bool ElcRTSPServer::setSource(const string& url, const string& iname)
{
LOG_INFO("ElcRTSPServer::setSource -url="<<url<<" -name="<<iname);
mUrl = url;
mName = iname;
Boolean reuseFirstSource = true;
if(!rtspServer)
{
LOG_ERROR("ElcRTSPServer::setSource failed! Rtspserver does not created propertly");
return false;
}
mediaSession = ServerMediaSession::createNew(*mEnv, mName.c_str(), mName.c_str(), "ELC");
if(!mediaSession)
return false;
// if multicast streaming
if(Config::isMulticast)
{
struct in_addr destinationAddress;
if(Config::streamingDestinationIp == "default" || Config::streamingDestinationIp =="")
destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*mEnv);
else
destinationAddress.s_addr = our_inet_addr(Config::streamingDestinationIp.c_str());
Port rtpPort(Config::rtpPortNum);
Port rtcpPort(Config::rtcpPortNum);
rtpGroupsock = new Groupsock(*mEnv, destinationAddress, rtpPort, 255/*Config::ttl*/);
rtpGroupsock->multicastSendOnly(); // we're a SSM source
rtcpGroupsock = new Groupsock(*mEnv, destinationAddress, rtcpPort, 255/*Config::ttl*/);
rtcpGroupsock->multicastSendOnly(); // we're a SSM source
videoSink = MPEG4ESVideoRTPSink::createNew(*mEnv, rtpGroupsock, 96);
const unsigned estimatedSessionBandwidth = 500; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
rtcp = RTCPInstance::createNew(*mEnv, rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
videoSink, NULL /* we're a server*/ ,
True /* we're a SSM source */);
mediaSession->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(mediaSession);
LOG_INFO("ElcRTSPServer::setSource OK");
}
else // unicast streaming
{
mediaSubSession = Mpeg4LiveServerMediaSubSession::createNew(*mEnv, reuseFirstSource, mUrl);
mediaSession->addSubsession(mediaSubSession);
rtspServer->addServerMediaSession(mediaSession);
LOG_INFO("ElcRTSPServer::setSource OK");
}
return true;
}
NetAddressList::NetAddressList(char const* hostname)
: fNumAddresses(0), fAddressArray(NULL) {
struct hostent* host;
// Check first whether "hostname" is an IP address string:
netAddressBits addr = our_inet_addr((char*)hostname);
if (addr != INADDR_NONE) { // yes it was an IP address string
//##### host = gethostbyaddr((char*)&addr, sizeof (netAddressBits), AF_INET);
host = NULL; // don't bother calling gethostbyaddr(); we only want 1 addr
if (host == NULL) {
// For some unknown reason, gethostbyaddr() failed, so just
// return a 1-element list with the address we were given:
fNumAddresses = 1;
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
fAddressArray[0] = new NetAddress((u_int8_t*)&addr,
sizeof (netAddressBits));
return;
}
} else { // Try resolving "hostname" as a real host name
#if defined(VXWORKS)
char hostentBuf[512];
host = (struct hostent*)resolvGetHostByName((char*)hostname,(char*)&hostentBuf,sizeof hostentBuf);
#else
host = our_gethostbyname((char*)hostname);
#endif
if (host == NULL) {
// It was a host name, and we couldn't resolve it. We're SOL.
return;
}
}
u_int8_t const** const hAddrPtr
= (u_int8_t const**)host->h_addr_list;
if (hAddrPtr != NULL) {
// First, count the number of addresses:
u_int8_t const** hAddrPtr1 = hAddrPtr;
while (*hAddrPtr1 != NULL) {
++fNumAddresses;
++hAddrPtr1;
}
// Next, set up the list:
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
for (unsigned i = 0; i < fNumAddresses; ++i) {
fAddressArray[i]
= new NetAddress(hAddrPtr[i], host->h_length);
}
}
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
// Create a 'groupsock' for the input multicast group,port:
char const* sessionAddressStr = "224.2.127.254";
struct in_addr sessionAddress;
sessionAddress.s_addr = our_inet_addr(sessionAddressStr);
const Port port(9875);
const unsigned char ttl = 0; // we're only reading from this mcast group
Groupsock inputGroupsock(*env, sessionAddress, port, ttl);
// Start reading and printing incoming packets
// (Because this is the only thing we do, we can just do this
// synchronously, in a loop, so we don't need to set up an asynchronous
// event handler like we do in most of the other test programs.)
unsigned packetSize;
struct sockaddr_in fromAddress;
while (inputGroupsock.handleRead(packet, maxPacketSize,
packetSize, fromAddress)) {
printf("\n[packet from %s (%d bytes)]\n",
our_inet_ntoa(fromAddress.sin_addr), packetSize);
// Ignore the first 8 bytes (SAP header).
