSize;
??? resetRequestBuffer(); // to prepare for any subsequent request
?
??? if (numBytesRemaining > 0) {
????? memmove(fRequestBuffer, &fRequestBuffer[requestSize], numBytesRemaining);
????? newBytesRead = numBytesRemaining;
??? }
? } while (numBytesRemaining > 0);
?
? --fRecursionCount;
? if (!fIsActive) {
??? if (fRecursionCount > 0) closeSockets();elsedeletethis;
??? // Note: The "fRecursionCount" test is for a pathological situation where we reenter the event loop and get called recursively
??? // while handling a command (e.g., while handling a "DESCRIBE", to get a SDP description).
??? // In such a case we don't want to actually delete ourself until we leave the outermost call.
? }
}
?
void RTSPServer::RTSPClientSession::noteLiveness() {
? if (fOurServer.fReclamationTestSeconds > 0) {
??? envir().taskScheduler()
????? .rescheduleDelayedTask(fLivenessCheckTask,
????????????? ???? fOurServer.fReclamationTestSeconds*1000000,
????????????? ???? (TaskFunc*)livenessTimeoutTask, this);
? }
}
?
noteLiveness該函數可以用來判斷流是不是斷開；這個相當重要，我們可以使用它判斷網絡是否斷開，尤其在客戶端可以使用這樣的方法來判斷網絡是否斷開，然后實現斷網重連的功能。
?
RTSPClientSession要提供什么功能呢,可以想象：需要監聽客戶端的rtsp請求并回應它，需要在DESCRIBE請求中返回所請求的流的信息，需要在SETUP請求中建立起RTP會話，需要在TEARDOWN請求中關閉RTP會話，等等;
?
下面在接著跟進到createNewClientSession會話的函數:
RTSPServer::RTSPClientSession*
RTSPServer::createNewClientSession(u_int32_t sessionId) {
? return new RTSPClientSession(*this, sessionId);
}
?
RTSPServer::RTSPClientSession
::RTSPClientSession(RTSPServer& ourServer, u_int32_t sessionId)
? : fOurServer(ourServer), fOurSessionId(sessionId), fOurServerMediaSession(NULL), fIsMulticast(False), fStreamAfterSETUP(False),
??? fTCPStreamIdCount(0), fLivenessCheckTask(NULL), fNumStreamStates(0), fStreamStates(NULL) {
? noteLiveness();
}
這個構造函數舊版本的live555和v0.78版本是不同的，舊版本的live555，在accept后就建立了rtsp會話，而新版本的是在收到setup請求后才建立的會話，所以這些地方都不同，在舊版本中RTSPClientSession會有一個回調函數，新版本中沒有，該回調函數在收到客戶端的Connect命令時設置；
?
下面在分析下服務端對Opinion各種命令的請求的處理的代碼；首先還是分析Opinion，該命令請求的作用是客戶端請求服務端支持哪些命令；Describe請求是得到會話描述信息，包括h264的sps，pps信息也可以在Describe的應答中發送；Setup命令是用來建立會話，服務端收到Setup請求后，建立會話，new 一個RTSPClientSession對象，該對象用來處理客戶端的各種Rtsp命令請求；同時服務端保存會話Id和會話對象，每次可以從表中取出RTSPClientSession對象；響應客戶端的請求；在收到Setup命令后;沒有等到客戶端的Play命令，就開始視頻流；
?
?? if (clientSession != NULL && clientSession->fStreamAfterSETUP && strcmp(cmdName,"SETUP") == 0) {
????? // The client has asked for streaming to commence now, rather than after a
????? // subsequent "PLAY" command.? So, simulate the effect of a "PLAY" command:
????? clientSession->handleCmd_withinSession(this,"PLAY", urlPreSuffix, urlSuffix, (charconst*)fRequestBuffer);
??? }
???
1)服務端對Opinion命令的處理;跟蹤源碼:
?
void RTSPServer::RTSPClientConnection::handleCmd_OPTIONS() {
? snprintf((char*)fResponseBuffer,sizeof fResponseBuffer,
???? ?? "RTSP/1.0 200 OK\r\nCSeq: %s\r\n%sPublic: %s\r\n\r\n",
???? ?? fCurrentCSeq, dateHeader(), fOurServer.allowedCommandNames());
}
?
1)????? 服務端對Describe命令的處理
void RTSPServer::RTSPClientConnection
::handleCmd_DESCRIBE(char const* urlPreSuffix, char const* urlSuffix, char const* fullRequestStr) {
? char* sdpDescription = NULL;
? char* rtspURL = NULL;
? do {
//整理一下下RTSP地址
??? char urlTotalSuffix[RTSP_PARAM_STRING_MAX];
??? if (strlen(urlPreSuffix) + strlen(urlSuffix) + 2 >sizeof urlTotalSuffix) {
????? handleCmd_bad();
????? break;
??? }
??? urlTotalSuffix[0] = '\0';
??? if (urlPreSuffix[0] != '\0') {
????? strcat(urlTotalSuffix, urlPreSuffix);
????? strcat(urlTotalSuffix, "/");
??? }
??? strcat(urlTotalSuffix, urlSuffix);
??? ?//鑒權
??? if (!authenticationOK("DESCRIBE", urlTotalSuffix, fullRequestStr))break;
???
??? // We should really check that the request contains an "Accept:" #####
??? // for "application/sdp", because that's what we're sending back #####
???
