예제 #1
0
  /** {@inheritDoc} */
  public synchronized Result createConference(String creator, String mucRoomName) {
    Conference conference = conferenceMap.get(mucRoomName);
    if (conference == null) {
      // Create new conference
      try {
        ApiResult result = api.createNewConference(creator, mucRoomName);

        if (result.getError() == null) {
          conference = result.getConference();
          conferenceMap.put(mucRoomName, conference);
        } else if (result.getStatusCode() == 409 && result.getError().getConflictId() != null) {
          Number conflictId = result.getError().getConflictId();

          // Conference already exists(check if we have it locally)
          conference = findConferenceForId(conflictId);

          logger.info("Conference '" + mucRoomName + "' already " + "allocated, id: " + conflictId);

          // do GET conflict conference
          if (conference == null) {
            ApiResult getResult = api.getConference(conflictId);
            if (getResult.getConference() != null) {
              conference = getResult.getConference();
              // Fill full room name as it is not transferred
              // over REST API
              conference.setMucRoomName(mucRoomName);

              conferenceMap.put(mucRoomName, conference);
            } else {
              logger.error("API error: " + result);
              return new Result(RESULT_INTERNAL_ERROR, result.getError().getMessage());
            }
          }
        } else {
          // Other error
          logger.error("API error: " + result);
          return new Result(RESULT_INTERNAL_ERROR, result.getError().getMessage());
        }
      } catch (FaultTolerantRESTRequest.RetryExhaustedException e) {
        logger.error(e, e);
        return new Result(RESULT_INTERNAL_ERROR, e.getMessage());
      } catch (UnsupportedEncodingException e) {
        logger.error(e, e);
        return new Result(RESULT_INTERNAL_ERROR, e.getMessage());
      }
    }

    // Verify owner == creator
    if (creator.equals(conference.getOwner())) {
      return new Result(RESULT_OK);
    } else {
      logger.error(
          "Room " + mucRoomName + ", conflict : " + creator + " != " + conference.getOwner());

      return new Result(RESULT_CONFLICT);
    }
  }
예제 #2
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  /**
   * 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;
  }
예제 #3
0
  /**
   * 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.
   * @return the sequence number which should be used for the next RTX packet sent using SSRC
   *     <tt>ssrc</tt>.
   */
  private int getNextRtxSequenceNumber(long ssrc) {
    Integer seq;
    synchronized (rtxSequenceNumbers) {
      seq = rtxSequenceNumbers.get(ssrc);
      if (seq == null) seq = new Random().nextInt(0xffff);
      else seq++;

      rtxSequenceNumbers.put(ssrc, seq);
    }

    return seq;
  }
예제 #4
0
 /**
  * Returns the <tt>SSRCDesc</tt> instance mapped to the SSRC <tt>ssrc</tt>. If no instance is
  * mapped to <tt>ssrc</tt>, create one and inserts it in the map. Always returns non-null.
  *
  * @param ssrc the ssrc to get the <tt>SSRCDesc</tt> for.
  * @return the <tt>SSRCDesc</tt> instance mapped to the SSRC <tt>ssrc</tt>.
  */
 private SSRCDesc getSSRCDesc(long ssrc) {
   SSRCDesc ssrcDesc = ssrcs.get(ssrc);
   if (ssrcDesc == null) {
     synchronized (ssrcs) {
       ssrcDesc = ssrcs.get(ssrc);
       if (ssrcDesc == null) {
         ssrcDesc = new SSRCDesc();
         ssrcs.put(ssrc, ssrcDesc);
       }
     }
   }
   return ssrcDesc;
 }
예제 #5
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  /**
   * Returns the <tt>Endpoint</tt> with id <tt>endpointId</tt>. Creates an <tt>Endpoint</tt> if
   * necessary. Always returns non-null.
   *
   * @param endpointId the string identifying the endpoint.
   * @return the <tt>Endpoint</tt> with id <tt>endpointId</tt>. Creates an <tt>Endpoint</tt> if
   *     necessary.
   */
  private Endpoint getEndpoint(String endpointId) {
    Endpoint endpoint = endpoints.get(endpointId);
    if (endpoint == null) {
      synchronized (endpoints) {
        endpoint = endpoints.get(endpointId);
        if (endpoint == null) {
          endpoint = new Endpoint();
          endpoints.put(endpointId, endpoint);
        }
      }
    }