if (packetSize < 8) {
*env << "Ignoring short packet from "
<< our_inet_ntoa(fromAddress.sin_addr) << "%s!\n";
continue;
}
// convert "application/sdp\0" -> "application/sdp\0x20"
// or all other nonprintable characters to blank, except new line
unsigned idx = 8;
while (idx < packetSize) {
if (packet[idx] < 0x20 && packet[idx] != '\n') packet[idx] = 0x20;
idx++;
}
packet[packetSize] = '\0'; // just in case
printf((char*)(packet+8));
}
return 0; // only to prevent compiler warning
}
FramedSource* MPEG2TransportUDPServerMediaSubsession
::createNewStreamSource(unsigned/* clientSessionId*/, unsigned& estBitrate) {
estBitrate = 5000; // kbps, estimate
if (fInputGroupsock == NULL) {
// Create a 'groupsock' object for receiving the input stream:
struct in_addr inputAddress;
inputAddress.s_addr = fInputAddressStr == NULL ? 0 : our_inet_addr(fInputAddressStr);
fInputGroupsock = new Groupsock(envir(), inputAddress, fInputPort, 255);
}
FramedSource* transportStreamSource;
if (fInputStreamIsRawUDP) {
transportStreamSource = BasicUDPSource::createNew(envir(), fInputGroupsock);
} else {
transportStreamSource = SimpleRTPSource::createNew(envir(), fInputGroupsock, 33, 90000, "video/MP2T", 0, False /*no 'M' bit*/);
}
return MPEG2TransportStreamFramer::createNew(envir(), transportStreamSource);
}
void CCreateRTPServer::createGroupsock()
{
m_prtcpGroupsockAudio = NULL;
m_prtpGroupsockAudio = NULL;
m_prtcpGroupsockVideo = NULL;
m_prtpGroupsockVideo = NULL;
struct in_addr destAddress;
//destAddress.s_addr = our_inet_addr(m_destAddressStr.c_str());
if(!m_rtpDestAddressStr.empty())
destAddress.s_addr = our_inet_addr(m_rtpDestAddressStr.c_str());
else
destAddress.s_addr = chooseRandomIPv4SSMAddress(*env);
if(m_pAudioSourceQueue != NULL)
{
const unsigned short rtpPortNumAudio = m_rtpPortNum;
const unsigned short rtcpPortNumAudio = rtpPortNumAudio + 1;
const Port rtpPortAudio(rtpPortNumAudio);
const Port rtcpPortAudio(rtcpPortNumAudio);
m_prtpGroupsockAudio = new Groupsock(*env, destAddress, rtpPortAudio, m_ttl);
m_prtcpGroupsockAudio = new Groupsock(*env, destAddress, rtcpPortAudio, m_ttl);
if(m_prtpGroupsockAudio && m_prtcpGroupsockAudio)
{
m_rtpPortNum += 2;
}
}
if(m_pVideoSourceQueue != NULL)
{
const unsigned short rtpPortNumVideo = m_rtpPortNum;
const unsigned short rtcpPortNumVideo = rtpPortNumVideo + 1;
const Port rtpPortVideo(rtpPortNumVideo);
const Port rtcpPortVideo(rtcpPortNumVideo);
m_prtpGroupsockVideo = new Groupsock(*env, destAddress, rtpPortVideo, m_ttl);
m_prtcpGroupsockVideo = new Groupsock(*env, destAddress, rtcpPortVideo, m_ttl);
}
}
char const* destinationAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= "239.255.42.42";
// Note: This is a multicast address. If you wish to stream using
// unicast instead, then replace this string with the unicast address
// of the (single) destination. (You may also need to make a similar
// change to the receiver program.)
#endif
const unsigned short rtpPortNum = 8888;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 7; // low, in case routers don't admin scope
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
#ifdef USE_SSM
rtpGroupsock.multicastSendOnly();
rtcpGroupsock.multicastSendOnly();
#endif
// Create a 'MPEG Video RTP' sink from the RTP 'groupsock':
videoSink = MPEG1or2VideoRTPSink::createNew(*env, &rtpGroupsock);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 4500; // in kbps; for RTCP b/w share
NetAddressList::NetAddressList(char const* hostname)
: fNumAddresses(0), fAddressArray(NULL) {
// First, check whether "hostname" is an IP address string:
netAddressBits addr = our_inet_addr((char*)hostname);
if (addr != INADDR_NONE) {
// Yes, it was an IP address string. Return a 1-element list with this address:
fNumAddresses = 1;
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
fAddressArray[0] = new NetAddress((u_int8_t*)&addr, sizeof (netAddressBits));
return;
}
// "hostname" is not an IP address string; try resolving it as a real host name instead:
#if defined(USE_GETHOSTBYNAME) || defined(VXWORKS)
struct hostent* host;
#if defined(VXWORKS)
char hostentBuf[512];
host = (struct hostent*)resolvGetHostByName((char*)hostname, (char*)&hostentBuf, sizeof hostentBuf);
#else
host = gethostbyname((char*)hostname);
#endif
if (host == NULL || host->h_length != 4 || host->h_addr_list == NULL) return; // no luck
u_int8_t const** const hAddrPtr = (u_int8_t const**)host->h_addr_list;
// First, count the number of addresses:
u_int8_t const** hAddrPtr1 = hAddrPtr;
while (*hAddrPtr1 != NULL) {
++fNumAddresses;
++hAddrPtr1;
}
// Next, set up the list:
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
for (unsigned i = 0; i < fNumAddresses; ++i) {
