/**
* Intercepts and handles outgoing RTX (RFC-4588) packets for an <tt>RtpChannel</tt>. Depending on
* whether the destination supports the RTX format (RFC-4588) either removes the RTX encapsulation
* (thus effectively retransmitting packets bit-by-bit) or updates the sequence number and SSRC
* fields taking into account the data sent to the particular <tt>RtpChannel</tt>.
*
* @author Boris Grozev
*/
public class RtxTransformer extends SinglePacketTransformerAdapter implements TransformEngine {
/**
* The <tt>Logger</tt> used by the <tt>RtxTransformer</tt> class and its instances to print debug
* information.
*/
private static final Logger logger = Logger.getLogger(RtxTransformer.class);
/** The <tt>RtpChannel</tt> for the transformer. */
private RtpChannel channel;
/** Maps an RTX SSRC to the last RTP sequence number sent with that SSRC. */
private final Map<Long, Integer> rtxSequenceNumbers = new HashMap<>();
/**
* Initializes a new <tt>RtxTransformer</tt> with a specific <tt>RtpChannel</tt>.
*
* @param channel the <tt>RtpChannel</tt> for the transformer.
*/
RtxTransformer(RtpChannel channel) {
this.channel = channel;
}
/** Implements {@link PacketTransformer#transform(RawPacket[])}. {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
byte rtxPt;
if (pkt != null
&& (rtxPt = channel.getRtxPayloadType()) != -1
&& pkt.getPayloadType() == rtxPt) {
pkt = handleRtxPacket(pkt);
}
return pkt;
}
/**
* Handles an RTX packet and returns it.
*
* @param pkt the packet to handle.
* @return the packet
*/
private RawPacket handleRtxPacket(RawPacket pkt) {
boolean destinationSupportsRtx = channel.getRtxPayloadType() != -1;
RawPacket mediaPacket = createMediaPacket(pkt);
if (mediaPacket != null) {
RawPacketCache cache = channel.getStream().getPacketCache();
if (cache != null) {
cache.cachePacket(mediaPacket);
}
}
if (destinationSupportsRtx) {
pkt.setSequenceNumber(
getNextRtxSequenceNumber(pkt.getSSRC() & 0xffffffffL, pkt.getSequenceNumber()));
} else {
// If the media packet was not reconstructed, drop the RTX packet
// (by returning null).
return mediaPacket;
}
return pkt;
}
/**
* Creates a {@code RawPacket} which represents the original packet encapsulated in {@code pkt}
* using the RTX format.
*
* @param pkt the packet from which to extract a media packet.
* @return the extracted media packet.
*/
private RawPacket createMediaPacket(RawPacket pkt) {
RawPacket mediaPacket = null;
long rtxSsrc = pkt.getSSRC() & 0xffffffffL;
// We need to know the SSRC paired with rtxSsrc *as seen by the
// receiver (i.e. this.channel)*. However, we only store SSRCs
// that endpoints *send* with.
// We therefore assume that SSRC re-writing has not introduced any
// new SSRCs and therefor the FID mappings known to the senders
// also apply to receivers.
RtpChannel sourceChannel = channel.getContent().findChannelByFidSsrc(rtxSsrc);
if (sourceChannel != null) {
long mediaSsrc = sourceChannel.getFidPairedSsrc(rtxSsrc);
if (mediaSsrc != -1) {
byte apt = sourceChannel.getRtxAssociatedPayloadType();
if (apt != -1) {
mediaPacket = new RawPacket(pkt.getBuffer().clone(), pkt.getOffset(), pkt.getLength());
// Remove the RTX header by moving the RTP header two bytes
// right.
byte[] buf = mediaPacket.getBuffer();
int off = mediaPacket.getOffset();
System.arraycopy(buf, off, buf, off + 2, mediaPacket.getHeaderLength());
mediaPacket.setOffset(off + 2);
mediaPacket.setLength(pkt.getLength() - 2);
mediaPacket.setSSRC((int) mediaSsrc);
mediaPacket.setSequenceNumber(pkt.getOriginalSequenceNumber());
mediaPacket.setPayloadType(apt);
}
}
}
return mediaPacket;
}
/** Implements {@link TransformEngine#getRTPTransformer()}. */
@Override
public PacketTransformer getRTPTransformer() {
return this;
}
/** Implements {@link TransformEngine#getRTCPTransformer()}. */
@Override
public PacketTransformer getRTCPTransformer() {
return null;
}
/**
* Returns the sequence number to use for a specific RTX packet, which is based on the packet's
* original sequence number.
*
* <p>Because we terminate the RTX format, and with simulcast we might translate RTX packets from
* multiple SSRCs into the same SSRC, we keep count of the RTX packets (and their sequence
* numbers) which we sent for each SSRC.