    return endpoint;
  }
예제 #6
0
  /**
   * Makes sure that conference is allocated for given <tt>room</tt>.
   *
   * @param room name of the MUC room of Jitsi Meet conference.
   * @param properties configuration properties, see {@link JitsiMeetConfig} for the list of valid
   *     properties.
   * @throws Exception if any error occurs.
   */
  private void createConference(String room, Map<String, String> properties) throws Exception {
    JitsiMeetConfig config = new JitsiMeetConfig(properties);

    JitsiMeetConference conference =
        new JitsiMeetConference(room, focusUserName, protocolProviderHandler, this, config);

    conferences.put(room, conference);

    StringBuilder options = new StringBuilder();
    for (Map.Entry<String, String> option : properties.entrySet()) {
      options.append("\n    ").append(option.getKey()).append(": ").append(option.getValue());
    }

    logger.info(
        "Created new focus for "
            + room
            + "@"
            + focusUserDomain
            + " conferences count: "
            + conferences.size()
            + " options:"
            + options.toString());

    // Send focus created event
    EventAdmin eventAdmin = FocusBundleActivator.getEventAdmin();
    if (eventAdmin != null) {
      eventAdmin.sendEvent(EventFactory.focusCreated(conference.getId(), conference.getRoomName()));
    }

    try {
      conference.start();
    } catch (Exception e) {
      logger.info("Exception while trying to start the conference", e);

      // stop() method is called by the conference automatically in order
      // to not release the lock on JitsiMeetConference instance and avoid
      // a deadlock. It may happen when this thread is about to call
      // conference.stop() and another thread has entered the method
      // before us. That other thread will try to call
      // FocusManager.conferenceEnded, but we're still holding the lock
      // on FocusManager instance.

      // conference.stop();

      throw e;
    }
  }
예제 #7
0
  /**
   * Makes sure that conference is allocated for given <tt>room</tt>.
   *
   * @param room name of the MUC room of Jitsi Meet conference.
   * @param properties configuration properties, see {@link JitsiMeetConfig} for the list of valid
   *     properties.
   * @throws Exception if any error occurs.
   */
  private void createConference(String room, Map<String, String> properties) throws Exception {
    JitsiMeetConfig config = new JitsiMeetConfig(properties);

    JitsiMeetConference conference =
        new JitsiMeetConference(room, focusUserName, protocolProviderHandler, this, config);

    conferences.put(room, conference);

    StringBuilder options = new StringBuilder();
    for (Map.Entry<String, String> option : properties.entrySet()) {
      options.append("\n    ").append(option.getKey()).append(": ").append(option.getValue());
    }

    logger.info(
        "Created new focus for "
            + room
            + "@"
            + focusUserDomain
            + " conferences count: "
            + conferences.size()
            + " options:"
            + options.toString());

    // Send focus created event
    EventAdmin eventAdmin = FocusBundleActivator.getEventAdmin();
    if (eventAdmin != null) {
      eventAdmin.sendEvent(EventFactory.focusCreated(conference.getId(), conference.getRoomName()));
    }

    try {
      conference.start();
    } catch (Exception e) {
      logger.info("Exception while trying to start the conference", e);

      conference.stop();