// Begin by looking up the "ServerMediaSession" object for the specified "urlTotalSuffix":
//跟據流的名字查找ServerMediaSession
??? ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlTotalSuffix);
??? if (session == NULL) {
????? handleCmd_notFound();
????? break;
??? }
???
??? // Then, assemble a SDP description for this session:
??? sdpDescription = session->generateSDPDescription();
??? if (sdpDescription == NULL) {
????? // This usually means that a file name that was specified for a
????? // "ServerMediaSubsession" does not exist.
????? setRTSPResponse("404 File Not Found, Or In Incorrect Format");
????? break;
??? }
??? unsigned sdpDescriptionSize = strlen(sdpDescription);
???
??? // Also, generate our RTSP URL, for the "Content-Base:" header
??? // (which is necessary to ensure that the correct URL gets used in subsequent "SETUP" requests).
??? rtspURL = fOurServer.rtspURL(session, fClientInputSocket);
???
??? snprintf((char*)fResponseBuffer,sizeof fResponseBuffer,
???? ???? "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
???? ???? "%s"
???? ???? "Content-Base: %s/\r\n"
???? ???? "Content-Type: application/sdp\r\n"
???? ???? "Content-Length: %d\r\n\r\n"
???? ???? "%s",
???? ???? fCurrentCSeq,
???? ???? dateHeader(),
???? ???? rtspURL,
???? ???? sdpDescriptionSize,
???? ???? sdpDescription);
? } while (0);
?
? delete[] sdpDescription;
? delete[] rtspURL;
}
?
ServerMediaSession*
DynamicRTSPServer::lookupServerMediaSession(charconst* streamName) {
? // First, check whether the specified "streamName" exists as a local file:
? FILE* fid = fopen(streamName, "rb");
? Boolean fileExists = fid != NULL;
?
? // Next, check whether we already have a "ServerMediaSession" for this file:
?//查找是否已經存在一個ServerMediaSession
? ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);
? Boolean smsExists = sms != NULL;
?
? // Handle the four possibilities for "fileExists" and "smsExists":
??
? if (!fileExists) {
??? //文件不存在
??? if (smsExists) {
????? // "sms" was created for a file that no longer exists. Remove it:
???? //刪除ServerMediaSession
????? removeServerMediaSession(sms);
??? }
??? return NULL;
? } else {
??? if (!smsExists) {
????? // Create a new "ServerMediaSession" object for streaming from the named file.
????? //如果ServerMediaSession不存在,新建一個ServerMediaSession
????? sms = createNewSMS(envir(), streamName, fid);
???? //將ServerMediaSession和會話關聯起來
????? addServerMediaSession(sms);
??? }
??? fclose(fid);
??? return sms;
? }
}
?
?
void RTSPServer::addServerMediaSession(ServerMediaSession* serverMediaSession) {
? if (serverMediaSession == NULL)return;
?
? char const* sessionName = serverMediaSession->streamName();
? if (sessionName == NULL) sessionName ="";
? removeServerMediaSession(sessionName); // in case an existing "ServerMediaSession" with this name already exists
?
? fServerMediaSessions->Add(sessionName, (void*)serverMediaSession);
}
2)????? 服務端對Setup命令的處理
?
void RTSPServer::RTSPClientSession
::handleCmd_SETUP(RTSPServer::RTSPClientConnection* ourClientConnection,
???????? ? char const* urlPreSuffix, char const* urlSuffix, char const* fullRequestStr) {
? // Normally, "urlPreSuffix" should be the session (stream) name, and "urlSuffix" should be the subsession (track) name.
? // However (being "liberal in what we accept"), we also handle 'aggregate' SETUP requests (i.e., without a track name),
? // in the special case where we have only a single track.?I.e., in this case, we also handle:
? //??? "urlPreSuffix" is empty and "urlSuffix" is the session (stream) name, or
? //??? "urlPreSuffix" concatenated with "urlSuffix" (with "/" inbetween) is the session (stream) name.
? char const* streamName = urlPreSuffix;// in the normal case
? char const* trackId = urlSuffix;// in the normal case
? char* concatenatedStreamName = NULL;// in the normal case
?
? noteLiveness();
? do {
// First, make sure the specified stream name exists:
//下面的注釋參數參考：
http://blog.csdn.net/niu_gao/article/details/6911130
每個ServerMediaSession中至少要包含一個?//ServerMediaSubsession。一個ServerMediaSession對應一個媒體，可以認為是Server上的一個文件，或一個實時獲取設備。其包含的每個ServerMediaSubSession代表媒體中的一個Track。所以一個ServerMediaSession對應一個媒體，如果客戶請求的媒體名相同，就使用已存在的ServerMediaSession，如果不同，就創建一個新的。一個流對應一個StreamState，StreamState與ServerMediaSubsession相關，但代表的是動態的，而ServerMediaSubsession代表靜態的。??
fOurServer.lookupServerMediaSession(streamName)中會在找不到同名ServerMediaSession時新建一個，代表一個RTP流的ServerMediaSession們是被RTSPServer管理的，而不是被RTSPClientSession擁有。為什么呢？因為ServerMediaSession代表的是一個靜態的流，也就是可以從它里面獲取一個流的各種信息，但不能獲取傳輸狀態。不同客戶可能連接到同一個流，所以ServerMediaSession應被RTSPServer所擁有。
?
