live555 学习笔记-建立RTSP连接的过程（RTSP服务器端）
监听
创建rtsp server，rtspserver的构造函数中，创建监听socket，添加到调度管理器BasicTaskScheduler的 HandlerSet* fHandlers当中，加入其中的handler的回调函数是incomingConnectionHandler。
RTSPServer::RTSPServer(UsageEnvironment& env,
int ourSocket, Port ourPort,
UserAuthenticationDatabase* authDatabase,
unsigned reclamationSeconds)
: GenericMediaServer(env, ourSocket, ourPort, reclamationSeconds),
fHTTPServerSocket(-1), fHTTPServerPort(0),
fClientConnectionsForHTTPTunneling(NULL), // will get created if needed
fTCPStreamingDatabase(HashTable::create(ONE_WORD_HASH_KEYS)),
fPendingRegisterOrDeregisterRequests(HashTable::create(ONE_WORD_HASH_KEYS)),
fRegisterOrDeregisterRequestCounter(0), fAuthDB(authDatabase), fAllowStreamingRTPOverTCP(True) {
}
GenericMediaServer
::GenericMediaServer(UsageEnvironment& env, int ourSocket, Port ourPort,
unsigned reclamationSeconds)
: Medium(env),
fServerSocket(ourSocket), fServerPort(ourPort), fReclamationSeconds(reclamationSeconds),
fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),
fClientConnections(HashTable::create(ONE_WORD_HASH_KEYS)),
fClientSessions(HashTable::create(STRING_HASH_KEYS)) {
ignoreSigPipeOnSocket(fServerSocket); // so that clients on the same host that are killed don't also kill us
// Arrange to handle connections from others:
env.taskScheduler().turnOnBackgroundReadHandling(fServerSocket, incomingConnectionHandler, this);
env << "GenericMediaServer constructor" << "add listenning socket" << "\n";
}
void GenericMediaServer::incomingConnectionHandler(void* instance, int /*mask*/) {
GenericMediaServer* server = (GenericMediaServer*)instance;
server->incomingConnectionHandler();
}
void GenericMediaServer::incomingConnectionHandler() {
incomingConnectionHandlerOnSocket(fServerSocket);
}
ClientConnection
incomingConnectionHandler中，在连接到来时创建一个clientconnection。
void GenericMediaServer::incomingConnectionHandlerOnSocket(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;
}
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
envir() << "incommingConnectionHandlerOnSocket" << "\n";
// Create a new object for handling this connection:
(void)createNewClientConnection(clientSocket, clientAddr);
}
GenericMediaServer::ClientConnection*
RTSPServer::createNewClientConnection(int clientSocket, struct sockaddr_in clientAddr) {
envir() << "new RTSPClinetConnection" << "\n";
return new RTSPClientConnection(*this, clientSocket, clientAddr);
}
RTSPServer::RTSPClientConnection
::RTSPClientConnection(RTSPServer& ourServer, int clientSocket, struct sockaddr_in clientAddr)
: GenericMediaServer::ClientConnection(ourServer, clientSocket, clientAddr),
fOurRTSPServer(ourServer), fClientInputSocket(fOurSocket), fClientOutputSocket(fOurSocket),
fIsActive(True), fRecursionCount(0), fOurSessionCookie(NULL) {
resetRequestBuffer();
}
GenericMediaServer::ClientConnection
::ClientConnection(GenericMediaServer& ourServer, int clientSocket, struct sockaddr_in clientAddr)
: fOurServer(ourServer), fOurSocket(clientSocket), fClientAddr(clientAddr) {
// Add ourself to our 'client connections' table:
fOurServer.fClientConnections->Add((char const*)this, this);
// Arrange to handle incoming requests:
resetRequestBuffer();
envir().taskScheduler()
.setBackgroundHandling(fOurSocket, SOCKET_READABLE|SOCKET_EXCEPTION, incomingRequestHandler, this);
}
request 的处理
incomingRequestHandler回调函数负责处理客户端发送的申请。
void RTSPServer::RTSPClientConnection::handleRequestBytes(int newBytesRead) {
int numBytesRemaining = 0;
++fRecursionCount;
envir() << "rtsp server handle request bytes" << "\n";
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;
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);
fLastCRLF[2] = '\r'; // restore its value
Boolean playAfterSetup = 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 (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) {
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) {
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 {
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
send(fClientOutputSocket, (char const*)fResponseBuffer, strlen((char*)fResponseBuffer), 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 (!fIsActive) {
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.