*
* @param ssrc the SSRC of the RTX stream for the packet.
* @param defaultSeq the default sequence number to use in case we don't (yet) have any
* information about <tt>ssrc</tt>.
* @return the sequence number which should be used for the next RTX packet sent using SSRC
* <tt>ssrc</tt>.
*/
private int getNextRtxSequenceNumber(long ssrc, int defaultSeq) {
Integer seq;
synchronized (rtxSequenceNumbers) {
seq = rtxSequenceNumbers.get(ssrc);
if (seq == null) seq = defaultSeq;
else seq++;
rtxSequenceNumbers.put(ssrc, seq);
}
return seq;
}
/**
* Returns the next RTP sequence number to use for the RTX stream for a particular SSRC.
*
* @param ssrc the SSRC.
* @return the next sequence number to use for SSRC <tt>ssrc</tt>.
*/
private int getNextRtxSequenceNumber(long ssrc) {
return getNextRtxSequenceNumber(ssrc, new Random().nextInt(1 << 16));
}
/**
* Tries to find an SSRC paired with {@code ssrc} in an FID group in one of the channels from
* {@link #channel}'s {@code Content}. Returns -1 on failure.
*
* @param ssrc the SSRC for which to find a paired SSRC.
* @return An SSRC paired with {@code ssrc} in an FID group, or -1.
*/
private long getPairedSsrc(long ssrc) {
RtpChannel sourceChannel = channel.getContent().findChannelByFidSsrc(ssrc);
if (sourceChannel != null) {
return sourceChannel.getFidPairedSsrc(ssrc);
}
return -1;
}
/**
* Retransmits a packet to {@link #channel}. If the destination supports the RTX format, the
* packet will be encapsulated in RTX, otherwise, the packet will be retransmitted as-is.
*
* @param pkt the packet to retransmit.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
public boolean retransmit(RawPacket pkt, TransformEngine after) {
boolean destinationSupportsRtx = channel.getRtxPayloadType() != -1;
boolean retransmitPlain;
if (destinationSupportsRtx) {
long rtxSsrc = getPairedSsrc(pkt.getSSRC());
if (rtxSsrc == -1) {
logger.warn("Cannot find SSRC for RTX, retransmitting plain.");
retransmitPlain = true;
} else {
retransmitPlain = !encapsulateInRtxAndTransmit(pkt, rtxSsrc);
}
} else {
retransmitPlain = true;
}
if (retransmitPlain) {
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
try {
mediaStream.injectPacket(pkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to retransmit a packet.");
return false;
}
}
}
return true;
}
/**
* Encapsulates {@code pkt} in the RTX format, using {@code rtxSsrc} as its SSRC, and transmits it
* to {@link #channel} by injecting it in the {@code MediaStream}.
*
* @param pkt the packet to transmit.
* @param rtxSsrc the SSRC for the RTX stream.
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
private boolean encapsulateInRtxAndTransmit(RawPacket pkt, long rtxSsrc) {
byte[] buf = pkt.getBuffer();
int len = pkt.getLength();
int off = pkt.getOffset();
byte[] newBuf = buf;
if (buf.length < len + 2) {
// FIXME The byte array newly allocated and assigned to newBuf must
// be made known to pkt eventually.
newBuf = new byte[len + 2];
}
int osn = pkt.getSequenceNumber();
int headerLength = pkt.getHeaderLength();
int payloadLength = len - headerLength;
System.arraycopy(buf, off, newBuf, 0, headerLength);
// FIXME If newBuf is actually buf, then we will override the first two
// bytes of the payload bellow.
newBuf[headerLength] = (byte) ((osn >> 8) & 0xff);
newBuf[headerLength + 1] = (byte) (osn & 0xff);
System.arraycopy(buf, off + headerLength, newBuf, headerLength + 2, payloadLength);
// FIXME We tried to extend the payload of pkt by two bytes above but
// we never told pkt that its length has increased by these two bytes.
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
pkt.setSSRC((int) rtxSsrc);
// Only call getNextRtxSequenceNumber() when we're sure we're going
// to transmit a packet, because it consumes a sequence number.
pkt.setSequenceNumber(getNextRtxSequenceNumber(rtxSsrc));
try {
mediaStream.injectPacket(pkt, /* data */ true, /* after */ null);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to transmit an RTX packet.");
return false;
}
}
return true;
}
}
/**
* The <tt>BasicRTCPTerminationStrategy</tt> "gateways" PLIs, FIRs, NACKs, etc, in the sense that it
* replaces the packet sender information in the PLIs, FIRs, NACKs, etc and it generates its own
* SRs/RRs/REMBs based on information that it collects and from information found in FMJ.
