void PassiveServerMediaSubsession::startStream(unsigned clientSessionId,
void* /*streamToken*/,
TaskFunc* rtcpRRHandler,
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp,
ServerRequestAlternativeByteHandler* /*serverRequestAlternativeByteHandler*/,
void* /*serverRequestAlternativeByteHandlerClientData*/) {
rtpSeqNum = fRTPSink.currentSeqNo();
rtpTimestamp = fRTPSink.presetNextTimestamp();
// Try to use a big send buffer for RTP - at least 0.1 second of
// specified bandwidth and at least 50 KB
unsigned streamBitrate = fRTCPInstance == NULL ? 50 : fRTCPInstance->totSessionBW(); // in kbps
unsigned rtpBufSize = streamBitrate * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
increaseSendBufferTo(envir(), fRTPSink.groupsockBeingUsed().socketNum(), rtpBufSize);
// Set up the handler for incoming RTCP "RR" packets from this client:
if (fRTCPInstance != NULL) {
RTCPSourceRecord* source = (RTCPSourceRecord*)(fClientRTCPSourceRecords->Lookup((char const*)clientSessionId));
if (source != NULL) {
fRTCPInstance->setSpecificRRHandler(source->addr, source->port,
rtcpRRHandler, rtcpRRHandlerClientData);
}
}
}
int HTTPSink::setUpOurSocket(UsageEnvironment& env, Port& ourPort) {
int ourSocket = -1;
do {
ourSocket = setupStreamSocket(env, ourPort);
if (ourSocket < 0) break;
// Make sure we have a big send buffer:
if (!increaseSendBufferTo(env, ourSocket, 50*1024)) break;
if (listen(ourSocket, 1) < 0) { // we allow only one connection
env.setResultErrMsg("listen() failed: ");
break;
}
if (ourPort.num() == 0) {
// bind() will have chosen a port for us; return it also:
if (!getSourcePort(env, ourSocket, ourPort)) break;
}
return ourSocket;
} while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
return -1;
}
int GenericMediaServer::setUpOurSocket(UsageEnvironment& env, Port& ourPort) {
int ourSocket = -1;
do {
// The following statement is enabled by default.
// Don't disable it (by defining ALLOW_SERVER_PORT_REUSE) unless you know what you're doing.
#if !defined(ALLOW_SERVER_PORT_REUSE) && !defined(ALLOW_RTSP_SERVER_PORT_REUSE)
// ALLOW_RTSP_SERVER_PORT_REUSE is for backwards-compatibility #####
NoReuse dummy(env); // Don't use this socket if there's already a local server using it
#endif
ourSocket = setupStreamSocket(env, ourPort);
if (ourSocket < 0) break;
// Make sure we have a big send buffer:
if (!increaseSendBufferTo(env, ourSocket, 50*1024)) break;
// Allow multiple simultaneous connections:
if (listen(ourSocket, LISTEN_BACKLOG_SIZE) < 0) {
env.setResultErrMsg("listen() failed: ");
break;
}
if (ourPort.num() == 0) {
// bind() will have chosen a port for us; return it also:
if (!getSourcePort(env, ourSocket, ourPort)) break;
}
return ourSocket;
} while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
return -1;
}
void RTSPServer::incomingConnectionHandler1() {
struct sockaddr_in clientAddr;
SOCKLEN_T clientAddrLen = sizeof clientAddr;
// printf("incomingConnectionHandler1\n"); //jay
int clientSocket = accept(fServerSocket, (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);
#if defined(DEBUG) || defined(DEBUG_CONNECTIONS)
envir() << "accept()ed connection from " << our_inet_ntoa(clientAddr.sin_addr) << '\n';
#endif
// Create a new object for this RTSP session.