}
}
命令交互
Describe
客户端Describe request
服务器Describe response
response中携带的SDP
v=0
o=- 1592096473074819 1 IN IP4 10.0.2.103
s=Session streamed by "testOnDemandRTSPServer"
i=h264ESVideoTest
t=0 0
a=tool:LIVE555 Streaming Media v2017.10.28
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "testOnDemandRTSPServer"
a=x-qt-text-inf:h264ESVideoTest
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=640028;sprop-parameter-sets=J2QAKKwa1AoAt03AQEBQAAA+gAAHUwBA,KO4E8sA=
a=control:track1
SDP
SDP是会话描述协议。一个SDP的返回表示一个会话(session), 以v=开头，会话内部会有一个或多个media信息，以m=开头，a=用来表示会话或者媒体的属性。
a=control:
a=control属性可用于session或media
如果在media中存在，则内容为控制指定流的URL。在Setup时就是用这个URL。
如果在session中存在，则内容为用于aggregate control的URL。在PLAY，TEARDOWN时使用。
a=control的值可能是相对的URL，也可能是绝对的URL。判断相对还是绝对的方法参见RFC1808。（估计就是是否有串“://”）
如果a=control的值是相对的则从Content-Base，Content-Location，request URL中去获得基路径。如果a=control的值为*，则使用的值即为Content-Base，Content-Location，request URL的值。
request URL指DESCRIBE时的url。因为a=control是在DESCRIBE后返回的sdp中存放的。
Describe 响应
DESCRIBE 响应详解
下面补充上文中描述不够详细的地方。
调用H264VideoFileServerMediaSubsession::sdplines()产生媒体相关的SDP信息，此函数的实现再父类OnDemandServerMediaSubsession中；OnDemandServerMediaSubsession::sdpLines() 中调用两个重要的虚函数createNewStreamSource(), createNewRTPSink()，它们的实现在H264VideoFileServerMediaSubsession类中。
H264VideoFileServerMediaSubsession ::createNewStreamSource() 返回H264VideoStreamFramer对象。
H264VideoFileServerMediaSubsession ::createNewRTPSink() 返回H264VideoRTPSink 对象。
H264VideoFileServerMediaSubsession ::createNewRTPSink()
createNewRTPSink()中比较重要的初始化参数，创建outputpacketbuffer，其中很重要的两个参数fpreferred 最佳payloadsize的默认值是 RTP_PAYLOAD_PREFERRED_SIZE = 1000，fMax 最大包长度是
RTP_PAYLOAD_MAX_SIZE =1456。
#ifndef RTP_PAYLOAD_MAX_SIZE
#define RTP_PAYLOAD_MAX_SIZE 1456
// Default max packet size (1500, minus allowance for IP, UDP, UMTP headers)
// (Also, make it a multiple of 4 bytes, just in case that matters.)
#endif
#ifndef RTP_PAYLOAD_PREFERRED_SIZE
#define RTP_PAYLOAD_PREFERRED_SIZE ((RTP_PAYLOAD_MAX_SIZE) < 1000 ? (RTP_PAYLOAD_MAX_SIZE) : 1000)
#endif
MultiFramedRTPSink::MultiFramedRTPSink(UsageEnvironment& env,
Groupsock* rtpGS,
unsigned char rtpPayloadType,
unsigned rtpTimestampFrequency,
char const* rtpPayloadFormatName,
unsigned numChannels)
: RTPSink(env, rtpGS, rtpPayloadType, rtpTimestampFrequency,
rtpPayloadFormatName, numChannels),
fOutBuf(NULL), fCurFragmentationOffset(0), fPreviousFrameEndedFragmentation(False),
fOnSendErrorFunc(NULL), fOnSendErrorData(NULL) {
setPacketSizes((RTP_PAYLOAD_PREFERRED_SIZE), (RTP_PAYLOAD_MAX_SIZE));
}
void MultiFramedRTPSink::setPacketSizes(unsigned preferredPacketSize,
unsigned maxPacketSize) {
if (preferredPacketSize > maxPacketSize || preferredPacketSize == 0) return;
// sanity check
delete fOutBuf;
fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);
fOurMaxPacketSize = maxPacketSize; // save value, in case subclasses need it
}
// OutPacketBuffer //
unsigned OutPacketBuffer::maxSize = 600000; // by default
OutPacketBuffer
::OutPacketBuffer(unsigned preferredPacketSize, unsigned maxPacketSize, unsigned maxBufferSize)
: fPreferred(preferredPacketSize), fMax(maxPacketSize),
fOverflowDataSize(0) {
if (maxBufferSize == 0) maxBufferSize = maxSize;
unsigned maxNumPackets = (maxBufferSize + (maxPacketSize-1))/maxPacketSize;
fLimit = maxNumPackets*maxPacketSize;
fBuf = new unsigned char[fLimit];
resetPacketStart();
resetOffset();
resetOverflowData();
}
获取sdpline
char const*