*
* @author George Politis
*/
public class BasicRTCPTerminationStrategy extends MediaStreamRTCPTerminationStrategy {
/**
* The <tt>Logger</tt> used by the <tt>BasicRTCPTerminationStrategy</tt> class and its instances
* to print debug information.
*/
private static final Logger logger = Logger.getLogger(BasicRTCPTerminationStrategy.class);
/** The maximum number of RTCP report blocks that an RR or an SR can contain. */
private static final int MAX_RTCP_REPORT_BLOCKS = 31;
/** The minimum number of RTCP report blocks that an RR or an SR can contain. */
private static final int MIN_RTCP_REPORT_BLOCKS = 0;
/**
* A reusable array that can be used to hold up to <tt>MAX_RTCP_REPORT_BLOCKS</tt>
* <tt>RTCPReportBlock</tt>s. It is assumed that a single thread is accessing this field at a
* given time.
*/
private final RTCPReportBlock[] MAX_RTCP_REPORT_BLOCKS_ARRAY =
new RTCPReportBlock[MAX_RTCP_REPORT_BLOCKS];
/** A reusable array that holds 0 <tt>RTCPReportBlock</tt>s. */
private static final RTCPReportBlock[] MIN_RTCP_REPORTS_BLOCKS_ARRAY =
new RTCPReportBlock[MIN_RTCP_REPORT_BLOCKS];
/**
* The RTP stats map that holds RTP statistics about all the streams that this
* <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
*/
private final RTPStatsMap rtpStatsMap = new RTPStatsMap();
/**
* The RTCP stats map that holds RTCP statistics about all the streams that this
* <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
*/
private final RemoteClockEstimator remoteClockEstimator = new RemoteClockEstimator();
/**
* The <tt>CNameRegistry</tt> holds the CNAMEs that this RTCP termination, seen as a
* TransformEngine, has seen.
*/
private final CNAMERegistry cnameRegistry = new CNAMERegistry();
/** The parser that parses <tt>RawPacket</tt>s to <tt>RTCPCompoundPacket</tt>s. */
private final RTCPPacketParserEx parser = new RTCPPacketParserEx();
/** The generator that generates <tt>RawPacket</tt>s from <tt>RTCPCompoundPacket</tt>s. */
private final RTCPGenerator generator = new RTCPGenerator();
/**
* The RTCP feedback gateway responsible for dropping all the stuff that we support in this RTCP
* termination strategy.
*/
private final FeedbackGateway feedbackGateway = new FeedbackGateway();
/** The garbage collector that cleans-up the state of this RTCP termination strategy. */
private final GarbageCollector garbageCollector = new GarbageCollector();
/** The RTP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
private final PacketTransformer rtpTransformer =
new SinglePacketTransformer() {
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
// Update our RTP stats map (packets/octet sent).
rtpStatsMap.apply(pkt);
return pkt;
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
// Let everything pass through.
return pkt;
}
};
/** The RTCP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
private final PacketTransformer rtcpTransformer =
new SinglePacketTransformer() {
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
if (pkt == null) {
return pkt;
}
RTCPCompoundPacket inPacket;
try {
inPacket =
(RTCPCompoundPacket)
parser.parse(pkt.getBuffer(), pkt.getOffset(), pkt.getLength());
} catch (BadFormatException e) {
logger.warn("Failed to terminate an RTCP packet. " + "Dropping packet.");
return null;
}
// Update our RTCP stats map (timestamps). This operation is
// read-only.
remoteClockEstimator.apply(inPacket);
cnameRegistry.update(inPacket);
// Remove SRs and RRs from the RTCP packet.
pkt = feedbackGateway.gateway(inPacket);
return pkt;
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
// Let everything pass through.
return pkt;
}
};
/** A counter that counts the number of times we've sent "full-blown" SDES. */
private int sdesCounter = 0;
/** {@inheritDoc} */
@Override
public PacketTransformer getRTPTransformer() {
return rtpTransformer;
}
/** {@inheritDoc} */
@Override
public PacketTransformer getRTCPTransformer() {
return rtcpTransformer;
}
/** {@inheritDoc} */
@Override
public RawPacket report() {
garbageCollector.cleanup();
// TODO Compound RTCP packets should not exceed the MTU of the network
// path.
//
// An individual RTP participant should send only one compound RTCP
// packet per report interval in order for the RTCP bandwidth per
// participant to be estimated correctly, except when the compound
// RTCP packet is split for partial encryption.
//
// If there are too many sources to fit all the necessary RR packets
// into one compound RTCP packet without exceeding the maximum
// transmission unit (MTU) of the network path, then only the subset
// that will fit into one MTU should be included in each interval. The
// subsets should be selected round-robin across multiple intervals so
// that all sources are reported.
