Boolean Groupsock::handleRead(unsigned char* buffer, unsigned bufferMaxSize,
unsigned& bytesRead,
struct sockaddr_in& fromAddress) {
// Read data from the socket, and relay it across any attached tunnels
//##### later make this code more general - independent of tunnels
bytesRead = 0;
int maxBytesToRead = bufferMaxSize - TunnelEncapsulationTrailerMaxSize;
int numBytes = readSocket(env(), socketNum(),
buffer, maxBytesToRead, fromAddress);
if (numBytes < 0) {
if (DebugLevel >= 0) { // this is a fatal error
env().setResultMsg("Groupsock read failed: ",
env().getResultMsg());
}
return False;
}
// If we're a SSM group, make sure the source address matches:
if (isSSM()
&& fromAddress.sin_addr.s_addr != sourceFilterAddress().s_addr) {
return True;
}
// We'll handle this data.
// Also write it (with the encapsulation trailer) to each member,
// unless the packet was originally sent by us to begin with.
bytesRead = numBytes;
int numMembers = 0;
if (!wasLoopedBackFromUs(env(), fromAddress)) {
statsIncoming.countPacket(numBytes);
statsGroupIncoming.countPacket(numBytes);
numMembers =
outputToAllMembersExcept(NULL, ttl(),
buffer, bytesRead,
fromAddress.sin_addr.s_addr);
if (numMembers > 0) {
statsRelayedIncoming.countPacket(numBytes);
statsGroupRelayedIncoming.countPacket(numBytes);
}
}
if (DebugLevel >= 3) {
env() << *this << ": read " << bytesRead << " bytes from ";
env() << our_inet_ntoa(fromAddress.sin_addr);
if (numMembers > 0) {
env() << "; relayed to " << numMembers << " members";
}
env() << "\n";
}
return True;
}
Groupsock::~Groupsock() {
if (isSSM()) {
if (!socketLeaveGroupSSM(env(), socketNum(), groupAddress().s_addr,
sourceFilterAddress().s_addr)) {
socketLeaveGroup(env(), socketNum(), groupAddress().s_addr);
}
} else {
socketLeaveGroup(env(), socketNum(), groupAddress().s_addr);
}
delete fDests;
if (DebugLevel >= 2) env() << *this << ": deleting\n";
}
void MediaSubsession::setDestinations(netAddressBits defaultDestAddress) {
// Get the destination address from the connection endpoint name
// (This will be 0 if it's not known, in which case we use the default)
netAddressBits destAddress = connectionEndpointAddress();
if (destAddress == 0) destAddress = defaultDestAddress;
struct in_addr destAddr; destAddr.s_addr = destAddress;
// The destination TTL remains unchanged:
int destTTL = ~0; // means: don't change
if (fRTPSocket != NULL) {
Port destPort(serverPortNum);
fRTPSocket->changeDestinationParameters(destAddr, destPort, destTTL);
}
if (fRTCPSocket != NULL && !isSSM()) {
// Note: For SSM sessions, the dest address for RTCP was already set.
Port destPort(serverPortNum+1);
fRTCPSocket->changeDestinationParameters(destAddr, destPort, destTTL);
}
}
int Groupsock::outputToAllMembersExcept(DirectedNetInterface* exceptInterface,
u_int8_t ttlToFwd,
unsigned char* data, unsigned size,
netAddressBits sourceAddr) {
// Don't forward TTL-0 packets
if (ttlToFwd == 0) return 0;
DirectedNetInterfaceSet::Iterator iter(members());
unsigned numMembers = 0;
DirectedNetInterface* interf;
while ((interf = iter.next()) != NULL) {
// Check whether we've asked to exclude this interface:
if (interf == exceptInterface)
continue;
// Check that the packet's source address makes it OK to
// be relayed across this interface:
UsageEnvironment& saveEnv = env();
// because the following call may delete "this"
if (!interf->SourceAddrOKForRelaying(saveEnv, sourceAddr)) {
if (strcmp(saveEnv.getResultMsg(), "") != 0) {
// Treat this as a fatal error
return -1;
} else {
continue;
}
}
if (numMembers == 0) {
// We know that we're going to forward to at least one
// member, so fill in the tunnel encapsulation trailer.
// (Note: Allow for it not being 4-byte-aligned.)