??? ServerMediaSession* sms = fOurServer.lookupServerMediaSession(streamName);
??? if (sms == NULL) {
????? // Check for the special case (noted above), before we give up:
????? if (urlPreSuffix[0] == '\0') {
???? streamName = urlSuffix;
????? } else {
???? concatenatedStreamName = newchar[strlen(urlPreSuffix) + strlen(urlSuffix) + 2];// allow for the "/" and the trailing '\0'
???? sprintf(concatenatedStreamName, "%s/%s", urlPreSuffix, urlSuffix);
???? streamName = concatenatedStreamName;
????? }
????? trackId = NULL;
?
????? // Check again:
????? sms = fOurServer.lookupServerMediaSession(streamName);
??? }
??? if (sms == NULL) {
????? if (fOurServerMediaSession == NULL) {
???? // The client asked for a stream that doesn't exist (and this session descriptor has not been used before):
???? ourClientConnection->handleCmd_notFound();
????? } else {
???? // The client asked for a stream that doesn't exist, but using a stream id for a stream that does exist. Bad request:
???? ourClientConnection->handleCmd_bad();
????? }
????? break;
??? } else {
????? if (fOurServerMediaSession == NULL) {
???? // We're accessing the "ServerMediaSession" for the first time.
???? fOurServerMediaSession = sms;
???? fOurServerMediaSession->incrementReferenceCount();
????? } else if (sms != fOurServerMediaSession) {
???? // The client asked for a stream that's different from the one originally requested for this stream id.?Bad request:
???? ourClientConnection->handleCmd_bad();
???? break;
????? }
??? }
?
??? if (fStreamStates == NULL) {
????? // This is the first "SETUP" for this session.?Set up our array of states for all of this session's subsessions (tracks):
????? ServerMediaSubsessionIterator iter(*fOurServerMediaSession);
????? for (fNumStreamStates = 0; iter.next() != NULL; ++fNumStreamStates) {}// begin by counting the number of subsessions (tracks)
?
????? fStreamStates = new struct streamState[fNumStreamStates];
?
????? iter.reset();
????? ServerMediaSubsession* subsession;
????? for (unsigned i = 0; i < fNumStreamStates; ++i) {
???? subsession = iter.next();
???? fStreamStates[i].subsession = subsession;
???? fStreamStates[i].streamToken = NULL; // for now; it may be changed by the "getStreamParameters()" call that comes later
?? ???}
??? }
?
??? // Look up information for the specified subsession (track):
??? ServerMediaSubsession* subsession = NULL;
??? unsigned streamNum;
??? if (trackId != NULL && trackId[0] !='\0') {// normal case
????? for (streamNum = 0; streamNum < fNumStreamStates; ++streamNum) {
???? subsession = fStreamStates[streamNum].subsession;
???? if (subsession != NULL && strcmp(trackId, subsession->trackId()) == 0)break;
????? }
????? if (streamNum >= fNumStreamStates) {
???? // The specified track id doesn't exist, so this request fails:
???? ourClientConnection->handleCmd_notFound();
???? break;
????? }
??? } else {
????? // Weird case: there was no track id in the URL.
????? // This works only if we have only one subsession:
????? if (fNumStreamStates != 1 || fStreamStates[0].subsession == NULL) {
???? ourClientConnection->handleCmd_bad();
???? break;
????? }
????? streamNum = 0;
????? subsession = fStreamStates[streamNum].subsession;
??? }
??? // ASSERT: subsession != NULL
?
??? // Look for a "Transport:" header in the request string, to extract client parameters:
??? StreamingMode streamingMode;
??? char* streamingModeString = NULL;// set when RAW_UDP streaming is specified
??? char* clientsDestinationAddressStr;
??? u_int8_t clientsDestinationTTL;
??? portNumBits clientRTPPortNum, clientRTCPPortNum;
??? unsigned char rtpChannelId, rtcpChannelId;
??? parseTransportHeader(fullRequestStr, streamingMode, streamingModeString,
????????????? ?clientsDestinationAddressStr, clientsDestinationTTL,
????????????? ?clientRTPPortNum, clientRTCPPortNum,
????????????? ?rtpChannelId, rtcpChannelId);
??? if ((streamingMode == RTP_TCP && rtpChannelId == 0xFF) ||
???? (streamingMode != RTP_TCP && ourClientConnection->fClientOutputSocket != ourClientConnection->fClientInputSocket)) {
????? // An anomolous situation, caused by a buggy client.?Either:
????? //???? 1/ TCP streaming was requested, but with no "interleaving=" fields.? (QuickTime Player sometimes does this.), or
????? //???? 2/ TCP streaming was not requested, but we're doing RTSP-over-HTTP tunneling (which implies TCP streaming).
?? ???// In either case, we assume TCP streaming, and set the RTP and RTCP channel ids to proper values:
????? streamingMode = RTP_TCP;
????? rtpChannelId = fTCPStreamIdCount; rtcpChannelId = fTCPStreamIdCount+1;
??? }
??? if (streamingMode == RTP_TCP) fTCPStreamIdCount += 2;
?
??? Port clientRTPPort(clientRTPPortNum);
??? Port clientRTCPPort(clientRTCPPortNum);
?
??? // Next, check whether a "Range:" or "x-playNow:" header is present in the request.
