本文介绍LIVE555 RTSP服务器的主要功能模块和工作流程，包括客户端连接建立、RTSP信令处理过程以及信令交互实例。深入分析了RTSPClientConnection类如何处理客户端请求，并解释了信令流程中各步骤的作用。
llive 555 信令类及消息
main 函数：
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;
#ifdef ACCESS_CONTROL
// To implement client access control to the RTSP server, do the following:
authDB = new UserAuthenticationDatabase;
authDB->addUserRecord("username1", "password1"); // replace these with real strings
// Repeat the above with each <username>, <password> that you wish to allow
// access to the server.
#endif
// 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;
/*创建rtsp 服务*/
rtspServer = DynamicRTSPServer::createNew(*env, rtspServerPortNum, authDB);
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);
}
*env << "LIVE555 Media Server\n";
*env << "\tversion " << MEDIA_SERVER_VERSION_STRING
<< " (LIVE555 Streaming Media library version "
<< LIVEMEDIA_LIBRARY_VERSION_STRING << ").\n";
*env << "Play streams from this server using the URL\n";
if (weHaveAnIPv4Address(*env)) {
char* rtspURLPrefix = rtspServer->ipv4rtspURLPrefix();
*env << "\t" << rtspURLPrefix << "<filename>\n";
delete[] rtspURLPrefix;
if (weHaveAnIPv6Address(*env)) *env << "or\n";
}
if (weHaveAnIPv6Address(*env)) {
char* rtspURLPrefix = rtspServer->ipv6rtspURLPrefix();
*env << "\t" << rtspURLPrefix << "<filename>\n";
delete[] rtspURLPrefix;
}
*env << "where <filename> is a file present in the current directory.\n";
*env << "Each file's type is inferred from its name suffix:\n";
*env << "\t\".264\" => a H.264 Video Elementary Stream file\n";
*env << "\t\".265\" => a H.265 Video Elementary Stream file\n";
*env << "\t\".aac\" => an AAC Audio (ADTS format) file\n";
*env << "\t\".ac3\" => an AC-3 Audio file\n";
*env << "\t\".amr\" => an AMR Audio file\n";
*env << "\t\".dv\" => a DV Video file\n";
*env << "\t\".m4e\" => a MPEG-4 Video Elementary Stream file\n";
*env << "\t\".mkv\" => a Matroska audio+video+(optional)subtitles file\n";
*env << "\t\".mp3\" => a MPEG-1 or 2 Audio file\n";
*env << "\t\".mpg\" => a MPEG-1 or 2 Program Stream (audio+video) file\n";
*env << "\t\".ogg\" or \".ogv\" or \".opus\" => an Ogg audio and/or video file\n";
*env << "\t\".ts\" => a MPEG Transport Stream file\n";
*env << "\t\t(a \".tsx\" index file - if present - provides server 'trick play' support)\n";
*env << "\t\".vob\" => a VOB (MPEG-2 video with AC-3 audio) file\n";
*env << "\t\".wav\" => a WAV Audio file\n";
*env << "\t\".webm\" => a WebM audio(Vorbis)+video(VP8) file\n";
*env << "See http://www.live555.com/mediaServer/ for additional documentation.\n";
// Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
// Try first with the default HTTP port (80), and then with the alternative HTTP
// port numbers (8000 and 8080).
if (rtspServer->setUpTunnelingOverHTTP(80) || rtspServer->setUpTunnelingOverHTTP(8000) || rtspServer->setUpTunnelingOverHTTP(8080)) {
*env << "(We use port " << rtspServer->httpServerPortNum() << " for optional RTSP-over-HTTP tunneling).)\n";
} else {
*env << "(RTSP-over-HTTP tunneling is not available.)\n";
}
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
main 函数中：1）new 一个DynamicRTSPServer类：DynamicRTSPServer 改类继承RTSPServer
2）setUpTunnelingOverHTTP，监听来自80等端口
Boolean RTSPServer::setUpTunnelingOverHTTP(Port httpPort) {
fHTTPServerSocketIPv4 = setUpOurSocket(envir(), httpPort, AF_INET);
fHTTPServerSocketIPv6 = setUpOurSocket(envir(), httpPort, AF_INET6);
if (fHTTPServerSocketIPv4 >= 0 || fHTTPServerSocketIPv6 >= 0) {
fHTTPServerPort = httpPort;
envir().taskScheduler().turnOnBackgroundReadHandling(fHTTPServerSocketIPv4,
incomingConnectionHandlerHTTPIPv4, this);
envir().taskScheduler().turnOnBackgroundReadHandling(fHTTPServerSocketIPv6,
incomingConnectionHandlerHTTPIPv6, this);
return True;
}
return False;
}
setUpTunnelingOverHTTP 函数一个任务，一直调用incomingConnectionHandlerHTTPIPv4或者
incomingConnectionHandlerHTTPIPv6
void RTSPServer::incomingConnectionHandlerHTTPIPv4(void* instance, int /*mask*/) {