TunnelEncapsulationTrailer* trailerInPacket
= (TunnelEncapsulationTrailer*)&data[size];
TunnelEncapsulationTrailer* trailer;
Boolean misaligned = ((unsigned long)trailerInPacket & 3) != 0;
unsigned trailerOffset;
u_int8_t tunnelCmd;
if (isSSM()) {
// add an 'auxilliary address' before the trailer
trailerOffset = TunnelEncapsulationTrailerAuxSize;
tunnelCmd = TunnelDataAuxCmd;
} else {
trailerOffset = 0;
tunnelCmd = TunnelDataCmd;
}
unsigned trailerSize = TunnelEncapsulationTrailerSize + trailerOffset;
unsigned tmpTr[TunnelEncapsulationTrailerMaxSize];
if (misaligned) {
trailer = (TunnelEncapsulationTrailer*)&tmpTr;
} else {
trailer = trailerInPacket;
}
trailer += trailerOffset;
if (fDests != NULL) {
trailer->address() = fDests->fGroupEId.groupAddress().s_addr;
trailer->port() = fDests->fPort; // structure copy, outputs in network order
}
trailer->ttl() = ttlToFwd;
trailer->command() = tunnelCmd;
if (isSSM()) {
trailer->auxAddress() = sourceFilterAddress().s_addr;
}
if (misaligned) {
memmove(trailerInPacket, trailer-trailerOffset, trailerSize);
}
size += trailerSize;
}
interf->write(data, size);
++numMembers;
}
return numMembers;
}
Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
if (fReadSource != NULL) return True; // has already been initiated
do {
if (fCodecName == NULL) {
env().setResultMsg("Codec is unspecified");
break;
}
// Create RTP and RTCP 'Groupsocks' on which to receive incoming data.
// (Groupsocks will work even for unicast addresses)
struct in_addr tempAddr;
tempAddr.s_addr = connectionEndpointAddress();
// This could get changed later, as a result of a RTSP "SETUP"
if (fClientPortNum != 0) {
// The sockets' port numbers were specified for us. Use these:
fClientPortNum = fClientPortNum&~1; // even
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, fClientPortNum);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("Failed to create RTP socket");
break;
}
// Set our RTCP port to be the RTP port +1
portNumBits const rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
if (fRTCPSocket == NULL) {
char tmpBuf[100];
sprintf(tmpBuf, "Failed to create RTCP socket (port %d)", rtcpPortNum);
env().setResultMsg(tmpBuf);
break;
}
} else {
// Port numbers were not specified in advance, so we use ephemeral port numbers.
// Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).
// We need to make sure that we don't keep trying to use the same bad port numbers over and over again.
// so we store bad sockets in a table, and delete them all when we're done.
HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);
if (socketHashTable == NULL) break;
Boolean success = False;
while (1) {
// Create a new socket:
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, 0);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");
break;
}
// Get the client port number, and check whether it's even (for RTP):
Port clientPort(0);
if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
break;
}
fClientPortNum = ntohs(clientPort.num());
if ((fClientPortNum&1) != 0) { // it's odd
// Record this socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
socketHashTable->Add((char const*)key, fRTPSocket);
continue;
}
// Make sure we can use the next (i.e., odd) port number, for RTCP:
portNumBits rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
if (fRTCPSocket != NULL) {
// Success! Use these two sockets (and delete any others that we've created):
Groupsock* oldGS;
while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
delete oldGS;
}
delete socketHashTable;
success = True;
break;
} else {
// We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
// Record the first socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
socketHashTable->Add((char const*)key, fRTPSocket);
continue;
}
}
if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
}
// ASSERT: fRTPSocket != NULL && fRTCPSocket != NULL
if (isSSM()) {
// Special case for RTCP SSM: Send RTCP packets back to the source via unicast:
fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
}
//////////////////////////////////////////////////////////////////////////
// 裁剪掉不需要的Source.
// Check "fProtocolName"
if (strcmp(fProtocolName, "UDP") == 0) {
#ifndef CUT_MIN_SIZE
// A UDP-packetized stream (*not* a RTP stream)
fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);
fRTPSource = NULL; // Note!