fAddressArray[i] = new NetAddress(hAddrPtr[i], host->h_length);
}
#else
// Use "getaddrinfo()" (rather than the older, deprecated "gethostbyname()"):
struct addrinfo addrinfoHints;
memset(&addrinfoHints, 0, sizeof addrinfoHints);
addrinfoHints.ai_family = AF_INET; // For now, we're interested in IPv4 addresses only
struct addrinfo* addrinfoResultPtr = NULL;
int result = getaddrinfo(hostname, NULL, &addrinfoHints, &addrinfoResultPtr);
if (result != 0 || addrinfoResultPtr == NULL) return; // no luck
// First, count the number of addresses:
const struct addrinfo* p = addrinfoResultPtr;
while (p != NULL) {
if (p->ai_addrlen < 4) continue; // sanity check: skip over addresses that are too small
++fNumAddresses;
p = p->ai_next;
}
// Next, set up the list:
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
unsigned i = 0;
p = addrinfoResultPtr;
while (p != NULL) {
if (p->ai_addrlen < 4) continue;
fAddressArray[i++] = new NetAddress((u_int8_t const*)&(((struct sockaddr_in*)p->ai_addr)->sin_addr.s_addr), 4);
p = p->ai_next;
}
// Finally, free the data that we had allocated by calling "getaddrinfo()":
freeaddrinfo(addrinfoResultPtr);
#endif
}
JNIEXPORT void JNICALL Java_com_parizene_streamer_Streamer_loop(JNIEnv *env,
jobject obj, jstring addr) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
uEnv = BasicUsageEnvironment::createNew(*scheduler);
// Create 'groupsocks' for RTP and RTCP:
struct in_addr destinationAddress;
const char *_addr = env->GetStringUTFChars(addr, NULL);
destinationAddress.s_addr = our_inet_addr(_addr); /*chooseRandomIPv4SSMAddress(*uEnv);*/
env->ReleaseStringUTFChars(addr, _addr);
// Note: This is a multicast address. If you wish instead to stream
// using unicast, then you should use the "testOnDemandRTSPServer"
// test program - not this test program - as a model.
const unsigned short rtpPortNum = 18888;
const unsigned short rtcpPortNum = rtpPortNum + 1;
const unsigned char ttl = 255;
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*uEnv, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*uEnv, destinationAddress, rtcpPort, ttl);
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
OutPacketBuffer::maxSize = 100000;
videoSink = H264VideoRTPSink::createNew(*uEnv, &rtpGroupsock, 96);
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 500; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen + 1];
gethostname((char*) CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
RTCPInstance* rtcp = RTCPInstance::createNew(*uEnv, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME, videoSink,
NULL /* we're a server */, True /* we're a SSM source */);
// Note: This starts RTCP running automatically
RTSPServer* rtspServer = RTSPServer::createNew(*uEnv, 8554);
if (rtspServer == NULL) {
LOGE("Failed to create RTSP server: %s", uEnv->getResultMsg());
exit(1);
}
ServerMediaSession* sms = ServerMediaSession::createNew(*uEnv, "streamer",
inputFilename, "Session streamed by \"testH264VideoStreamer\"",
True /*SSM*/);
sms->addSubsession(
PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
LOGI("Play this stream using the URL \"%s\"", url);
delete[] url;
// Start the streaming:
LOGI("Beginning streaming...\n");
play();
uEnv->taskScheduler().doEventLoop(); // does not return
}
netAddressBits ourIPAddress(UsageEnvironment& env) {
static netAddressBits ourAddress = 0;
int sock = -1;
struct in_addr testAddr;
if (ourAddress == 0) {
// We need to find our source address
struct sockaddr_in fromAddr;
fromAddr.sin_addr.s_addr = 0;
// Get our address by sending a (0-TTL) multicast packet,
// receiving it, and looking at the source address used.
// (This is kinda bogus, but it provides the best guarantee
// that other nodes will think our address is the same as we do.)
do {
loopbackWorks = 0; // until we learn otherwise
testAddr.s_addr = our_inet_addr("228.67.43.91"); // arbitrary
Port testPort(15947); // ditto
sock = setupDatagramSocket(env, testPort);
if (sock < 0) break;
if (!socketJoinGroup(env, sock, testAddr.s_addr)) break;
unsigned char testString[] = "hostIdTest";
unsigned testStringLength = sizeof testString;
if (!writeSocket(env, sock, testAddr, testPort, 0,
testString, testStringLength)) break;
unsigned char readBuffer[20];
struct timeval timeout;
timeout.tv_sec = 5;
timeout.tv_usec = 0;
int bytesRead = readSocket(env, sock,
readBuffer, sizeof readBuffer,
fromAddr, &timeout);
if (bytesRead == 0 // timeout occurred
|| bytesRead != (int)testStringLength
|| strncmp((char*)readBuffer, (char*)testString,
testStringLength) != 0) {
break;
}
loopbackWorks = 1;
} while (0);
if (!loopbackWorks) do {
// We couldn't find our address using multicast loopback
// so try instead to look it up directly.
