void RTSPServerSupportingHTTPStreaming::RTSPClientConnectionSupportingHTTPStreaming
::handleHTTPCmd_StreamingGET(char const* urlSuffix, char const* /*fullRequestStr*/) {
// If "urlSuffix" ends with "?segment=<offset-in-seconds>,<duration-in-seconds>", then strip this off, and send the
// specified segment. Otherwise, construct and send a playlist that consists of segments from the specified file.
do {
char const* questionMarkPos = strrchr(urlSuffix, '?');
if (questionMarkPos == NULL) break;
unsigned offsetInSeconds, durationInSeconds;
if (sscanf(questionMarkPos, "?segment=%u,%u", &offsetInSeconds, &durationInSeconds) != 2) break;
char* streamName = strDup(urlSuffix);
streamName[questionMarkPos-urlSuffix] = '\0';
do {
ServerMediaSession* session = fOurServer.lookupServerMediaSession(streamName);
if (session == NULL) {
handleHTTPCmd_notFound();
break;
}
// We can't send multi-subsession streams over HTTP (because there's no defined way to multiplex more than one subsession).
// Therefore, use the first (and presumed only) substream:
ServerMediaSubsessionIterator iter(*session);
ServerMediaSubsession* subsession = iter.next();
if (subsession == NULL) {
// Treat an 'empty' ServerMediaSession the same as one that doesn't exist at all:
handleHTTPCmd_notFound();
break;
}
// Call "getStreamParameters()" to create the stream's source. (Because we're not actually streaming via RTP/RTCP, most
// of the parameters to the call are dummy.)
++fClientSessionId;
Port clientRTPPort(0), clientRTCPPort(0), serverRTPPort(0), serverRTCPPort(0);
netAddressBits destinationAddress = 0;
u_int8_t destinationTTL = 0;
Boolean isMulticast = False;
void* streamToken;
subsession->getStreamParameters(fClientSessionId, 0, clientRTPPort,clientRTCPPort, -1,0,0, destinationAddress,destinationTTL, isMulticast, serverRTPPort,serverRTCPPort, streamToken);
// Seek the stream source to the desired place, with the desired duration, and (as a side effect) get the number of bytes:
double dOffsetInSeconds = (double)offsetInSeconds;
u_int64_t numBytes;
subsession->seekStream(fClientSessionId, streamToken, dOffsetInSeconds, (double)durationInSeconds, numBytes);
unsigned numTSBytesToStream = (unsigned)numBytes;
if (numTSBytesToStream == 0) {
// For some reason, we do not know the size of the requested range. We can't handle this request:
handleHTTPCmd_notSupported();
break;
}
// Construct our response:
snprintf((char*)fResponseBuffer, sizeof fResponseBuffer,
"HTTP/1.1 200 OK\r\n"
"%s"
"Server: LIVE555 Streaming Media v%s\r\n"
"%s"
"Content-Length: %d\r\n"
"Content-Type: text/plain; charset=ISO-8859-1\r\n"
"\r\n",
dateHeader(),
LIVEMEDIA_LIBRARY_VERSION_STRING,
lastModifiedHeader(streamName),
numTSBytesToStream);
// Send the response now, because we're about to add more data (from the source):
send(fClientOutputSocket, (char const*)fResponseBuffer, strlen((char*)fResponseBuffer), 0);
fResponseBuffer[0] = '\0'; // We've already sent the response. This tells the calling code not to send it again.
// Ask the media source to deliver - to the TCP sink - the desired data:
if (fStreamSource != NULL) { // sanity check
if (fTCPSink != NULL) fTCPSink->stopPlaying();
Medium::close(fStreamSource);
}
fStreamSource = subsession->getStreamSource(streamToken);
if (fStreamSource != NULL) {
if (fTCPSink == NULL) fTCPSink = TCPStreamSink::createNew(envir(), fClientOutputSocket);
fTCPSink->startPlaying(*fStreamSource, afterStreaming, this);
}
} while(0);
delete[] streamName;
return;
} while (0);
// "urlSuffix" does not end with "?segment=<offset-in-seconds>,<duration-in-seconds>".
// Construct and send a playlist that describes segments from the specified file.