??? // This isn't legal, but some clients do this to combine "SETUP" and "PLAY":
??? double rangeStart = 0.0, rangeEnd = 0.0;
??? char* absStart = NULL; char* absEnd = NULL;
??? if (parseRangeHeader(fullRequestStr, rangeStart, rangeEnd, absStart, absEnd)) {
????? delete[] absStart; delete[] absEnd;
????? fStreamAfterSETUP = True;
??? } else if (parsePlayNowHeader(fullRequestStr)) {
????? fStreamAfterSETUP = True;
??? } else {
????? fStreamAfterSETUP = False;
??? }
?
??? // Then, get server parameters from the 'subsession':
??? int tcpSocketNum = streamingMode == RTP_TCP ? ourClientConnection->fClientOutputSocket : -1;
??? netAddressBits destinationAddress = 0;
??? u_int8_t destinationTTL = 255;
#ifdef RTSP_ALLOW_CLIENT_DESTINATION_SETTING
??? if (clientsDestinationAddressStr != NULL) {
????? // Use the client-provided "destination" address.
????? // Note: This potentially allows the server to be used in denial-of-service
????? // attacks, so don't enable this code unless you're sure that clients are
????? // trusted.
????? destinationAddress = our_inet_addr(clientsDestinationAddressStr);
??? }
??? // Also use the client-provided TTL.
??? destinationTTL = clientsDestinationTTL;
#endif
??? delete[] clientsDestinationAddressStr;
??? Port serverRTPPort(0);
??? Port serverRTCPPort(0);
?
??? // Make sure that we transmit on the same interface that's used by the client (in case we're a multi-homed server):
??? struct sockaddr_in sourceAddr; SOCKLEN_T namelen =sizeof sourceAddr;
??? getsockname(ourClientConnection->fClientInputSocket, (struct sockaddr*)&sourceAddr, &namelen);
??? netAddressBits origSendingInterfaceAddr = SendingInterfaceAddr;
??? netAddressBits origReceivingInterfaceAddr = ReceivingInterfaceAddr;
??? // NOTE: The following might not work properly, so we ifdef it out for now:
#ifdef HACK_FOR_MULTIHOMED_SERVERS
??? ReceivingInterfaceAddr = SendingInterfaceAddr = sourceAddr.sin_addr.s_addr;
#endif
?
??? subsession->getStreamParameters(fOurSessionId, ourClientConnection->fClientAddr.sin_addr.s_addr,
?????????????????? ??? clientRTPPort, clientRTCPPort,
?????????????????? ??? tcpSocketNum, rtpChannelId, rtcpChannelId,
?????????????????? ??? destinationAddress, destinationTTL, fIsMulticast,
?????????????????? ??? serverRTPPort, serverRTCPPort,
?????????????????? ??? fStreamStates[streamNum].streamToken);
??? SendingInterfaceAddr = origSendingInterfaceAddr;
??? ReceivingInterfaceAddr = origReceivingInterfaceAddr;
???
??? AddressString destAddrStr(destinationAddress);
??? AddressString sourceAddrStr(sourceAddr);
??? if (fIsMulticast) {
????? switch (streamingMode) {
??????? case RTP_UDP:
???? ? snprintf((char*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???????? ?? "RTSP/1.0 200 OK\r\n"
???????? ?? "CSeq: %s\r\n"
???????? ?? "%s"
???????? ?? "Transport: RTP/AVP;multicast;destination=%s;source=%s;port=%d-%d;ttl=%d\r\n"
???????? ?? "Session: %08X\r\n\r\n",
???????? ?? ourClientConnection->fCurrentCSeq,
???????? ?? dateHeader(),
???????? ?? destAddrStr.val(), sourceAddrStr.val(), ntohs(serverRTPPort.num()), ntohs(serverRTCPPort.num()), destinationTTL,
???????? ?? fOurSessionId);
???? ? break;
??????? case RTP_TCP:
???? ? // multicast streams can't be sent via TCP
???? ? ourClientConnection->handleCmd_unsupportedTransport();
???? ? break;
??????? case RAW_UDP:
???? ? snprintf((char*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???????? ?? "RTSP/1.0 200 OK\r\n"
???????? ?? "CSeq: %s\r\n"
???????? ?? "%s"
???????? ?? "Transport: %s;multicast;destination=%s;source=%s;port=%d;ttl=%d\r\n"
???????? ?? "Session: %08X\r\n\r\n",
???????? ?? ourClientConnection->fCurrentCSeq,
???????? ?? dateHeader(),
???????? ?? streamingModeString, destAddrStr.val(), sourceAddrStr.val(), ntohs(serverRTPPort.num()), destinationTTL,
???????? ?? fOurSessionId);
???? ? break;
????? }
??? } else {
????? switch (streamingMode) {
??????? case RTP_UDP: {
???? ? snprintf((char*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???????? ?? "RTSP/1.0 200 OK\r\n"
???????? ?? "CSeq: %s\r\n"
???????? ?? "%s"
???????? ?? "Transport: RTP/AVP;unicast;destination=%s;source=%s;client_port=%d-%d;server_port=%d-%d\r\n"
???????? ?? "Session: %08X\r\n\r\n",
???????? ?? ourClientConnection->fCurrentCSeq,
???????? ?? dateHeader(),
???????? ?? destAddrStr.val(), sourceAddrStr.val(), ntohs(clientRTPPort.num()), ntohs(clientRTCPPort.num()), ntohs(serverRTPPort.num()), ntohs(serverRTCPPort.num()),
???????? ?? fOurSessionId);
???? ? break;
???? }
??????? case RTP_TCP: {
???? ? snprintf((char*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???????? ?? "RTSP/1.0 200 OK\r\n"
???????? ?? "CSeq: %s\r\n"
???????? ?? "%s"
???????? ?? "Transport: RTP/AVP/TCP;unicast;destination=%s;source=%s;interleaved=%d-%d\r\n"
???????? ?? "Session: %08X\r\n\r\n",
???????? ?? ourClientConnection->fCurrentCSeq,