OnDemandServerMediaSubsession::sdpLines() {
if (fSDPLines == NULL) {
// We need to construct a set of SDP lines that describe this
// subsession (as a unicast stream). To do so, we first create
// dummy (unused) source and "RTPSink" objects,
// whose parameters we use for the SDP lines:
unsigned estBitrate;
//创建H264VideoStreamFramer对象
FramedSource* inputSource = createNewStreamSource(0, estBitrate);
if (inputSource == NULL) return NULL; // file not found
struct in_addr dummyAddr;
dummyAddr.s_addr = 0;
Groupsock* dummyGroupsock = createGroupsock(dummyAddr, 0);
unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
RTPSink* dummyRTPSink = createNewRTPSink(dummyGroupsock, rtpPayloadType, inputSource);
if (dummyRTPSink != NULL && dummyRTPSink->estimatedBitrate() > 0) estBitrate = dummyRTPSink->estimatedBitrate();
setSDPLinesFromRTPSink(dummyRTPSink, inputSource, estBitrate);
Medium::close(dummyRTPSink);
delete dummyGroupsock;
closeStreamSource(inputSource);
}
return fSDPLines;
}
创建H264VideoStreamFramer对象时，同时创建关联的H264or5VideoStreamParser对象，H264VideoStreamFramer还关联ByteStreamFileSource对象。
OnDemandServerMediaSubsession::setSDPLinesFromRTPSink函数，在这个函数中，我们通过创建的FramedSource对象和RTPSink对象将文件播放一段以便产生出sdp信息。在此，我要插一下Live555 RTSPServer播放媒体资源的一个大体流程：RTSPServer使用RTPSink获得和保存RTP包，RTPSink不断地向FramedSource请求帧数据，FramedSource取得帧数据后就调用回调函数把数据给RTPSink处理，RTPSink在回调函数中将数据发送给客户端。
void OnDemandServerMediaSubsession
::setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource, unsigned estBitrate) {
if (rtpSink == NULL) return;
char const* mediaType = rtpSink->sdpMediaType();
unsigned char rtpPayloadType = rtpSink->rtpPayloadType();
AddressString ipAddressStr(fServerAddressForSDP);
char* rtpmapLine = rtpSink->rtpmapLine();
char const* rtcpmuxLine = fMultiplexRTCPWithRTP ? "a=rtcp-mux\r\n" : "";
char const* rangeLine = rangeSDPLine();
char const* auxSDPLine = getAuxSDPLine(rtpSink, inputSource);
if (auxSDPLine == NULL) auxSDPLine = "";
char const* const sdpFmt =
"m=%s %u RTP/AVP %d\r\n"
"c=IN IP4 %s\r\n"
"b=AS:%u\r\n"
"%s"
"%s"
"%s"
"%s"
"a=control:%s\r\n";
unsigned sdpFmtSize = strlen(sdpFmt)
+ strlen(mediaType) + 5 /* max short len */ + 3 /* max char len */
+ strlen(ipAddressStr.val())
+ 20 /* max int len */
+ strlen(rtpmapLine)
+ strlen(rtcpmuxLine)
+ strlen(rangeLine)
+ strlen(auxSDPLine)
+ strlen(trackId());
char* sdpLines = new char[sdpFmtSize];
sprintf(sdpLines, sdpFmt,
mediaType, // m= <media>
fPortNumForSDP, // m= <port>
rtpPayloadType, // m= <fmt list>
ipAddressStr.val(), // c= address
estBitrate, // b=AS:<bandwidth>
rtpmapLine, // a=rtpmap:... (if present)
rtcpmuxLine, // a=rtcp-mux:... (if present)
rangeLine, // a=range:... (if present)
auxSDPLine, // optional extra SDP line
trackId()); // a=control:<track-id>
delete[] (char*)rangeLine; delete[] rtpmapLine;
fSDPLines = strDup(sdpLines);
delete[] sdpLines;
}
char const* H264VideoFileServerMediaSubsession::getAuxSDPLine(RTPSink* rtpSink, FramedSource* inputSource) {
if (fAuxSDPLine != NULL) return fAuxSDPLine; // it's already been set up (for a previous client)
if (fDummyRTPSink == NULL) { // we're not already setting it up for another, concurrent stream
// Note: For H264 video files, the 'config' information ("profile-level-id" and "sprop-parameter-sets") isn't known
// until we start reading the file. This means that "rtpSink"s "auxSDPLine()" will be NULL initially,
// and we need to start reading data from our file until this changes.