char hostname[100];
hostname[0] = '\0';
#ifndef CRIS
gethostname(hostname, sizeof hostname);
#endif
if (hostname[0] == '\0') {
env.setResultErrMsg("initial gethostname() failed");
break;
}
#if defined(VXWORKS)
#include <hostLib.h>
if (ERROR == (ourAddress = hostGetByName( hostname ))) break;
#else
struct hostent* hstent
= (struct hostent*)gethostbyname(hostname);
if (hstent == NULL || hstent->h_length != 4) {
env.setResultErrMsg("initial gethostbyname() failed");
break;
}
// Take the first address that's not bad
// (This code, like many others, won't handle IPv6)
netAddressBits addr = 0;
for (unsigned i = 0; ; ++i) {
char* addrPtr = hstent->h_addr_list[i];
if (addrPtr == NULL) break;
netAddressBits a = *(netAddressBits*)addrPtr;
if (!badAddress(a)) {
addr = a;
break;
}
}
if (addr != 0) {
fromAddr.sin_addr.s_addr = addr;
} else {
env.setResultMsg("no address");
break;
}
} while (0);
// Make sure we have a good address:
netAddressBits from = fromAddr.sin_addr.s_addr;
if (badAddress(from)) {
char tmp[100];
sprintf(tmp,
"This computer has an invalid IP address: 0x%x",
(netAddressBits)(ntohl(from)));
env.setResultMsg(tmp);
from = 0;
}
ourAddress = from;
#endif
if (sock >= 0) {
socketLeaveGroup(env, sock, testAddr.s_addr);
closeSocket(sock);
}
// Use our newly-discovered IP address, and the current time,
// to initialize the random number generator's seed:
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
unsigned seed = ourAddress^timeNow.tv_sec^timeNow.tv_usec;
our_srandom(seed);
}
return ourAddress;
}
bool IPTVFeederRTP::Open(const QString &url)
{
LOG(VB_RECORD, LOG_INFO, LOC + QString("Open(%1) -- begin").arg(url));
QMutexLocker locker(&_lock);
if (_source)
{
LOG(VB_RECORD, LOG_INFO, LOC + "Open() -- end 1");
return true;
}
QUrl parse(url);
if (!parse.isValid() || parse.host().isEmpty() || (-1 == parse.port()))
{
LOG(VB_RECORD, LOG_INFO, LOC + "Open() -- end 2");
return false;
}
struct in_addr addr;
QByteArray host = parse.host().toLatin1();
addr.s_addr = our_inet_addr(host.constData());
// Begin by setting up our usage environment:
if (!InitEnv())
return false;
ReceivingSocketAddr = our_inet_addr(parse.host().toLatin1());
Groupsock *socket = new Groupsock(*_live_env, addr, parse.port(), 0);
if (!socket)
{
LOG(VB_GENERAL, LOG_ERR, LOC + "Failed to create Live RTP Socket.");
FreeEnv();
return false;
}
_source = SimpleRTPSource::createNew(*_live_env, socket, 33, 90000,
"video/MP2T", 0, False);
if (!_source)
{
LOG(VB_GENERAL, LOG_ERR, LOC + "Failed to create Live RTP Source.");
if (socket)
delete socket;
FreeEnv();
return false;
}
_sink = IPTVMediaSink::CreateNew(*_live_env, TSPacket::kSize * 128*1024);
if (!_sink)
{
LOG(VB_GENERAL, LOG_ERR, QString("IPTV # Failed to create sink: %1")
.arg(_live_env->getResultMsg()));
Medium::close(_source);
_source = NULL;
if (socket)
delete socket;
FreeEnv();
return false;
}
_sink->startPlaying(*_source, NULL, NULL);
vector<TSDataListener*>::iterator it = _listeners.begin();
for (; it != _listeners.end(); ++it)
_sink->AddListener(*it);
LOG(VB_RECORD, LOG_INFO, LOC + "Open() -- end");
return true;
}
int CMPIPTV_RTSP::OpenConnection(void)
{
this->logger.Log(LOGGER_INFO, METHOD_START_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
int result = STATUS_OK;
this->rtspClient = RTSPClient::createNew(*this->rtspEnvironment);
result |= (this->rtspClient == NULL);
if (result == STATUS_OK)
{
// RTSPClient works with char, not with TCHAR
char *tempRtspUrl = ConvertToMultiByte(this->rtspUrl);
result |= (tempRtspUrl == NULL);
if (result == STATUS_OK)
{
char* optionsResult = this->rtspClient->sendOptionsCmd(tempRtspUrl, NULL, NULL, NULL, this->receiveDataTimeout / 2000);
result |= (optionsResult == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending OPTIONS command"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("OPTIONS result"));
this->LogFullRtspMessage(LOGGER_VERBOSE, message, optionsResult);
FREE_MEM(message);
char *describeResult = this->rtspClient->describeURL(tempRtspUrl, NULL, FALSE, this->receiveDataTimeout / 2000);
result |= (describeResult == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while sending DESCRIBE command"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("DESCRIBE result"));
this->LogFullRtspMessage(LOGGER_VERBOSE, message, describeResult);
FREE_MEM(message);
this->rtspSession = MediaSession::createNew(*this->rtspEnvironment, describeResult);
result |= (this->rtspSession == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while creating new session"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
result |= (!this->rtspSession->hasSubsessions());
if (result != STATUS_OK)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("session doesn't have subsessions"));
}
else
{
// Then, setup the "RTPSource"s for the session:
MediaSubsessionIterator iter(*this->rtspSession);