???????? ?? dateHeader(),
???????? ?? destAddrStr.val(), sourceAddrStr.val(), rtpChannelId, rtcpChannelId,
???????? ?? fOurSessionId);
???? ? break;
???? }
??????? case RAW_UDP: {
???? ? snprintf((char*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???????? ?? "RTSP/1.0 200 OK\r\n"
???????? ?? "CSeq: %s\r\n"
???????? ?? "%s"
???????? ?? "Transport: %s;unicast;destination=%s;source=%s;client_port=%d;server_port=%d\r\n"
???????? ?? "Session: %08X\r\n\r\n",
???????? ?? ourClientConnection->fCurrentCSeq,
???????? ?? dateHeader(),
???????? ?? streamingModeString, destAddrStr.val(), sourceAddrStr.val(), ntohs(clientRTPPort.num()), ntohs(serverRTPPort.num()),
???????? ?? fOurSessionId);
???? ? break;
???? }
????? }
??? }
??? delete[] streamingModeString;
? } while (0);
?
? delete[] concatenatedStreamName;
}
?
//新建ServerMediaSession的源代碼如下：
static ServerMediaSession* createNewSMS(UsageEnvironment& env,
?????????????????????? char const* fileName, FILE* /*fid*/) {
? // Use the file name extension to determine the type of "ServerMediaSession":
? char const* extension = strrchr(fileName,'.');
? if (extension == NULL) return NULL;
?
? ServerMediaSession* sms = NULL;
? Boolean const reuseSource = False;
? if (strcmp(extension, ".aac") == 0) {
??? // Assumed to be an AAC Audio (ADTS format) file:
??? NEW_SMS("AAC Audio");
??? sms->addSubsession(ADTSAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".amr") == 0) {
??? // Assumed to be an AMR Audio file:
??? NEW_SMS("AMR Audio");
??? sms->addSubsession(AMRAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".ac3") == 0) {
??? // Assumed to be an AC-3 Audio file:
??? NEW_SMS("AC-3 Audio");
??? sms->addSubsession(AC3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".m4e") == 0) {
??? // Assumed to be a MPEG-4 Video Elementary Stream file:
??? NEW_SMS("MPEG-4 Video");
??? sms->addSubsession(MPEG4VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".264") == 0) {
??? // Assumed to be a H.264 Video Elementary Stream file:
??? NEW_SMS("H.264 Video");
??? OutPacketBuffer::maxSize = 100000; // allow for some possibly large H.264 frames
??? sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".mp3") == 0) {
??? // Assumed to be a MPEG-1 or 2 Audio file:
??? NEW_SMS("MPEG-1 or 2 Audio");
??? // To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
??? // To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
??? // (For more information about ADUs and interleaving,
??? //? see <http://www.live555.com/rtp-mp3/>)
??? Boolean useADUs = False;
??? Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
??? useADUs = True;
#ifdef INTERLEAVE_ADUS
??? unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
??? unsigned const interleaveCycleSize
????? = (sizeof interleaveCycle)/(sizeof (unsigned char));
??? interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
??? sms->addSubsession(MP3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, useADUs, interleaving));
? } else if (strcmp(extension, ".mpg") == 0) {
??? // Assumed to be a MPEG-1 or 2 Program Stream (audio+video) file:
??? NEW_SMS("MPEG-1 or 2 Program Stream");
??? MPEG1or2FileServerDemux* demux
????? = MPEG1or2FileServerDemux::createNew(env, fileName, reuseSource);
??? sms->addSubsession(demux->newVideoServerMediaSubsession());
??? sms->addSubsession(demux->newAudioServerMediaSubsession());
? } else if (strcmp(extension, ".vob") == 0) {
??? // Assumed to be a VOB (MPEG-2 Program Stream, with AC-3 audio) file:
??? NEW_SMS("VOB (MPEG-2 video with AC-3 audio)");
??? MPEG1or2FileServerDemux* demux
????? = MPEG1or2FileServerDemux::createNew(env, fileName, reuseSource);
??? sms->addSubsession(demux->newVideoServerMediaSubsession());
??? sms->addSubsession(demux->newAC3AudioServerMediaSubsession());
? } else if (strcmp(extension, ".ts") == 0) {
??? // Assumed to be a MPEG Transport Stream file:
??? // Use an index file name that's the same as the TS file name, except with ".tsx":
??? unsigned indexFileNameLen = strlen(fileName) + 2;// allow for trailing "x\0"
??? char* indexFileName = new char[indexFileNameLen];
??? sprintf(indexFileName, "%sx", fileName);
??? NEW_SMS("MPEG Transport Stream");
??? sms->addSubsession(MPEG2TransportFileServerMediaSubsession::createNew(env, fileName, indexFileName, reuseSource));
??? delete[] indexFileName;
? } else if (strcmp(extension, ".wav") == 0) {
??? // Assumed to be a WAV Audio file:
??? NEW_SMS("WAV Audio Stream");
??? // To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
??? // change the following to True:
??? Boolean convertToULaw = False;
??? sms->addSubsession(WAVAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, convertToULaw));
? } else if (strcmp(extension, ".dv") == 0) {
??? // Assumed to be a DV Video file
??? // First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
??? OutPacketBuffer::maxSize = 300000;
?