fDummyRTPSink = rtpSink;
// Start reading the file:
fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
// Check whether the sink's 'auxSDPLine()' is ready:
checkForAuxSDPLine(this);
}
envir().taskScheduler().doEventLoop(&fDoneFlag);
return fAuxSDPLine;
}
播放的具体过程参考播放过程
checkForAuxSDPLine()函数检查是否获取到附加sdpline，如果没有获取到，程序就停在这里进入事件调度主循环，不断地播放和检查。获取到时跳出主循环getAuxSDPLine()函数返回。
static void checkForAuxSDPLine(void* clientData) {
H264VideoFileServerMediaSubsession* subsess = (H264VideoFileServerMediaSubsession*)clientData;
subsess->checkForAuxSDPLine1();
}
void H264VideoFileServerMediaSubsession::checkForAuxSDPLine1() {
nextTask() = NULL;
char const* dasl;
if (fAuxSDPLine != NULL) {
// Signal the event loop that we're done:
setDoneFlag();
} else if (fDummyRTPSink != NULL && (dasl = fDummyRTPSink->auxSDPLine()) != NULL) {
fAuxSDPLine = strDup(dasl);
fDummyRTPSink = NULL;
// Signal the event loop that we're done:
setDoneFlag();
} else if (!fDoneFlag) {
// try again after a brief delay:
int uSecsToDelay = 100000; // 100 ms
nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecsToDelay,
(TaskFunc*)checkForAuxSDPLine, this);
}
}
char const* H264VideoRTPSink::auxSDPLine() {
// Generate a new "a=fmtp:" line each time, using our SPS and PPS (if we have them),
// otherwise parameters from our framer source (in case they've changed since the last time that
// we were called):
H264or5VideoStreamFramer* framerSource = NULL;
u_int8_t* vpsDummy = NULL; unsigned vpsDummySize = 0;
u_int8_t* sps = fSPS; unsigned spsSize = fSPSSize;
u_int8_t* pps = fPPS; unsigned ppsSize = fPPSSize;
if (sps == NULL || pps == NULL) {
// We need to get SPS and PPS from our framer source:
if (fOurFragmenter == NULL) return NULL; // we don't yet have a fragmenter (and therefore not a source)
framerSource = (H264or5VideoStreamFramer*)(fOurFragmenter->inputSource());
if (framerSource == NULL) return NULL; // we don't yet have a source
framerSource->getVPSandSPSandPPS(vpsDummy, vpsDummySize, sps, spsSize, pps, ppsSize);
if (sps == NULL || pps == NULL) return NULL; // our source isn't ready
}
// Set up the "a=fmtp:" SDP line for this stream:
u_int8_t* spsWEB = new u_int8_t[spsSize]; // "WEB" means "Without Emulation Bytes"
unsigned spsWEBSize = removeH264or5EmulationBytes(spsWEB, spsSize, sps, spsSize);
if (spsWEBSize < 4) { // Bad SPS size => assume our source isn't ready
delete[] spsWEB;
return NULL;
}
u_int32_t profileLevelId = (spsWEB[1]<<16) | (spsWEB[2]<<8) | spsWEB[3];
delete[] spsWEB;
char* sps_base64 = base64Encode((char*)sps, spsSize);