//
// It is impossible to know in advance what the MTU of path will be.
// There are various algorithms for experimenting to find out, but many
// devices do not properly implement (or deliberately ignore) the
// necessary standards so it all comes down to trial and error. For that
// reason, we can just guess 1200 or 1500 bytes per message.
long time = System.currentTimeMillis();
Collection<RTCPPacket> packets = new ArrayList<RTCPPacket>();
// First, we build the RRs.
Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(time);
if (rrPackets != null && rrPackets.size() != 0) {
packets.addAll(rrPackets);
}
// Next, we build the SRs.
Collection<RTCPSRPacket> srPackets = makeRTCPSRPackets(time);
if (srPackets != null && srPackets.size() != 0) {
packets.addAll(srPackets);
}
// Bail out if we have nothing to report.
if (packets.size() == 0) {
return null;
}
// Next, we build the REMB.
RTCPREMBPacket rembPacket = makeRTCPREMBPacket();
if (rembPacket != null) {
packets.add(rembPacket);
}
// Finally, we add an SDES packet.
RTCPSDESPacket sdesPacket = makeSDESPacket();
if (sdesPacket != null) {
packets.add(sdesPacket);
}
// Prepare the <tt>RTCPCompoundPacket</tt> to return.
RTCPPacket rtcpPackets[] = packets.toArray(new RTCPPacket[packets.size()]);
RTCPCompoundPacket cp = new RTCPCompoundPacket(rtcpPackets);
// Build the <tt>RTCPCompoundPacket</tt> and return the
// <tt>RawPacket</tt> to inject to the <tt>MediaStream</tt>.
return generator.apply(cp);
}
/**
* (attempts) to get the local SSRC that will be used in the media sender SSRC field of the RTCP
* reports. TAG(cat4-local-ssrc-hurricane)
*
* @return
*/
private long getLocalSSRC() {
return getStream().getStreamRTPManager().getLocalSSRC();
}
/**
* Makes <tt>RTCPRRPacket</tt>s using information in FMJ.
*
* @param time
* @return A <tt>Collection</tt> of <tt>RTCPRRPacket</tt>s to inject to the <tt>MediaStream</tt>.
*/
private Collection<RTCPRRPacket> makeRTCPRRPackets(long time) {
RTCPReportBlock[] reportBlocks = makeRTCPReportBlocks(time);
if (reportBlocks == null || reportBlocks.length == 0) {
return null;
}
Collection<RTCPRRPacket> rrPackets = new ArrayList<RTCPRRPacket>();
// We use the stream's local source ID (SSRC) as the SSRC of packet
// sender.
long streamSSRC = getLocalSSRC();
// Since a maximum of 31 reception report blocks will fit in an SR
// or RR packet, additional RR packets SHOULD be stacked after the
// initial SR or RR packet as needed to contain the reception
// reports for all sources heard during the interval since the last
// report.
if (reportBlocks.length > MAX_RTCP_REPORT_BLOCKS) {
for (int offset = 0; offset < reportBlocks.length; offset += MAX_RTCP_REPORT_BLOCKS) {
RTCPReportBlock[] blocks =
(reportBlocks.length - offset < MAX_RTCP_REPORT_BLOCKS)
? new RTCPReportBlock[reportBlocks.length - offset]
: MAX_RTCP_REPORT_BLOCKS_ARRAY;
System.arraycopy(reportBlocks, offset, blocks, 0, blocks.length);
RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, blocks);
rrPackets.add(rr);
}
} else {
RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, reportBlocks);
rrPackets.add(rr);
}
return rrPackets;
}
/**
* Iterate through all the <tt>ReceiveStream</tt>s that this <tt>MediaStream</tt> has and make
* <tt>RTCPReportBlock</tt>s for all of them.
*
* @param time
* @return
*/
private RTCPReportBlock[] makeRTCPReportBlocks(long time) {
MediaStream stream = getStream();
// State validation.
if (stream == null) {
logger.warn("stream is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
StreamRTPManager streamRTPManager = stream.getStreamRTPManager();
if (streamRTPManager == null) {
logger.warn("streamRTPManager is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
Collection<ReceiveStream> receiveStreams = streamRTPManager.getReceiveStreams();
if (receiveStreams == null || receiveStreams.size() == 0) {
logger.info("There are no receive streams to build report " + "blocks for.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
SSRCCache cache = streamRTPManager.getSSRCCache();
if (cache == null) {
logger.info("cache is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
// Create the return object.
Collection<RTCPReportBlock> rtcpReportBlocks = new ArrayList<RTCPReportBlock>();
// Populate the return object.
for (ReceiveStream receiveStream : receiveStreams) {
// Dig into the guts of FMJ and get the stats for the current
// receiveStream.