RTSPServer* server = (RTSPServer*)instance;
server->incomingConnectionHandlerHTTPIPv4();
}
void RTSPServer::incomingConnectionHandlerHTTPIPv4() {
incomingConnectionHandlerOnSocket(fHTTPServerSocketIPv4);
}
void RTSPServer::incomingConnectionHandlerHTTPIPv6(void* instance, int /*mask*/) {
RTSPServer* server = (RTSPServer*)instance;
server->incomingConnectionHandlerHTTPIPv6();
}
void RTSPServer::incomingConnectionHandlerHTTPIPv6() {
incomingConnectionHandlerOnSocket(fHTTPServerSocketIPv6);
}
incomingConnectionHandlerHTTPIPv4 调用incomingConnectionHandlerOnSocket函数
void GenericMediaServer::incomingConnectionHandlerOnSocket(int serverSocket) {
struct sockaddr_storage 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;
}
ignoreSigPipeOnSocket(clientSocket); // so that clients on the same host that are killed don't also kill us
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 connection:
(void)createNewClientConnection(clientSocket, clientAddr);
}
调用accept 函数，有网络请求，就调用createNewClientConnection创建RTSPClientConnection
GenericMediaServer::ClientConnection*
RTSPServer::createNewClientConnection(int clientSocket, struct sockaddr_storage const& clientAddr) {
return new RTSPClientConnection(*this, clientSocket, clientAddr, fOurConnectionsUseTLS);
}
RTSPServer::RTSPClientConnection
::RTSPClientConnection(RTSPServer& ourServer,
int clientSocket, struct sockaddr_storage const& clientAddr,
Boolean useTLS)
: GenericMediaServer::ClientConnection(ourServer, clientSocket, clientAddr, useTLS),
fOurRTSPServer(ourServer), fClientInputSocket(fOurSocket), fClientOutputSocket(fOurSocket),
fAddressFamily(clientAddr.ss_family),
fIsActive(True), fRecursionCount(0), fOurSessionCookie(NULL), fScheduledDelayedTask(0) {
resetRequestBuffer();
}
RTSPClientConnection 类构造函数：调用GenericMediaServer::ClientConnection(ourServer, clientSocket, clientAddr, useTLS)；
GenericMediaServer::ClientConnection
::ClientConnection(GenericMediaServer& ourServer,
int clientSocket, struct sockaddr_storage const& clientAddr,
Boolean useTLS)
: fOurServer(ourServer), fOurSocket(clientSocket), fClientAddr(clientAddr), fTLS(envir()) {
// Add ourself to our 'client connections' table:
fOurServer.fClientConnections->Add((char const*)this, this);
if (useTLS) {
// Perform extra processing to handle a TLS connection:
fTLS.setCertificateAndPrivateKeyFileNames(ourServer.fTLSCertificateFileName,
ourServer.fTLSPrivateKeyFileName);
fTLS.isNeeded = True;
fTLS.tlsAcceptIsNeeded = True; // call fTLS.accept() the next time the socket is readable
}
// Arrange to handle incoming requests:
resetRequestBuffer();
envir().taskScheduler()
.setBackgroundHandling(fOurSocket, SOCKET_READABLE|SOCKET_EXCEPTION, incomingRequestHandler, this);
}
ClientConnection 构造函数：起一个任务一直调用incomingRequestHandler 函数
void GenericMediaServer::ClientConnection::incomingRequestHandler(void* instance, int /*mask*/) {
ClientConnection* connection = (ClientConnection*)instance;
connection->incomingRequestHandler();
}
void GenericMediaServer::ClientConnection::incomingRequestHandler() {
if (fTLS.tlsAcceptIsNeeded) { // we need to successfully call fTLS.accept() first:
if (fTLS.accept(fOurSocket) <= 0) return; // either an error, or we need to try again later
fTLS.tlsAcceptIsNeeded = False;
// We can now read data, as usual:
}
int bytesRead;
if (fTLS.isNeeded) {
bytesRead = fTLS.read(&fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft);
} else {
struct sockaddr_storage dummy; // 'from' address, meaningless in this case
bytesRead = readSocket(envir(), fOurSocket, &fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft, dummy);
}
handleRequestBytes(bytesRead);
}
incomingRequestHandler 函数调用handleRequestBytes 函数
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 && numBytesRemaining == 0) {
// 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 - fBase64RemainderCount + newBase64RemainderCount;
// adjust to allow for the size of the new decoded data (+ remainder)
delete[] decodedBytes;
}