// First, make sure that the named file exists, and is streamable:
ServerMediaSession* session = fOurServer.lookupServerMediaSession(urlSuffix);
if (session == NULL) {
handleHTTPCmd_notFound();
return;
}
// To be able to construct a playlist for the requested file, we need to know its duration:
float duration = session->duration();
if (duration <= 0.0) {
// We can't handle this request:
handleHTTPCmd_notSupported();
return;
}
// Now, construct the playlist. It will consist of a prefix, one or more media file specifications, and a suffix:
unsigned const maxIntLen = 10; // >= the maximum possible strlen() of an integer in the playlist
char const* const playlistPrefixFmt =
"#EXTM3U\r\n"
"#EXT-X-ALLOW-CACHE:YES\r\n"
"#EXT-X-MEDIA-SEQUENCE:0\r\n"
"#EXT-X-TARGETDURATION:%d\r\n";
unsigned const playlistPrefixFmt_maxLen = strlen(playlistPrefixFmt) + maxIntLen;
char const* const playlistMediaFileSpecFmt =
"#EXTINF:%d,\r\n"
"%s?segment=%d,%d\r\n";
unsigned const playlistMediaFileSpecFmt_maxLen = strlen(playlistMediaFileSpecFmt) + maxIntLen + strlen(urlSuffix) + 2*maxIntLen;
char const* const playlistSuffixFmt =
"#EXT-X-ENDLIST\r\n";
unsigned const playlistSuffixFmt_maxLen = strlen(playlistSuffixFmt);
// Figure out the 'target duration' that will produce a playlist that will fit in our response buffer. (But make it at least 10s.)
unsigned const playlistMaxSize = 10000;
unsigned const mediaFileSpecsMaxSize = playlistMaxSize - (playlistPrefixFmt_maxLen + playlistSuffixFmt_maxLen);
unsigned const maxNumMediaFileSpecs = mediaFileSpecsMaxSize/playlistMediaFileSpecFmt_maxLen;
unsigned targetDuration = (unsigned)(duration/maxNumMediaFileSpecs + 1);
if (targetDuration < 10) targetDuration = 10;
char* playlist = new char[playlistMaxSize];
char* s = playlist;
sprintf(s, playlistPrefixFmt, targetDuration);
s += strlen(s);
unsigned durSoFar = 0;
while (1) {
unsigned dur = targetDuration < duration ? targetDuration : (unsigned)duration;
duration -= dur;
sprintf(s, playlistMediaFileSpecFmt, dur, urlSuffix, durSoFar, dur);
s += strlen(s);
if (duration < 1.0) break;
durSoFar += dur;
}
sprintf(s, playlistSuffixFmt);
s += strlen(s);
unsigned playlistLen = s - playlist;
// Construct our response:
snprintf((char*)fResponseBuffer, sizeof fResponseBuffer,
"HTTP/1.1 200 OK\r\n"
"%s"
"Server: LIVE555 Streaming Media v%s\r\n"
"%s"
"Content-Length: %d\r\n"
"Content-Type: application/vnd.apple.mpegurl\r\n"
"\r\n",
dateHeader(),
LIVEMEDIA_LIBRARY_VERSION_STRING,
lastModifiedHeader(urlSuffix),
playlistLen);
// Send the response header now, because we're about to add more data (the playlist):
send(fClientOutputSocket, (char const*)fResponseBuffer, strlen((char*)fResponseBuffer), 0);
fResponseBuffer[0] = '\0'; // We've already sent the response. This tells the calling code not to send it again.
// Then, send the playlist. Because it's large, we don't do so using "send()", because that might not send it all at once.
// Instead, we stream the playlist over the TCP socket:
if (fPlaylistSource != NULL) { // sanity check
if (fTCPSink != NULL) fTCPSink->stopPlaying();
Medium::close(fPlaylistSource);
}
fPlaylistSource = ByteStreamMemoryBufferSource::createNew(envir(), (u_int8_t*)playlist, playlistLen);
if (fTCPSink == NULL) fTCPSink = TCPStreamSink::createNew(envir(), fClientOutputSocket);
fTCPSink->startPlaying(*fPlaylistSource, afterStreaming, this);
}
void RTSPServer::RTSPClientSession
::handleCmd_SETUP(char const* cseq,
char const* urlPreSuffix, char const* urlSuffix,
char const* fullRequestStr) {
// "urlPreSuffix" should be the session (stream) name, and
// "urlSuffix" should be the subsession (track) name.
char const* streamName = urlPreSuffix;
char const* trackId = urlSuffix;
// Check whether we have existing session state, and, if so, whether it's
// for the session that's named in "streamName". (Note that we don't
// support more than one concurrent session on the same client connection.) #####
if (fOurServerMediaSession != NULL
&& strcmp(streamName, fOurServerMediaSession->streamName()) != 0) {
fOurServerMediaSession = NULL;
}
if (fOurServerMediaSession == NULL) {
// Set up this session's state.