SSRCInfo info = cache.cache.get((int) receiveStream.getSSRC());
if (!info.ours && info.sender) {
RTCPReportBlock rtcpReportBlock = info.makeReceiverReport(time);
rtcpReportBlocks.add(rtcpReportBlock);
}
}
return rtcpReportBlocks.toArray(new RTCPReportBlock[rtcpReportBlocks.size()]);
}
/**
* Makes an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which we
* receive.
*
* @return an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which
* we receive.
*/
private RTCPREMBPacket makeRTCPREMBPacket() {
// TODO we should only make REMBs if REMB support has been advertised.
// Destination
RemoteBitrateEstimator remoteBitrateEstimator =
((VideoMediaStream) getStream()).getRemoteBitrateEstimator();
Collection<Integer> ssrcs = remoteBitrateEstimator.getSsrcs();
// TODO(gp) intersect with SSRCs from signaled simulcast layers
// NOTE(gp) The Google Congestion Control algorithm (sender side)
// doesn't seem to care about the SSRCs in the dest field.
long[] dest = new long[ssrcs.size()];
int i = 0;
for (Integer ssrc : ssrcs) dest[i++] = ssrc & 0xFFFFFFFFL;
// Exp & mantissa
long bitrate = remoteBitrateEstimator.getLatestEstimate();
if (bitrate == -1) return null;
if (logger.isDebugEnabled()) logger.debug("Estimated bitrate: " + bitrate);
// Create and return the packet.
// We use the stream's local source ID (SSRC) as the SSRC of packet
// sender.
long streamSSRC = getLocalSSRC();
return new RTCPREMBPacket(streamSSRC, /* mediaSSRC */ 0L, bitrate, dest);
}
/**
* Makes <tt>RTCPSRPacket</tt>s for all the RTP streams that we're sending.
*
* @return a <tt>List</tt> of <tt>RTCPSRPacket</tt> for all the RTP streams that we're sending.
*/
private Collection<RTCPSRPacket> makeRTCPSRPackets(long time) {
Collection<RTCPSRPacket> srPackets = new ArrayList<RTCPSRPacket>();
for (RTPStatsEntry rtpStatsEntry : rtpStatsMap.values()) {
int ssrc = rtpStatsEntry.getSsrc();
RemoteClock estimate = remoteClockEstimator.estimate(ssrc, time);
if (estimate == null) {
// We're not going to go far without an estimate..
continue;
}
RTCPSRPacket srPacket = new RTCPSRPacket(ssrc, MIN_RTCP_REPORTS_BLOCKS_ARRAY);
// Set the NTP timestamp for this SR.
long estimatedRemoteTime = estimate.getRemoteTime();
long secs = estimatedRemoteTime / 1000L;
double fraction = (estimatedRemoteTime - secs * 1000L) / 1000D;
srPacket.ntptimestamplsw = (int) (fraction * 4294967296D);
srPacket.ntptimestampmsw = secs;
// Set the RTP timestamp.
srPacket.rtptimestamp = estimate.getRtpTimestamp();
// Fill-in packet and octet send count.
srPacket.packetcount = rtpStatsEntry.getPacketsSent();
srPacket.octetcount = rtpStatsEntry.getBytesSent();
srPackets.add(srPacket);
}
return srPackets;
}
/**
* Makes <tt>RTCPSDES</tt> packets for all the RTP streams that we're sending.
*
* @return a <tt>List</tt> of <tt>RTCPSDES</tt> packets for all the RTP streams that we're
* sending.
*/
private RTCPSDESPacket makeSDESPacket() {
Collection<RTCPSDES> sdesChunks = new ArrayList<RTCPSDES>();
// Create an SDES for our own SSRC.
RTCPSDES ownSDES = new RTCPSDES();
SSRCInfo ourinfo = getStream().getStreamRTPManager().getSSRCCache().ourssrc;
ownSDES.ssrc = (int) getLocalSSRC();
Collection<RTCPSDESItem> ownItems = new ArrayList<RTCPSDESItem>();
ownItems.add(new RTCPSDESItem(RTCPSDESItem.CNAME, ourinfo.sourceInfo.getCNAME()));
// Throttle the source description bandwidth. See RFC3550#6.3.9
// Allocation of Source Description Bandwidth.