// (Choose a random 32-bit integer for the session id (it will be encoded as a 8-digit hex number). We don't bother checking for
// a collision; the probability of two concurrent sessions getting the same session id is very low.)
unsigned sessionId = (unsigned)our_random();
(void)createNewClientSession(sessionId, clientSocket, clientAddr);
}
int RTSPServer::setUpOurSocket(UsageEnvironment& env, Port& ourPort) {
int ourSocket = -1;
do {
ourSocket = setupStreamSocket(env, ourPort);
if (ourSocket < 0) break;
// Make sure we have a big send buffer:
if (!increaseSendBufferTo(env, ourSocket, 50*1024)) break;
// Allow multiple simultaneous connections:
if (listen(ourSocket, LISTEN_BACKLOG_SIZE) < 0) {
env.setResultErrMsg("listen() failed: ");
break;
}
if (ourPort.num() == 0) {
// bind() will have chosen a port for us; return it also:
if (!getSourcePort(env, ourSocket, ourPort)) break;
}
return ourSocket;
} while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
return -1;
}
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
// Create a new object for handling this connection:
(void)createNewClientConnection(clientSocket, clientAddr);
}
void RTSPServer::incomingConnectionHandler1() {
struct sockaddr_in clientAddr;
SOCKLEN_T clientAddrLen = sizeof clientAddr;
int clientSocket = accept(fServerSocket, (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);
#if defined(DEBUG) || defined(DEBUG_CONNECTIONS)
fprintf(stderr, "accept()ed connection from %s\n", our_inet_ntoa(clientAddr.sin_addr));
#endif
// Create a new object for this RTSP session:
new RTSPClientSession(*this, ++fSessionIdCounter,
clientSocket, clientAddr);
}
void handleResponse(int resultCode, char* resultString) {
if (resultCode == 0) {
// The "REGISTER" request succeeded, so use the still-open RTSP socket to await incoming commands from the remote endpoint:
int sock;
struct sockaddr_in remoteAddress;
grabConnection(sock, remoteAddress);
if (sock >= 0) {
increaseSendBufferTo(envir(), sock, 50*1024); // in anticipation of streaming over it
(void)fOurServer.createNewClientConnection(sock, remoteAddress);
}
}
if (fResponseHandler != NULL) {
// Call our (REGISTER-specific) response handler now:
(*fResponseHandler)(&fOurServer, fRequestId, resultCode, resultString);
} else {
// We need to delete[] "resultString" before we leave:
delete[] resultString;
}
// We're completely done with the REGISTER command now, so delete ourself now:
delete this;
}
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;
rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
if (rtpGroupsock->socketNum() < 0) {
delete rtpGroupsock;
continue; // try again
}
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((char const*)clientSessionId, destinations);
}
Boolean DarwinInjector
::setDestination(char const* remoteRTSPServerNameOrAddress,
char const* remoteFileName,
char const* sessionName,
char const* sessionInfo,
portNumBits remoteRTSPServerPortNumber,
char const* remoteUserName,
char const* remotePassword,
char const* sessionAuthor,
char const* sessionCopyright,
int timeout) {
char* sdp = NULL;
char* url = NULL;
Boolean success = False; // until we learn otherwise
do {
// Construct a RTSP URL for the remote stream:
char const* const urlFmt = "rtsp://%s:%u/%s";
unsigned urlLen
= strlen(urlFmt) + strlen(remoteRTSPServerNameOrAddress) + 5 /* max short len */ + strlen(remoteFileName);
url = new char[urlLen];
sprintf(url, urlFmt, remoteRTSPServerNameOrAddress, remoteRTSPServerPortNumber, remoteFileName);
// Begin by creating our RTSP client object:
fRTSPClient = new RTSPClientForDarwinInjector(envir(), url, fVerbosityLevel, fApplicationName, this);
if (fRTSPClient == NULL) break;
// Get the remote RTSP server's IP address:
struct in_addr addr;
{
NetAddressList addresses(remoteRTSPServerNameOrAddress);