if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
}
#endif
} else {
// Check "fCodecName" against the set of codecs that we support,
// and create our RTP source accordingly
// (Later make this code more efficient, as this set grows #####)
// (Also, add more fmts that can be implemented by SimpleRTPSource#####)
Boolean createSimpleRTPSource = False;
Boolean doNormalMBitRule = False; // used if "createSimpleRTPSource"
if (strcmp(fCodecName, "H264") == 0) {
fReadSource = fRTPSource
= H264VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
}
else if (strcmp(fCodecName, "MP4V-ES") == 0)
{ // MPEG-4 Elem Str vid
fReadSource = fRTPSource
= MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
}
#ifndef CUT_MIN_SIZE
else if (strcmp(fCodecName, "QCELP") == 0) { // QCELP audio
fReadSource =
QCELPAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat,
fRTPTimestampFrequency);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 0 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)
fReadSource =
AMRAudioRTPSource::createNew(env(), fRTPSocket, fRTPSource,
fRTPPayloadFormat, 1 /*isWideband*/,
fNumChannels, fOctetalign, fInterleaving,
fRobustsorting, fCRC);
// Note that fReadSource will differ from fRTPSource in this case
} else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audio
fReadSource = fRTPSource
= MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audio
fRTPSource
= MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency);
if (fRTPSource == NULL) break;
// Add a filter that deinterleaves the ADUs after depacketizing them:
MP3ADUdeinterleaver* deinterleaver
= MP3ADUdeinterleaver::createNew(env(), fRTPSource);
if (deinterleaver == NULL) break;
// Add another filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
} else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
// a non-standard variant of "MPA-ROBUST" used by RealNetworks
// (one 'ADU'ized MP3 frame per packet; no headers)
fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency,
"audio/MPA-ROBUST" /*hack*/);
if (fRTPSource == NULL) break;
// Add a filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
False /*no ADU header*/);
} else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audio
fReadSource = fRTPSource
= MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "AC3") == 0) { // AC3 audio
fReadSource = fRTPSource
= AC3AudioRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {
fReadSource = fRTPSource
= MPEG4GenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
fMediumName, fMode,
fSizelength, fIndexlength,
fIndexdeltalength);
} else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 video
fReadSource = fRTPSource
= MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport Stream
fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, "video/MP2T",
0, False);
fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);
// this sets "durationInMicroseconds" correctly, based on the PCR values
} else if (strcmp(fCodecName, "H261") == 0) { // H.261
fReadSource = fRTPSource
= H261VideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "H263-1998") == 0 ||
strcmp(fCodecName, "H263-2000") == 0) { // H.263+
fReadSource = fRTPSource
= H263plusVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency);
} else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEG
fReadSource = fRTPSource
= JPEGVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
videoWidth(),
videoHeight());
} else if (strcmp(fCodecName, "X-QT") == 0
|| strcmp(fCodecName, "X-QUICKTIME") == 0) {
// Generic QuickTime streams, as defined in
// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= QuickTimeGenericRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
fRTPTimestampFrequency,
mimeType);
delete[] mimeType;
#ifdef SUPPORT_REAL_RTSP
} else if (strcmp(fCodecName, "X-PN-REALAUDIO") == 0 ||
strcmp(fCodecName, "X-PN-MULTIRATE-REALAUDIO-LIVE") == 0 ||
strcmp(fCodecName, "X-PN-REALVIDEO") == 0 ||
strcmp(fCodecName, "X-PN-MULTIRATE-REALVIDEO-LIVE") == 0) {
// A RealNetworks 'RDT' stream (*not* a RTP stream)
fReadSource = RealRDTSource::createNew(env());
fRTPSource = NULL; // Note!
parentSession().isRealNetworksRDT = True;
#endif
}
#endif // CUT_MIN_SIZE
else if ( strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio
|| strcmp(fCodecName, "GSM") == 0 // GSM audio
|| strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio
|| strcmp(fCodecName, "L16") == 0 // 16-bit linear audio
|| strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream
|| strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream
|| strcmp(fCodecName, "L8") == 0 // 8-bit linear audio
|| strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps
|| strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps
|| strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps
|| strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps
|| strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio
) {
createSimpleRTPSource = True;
useSpecialRTPoffset = 0;
} else if (useSpecialRTPoffset >= 0) {
// We don't know this RTP payload format, but try to receive
// it using a 'SimpleRTPSource' with the specified header offset:
createSimpleRTPSource = True;
} else {
env().setResultMsg("RTP payload format unknown or not supported");
break;
}
if (createSimpleRTPSource) {
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
sprintf(mimeType, "%s/%s", mediumName(), codecName());
fReadSource = fRTPSource
= SimpleRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency, mimeType,
(unsigned)useSpecialRTPoffset,
doNormalMBitRule);
delete[] mimeType;
}
}
if (fReadSource == NULL) {
env().setResultMsg("Failed to create read source");
break;
}
// Finally, create our RTCP instance. (It starts running automatically)
if (fRTPSource != NULL) {
unsigned totSessionBandwidth = 500; // HACK - later get from SDP#####
fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
totSessionBandwidth,
(unsigned char const*)
fParent.CNAME(),
NULL /* we're a client */,
fRTPSource);
if (fRTCPInstance == NULL) {
env().setResultMsg("Failed to create RTCP instance");
break;
}
}
return True;
} while (0);
delete fRTPSocket; fRTPSocket = NULL;
delete fRTCPSocket; fRTCPSocket = NULL;
Medium::close(fRTCPInstance); fRTCPInstance = NULL;
Medium::close(fReadSource); fReadSource = fRTPSource = NULL;
fClientPortNum = 0;
return False;
}
Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
if (fReadSource != NULL) return True; // has already been initiated
do {
if (fCodecName == NULL) {
env().setResultMsg("Codec is unspecified");
break;
}
// Create RTP and RTCP 'Groupsocks' on which to receive incoming data.