MediaSubsession *subsession = NULL;
while ((result == STATUS_OK) && ((subsession = iter.next()) != NULL))
{
char *tempSubSessionName = (char *)subsession->mediumName();
char *tempSubSessionCodecName = (char *)subsession->codecName();
#ifdef _MBCS
TCHAR *subSessionName = ConvertToMultiByteA(tempSubSessionName);
TCHAR *subSessionCodecName = ConvertToMultiByteA(tempSubSessionCodecName);
#else
TCHAR *subSessionName = ConvertToUnicodeA(tempSubSessionName);
TCHAR *subSessionCodecName = ConvertToUnicodeA(tempSubSessionCodecName);
#endif
if (!subsession->initiate())
{
result = STATUS_ERROR;
TCHAR *message = FormatString(_T("%s: %s: unable to create receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: created receiver for subsession '%s', codec '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
// set session ID, doesn't matter what
subsession->sessionId = tempSubSessionName;
if (subsession->rtpSource() != NULL)
{
// because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
// set the RTP source's OS socket buffer size as appropriate
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned int currentBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (this->defaultBufferSize > currentBufferSize)
{
setReceiveBufferTo(*this->rtspEnvironment, socketNum, this->defaultBufferSize);
unsigned setBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (setBufferSize == this->defaultBufferSize)
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: set buffer size for subsession '%s' successful, previous size: %i, requested size: %i, current size: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
else
{
result = STATUS_ERROR;
this->logger.Log(LOGGER_ERROR, _T("%s: %s: set buffer size for subsession '%s' failed, previous size: %i, requested size: %i, current size: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
}
if (_tcsncicmp(subSessionName, _T("audio"), 5) == 0)
{
// audio
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: audio subsession '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
result |= (!rtspClient->setupMediaSubsession(*subsession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
this->logger.Log(LOGGER_WARNING, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("subsession audio codec not supported"));
}
}
else if (_tcsncicmp(subSessionName, _T("video"), 5) == 0)
{
// video
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: video subsession '%s'"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
result |= (!rtspClient->setupMediaSubsession(*subsession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while setup subsession"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
if (_tcsncicmp(subSessionCodecName, _T("MP2T"), 4) == 0)
{
// MPEG2 Transport Stream
// set new RTSP source
this->rtspSource = subsession->rtpSource();
if (subsession->rtcpInstance() != NULL)
{
this->logger.Log(LOGGER_VERBOSE, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("set subsession 'Bye' handler"));
subsession->rtcpInstance()->setByeHandler(SubsessionByeHandler, this);
}
}
else if (_tcsncicmp(subSessionCodecName, _T("H264"), 4) == 0)
{
// H264 codec, HD TV
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("H264 not supported"));
result = STATUS_ERROR;
}
else
{
// SD TV
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("other subsession video codec than MP2T not supported"));
result = STATUS_ERROR;
}
}
}
else
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: unknown subsession '%s', ignored"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
}
}
else
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: subsession '%s' doesn't have RTP source"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName);
}
}
// free subsession name and codec name
FREE_MEM(subSessionName);
FREE_MEM(subSessionCodecName);
}
// we should have some RTSP source
result |= (this->rtspSource == NULL);
if (result == STATUS_OK)
{
result |= (!this->rtspClient->playMediaSession(*this->rtspSession));
if (result != STATUS_OK)
{
// error occured
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error while playing session"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
// create UDP socket and start playing
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr("127.0.0.1");
unsigned int port = this->rtspUdpPortRangeStart;
do
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: UDP port %u"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
// special construction force not reuse same UDP port
{
NoReuse noReuse;
this->rtspUdpGroupsock = new Groupsock(*this->rtspEnvironment, destinationAddress, port, 1);
}
if (this->rtspUdpGroupsock->socketNum() == (-1))
{
this->logger.Log(LOGGER_ERROR, _T("%s: %s: UDP port %u occupied, trying another port"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
port++;
delete this->rtspUdpGroupsock;