char* pps_base64 = base64Encode((char*)pps, ppsSize);
char const* fmtpFmt =
"a=fmtp:%d packetization-mode=1"
";profile-level-id=%06X"
";sprop-parameter-sets=%s,%s\r\n";
unsigned fmtpFmtSize = strlen(fmtpFmt)
+ 3 /* max char len */
+ 6 /* 3 bytes in hex */
+ strlen(sps_base64) + strlen(pps_base64);
char* fmtp = new char[fmtpFmtSize];
sprintf(fmtp, fmtpFmt,
rtpPayloadType(),
profileLevelId,
sps_base64, pps_base64);
delete[] sps_base64;
delete[] pps_base64;
delete[] fFmtpSDPLine; fFmtpSDPLine = fmtp;
return fFmtpSDPLine;
}
SETUP
RTSP客户端 —> RTSP服务器端 SETUP命令 请求建立对某个媒体资源的连接
RTSP服务器端 —> RTSP客户端 回复SETUP命令 回复建立连接的结果
SETUP请求服务器建立媒体session
request
SETUP rtsp://172.29.64.90:8554/h264ESVideoTest/track1 RTSP/1.0
CSeq: 4
User-Agent: LibVLC/3.0.8 (LIVE555 Streaming Media v2016.11.28)
Transport: RTP/AVP;unicast;client_port=57020-57021
response
RTSP/1.0 200 OK
CSeq: 4
Date: Mon, Jun 22 2020 06:00:30 GMT
Transport: RTP/AVP;unicast;destination=172.29.65.253;source=172.29.64.90;client_port=57020-57021;server_port=6970-6971
Session: B1BFFAF6;timeout=65
其中Transport 头value的格式如下：
Transport = "Transport" ":"
1\#transport-spec
transport-spec = transport-protocol/profile[/lower-transport]
*parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
parameter = ( "unicast" | "multicast" )
| ";" "destination" [ "=" address ]
| ";" "interleaved" "=" channel [ "-" channel ]
| ";" "append"
| ";" "ttl" "=" ttl
| ";" "layers" "=" 1*DIGIT
| ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ]
| ";" "server_port" "=" port [ "-" port ]
| ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1\#mode <">
ttl = 1*3(DIGIT)
port = 1*5(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
address = host
mode = <"> *Method <"> | Method
一个Transport头只用来描述一个RTP 流。session 头包含session-id和超时时间。
(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) {
handleCmd_sessionNotFound();
}
}
收到SETUP命令时创建ClientSession，handleCmd_SETUP开始处理SETUP请求。
剩下的主要动作在getStreamParameters中
- createNewStreamSource()
- 两次createGroupsock() 创建server端udp socket
- createNewRTPSink()
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 = NULL;
BasicUDPSink* udpSink = NULL;
Groupsock* rtpGroupsock = NULL;
Groupsock* rtcpGroupsock = NULL;
if (clientRTPPort.num() != 0 || tcpSocketNum >= 0) { // Normal case: Create destinations
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 = createGroupsock(dummyAddr, serverRTPPort);
if (rtpGroupsock->socketNum() >= 0) break; // success
}
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).
// (If we're multiplexing RTCP and RTP over the same port number, it can be odd or even.)