// (Groupsocks will work even for unicast addresses)
struct in_addr tempAddr;
tempAddr.s_addr = connectionEndpointAddress();
// This could get changed later, as a result of a RTSP "SETUP"
if (fClientPortNum != 0) {
// The sockets' port numbers were specified for us. Use these:
Boolean const protocolIsRTP = strcmp(fProtocolName, "RTP") == 0;
if (protocolIsRTP) {
fClientPortNum = fClientPortNum&~1; // use an even-numbered port for RTP, and the next (odd-numbered) port for RTCP
}
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, fClientPortNum);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("Failed to create RTP socket");
break;
}
if (protocolIsRTP) {
// Set our RTCP port to be the RTP port +1
portNumBits const rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
}
} else {
// Port numbers were not specified in advance, so we use ephemeral port numbers.
// Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).
// We need to make sure that we don't keep trying to use the same bad port numbers over and over again.
// so we store bad sockets in a table, and delete them all when we're done.
HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);
if (socketHashTable == NULL) break;
Boolean success = False;
NoReuse dummy(env()); // ensures that our new ephemeral port number won't be one that's already in use
while (1) {
// Create a new socket:
if (isSSM()) {
fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, 0);
} else {
fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);
}
if (fRTPSocket == NULL) {
env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");
break;
}
// Get the client port number, and check whether it's even (for RTP):
Port clientPort(0);
if (!getSourcePort(env(), fRTPSocket->socketNum(), clientPort)) {
break;
}
fClientPortNum = ntohs(clientPort.num());
if ((fClientPortNum&1) != 0) { // it's odd
// Record this socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
delete existing; // in case it wasn't NULL
continue;
}
// Make sure we can use the next (i.e., odd) port number, for RTCP:
portNumBits rtcpPortNum = fClientPortNum|1;
if (isSSM()) {
fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr, rtcpPortNum);
} else {
fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);
}
if (fRTCPSocket != NULL && fRTCPSocket->socketNum() >= 0) {
// Success! Use these two sockets.
success = True;
break;
} else {
// We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).
delete fRTCPSocket;
// Record the first socket in our table, and keep trying:
unsigned key = (unsigned)fClientPortNum;
Groupsock* existing = (Groupsock*)socketHashTable->Add((char const*)key, fRTPSocket);
delete existing; // in case it wasn't NULL
continue;
}
}
// Clean up the socket hash table (and contents):
Groupsock* oldGS;
while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
delete oldGS;
}
delete socketHashTable;
if (!success) break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue
}
// Try to use a big receive buffer for RTP - at least 0.1 second of
// specified bandwidth and at least 50 KB
unsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
if (rtpBufSize < 50 * 1024)
rtpBufSize = 50 * 1024;
increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);
if (isSSM() && fRTCPSocket != NULL) {
// Special case for RTCP SSM: Send RTCP packets back to the source via unicast:
fRTCPSocket->changeDestinationParameters(fSourceFilterAddr,0,~0);
}
// Create "fRTPSource" and "fReadSource":
if (!createSourceObjects(useSpecialRTPoffset)) break;
if (fReadSource == NULL) {
env().setResultMsg("Failed to create read source");
break;
}
// Finally, create our RTCP instance. (It starts running automatically)
if (fRTPSource != NULL && fRTCPSocket != NULL) {
// If bandwidth is specified, use it and add 5% for RTCP overhead.
// Otherwise make a guess at 500 kbps.
unsigned totSessionBandwidth
= fBandwidth ? fBandwidth + fBandwidth / 20 : 500;
fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,
totSessionBandwidth,
(unsigned char const*)
fParent.CNAME(),
NULL /* we're a client */,
fRTPSource);
if (fRTCPInstance == NULL) {
env().setResultMsg("Failed to create RTCP instance");
break;
}
}
return True;
} while (0);
delete fRTPSocket; fRTPSocket = NULL;
delete fRTCPSocket; fRTCPSocket = NULL;
Medium::close(fRTCPInstance); fRTCPInstance = NULL;
Medium::close(fReadSource); fReadSource = fRTPSource = NULL;
fClientPortNum = 0;
return False;
}