// Look up the "ServerMediaSession" object for the specified stream:
if (streamName[0] != '\0' ||
fOurServer.lookupServerMediaSession("") != NULL) { // normal case
} else { // weird case: there was no track id in the URL
streamName = urlSuffix;
trackId = NULL;
}
fOurServerMediaSession = fOurServer.lookupServerMediaSession(streamName);
if (fOurServerMediaSession == NULL) {
handleCmd_notFound(cseq);
return;
}
fOurServerMediaSession->incrementReferenceCount();
// Set up our array of states for this session's subsessions (tracks):
reclaimStreamStates();
ServerMediaSubsessionIterator iter(*fOurServerMediaSession);
for (fNumStreamStates = 0; iter.next() != NULL; ++fNumStreamStates) {}
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; reset by SETUP 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:
handleCmd_notFound(cseq);
return;
}
} else {
// Weird case: there was no track id in the URL.
// This works only if we have only one subsession:
if (fNumStreamStates != 1) {
handleCmd_bad(cseq);
return;
}
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, fIsMulticast,
clientsDestinationAddressStr, clientsDestinationTTL,
clientRTPPortNum, clientRTCPPortNum,
rtpChannelId, rtcpChannelId);
if (streamingMode == RTP_TCP && rtpChannelId == 0xFF) {
// TCP streaming was requested, but with no "interleaving=" fields.
// (QuickTime Player sometimes does this.) Set the RTP and RTCP channel ids to
// proper values:
rtpChannelId = fTCPStreamIdCount; rtcpChannelId = fTCPStreamIdCount+1;
}
fTCPStreamIdCount += 2;
Port clientRTPPort(clientRTPPortNum);
Port clientRTCPPort(clientRTCPPortNum);
// Next, check whether a "Range:" 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;
fStreamAfterSETUP = parseRangeHeader(fullRequestStr, rangeStart, rangeEnd) ||
parsePlayNowHeader(fullRequestStr);
// Then, get server parameters from the 'subsession':
int tcpSocketNum = streamingMode == RTP_TCP ? fClientSocket : -1;
netAddressBits destinationAddress = 0;
u_int8_t destinationTTL = 5;
#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);
subsession->getStreamParameters(fOurSessionId, fClientAddr.sin_addr.s_addr,
clientRTPPort, clientRTCPPort,
tcpSocketNum, rtpChannelId, rtcpChannelId,
destinationAddress, destinationTTL, fIsMulticast,
serverRTPPort, serverRTCPPort,
fStreamStates[streamNum].streamToken);
struct in_addr destinationAddr; destinationAddr.s_addr = destinationAddress;
char* destAddrStr = strDup(our_inet_ntoa(destinationAddr));
struct sockaddr_in sourceAddr; SOCKLEN_T namelen = sizeof sourceAddr;
getsockname(fClientSocket, (struct sockaddr*)&sourceAddr, &namelen);
char* sourceAddrStr = strDup(our_inet_ntoa(sourceAddr.sin_addr));
if (fIsMulticast) {
switch (streamingMode) {
case RTP_UDP:
snprintf((char*)fResponseBuffer, sizeof 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",
cseq,
dateHeader(),
destAddrStr, sourceAddrStr, ntohs(serverRTPPort.num()), ntohs(serverRTCPPort.num()), destinationTTL,
fOurSessionId);
break;
case RTP_TCP:
// multicast streams can't be sent via TCP
handleCmd_unsupportedTransport(cseq);
break;
case RAW_UDP:
snprintf((char*)fResponseBuffer, sizeof 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",
cseq,
dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr, ntohs(serverRTPPort.num()), destinationTTL,
fOurSessionId);
break;
}
} else {
switch (streamingMode) {
case RTP_UDP: {
snprintf((char*)fResponseBuffer, sizeof 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",
cseq,
dateHeader(),
destAddrStr, sourceAddrStr, ntohs(clientRTPPort.num()), ntohs(clientRTCPPort.num()), ntohs(serverRTPPort.num()), ntohs(serverRTCPPort.num()),
fOurSessionId);
break;
}
case RTP_TCP: {
snprintf((char*)fResponseBuffer, sizeof 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",
cseq,
dateHeader(),
destAddrStr, sourceAddrStr, rtpChannelId, rtcpChannelId,
fOurSessionId);
break;
}
case RAW_UDP: {
snprintf((char*)fResponseBuffer, sizeof 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",
cseq,
dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr, ntohs(clientRTPPort.num()), ntohs(serverRTPPort.num()),
fOurSessionId);
break;
}
}
}
delete[] destAddrStr; delete[] sourceAddrStr; delete[] streamingModeString;
}