if (sdesCounter % 3 == 0) {
if (ourinfo.name != null && ourinfo.name.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.NAME, ourinfo.name.getDescription()));
if (ourinfo.email != null && ourinfo.email.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.EMAIL, ourinfo.email.getDescription()));
if (ourinfo.phone != null && ourinfo.phone.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.PHONE, ourinfo.phone.getDescription()));
if (ourinfo.loc != null && ourinfo.loc.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.LOC, ourinfo.loc.getDescription()));
if (ourinfo.tool != null && ourinfo.tool.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.TOOL, ourinfo.tool.getDescription()));
if (ourinfo.note != null && ourinfo.note.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.NOTE, ourinfo.note.getDescription()));
}
sdesCounter++;
ownSDES.items = ownItems.toArray(new RTCPSDESItem[ownItems.size()]);
sdesChunks.add(ownSDES);
for (Map.Entry<Integer, byte[]> entry : cnameRegistry.entrySet()) {
RTCPSDES sdes = new RTCPSDES();
sdes.ssrc = entry.getKey();
sdes.items = new RTCPSDESItem[] {new RTCPSDESItem(RTCPSDESItem.CNAME, entry.getValue())};
}
RTCPSDES[] sps = sdesChunks.toArray(new RTCPSDES[sdesChunks.size()]);
RTCPSDESPacket sp = new RTCPSDESPacket(sps);
return sp;
}
/**
* The garbage collector runs at each reporting interval and cleans up the data structures of this
* RTCP termination strategy based on the SSRCs that the owner <tt>MediaStream</tt> is still
* sending.
*/
class GarbageCollector {
public void cleanup() {
// TODO We need to fix TAG(cat4-local-ssrc-hurricane) and
// TAG(cat4-remote-ssrc-hurricane) first. The idea is to remove
// from our data structures everything that is not listed in as
// a remote SSRC.
}
}
/**
* Removes receiver and sender feedback from RTCP packets. Typically this means dropping SRs, RR
* report blocks and REMBs. It needs to pass through PLIs, FIRs, NACKs, etc.
*/
class FeedbackGateway {
/**
* Removes receiver and sender feedback from RTCP packets.
*
* @param inPacket the <tt>RTCPCompoundPacket</tt> to filter.
* @return the filtered <tt>RawPacket</tt>.
*/
public RawPacket gateway(RTCPCompoundPacket inPacket) {
if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
logger.info("Ignoring empty RTCP packet.");
return null;
}
ArrayList<RTCPPacket> outPackets = new ArrayList<RTCPPacket>(inPacket.packets.length);
for (RTCPPacket p : inPacket.packets) {
switch (p.type) {
case RTCPPacket.RR:
case RTCPPacket.SR:
case RTCPPacket.SDES:
// We generate our own RR/SR/SDES packets. We only want
// to forward NACKs/PLIs/etc.
break;
case RTCPFBPacket.PSFB:
RTCPFBPacket psfb = (RTCPFBPacket) p;
switch (psfb.fmt) {
case RTCPREMBPacket.FMT:
// We generate its own REMB packets.
break;
default:
// We let through everything else, like NACK
// packets.
outPackets.add(psfb);
break;
}
break;
default:
// We let through everything else, like BYE and APP
// packets.
outPackets.add(p);
break;
}
}
if (outPackets.size() == 0) {
return null;
}
// We have feedback messages to send. Pack them in a compound
// RR and send them. TODO Use RFC5506 Reduced-Size RTCP, if the
// receiver supports it.
Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(System.currentTimeMillis());
if (rrPackets != null && rrPackets.size() != 0) {
outPackets.addAll(0, rrPackets);
} else {
logger.warn("We might be sending invalid RTCPs.");
}
RTCPPacket[] pkts = outPackets.toArray(new RTCPPacket[outPackets.size()]);
RTCPCompoundPacket outPacket = new RTCPCompoundPacket(pkts);
return generator.apply(outPacket);
}
}
/** Holds the NTP timestamp and the associated RTP timestamp for a given RTP stream. */
class RemoteClock {
/**
* Ctor.
*
* @param remoteTime
* @param rtpTimestamp
*/
public RemoteClock(long remoteTime, int rtpTimestamp) {
this.remoteTime = remoteTime;
this.rtpTimestamp = rtpTimestamp;
}
/**
* The last NTP timestamp that we received for {@link this.ssrc} expressed in millis. Should be
* treated a signed long.
*/
private final long remoteTime;
/**
* The RTP timestamp associated to {@link this.ntpTimestamp}. The RTP timestamp is an unsigned
* int.
*/
private final int rtpTimestamp;
/** @return */
public int getRtpTimestamp() {
return rtpTimestamp;
}
/** @return */
public long getRemoteTime() {
return remoteTime;
}
}
/** */
class ReceivedRemoteClock {
/** The SSRC. */
private final int ssrc;
/**
* The <tt>RemoteClock</tt> which was received at {@link this.receivedTime} for this RTP stream.
*/
private final RemoteClock remoteClock;
/**
* The local time in millis when we received the RTCP report with the RTP/NTP timestamps. It's a
* signed long.