if (addresses.numAddresses() == 0) break;
NetAddress const* address = addresses.firstAddress();
addr.s_addr = *(unsigned*)(address->data());
}
AddressString remoteRTSPServerAddressStr(addr);
// Construct a SDP description for the session that we'll be streaming:
char const* const sdpFmt =
"v=0\r\n"
"o=- %u %u IN IP4 127.0.0.1\r\n"
"s=%s\r\n"
"i=%s\r\n"
"c=IN IP4 %s\r\n"
"t=0 0\r\n"
"a=x-qt-text-nam:%s\r\n"
"a=x-qt-text-inf:%s\r\n"
"a=x-qt-text-cmt:source application:%s\r\n"
"a=x-qt-text-aut:%s\r\n"
"a=x-qt-text-cpy:%s\r\n";
// plus, %s for each substream SDP
unsigned sdpLen = strlen(sdpFmt)
+ 20 /* max int len */ + 20 /* max int len */
+ strlen(sessionName)
+ strlen(sessionInfo)
+ strlen(remoteRTSPServerAddressStr.val())
+ strlen(sessionName)
+ strlen(sessionInfo)
+ strlen(fApplicationName)
+ strlen(sessionAuthor)
+ strlen(sessionCopyright)
+ fSubstreamSDPSizes;
unsigned const sdpSessionId = our_random32();
unsigned const sdpVersion = sdpSessionId;
sdp = new char[sdpLen];
sprintf(sdp, sdpFmt,
sdpSessionId, sdpVersion, // o= line
sessionName, // s= line
sessionInfo, // i= line
remoteRTSPServerAddressStr.val(), // c= line
sessionName, // a=x-qt-text-nam: line
sessionInfo, // a=x-qt-text-inf: line
fApplicationName, // a=x-qt-text-cmt: line
sessionAuthor, // a=x-qt-text-aut: line
sessionCopyright // a=x-qt-text-cpy: line
);
char* p = &sdp[strlen(sdp)];
SubstreamDescriptor* ss;
for (ss = fHeadSubstream; ss != NULL; ss = ss->next()) {
sprintf(p, "%s", ss->sdpLines());
p += strlen(p);
}
// Do a RTSP "ANNOUNCE" with this SDP description:
Authenticator auth;
Authenticator* authToUse = NULL;
if (remoteUserName[0] != '\0' || remotePassword[0] != '\0') {
auth.setUsernameAndPassword(remoteUserName, remotePassword);
authToUse = &auth;
}
fWatchVariable = 0;
(void)fRTSPClient->sendAnnounceCommand(sdp, genericResponseHandler, authToUse);
// Now block (but handling events) until we get a response:
envir().taskScheduler().doEventLoop(&fWatchVariable);
delete[] fResultString;
if (fResultCode != 0) break; // an error occurred with the RTSP "ANNOUNCE" command
// Next, tell the remote server to start receiving the stream from us.
// (To do this, we first create a "MediaSession" object from the SDP description.)
fSession = MediaSession::createNew(envir(), sdp);
if (fSession == NULL) break;
ss = fHeadSubstream;
MediaSubsessionIterator iter(*fSession);
MediaSubsession* subsession;
ss = fHeadSubstream;
unsigned streamChannelId = 0;
while ((subsession = iter.next()) != NULL) {
if (!subsession->initiate()) break;
fWatchVariable = 0;
(void)fRTSPClient->sendSetupCommand(*subsession, genericResponseHandler,
True /*streamOutgoing*/,
True /*streamUsingTCP*/);
// Now block (but handling events) until we get a response:
envir().taskScheduler().doEventLoop(&fWatchVariable);
delete[] fResultString;
if (fResultCode != 0) break; // an error occurred with the RTSP "SETUP" command
// Tell this subsession's RTPSink and RTCPInstance to use
// the RTSP TCP connection:
ss->rtpSink()->setStreamSocket(fRTSPClient->socketNum(), streamChannelId++);
if (ss->rtcpInstance() != NULL) {
ss->rtcpInstance()->setStreamSocket(fRTSPClient->socketNum(),
streamChannelId++);
}
ss = ss->next();
}
if (subsession != NULL) break; // an error occurred above
// Tell the RTSP server to start:
fWatchVariable = 0;
(void)fRTSPClient->sendPlayCommand(*fSession, genericResponseHandler);
// Now block (but handling events) until we get a response:
envir().taskScheduler().doEventLoop(&fWatchVariable);
delete[] fResultString;
if (fResultCode != 0) break; // an error occurred with the RTSP "PLAY" command
// Finally, make sure that the output TCP buffer is a reasonable size:
increaseSendBufferTo(envir(), fRTSPClient->socketNum(), 100*1024);
success = True;
} while (0);
delete[] sdp;
delete[] url;
return success;
}