this->rtspUdpGroupsock = NULL;
}
}
while ((this->rtspUdpGroupsock == NULL) && (port <= this->rtspUdpPortRangeEnd));
result |= (this->rtspUdpGroupsock == NULL);
if (result != STATUS_OK)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("cannot create UDP sink, no free port"));
}
else
{
this->rtspUdpSink = BasicUDPSink::createNew(*this->rtspEnvironment, this->rtspUdpGroupsock, this->rtspUdpSinkMaxPayloadSize);
result |= (this->rtspUdpSink == NULL);
if (result != STATUS_OK)
{
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("cannot create UDP sink"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
else
{
if (this->rtspUdpSink->startPlaying(*this->rtspSource, NULL, NULL))
{
this->logger.Log(LOGGER_INFO, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("playing started"));
// now create UDP connection
TCHAR *url = FormatString(_T("udp://@127.0.0.1:%u"), port);
result |= (url == NULL);
if (result == STATUS_OK)
{
// parse UDP url
// ParseURL calls ClearSession and IsConnected must return FALSE
// in another case will be current RTSP connection closed
result = this->CMPIPTV_UDP::ParseUrl(url, NULL);
if (result == STATUS_OK)
{
// connect to UDP url
result = this->CMPIPTV_UDP::OpenConnection();
}
}
FREE_MEM(url);
}
else
{
result = STATUS_ERROR;
TCHAR *message = FormatString(METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("error occured while starting playing"));
this->LogRtspMessage(LOGGER_ERROR, message);
FREE_MEM(message);
}
}
}
}
}
}
}
}
}
if (optionsResult != NULL)
{
delete[] optionsResult;
optionsResult = NULL;
}
}
FREE_MEM(tempRtspUrl);
}
if (result == STATUS_OK)
{
// start winsock worker thread
this->rtspSchedulerThreadHandle = CreateThread(
NULL, // default security attributes
0, // use default stack size
&CMPIPTV_RTSP::RtspSchedulerWorker, // thread function name
this, // argument to thread function
0, // use default creation flags
&this->rtspSchedulerThreadId); // returns the thread identifier
if (this->rtspSchedulerThreadHandle == NULL)
{
// thread not created
result = STATUS_ERROR;
this->logger.Log(LOGGER_ERROR, _T("%s: %s: cannot create RTSP scheduler thread, error: %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, GetLastError());
}
}
if (result != STATUS_OK)
{
// if failed opening connection, than close connection
this->CloseConnection();
}
this->logger.Log(LOGGER_INFO, (result == STATUS_OK) ? METHOD_END_FORMAT : METHOD_END_FAIL_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
return (result == STATUS_OK) ? STATUS_OK : STATUS_ERROR;
}
netAddressBits ourIPAddress(UsageEnvironment& env) {
static netAddressBits ourAddress = 0;
int sock = -1;
struct in_addr testAddr;
if (ReceivingInterfaceAddr != INADDR_ANY) {
// Hack: If we were told to receive on a specific interface address, then
// define this to be our ip address:
ourAddress = ReceivingInterfaceAddr;
}
if (ourAddress == 0) {
// We need to find our source address
struct sockaddr_in fromAddr;
fromAddr.sin_addr.s_addr = 0;
// Get our address by sending a (0-TTL) multicast packet,
// receiving it, and looking at the source address used.
// (This is kinda bogus, but it provides the best guarantee
// that other nodes will think our address is the same as we do.)
do {
loopbackWorks = 0; // until we learn otherwise
testAddr.s_addr = our_inet_addr("228.67.43.91"); // arbitrary
Port testPort(15947); // ditto
sock = setupDatagramSocket(env, testPort);
if (sock < 0) break;
if (!socketJoinGroup(env, sock, testAddr.s_addr)) break;
unsigned char testString[] = "hostIdTest";
unsigned testStringLength = sizeof testString;
if (!writeSocket(env, sock, testAddr, testPort.num(), 0,
testString, testStringLength)) break;
// Block until the socket is readable (with a 5-second timeout):
fd_set rd_set;
FD_ZERO(&rd_set);
FD_SET((unsigned)sock, &rd_set);
const unsigned numFds = sock+1;
struct timeval timeout;
timeout.tv_sec = 5;
timeout.tv_usec = 0;
int result = select(numFds, &rd_set, NULL, NULL, &timeout);
if (result <= 0) break;
unsigned char readBuffer[20];
int bytesRead = readSocket(env, sock,
readBuffer, sizeof readBuffer,
fromAddr);
if (bytesRead != (int)testStringLength
|| strncmp((char*)readBuffer, (char*)testString, testStringLength) != 0) {
break;
}
// We use this packet's source address, if it's good:
loopbackWorks = !badAddressForUs(fromAddr.sin_addr.s_addr);
} while (0);
if (sock >= 0) {
socketLeaveGroup(env, sock, testAddr.s_addr);
closeSocket(sock);
}
if (!loopbackWorks) do {
// We couldn't find our address using multicast loopback,
// so try instead to look it up directly - by first getting our host name, and then resolving this host name
char hostname[100];
hostname[0] = '\0';
int result = gethostname(hostname, sizeof hostname);
if (result != 0 || hostname[0] == '\0') {
env.setResultErrMsg("initial gethostname() failed");
break;
}
// Try to resolve "hostname" to an IP address:
NetAddressList addresses(hostname);
NetAddressList::Iterator iter(addresses);
NetAddress const* address;
// Take the first address that's not bad:
netAddressBits addr = 0;
while ((address = iter.nextAddress()) != NULL) {
netAddressBits a = *(netAddressBits*)(address->data());
if (!badAddressForUs(a)) {
addr = a;
break;
}
}
// Assign the address that we found to "fromAddr" (as if the 'loopback' method had worked), to simplify the code below:
fromAddr.sin_addr.s_addr = addr;
} while (0);
// Make sure we have a good address:
netAddressBits from = fromAddr.sin_addr.s_addr;
if (badAddressForUs(from)) {
char tmp[100];
sprintf(tmp, "This computer has an invalid IP address: %s", AddressString(from).val());
env.setResultMsg(tmp);
from = 0;
}
ourAddress = from;
// Use our newly-discovered IP address, and the current time,
// to initialize the random number generator's seed:
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
unsigned seed = ourAddress^timeNow.tv_sec^timeNow.tv_usec;
our_srandom(seed);
}
return ourAddress;
}
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;
}
int CMPIPTV_RTSP::OpenConnection(void)
{
this->logger.Log(LOGGER_INFO, METHOD_START_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
this->isRtspSessionSetup = false;
// LIVE555 works with char, not with TCHAR
char *tempRtspUrl = ConvertToMultiByte(this->rtspUrl);
if (tempRtspUrl == NULL)
{
return STATUS_ERROR;
}
// start LIVE555 worker thread
this->rtspSchedulerThreadHandle = CreateThread(
NULL, // default security attributes
0, // use default stack size
&CMPIPTV_RTSP::RtspSchedulerWorker, // thread function name
this, // argument to thread function
0, // use default creation flags
&this->rtspSchedulerThreadId); // returns the thread identifier
if (this->rtspSchedulerThreadHandle == NULL)
{
this->logger.Log(LOGGER_ERROR, _T("%s: %s: failed to create RTSP scheduler thread, error = %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, GetLastError());
return STATUS_ERROR;
}
this->rtspClient = MPRTSPClient::createNew(this, *this->rtspEnvironment, tempRtspUrl);
FREE_MEM(tempRtspUrl);
if (
this->rtspClient == NULL ||
SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("OPTIONS")) != STATUS_OK ||
SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("DESCRIBE")) != STATUS_OK
)
{
CloseConnection();
return STATUS_ERROR;
}
this->rtspSession = MediaSession::createNew(*this->rtspEnvironment, this->rtspResponseResultString);
if (this->rtspSession == NULL || !this->rtspSession->hasSubsessions())
{
this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, this->rtspSession == NULL ? _T("failed to create session") : _T("session doesn't have sub-sessions"));
CloseConnection();
return STATUS_ERROR;
}
// Setup the RTP source for the session. Only one sub-session expected/supported.
MediaSubsessionIterator iter(*this->rtspSession);
MediaSubsession *subsession = NULL;
FramedSource *rtspSource = NULL;
while ((subsession = iter.next()) != NULL)
{
#ifdef _MBCS
TCHAR *subSessionName = ConvertToMultiByteA(subsession->mediumName());
TCHAR *subSessionCodecName = ConvertToMultiByteA(subsession->codecName());
#else
TCHAR *subSessionName = ConvertToUnicodeA(subsession->mediumName());
TCHAR *subSessionCodecName = ConvertToUnicodeA(subsession->codecName());
#endif
if (_tcsncicmp(subSessionName, _T("video"), 5) != 0 || _tcsncicmp(subSessionCodecName, _T("MP2T"), 4) != 0)
{
TCHAR *message = FormatString(_T("sub-session medium or codec not supported, medium = %s, codec = %s"), subSessionName, subSessionCodecName);
this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, message);
FREE_MEM(message);
FREE_MEM(subSessionName);
FREE_MEM(subSessionCodecName);
continue;
}
// If a client port is configured, find a free pair of ports in the range.
// The first port is used for RTP; the second port is used for RTCP. Once
// we find one free port, we assume the next one is also free.
if (this->rtspRtpClientPortRangeStart > 0)
{
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr("127.0.0.1");
unsigned int port = this->rtspRtpClientPortRangeStart;
Groupsock *groupsock = NULL;
do
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: RTP client port %u"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
// special construction force not reuse same UDP port
{
NoReuse noReuse(*this->rtspEnvironment);
groupsock = new Groupsock(*this->rtspEnvironment, destinationAddress, port, 1);
}
if (groupsock == NULL || groupsock->socketNum() == -1)
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: RTP client port %u occupied, trying next even port"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
port += 2;
if (groupsock != NULL)
{
delete groupsock;
groupsock = NULL;
}
}
}
while ((groupsock == NULL) && (port <= this->rtspRtpClientPortRangeEnd));
// Did we find a free port? If not, we fall back to a random port chosen
// by LIVE555.