??? NEW_SMS("DV Video");
??? sms->addSubsession(DVVideoFileServerMediaSubsession::createNew(env, fileName, reuseSource));
? } else if (strcmp(extension, ".mkv") == 0 || strcmp(extension,".webm") == 0) {
??? // Assumed to be a Matroska file (note that WebM ('.webm') files are also Matroska files)
??? NEW_SMS("Matroska video+audio+(optional)subtitles");
?
??? // Create a Matroska file server demultiplexor for the specified file.?(We enter the event loop to wait for this to complete.)
??? newMatroskaDemuxWatchVariable = 0;
??? MatroskaFileServerDemux::createNew(env, fileName, onMatroskaDemuxCreation, NULL);
??? env.taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);
?
??? ServerMediaSubsession* smss;
??? while ((smss = demux->newServerMediaSubsession()) != NULL) {
????? sms->addSubsession(smss);
??? }
? }
?
? return sms;
}
?
?
?
3)????? 服務端對Play命令的處理
?
?
void RTSPServer::RTSPClientSession
::handleCmd_withinSession(RTSPServer::RTSPClientConnection* ourClientConnection,
????????????? ? char const* cmdName,
????????????? ? char const* urlPreSuffix, char const* urlSuffix,
????????????? ? char const* fullRequestStr) {
? // This will either be:
? // - a non-aggregated operation, if "urlPreSuffix" is the session (stream)
? //?? name and "urlSuffix" is the subsession (track) name, or
? // - an aggregated operation, if "urlSuffix" is the session (stream) name,
? //?? or "urlPreSuffix" is the session (stream) name, and "urlSuffix" is empty,
? //?? or "urlPreSuffix" and "urlSuffix" are both nonempty, but when concatenated, (with "/") form the session (stream) name.
? // Begin by figuring out which of these it is:
? ServerMediaSubsession* subsession;
?
? noteLiveness();
? if (fOurServerMediaSession == NULL) {// There wasn't a previous SETUP!
??? ourClientConnection->handleCmd_notSupported();
??? return;
? } else if (urlSuffix[0] != '\0' && strcmp(fOurServerMediaSession->streamName(), urlPreSuffix) == 0) {
??? // Non-aggregated operation.
??? // Look up the media subsession whose track id is "urlSuffix":
??? ServerMediaSubsessionIterator iter(*fOurServerMediaSession);
??? while ((subsession = iter.next()) != NULL) {
????? if (strcmp(subsession->trackId(), urlSuffix) == 0)break;// success
??? }
??? if (subsession == NULL) { // no such track!
????? ourClientConnection->handleCmd_notFound();
????? return;
??? }
? } else if (strcmp(fOurServerMediaSession->streamName(), urlSuffix) == 0 ||
???? ???? (urlSuffix[0] == '\0' && strcmp(fOurServerMediaSession->streamName(), urlPreSuffix) == 0)) {
??? // Aggregated operation
??? subsession = NULL;
? } else if (urlPreSuffix[0] != '\0' && urlSuffix[0] !='\0') {
??? // Aggregated operation, if <urlPreSuffix>/<urlSuffix> is the session (stream) name:
??? unsigned const urlPreSuffixLen = strlen(urlPreSuffix);
??? if (strncmp(fOurServerMediaSession->streamName(), urlPreSuffix, urlPreSuffixLen) == 0 &&
???? fOurServerMediaSession->streamName()[urlPreSuffixLen] == '/' &&
???? strcmp(&(fOurServerMediaSession->streamName())[urlPreSuffixLen+1], urlSuffix) == 0) {
????? subsession = NULL;
??? } else {
????? ourClientConnection->handleCmd_notFound();
????? return;
??? }
? } else { // the request doesn't match a known stream and/or track at all!
??? ourClientConnection->handleCmd_notFound();
??? return;
? }
?
? if (strcmp(cmdName, "TEARDOWN") == 0) {
??? handleCmd_TEARDOWN(ourClientConnection, subsession);
? } else if (strcmp(cmdName, "PLAY") == 0) {
??? handleCmd_PLAY(ourClientConnection, subsession, fullRequestStr);
? } else if (strcmp(cmdName, "PAUSE") == 0) {
??? handleCmd_PAUSE(ourClientConnection, subsession);
? } else if (strcmp(cmdName, "GET_PARAMETER") == 0) {
??? handleCmd_GET_PARAMETER(ourClientConnection, subsession, fullRequestStr);
? } else if (strcmp(cmdName, "SET_PARAMETER") == 0) {
??? handleCmd_SET_PARAMETER(ourClientConnection, subsession, fullRequestStr);
? }
}
?
?
void RTSPServer::RTSPClientSession
::handleCmd_PLAY(RTSPServer::RTSPClientConnection* ourClientConnection,
???????? ?ServerMediaSubsession* subsession, char const* fullRequestStr) {
? char* rtspURL = fOurServer.rtspURL(fOurServerMediaSession, ourClientConnection->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);
?