NoReuse dummy(envir()); // ensures that we skip over ports that are already in use
for (portNumBits serverPortNum = fInitialPortNum; ; ++serverPortNum) {
struct in_addr dummyAddr; dummyAddr.s_addr = 0;
serverRTPPort = serverPortNum;
rtpGroupsock = createGroupsock(dummyAddr, serverRTPPort);
if (rtpGroupsock->socketNum() < 0) {
delete rtpGroupsock;
continue; // try again
}
if (fMultiplexRTCPWithRTP) {
// Use the RTP 'groupsock' object for RTCP as well:
serverRTCPPort = serverRTPPort;
rtcpGroupsock = rtpGroupsock;
} else {
// Create a separate 'groupsock' object (with the next (odd) port number) for RTCP:
serverRTCPPort = ++serverPortNum;
rtcpGroupsock = createGroupsock(dummyAddr, serverRTCPPort);
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);
if (rtpSink != NULL && rtpSink->estimatedBitrate() > 0) streamBitrate = rtpSink->estimatedBitrate();
}
// 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:
// key 为sessionid， value是 客户端目的地址，客户rtpport 客户端rtcpport
Destinations* destinations;
if (tcpSocketNum < 0) { // UDP
destinations = new Destinations(destinationAddr, clientRTPPort, clientRTCPPort);
} else { // TCP
destinations = new Destinations(tcpSocketNum, rtpChannelId, rtcpChannelId);
}
fDestinationsHashTable->Add((char const*)clientSessionId, destinations);
}
PLAY
request
PLAY rtsp://10.0.15.50:8554/h264ESVideoTest/ RTSP/1.0
CSeq: 5
User-Agent: LibVLC/3.0.8 (LIVE555 Streaming Media v2016.11.28)
Session: B6A584EF
Range: npt=0.000-
response
RTSP/1.0 200 OK
CSeq: 5
Date: Tue, Jun 30 2020 08:46:58 GMT
Range: npt=0.000-
Session: B6A584EF
RTP-Info: url=rtsp://10.0.15.50:8554/h264ESVideoTest/track1;seq=38987;rtptime=380618412
经过上面的步骤后，服务器端就已经准备好向客户端传送RTP包以及RTCP包了，等待客户端发送PLAY命令后开始传输。服务器端收到PLAY命令后，调用RTSPClientSession::handleCmd_PLAY函数处理。在handleCmd_PLAY函数中，首先提取Scale，表示客户端期望的播放速度（正常、快进、快退），然后提取Range，表示客户端期望的播放起止范围，根据这两个参数分别调用ServerMediaSubsession::setStreamScale函数和ServerMediaSubsession::seekStream函数，最后调用ServerMediaSubsession::startStream函数开始传输数据。实际调用的是OnDemandServerMediaSubsession::startStream函数，看一下这个函数的内容：
void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,
void* streamToken,
TaskFunc* rtcpRRHandler,
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData) {
StreamState* streamState = (StreamState*)streamToken;
Destinations* destinations
= (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
if (streamState != NULL) {
streamState->startPlaying(destinations, clientSessionId,
rtcpRRHandler, rtcpRRHandlerClientData,
serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
RTPSink* rtpSink = streamState->rtpSink(); // alias
if (rtpSink != NULL) {
rtpSeqNum = rtpSink->currentSeqNo();
rtpTimestamp = rtpSink->presetNextTimestamp();
}
}
}
在OnDemandServerMediaSubsessionstartStream函数中，主要是调用了StreamState::startPlaying函数，来看一下这个函数：
void StreamState
::startPlaying(Destinations* dests, unsigned clientSessionId,
TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData) {
if (dests == NULL) return;
if (fRTCPInstance == NULL && fRTPSink != NULL) {
// Create (and start) a 'RTCP instance' for this RTP sink:
fRTCPInstance = fMaster.createRTCP(fRTCPgs, fTotalBW, (unsigned char*)fMaster.fCNAME, fRTPSink);
// Note: This starts RTCP running automatically
fRTCPInstance->setAppHandler(fMaster.fAppHandlerTask, fMaster.fAppHandlerClientData);
}
if (dests->isTCP) {
// Change RTP and RTCP to use the TCP socket instead of UDP:
if (fRTPSink != NULL) {
fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
RTPInterface
::setServerRequestAlternativeByteHandler(fRTPSink->envir(), dests->tcpSocketNum,
serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
// So that we continue to handle RTSP commands from the client
}
if (fRTCPInstance != NULL) {
fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
rtcpRRHandler, rtcpRRHandlerClientData);
}
} else {
// 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, clientSessionId);
if (fRTCPgs != NULL && !(fRTCPgs == fRTPgs && dests->rtcpPort.num() == dests->rtpPort.num())) {
fRTCPgs->addDestination(dests->addr, dests->rtcpPort, clientSessionId);
}
if (fRTCPInstance != NULL) {
fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
rtcpRRHandler, rtcpRRHandlerClientData);
}
}
if (fRTCPInstance != NULL) {
// Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to
// get RTCP-synchronized presentation times immediately:
fRTCPInstance->sendReport();
}
if (!fAreCurrentlyPlaying && fMediaSource != NULL) {
if (fRTPSink != NULL) {
fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
fAreCurrentlyPlaying = True;
} else if (fUDPSink != NULL) {
fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
fAreCurrentlyPlaying = True;
}
}
}
在StreamState::startPlaying函数中，将客户端添加到目的客户端列表中去，然后调用RTPSink::startPlaying函数，实际调用的是MediaSink::startPlaying函数.
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