fBase64RemainderCount = newBase64RemainderCount;
}
unsigned char* tmpPtr = fLastCRLF + 2;
if (fBase64RemainderCount == 0) { // no more Base-64 bytes remain to be read/decoded
// Look for the end of the message: <CR><LF><CR><LF>
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;
Boolean urlIsRTSPS;
Boolean playAfterSetup = False;
fLastCRLF[2] = '\0'; // temporarily, for parsing
Boolean parseSucceeded = parseRTSPRequestString((char*)fRequestBuffer, fLastCRLF+2 - fRequestBuffer,
cmdName, sizeof cmdName,
urlPreSuffix, sizeof urlPreSuffix,
urlSuffix, sizeof urlSuffix,
cseq, sizeof cseq,
sessionIdStr, sizeof sessionIdStr,
contentLength, urlIsRTSPS);
fLastCRLF[2] = '\r'; // restore its value
// Check first for a bogus "Content-Length" value that would cause a pointer wraparound:
if (tmpPtr + 2 + contentLength < tmpPtr + 2) {
#ifdef DEBUG
fprintf(stderr, "parseRTSPRequestString() returned a bogus \"Content-Length:\" value: 0x%x (%d)\n", contentLength, (int)contentLength);
#endif
contentLength = 0;
parseSucceeded = False;
}
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
// 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:
Boolean const requestIncludedSessionId = sessionIdStr[0] != '\0';
if (requestIncludedSessionId) {
clientSession
= (RTSPServer::RTSPClientSession*)(fOurRTSPServer.lookupClientSession(sessionIdStr));
if (clientSession != NULL) clientSession->noteLiveness();
}
// We now have a complete RTSP request.
// Handle the specified command (beginning with commands that are session-independent):
fCurrentCSeq = cseq;
// If the request specified the wrong type of URL
// (i.e., "rtsps" instead of "rtsp", or vice versa), then send back a 'redirect':
if (urlIsRTSPS != fOurRTSPServer.fWeServeSRTP) {
#ifdef DEBUG
fprintf(stderr, "Calling handleCmd_redirect()\n");
#endif
handleCmd_redirect(urlSuffix);
} else if (strcmp(cmdName, "OPTIONS") == 0) {
// If the "OPTIONS" command included a "Session:" id for a session that doesn't exist,
// then treat this as an error:
if (requestIncludedSessionId && clientSession == NULL) {
#ifdef DEBUG
fprintf(stderr, "Calling handleCmd_sessionNotFound() (case 1)\n");
#endif
handleCmd_sessionNotFound();
} else {
// Normal case:
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((char const*)fRequestBuffer);
} else if (strcmp(cmdName, "SET_PARAMETER") == 0) {
handleCmd_SET_PARAMETER((char const*)fRequestBuffer);
} else {
handleCmd_notSupported();
}
} else if (strcmp(cmdName, "DESCRIBE") == 0) {
handleCmd_DESCRIBE(urlPreSuffix, urlSuffix, (char const*)fRequestBuffer);
} else if (strcmp(cmdName, "SETUP") == 0) {
Boolean areAuthenticated = True;
if (!requestIncludedSessionId) {
// No session id was present in the request.
// So create a new "RTSPClientSession" object for this request.
// But first, make sure that we're authenticated to perform this command:
char urlTotalSuffix[2*RTSP_PARAM_STRING_MAX];
// enough space for urlPreSuffix/urlSuffix'\0'
urlTotalSuffix[0] = '\0';
if (urlPreSuffix[0] != '\0') {
strcat(urlTotalSuffix, urlPreSuffix);
strcat(urlTotalSuffix, "/");
}
strcat(urlTotalSuffix, urlSuffix);
if (authenticationOK("SETUP", urlTotalSuffix, (char const*)fRequestBuffer)) {
clientSession
= (RTSPServer::RTSPClientSession*)fOurRTSPServer.createNewClientSessionWithId();
} else {
areAuthenticated = False;
}
}
if (clientSession != NULL) {
clientSession->handleCmd_SETUP(this, urlPreSuffix, urlSuffix, (char const*)fRequestBuffer);
playAfterSetup = clientSession->fStreamAfterSETUP;
} else if (areAuthenticated) {
#ifdef DEBUG
fprintf(stderr, "Calling handleCmd_sessionNotFound() (case 2)\n");
#endif
handleCmd_sessionNotFound();
}
} else if (strcmp(cmdName, "TEARDOWN") == 0
|| strcmp(cmdName, "PLAY") == 0
|| strcmp(cmdName, "PAUSE") == 0
|| strcmp(cmdName, "GET_PARAMETER") == 0
|| strcmp(cmdName, "SET_PARAMETER") == 0) {
if (clientSession != NULL) {
clientSession->handleCmd_withinSession(this, cmdName, urlPreSuffix, urlSuffix, (char const*)fRequestBuffer);
} else {
#ifdef DEBUG