? // Parse the client's "Range:" header, if any:
? float duration = 0.0;
? double rangeStart = 0.0, rangeEnd = 0.0;
? char* absStart = NULL; char* absEnd = NULL;
? Boolean sawRangeHeader = parseRangeHeader(fullRequestStr, rangeStart, rangeEnd, absStart, absEnd);
?
? if (sawRangeHeader && absStart == NULL/*not seeking by 'absolute' time*/) {
??? // Use this information, plus the stream's duration (if known), to create our own "Range:" header, for the response:
??? 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;
??? }
? }
?
? // 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.
? // (However, we don't do this if the "PLAY" request was for just a single subsession of a multiple-subsession stream;
? //? for such streams, seeking/scaling can be done only with an aggregate "PLAY".)
? for (i = 0; i < fNumStreamStates; ++i) {
??? if (subsession == NULL /* means: aggregated operation */ || fNumStreamStates == 1) {
????? if (sawScaleHeader) {
???? if (fStreamStates[i].subsession != NULL) {
???? ? fStreamStates[i].subsession->setStreamScale(fOurSessionId, fStreamStates[i].streamToken, scale);
???? }
????? }
????? if (sawRangeHeader) {
???? if (absStart != NULL) {
???? ? // Special case handling for seeking by 'absolute' time:
?
???? ? if (fStreamStates[i].subsession != NULL) {
???? ??? fStreamStates[i].subsession->seekStream(fOurSessionId, fStreamStates[i].streamToken, absStart, absEnd);
???? ? }
???? } else {
???? ? // Seeking by relative (NPT) time:
?
???? ? 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
???? ? }
???? ? if (fStreamStates[i].subsession != NULL) {
???? ??? u_int64_t numBytes;
???????? //查找流
???? ??? fStreamStates[i].subsession->seekStream(fOurSessionId, fStreamStates[i].streamToken,
??????????????????????????? ??? rangeStart, streamDuration, numBytes);
???? ? }
???? }
????? } else {
???? // No "Range:" header was specified in the "PLAY", so we do a 'null' seek (i.e., we don't seek at all):
???? if (fStreamStates[i].subsession != NULL) {
???? ? fStreamStates[i].subsession->nullSeekStream(fOurSessionId, fStreamStates[i].streamToken);
???? }
????? }
??? }
? }
?
? // Create the "Range:" header that we'll send back in our response.
? // (Note that we do this after seeking, in case the seeking operation changed the range start time.)
? char* rangeHeader;
? if (!sawRangeHeader) {
??? // There wasn't a "Range:" header in the request, so, in our response, begin the range with the current NPT (normal play time):
??? float curNPT = 0.0;
??? for (i = 0; i < fNumStreamStates; ++i) {
????? if (subsession == NULL /* means: aggregated operation */
???? ? || subsession == fStreamStates[i].subsession) {
???? if (fStreamStates[i].subsession == NULL)continue;
???? float npt = fStreamStates[i].subsession->getCurrentNPT(fStreamStates[i].streamToken);
???? if (npt > curNPT) curNPT = npt;
???? // Note: If this is an aggregate "PLAY" on a multi-subsession stream, then it's conceivable that the NPTs of each subsession
???? // may differ (if there has been a previous seek on just one subsession).?In this (unusual) case, we just return the
???? // largest NPT; I hope that turns out OK...
????? }
??? }
?
??? sprintf(buf, "Range: npt=%.3f-\r\n", curNPT);
? } else if (absStart != NULL) {
??? // We're seeking by 'absolute' time:
??? if (absEnd == NULL) {
????? sprintf(buf, "Range: clock=%s-\r\n", absStart);
??? } else {
????? sprintf(buf, "Range: clock=%s-%s\r\n", absStart, absEnd);
??? }
??? delete[] absStart; delete[] absEnd;
? } else {
??? // We're seeking by relative (NPT) time:
??? 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);
?
? // 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;
????? if (fStreamStates[i].subsession == NULL)continue;
????? fStreamStates[i].subsession->startStream(fOurSessionId,
?????????????????????? ?????? fStreamStates[i].streamToken,
?????????????????????? ?????? (TaskFunc*)noteClientLiveness, this,
?????????????????????? ?????? rtpSeqNum, rtpTimestamp,
???? ?????????????????? ?????? RTSPServer::RTSPClientConnection::handleAlternativeRequestByte, ourClientConnection);
????? 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*)ourClientConnection->fResponseBuffer,sizeof ourClientConnection->fResponseBuffer,
???? ?? "RTSP/1.0 200 OK\r\n"
???? ?? "CSeq: %s\r\n"
???? ?? "%s"
???? ?? "%s"
???? ?? "%s"
???? ?? "Session: %08X\r\n"
???? ?? "%s\r\n",
???? ?? ourClientConnection->fCurrentCSeq,
???? ?? dateHeader(),
???? ?? scaleHeader,
???? ?? rangeHeader,
???? ?? fOurSessionId,
???? ?? rtpInfo);
? delete[] rtpInfo; delete[] rangeHeader;
? delete[] scaleHeader; delete[] rtspURL;
}
?
Live555 RTP建立流程
?