*/
private final long receivedTime;
/**
* The clock rate for {@link.ssrc}. We need to have received at least two SRs in order to be
* able to calculate this. Unsigned short.
*/
private final int frequencyHz;
/**
* Ctor.
*
* @param ssrc
* @param remoteTime
* @param rtpTimestamp
* @param frequencyHz
*/
ReceivedRemoteClock(int ssrc, long remoteTime, int rtpTimestamp, int frequencyHz) {
this.ssrc = ssrc;
this.remoteClock = new RemoteClock(remoteTime, rtpTimestamp);
this.frequencyHz = frequencyHz;
this.receivedTime = System.currentTimeMillis();
}
/** @return */
public RemoteClock getRemoteClock() {
return remoteClock;
}
/** @return */
public long getReceivedTime() {
return receivedTime;
}
/** @return */
public int getSsrc() {
return ssrc;
}
/** @return */
public int getFrequencyHz() {
return frequencyHz;
}
}
/** The <tt>RTPStatsEntry</tt> class contains information about an outgoing SSRC. */
class RTPStatsEntry {
/** The SSRC of the stream that this instance tracks. */
private final int ssrc;
/**
* The total number of _payload_ octets (i.e., not including header or padding) transmitted in
* RTP data packets by the sender since starting transmission up until the time this SR packet
* was generated. This should be treated as an unsigned int.
*/
private final int bytesSent;
/**
* The total number of RTP data packets transmitted by the sender (including re-transmissions)
* since starting transmission up until the time this SR packet was generated. Re-transmissions
* using an RTX stream are tracked in the RTX SSRC. This should be treated as an unsigned int.
*/
private final int packetsSent;
/** @return */
public int getSsrc() {
return ssrc;
}
/** @return */
public int getBytesSent() {
return bytesSent;
}
/** @return */
public int getPacketsSent() {
return packetsSent;
}
/**
* Ctor.
*
* @param ssrc
* @param bytesSent
*/
RTPStatsEntry(int ssrc, int bytesSent, int packetsSent) {
this.ssrc = ssrc;
this.bytesSent = bytesSent;
this.packetsSent = packetsSent;
}
}
/**
* The <tt>RtpStatsMap</tt> gathers stats from RTP packets that the <tt>RTCPReportBuilder</tt>
* uses to build its reports.
*/
class RTPStatsMap extends ConcurrentHashMap<Integer, RTPStatsEntry> {
/**
* Updates this <tt>RTPStatsMap</tt> with information it gets from the <tt>RawPacket</tt>.
*
* @param pkt the <tt>RawPacket</tt> that is being transmitted.
*/
public void apply(RawPacket pkt) {
int ssrc = pkt.getSSRC();
if (this.containsKey(ssrc)) {
RTPStatsEntry oldRtpStatsEntry = this.get(ssrc);
// Replace whatever was in there before. A feature of the two's
// complement encoding (which is used by Java integers) is that
// the bitwise results for add, subtract, and multiply are the
// same if both inputs are interpreted as signed values or both
// inputs are interpreted as unsigned values. (Other encodings
// like one's complement and signed magnitude don't have this
// properly.)
this.put(
ssrc,
new RTPStatsEntry(
ssrc,
oldRtpStatsEntry.getBytesSent()
+ pkt.getLength()
- pkt.getHeaderLength()
- pkt.getPaddingSize(),
oldRtpStatsEntry.getPacketsSent() + 1));
} else {
// Add a new <tt>RTPStatsEntry</tt> in this map.
this.put(
ssrc,
new RTPStatsEntry(
ssrc, pkt.getLength() - pkt.getHeaderLength() - pkt.getPaddingSize(), 1));
}
}
}
/** A class that can be used to estimate the remote time at a given local time. */
class RemoteClockEstimator {
/** base: 7-Feb-2036 @ 06:28:16 UTC */
private static final long msb0baseTime = 2085978496000L;
/** base: 1-Jan-1900 @ 01:00:00 UTC */
private static final long msb1baseTime = -2208988800000L;
/** A map holding the received remote clocks. */
private Map<Integer, ReceivedRemoteClock> receivedClocks =
new ConcurrentHashMap<Integer, ReceivedRemoteClock>();
/**
* Inspect an <tt>RTCPCompoundPacket</tt> and build-up the state for future estimations.
*
* @param pkt
*/
public void apply(RTCPCompoundPacket pkt) {
if (pkt == null || pkt.packets == null || pkt.packets.length == 0) {
return;
}
for (RTCPPacket rtcpPacket : pkt.packets) {
switch (rtcpPacket.type) {
case RTCPPacket.SR:
RTCPSRPacket srPacket = (RTCPSRPacket) rtcpPacket;
// The media sender SSRC.
int ssrc = srPacket.ssrc;
// Convert 64-bit NTP timestamp to Java standard time.