fprintf(stderr, "Calling handleCmd_sessionNotFound() (case 3)\n");
#endif
handleCmd_sessionNotFound();
}
} else if (strcmp(cmdName, "REGISTER") == 0 || strcmp(cmdName, "DEREGISTER") == 0) {
// Because - unlike other commands - an implementation of this command 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) {
// Check for special command-specific parameters in a "Transport:" header:
Boolean reuseConnection, deliverViaTCP;
char* proxyURLSuffix;
parseTransportHeaderForREGISTER((const char*)fRequestBuffer, reuseConnection, deliverViaTCP, proxyURLSuffix);
handleCmd_REGISTER(cmdName, url, urlSuffix, (char const*)fRequestBuffer, reuseConnection, deliverViaTCP, proxyURLSuffix);
delete[] proxyURLSuffix;
} 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 (strcmp(cmdName, "OPTIONS") == 0) {
handleHTTPCmd_OPTIONS();
} else 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, (char const*)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
unsigned const numBytesToWrite = strlen((char*)fResponseBuffer);
if (fTLS.isNeeded) {
fTLS.write((char const*)fResponseBuffer, numBytesToWrite);
} else {
send(fClientOutputSocket, (char const*)fResponseBuffer, numBytesToWrite, 0);
}
if (playAfterSetup) {
// 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, (char const*)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 it has a scheduledDelayedTask, don't delete the instance or close the sockets. The sockets can be reused in the task.
if (!fIsActive && fScheduledDelayedTask <= 0) {
if (fRecursionCount > 0) closeSockets(); else delete this;
// 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.
}
}
handleRequestBytes 处理rtsp 信令流程。
信息令抓包数据
OPTIONS rtsp://192.168.31.188:8000/slamtv60.264 RTSP/1.0
CSeq: 2
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
RTSP/1.0 200 OK
CSeq: 2
Date: Sat, Sep 02 2017 07:28:23 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, GET_PARAMETER, SET_PARAMETER
DESCRIBE rtsp://192.168.31.188:8000/slamtv60.264 RTSP/1.0
CSeq: 3
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
Accept: application/sdp
RTSP/1.0 200 OK
CSeq: 3
Date: Sat, Sep 02 2017 07:28:23 GMT
Content-Base: rtsp://192.168.31.188:8554/slamtv60.264/
Content-Type: application/sdp
Content-Length: 526
v=0
o=- 1504337303775687 1 IN IP4 192.168.31.188
s=H.264 Video, streamed by the LIVE555 Media Server
i=slamtv60.264
t=0 0
a=tool:LIVE555 Streaming Media v2017.07.18
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:H.264 Video, streamed by the LIVE555 Media Server
a=x-qt-text-inf:slamtv60.264
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
b=AS:500
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=4D4033;sprop-parameter-sets=Z01AM5JUDAS0IAAAAwBAAAAM0eMGVA==,aO48gA==
a=control:track1
SETUP rtsp://192.168.31.188:8554/slamtv60.264/track1 RTSP/1.0
CSeq: 4
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
Transport: RTP/AVP;unicast;client_port=49484-49485
RTSP/1.0 200 OK
CSeq: 4
Date: Sat, Sep 02 2017 07:28:23 GMT
Transport: RTP/AVP;unicast;destination=192.168.31.146;source=192.168.31.188;client_port=49484-49485;server_port=6970-6971
Session: EF285CD9;timeout=65
PLAY rtsp://192.168.31.188:8554/slamtv60.264/ RTSP/1.0
CSeq: 5
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
Session: EF285CD9
Range: npt=0.000-
RTSP/1.0 200 OK
CSeq: 5
Date: Sat, Sep 02 2017 07:28:23 GMT
Range: npt=0.000-
Session: EF285CD9
RTP-Info: url=rtsp://192.168.31.188:8554/slamtv60.264/track1;seq=7267;rtptime=3188764371
GET_PARAMETER rtsp://192.168.31.188:8554/slamtv60.264/ RTSP/1.0
CSeq: 6
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
Session: EF285CD9
RTSP/1.0 200 OK
CSeq: 6
Date: Sat, Sep 02 2017 07:28:23 GMT
Session: EF285CD9
Content-Length: 10
TEARDOWN rtsp://192.168.31.188:8554/slamtv60.264/ RTSP/1.0
CSeq: 7
User-Agent: LibVLC/2.2.6 (LIVE555 Streaming Media v2016.02.22)
Session: EF285CD9
RTSP/1.0 200 OK
CSeq: 7
Date: Sat, Sep 02 2017 07:28:34 GMT
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