if (groupsock != NULL)
{
delete groupsock;
groupsock = NULL;
subsession->setClientPortNum(port);
}
}
if (!subsession->initiate() || subsession->rtpSource() == NULL)
{
TCHAR *message = FormatString(_T("failed to create receiver for sub-session, medium = %s, codec = %s"), subSessionName, subSessionCodecName);
this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, message);
FREE_MEM(message);
FREE_MEM(subSessionName);
FREE_MEM(subSessionCodecName);
continue;
}
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: created receiver for sub-session, medium = %s, codec = %s"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, subSessionCodecName);
FREE_MEM(subSessionName);
FREE_MEM(subSessionCodecName);
// set session ID, doesn't matter what
subsession->setSessionId(subsession->mediumName());
// because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// for reordering misordered incoming packets:
subsession->rtpSource()->setPacketReorderingThresholdTime(1000000); // 1 second
// set the RTP source's OS socket buffer size as appropriate
int socketNum = subsession->rtpSource()->RTPgs()->socketNum();
unsigned int currentBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (this->defaultBufferSize > currentBufferSize)
{
setReceiveBufferTo(*this->rtspEnvironment, socketNum, this->defaultBufferSize);
unsigned setBufferSize = getReceiveBufferSize(*this->rtspEnvironment, socketNum);
if (setBufferSize == this->defaultBufferSize)
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: set buffer size for sub-session, previous size = %i, requested size = %i, current size = %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
else
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: failed to set buffer size for sub-session, previous size = %i, requested size = %i, current size = %i"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, subSessionName, currentBufferSize, this->defaultBufferSize, setBufferSize);
}
}
if (SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("SETUP"), subsession) != STATUS_OK)
{
CloseConnection();
return STATUS_ERROR;
}
rtspSource = subsession->rtpSource();
break;
}
// If we don't have an RTSP source then we can't continue.
if (rtspSource == NULL)
{
CloseConnection();
return STATUS_ERROR;
}
this->isRtspSessionSetup = true;
if (SendRtspCommand(METHOD_OPEN_CONNECTION_NAME, _T("PLAY")) != STATUS_OK)
{
CloseConnection();
return STATUS_ERROR;
}
// create UDP socket and start playing
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr("127.0.0.1");
unsigned int port = this->rtspUdpPortRangeStart;
do
{
this->logger.Log(LOGGER_VERBOSE, _T("%s: %s: UDP port %u"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
// special construction force not reuse same UDP port
{
NoReuse noReuse(*this->rtspEnvironment);
this->rtspUdpGroupsock = new Groupsock(*this->rtspEnvironment, destinationAddress, port, 1);
}
if (this->rtspUdpGroupsock == NULL || this->rtspUdpGroupsock->socketNum() == -1)
{
this->logger.Log(LOGGER_WARNING, _T("%s: %s: UDP port %u occupied, trying another port"), PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, port);
port++;
if (this->rtspUdpGroupsock != NULL)
{
delete this->rtspUdpGroupsock;
this->rtspUdpGroupsock = NULL;
}
}
}
while ((this->rtspUdpGroupsock == NULL) && (port <= this->rtspUdpPortRangeEnd));
if (this->rtspUdpGroupsock == NULL)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("failed to create UDP socket, no free port"));
CloseConnection();
return STATUS_ERROR;
}
this->rtspUdpSink = BasicUDPSink::createNew(*this->rtspEnvironment, this->rtspUdpGroupsock, this->rtspUdpSinkMaxPayloadSize);
if (this->rtspUdpSink == NULL)
{
this->logger.Log(LOGGER_ERROR, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("failed to create UDP sink"));
CloseConnection();
return STATUS_ERROR;
}
if (!this->rtspUdpSink->startPlaying(*rtspSource, NULL, NULL))
{
this->LogRtspMessage(LOGGER_ERROR, METHOD_OPEN_CONNECTION_NAME, _T("failed to start UDP sink"));
CloseConnection();
return STATUS_ERROR;
}
this->logger.Log(LOGGER_INFO, METHOD_MESSAGE_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME, _T("streaming started"));
// create a UDP connection to the local stream
TCHAR *url = FormatString(_T("udp://@127.0.0.1:%u"), port);
if (
url == NULL ||
this->CMPIPTV_UDP::ParseUrl(url, NULL) != STATUS_OK ||
this->CMPIPTV_UDP::OpenConnection() != STATUS_OK
)
{
FREE_MEM(url);
this->logger.Log(LOGGER_INFO, METHOD_END_FAIL_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
CloseConnection();
return STATUS_ERROR;
}
FREE_MEM(url);
this->logger.Log(LOGGER_INFO, METHOD_END_FORMAT, PROTOCOL_IMPLEMENTATION_NAME, METHOD_OPEN_CONNECTION_NAME);
return STATUS_OK;
}
void startReplicaFileSink(StreamReplicator* replicator, char const* outputFileName); // forward
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
// Create a 'groupsock' for the input multicast group,port:
char const* inputAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= "239.255.42.42";
#endif
struct in_addr inputAddress;
inputAddress.s_addr = our_inet_addr(inputAddressStr);
Port const inputPort(8888);
unsigned char const inputTTL = 0; // we're only reading from this mcast group
#ifdef USE_SSM
char* sourceAddressStr = "aaa.bbb.ccc.ddd";
// replace this with the real source address
struct in_addr sourceFilterAddress;
sourceFilterAddress.s_addr = our_inet_addr(sourceAddressStr);
Groupsock inputGroupsock(*env, inputAddress, sourceFilterAddress, inputPort);
#else
Groupsock inputGroupsock(*env, inputAddress, inputPort, inputTTL);
#endif