// Note that java time (milliseconds) by definition has
// less precision then NTP time (picoseconds) so
// converting NTP timestamp to java time and back to NTP
// timestamp loses precision. For example, Tue, Dec 17
// 2002 09:07:24.810 EST is represented by a single
// Java-based time value of f22cd1fc8a, but its NTP
// equivalent are all values ranging from
// c1a9ae1c.cf5c28f5 to c1a9ae1c.cf9db22c.
// Use round-off on fractional part to preserve going to
// lower precision
long fraction = Math.round(1000D * srPacket.ntptimestamplsw / 0x100000000L);
/*
* If the most significant bit (MSB) on the seconds
* field is set we use a different time base. The
* following text is a quote from RFC-2030 (SNTP v4):
*
* If bit 0 is set, the UTC time is in the range
* 1968-2036 and UTC time is reckoned from 0h 0m 0s UTC
* on 1 January 1900. If bit 0 is not set, the time is
* in the range 2036-2104 and UTC time is reckoned from
* 6h 28m 16s UTC on 7 February 2036.
*/
long msb = srPacket.ntptimestampmsw & 0x80000000L;
long remoteTime =
(msb == 0)
// use base: 7-Feb-2036 @ 06:28:16 UTC
? msb0baseTime + (srPacket.ntptimestampmsw * 1000) + fraction
// use base: 1-Jan-1900 @ 01:00:00 UTC
: msb1baseTime + (srPacket.ntptimestampmsw * 1000) + fraction;
// Estimate the clock rate of the sender.
int frequencyHz = -1;
if (receivedClocks.containsKey(ssrc)) {
// Calculate the clock rate.
ReceivedRemoteClock oldStats = receivedClocks.get(ssrc);
RemoteClock oldRemoteClock = oldStats.getRemoteClock();
frequencyHz =
Math.round(
(float)
(((int) srPacket.rtptimestamp - oldRemoteClock.getRtpTimestamp())
& 0xffffffffl)
/ (remoteTime - oldRemoteClock.getRemoteTime()));
}
// Replace whatever was in there before.
receivedClocks.put(
ssrc,
new ReceivedRemoteClock(
ssrc, remoteTime, (int) srPacket.rtptimestamp, frequencyHz));
break;
case RTCPPacket.SDES:
break;
}
}
}
/**
* Estimate the <tt>RemoteClock</tt> of a given RTP stream (identified by its SSRC) at a given
* time.
*
* @param ssrc the SSRC of the RTP stream whose <tt>RemoteClock</tt> we want to estimate.
* @param time the local time that will be mapped to a remote time.
* @return An estimation of the <tt>RemoteClock</tt> at time "time".
*/
public RemoteClock estimate(int ssrc, long time) {
ReceivedRemoteClock receivedRemoteClock = receivedClocks.get(ssrc);
if (receivedRemoteClock == null || receivedRemoteClock.getFrequencyHz() == -1) {
// We can't continue if we don't have NTP and RTP timestamps
// and/or the original sender frequency, so move to the next
// one.
return null;
}
long delayMillis = time - receivedRemoteClock.getReceivedTime();
// Estimate the remote wall clock.
long remoteTime = receivedRemoteClock.getRemoteClock().getRemoteTime();
long estimatedRemoteTime = remoteTime + delayMillis;
// Drift the RTP timestamp.
int rtpTimestamp =
receivedRemoteClock.getRemoteClock().getRtpTimestamp()
+ ((int) delayMillis) * (receivedRemoteClock.getFrequencyHz() / 1000);
return new RemoteClock(estimatedRemoteTime, rtpTimestamp);
}
}
/** Keeps track of the CNAMEs of the RTP streams that we've seen. */
class CNAMERegistry extends ConcurrentHashMap<Integer, byte[]> {
/** @param inPacket */
public void update(RTCPCompoundPacket inPacket) {
// Update CNAMEs.
if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
return;
}
for (RTCPPacket p : inPacket.packets) {
switch (p.type) {
case RTCPPacket.SDES:
RTCPSDESPacket sdesPacket = (RTCPSDESPacket) p;
if (sdesPacket.sdes == null || sdesPacket.sdes.length == 0) {
continue;
}
for (RTCPSDES chunk : sdesPacket.sdes) {
if (chunk.items == null || chunk.items.length == 0) {
continue;
}
for (RTCPSDESItem sdesItm : chunk.items) {
if (sdesItm.type != RTCPSDESItem.CNAME) {
continue;
}
this.put(chunk.ssrc, sdesItm.data);
}
}
break;
}
}
}
}
}