RTP的建立流程在客戶端發送Setup請求開始建立，客戶端發送Setup請求時，會將RTP/RTCP的端口號告訴服務端，也會將Rtp over tcp還是udp的方式告訴到服務端，服務端收到Setup請求時，根據端口號建立socket，在收到客戶端的Play命令時，啟動流傳輸；啟動流傳輸的代碼如下：
?
void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,
??????????????????????????? void* streamToken,
??????????????????????????? TaskFunc* rtcpRRHandler,
??????????????????????????? void* rtcpRRHandlerClientData,
??????????????????????????? unsignedshort& rtpSeqNum,
??????????????????????????? unsigned& rtpTimestamp,
??????????????????????????? ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
??????????????????????????? void* serverRequestAlternativeByteHandlerClientData) {
? StreamState* streamState = (StreamState*)streamToken;
? Destinations* destinations
??? = (Destinations*)(fDestinationsHashTable->Lookup((charconst*)clientSessionId));
? if (streamState != NULL) {
??? streamState->startPlaying(destinations,
????????????? ????? rtcpRRHandler, rtcpRRHandlerClientData,
????????????? ????? serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
??? RTPSink* rtpSink = streamState->rtpSink(); // alias
??? if (rtpSink != NULL) {
????? rtpSeqNum = rtpSink->currentSeqNo();
????? rtpTimestamp = rtpSink->presetNextTimestamp();
??? }
? }
}
?
//
Live555 rtsp/rtp是同一個socket，但端口號不同嗎？
看源碼：
?
void 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;
?
? if (fLastStreamToken != NULL && fReuseFirstSource) {
??? // 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(envir()); // ensures that we skip over ports that are already in use
????? for (serverPortNum = fInitialPortNum; ; ++serverPortNum) {
???? struct in_addr dummyAddr; dummyAddr.s_addr = 0;
?
???? serverRTPPort = serverPortNum;
???? rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
???? if (rtpGroupsock->socketNum() >= 0)break;// success
????? }
?
????? rtcpGroupsock = NULL;
????? rtpSink = NULL;
????? udpSink = BasicUDPSink::createNew(envir(), rtpGroupsock);
??? } else {
????? // Normal case: We're streaming RTP (over UDP or TCP).?Create a pair of
????? // groupsocks (RTP and RTCP), with adjacent port numbers (RTP port number even):
????? NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
????? for (portNumBits serverPortNum = fInitialPortNum; ; serverPortNum += 2) {
???? struct in_addr dummyAddr; dummyAddr.s_addr = 0;
?
???? serverRTPPort = serverPortNum;
?
???? //建立RTPsocket
???? rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
???? if (rtpGroupsock->socketNum() < 0) {
???? ? delete rtpGroupsock;
???? ? continue; // try again
???? }
????
? ???//建立Rtcp socket
???? serverRTCPPort = serverPortNum+1;
???? rtcpGroupsock = new Groupsock(envir(), dummyAddr, serverRTCPPort, 255);
???? if (rtcpGroupsock->socketNum() < 0) {
???? ? delete rtpGroupsock;
???? ? delete rtcpGroupsock;
???? ? continue; // try again
???? }
?
???? break; // success
????? }
?
????? unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
????? rtpSink = createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);
????? udpSink = NULL;
??? }
?
??? // Turn off the destinations for each groupsock.?They'll get set later
??? // (unless TCP is used instead):
??? if (rtpGroupsock != NULL) rtpGroupsock->removeAllDestinations();
??? if (rtcpGroupsock != NULL) rtcpGroupsock->removeAllDestinations();
?
??? if (rtpGroupsock != NULL) {
????? // Try to use a big send buffer for RTP -?at least 0.1 second of
????? // specified bandwidth and at least 50 KB
????? unsigned rtpBufSize = streamBitrate * 25 / 2;// 1 kbps * 0.1 s = 12.5 bytes
????? if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
????? increaseSendBufferTo(envir(), rtpGroupsock->socketNum(), rtpBufSize);
??? }
?
??? // Set up the state of the stream.?The stream will get started later:
??? streamToken = fLastStreamToken
????? = new StreamState(*this, serverRTPPort, serverRTCPPort, rtpSink, udpSink,
????????????? streamBitrate, mediaSource,
????????????? rtpGroupsock, rtcpGroupsock);
? }
?
? // Record these destinations as being for this client session id:
? Destinations* destinations;
? if (tcpSocketNum < 0) { // UDP
??? destinations = new Destinations(destinationAddr, clientRTPPort, clientRTCPPort);
? } else { // TCP
??? destinations = new Destinations(tcpSocketNum, rtpChannelId, rtcpChannelId);
? }
? fDestinationsHashTable->Add((charconst*)clientSessionId, destinations);
}
?
//從這段代碼中可以看到rtsp，rtp，rtcp的socket是不同的；同時分析了客戶端的源碼，socket也是不一樣的，初始化subsession時，在其中會建立RTP/RTCP?socket以及RTPSource。對于每個subsession都會建立不同的socket。
?
?
3)MediaSession和socket的關系？一個MediaSession包括多個連接，關聯到多個socket嗎？
MediaSession 包括多個MediaSubSession,每個MediaSubSession對應相應的socket，source和sink，形成一個數據流！
?
總結
以上是生活随笔為你收集整理的庖丁解牛-----Live555源码彻底解密(根据MediaServer讲解Rtsp的建立过程)的全部內容，希望文章能夠幫你解決所遇到的問題。
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