      throw e;
    }
  }
예제 #8
0
  /**
   * Opens new WebRTC data channel using specified parameters.
   *
   * @param type channel type as defined in control protocol description. Use 0 for "reliable".
   * @param prio channel priority. The higher the number, the lower the priority.
   * @param reliab Reliability Parameter<br>
   *     This field is ignored if a reliable channel is used. If a partial reliable channel with
   *     limited number of retransmissions is used, this field specifies the number of
   *     retransmissions. If a partial reliable channel with limited lifetime is used, this field
   *     specifies the maximum lifetime in milliseconds. The following table summarizes this:<br>
   *     </br>
   *     <p>+------------------------------------------------+------------------+ | Channel Type |
   *     Reliability | | | Parameter |
   *     +------------------------------------------------+------------------+ |
   *     DATA_CHANNEL_RELIABLE | Ignored | | DATA_CHANNEL_RELIABLE_UNORDERED | Ignored | |
   *     DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT | Number of RTX | |
   *     DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED | Number of RTX | |
   *     DATA_CHANNEL_PARTIAL_RELIABLE_TIMED | Lifetime in ms | |
   *     DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED | Lifetime in ms |
   *     +------------------------------------------------+------------------+
   * @param sid SCTP stream id that will be used by new channel (it must not be already used).
   * @param label text label for the channel.
   * @return new instance of <tt>WebRtcDataStream</tt> that represents opened WebRTC data channel.
   * @throws IOException if IO error occurs.
   */
  public synchronized WebRtcDataStream openChannel(
      int type, int prio, long reliab, int sid, String label) throws IOException {
    if (channels.containsKey(sid)) {
      throw new IOException("Channel on sid: " + sid + " already exists");
    }

    // Label Length & Label
    byte[] labelBytes;
    int labelByteLength;

    if (label == null) {
      labelBytes = null;
      labelByteLength = 0;
    } else {
      labelBytes = label.getBytes("UTF-8");
      labelByteLength = labelBytes.length;
      if (labelByteLength > 0xFFFF) labelByteLength = 0xFFFF;
    }

    // Protocol Length & Protocol
    String protocol = WEBRTC_DATA_CHANNEL_PROTOCOL;
    byte[] protocolBytes;
    int protocolByteLength;

    if (protocol == null) {
      protocolBytes = null;
      protocolByteLength = 0;
    } else {
      protocolBytes = protocol.getBytes("UTF-8");
      protocolByteLength = protocolBytes.length;
      if (protocolByteLength > 0xFFFF) protocolByteLength = 0xFFFF;
    }

    ByteBuffer packet = ByteBuffer.allocate(12 + labelByteLength + protocolByteLength);

    // Message open new channel on current sid
    // Message Type
    packet.put((byte) MSG_OPEN_CHANNEL);
    // Channel Type
    packet.put((byte) type);
    // Priority
    packet.putShort((short) prio);
    // Reliability Parameter
    packet.putInt((int) reliab);
    // Label Length
    packet.putShort((short) labelByteLength);
    // Protocol Length
    packet.putShort((short) protocolByteLength);
    // Label
    if (labelByteLength != 0) {
      packet.put(labelBytes, 0, labelByteLength);
    }
    // Protocol
    if (protocolByteLength != 0) {
      packet.put(protocolBytes, 0, protocolByteLength);
    }

    int sentCount = sctpSocket.send(packet.array(), true, sid, WEB_RTC_PPID_CTRL);

    if (sentCount != packet.capacity()) {
      throw new IOException("Failed to open new chanel on sid: " + sid);
    }

    WebRtcDataStream channel = new WebRtcDataStream(sctpSocket, sid, label, false);

    channels.put(sid, channel);

    return channel;
  }
예제 #9
0
  /**
   * Handles control packet.
   *
   * @param data raw packet data that arrived on control PPID.
   * @param sid SCTP stream id on which the data has arrived.
   */
  private synchronized void onCtrlPacket(byte[] data, int sid) throws IOException {
    ByteBuffer buffer = ByteBuffer.wrap(data);
    int messageType = /* 1 byte unsigned integer */ 0xFF & buffer.get();

    if (messageType == MSG_CHANNEL_ACK) {
      if (logger.isDebugEnabled()) {
        logger.debug(getEndpoint().getID() + " ACK received SID: " + sid);
      }
      // Open channel ACK
      WebRtcDataStream channel = channels.get(sid);
      if (channel != null) {
        // Ack check prevents from firing multiple notifications
        // if we get more than one ACKs (by mistake/bug).
        if (!channel.isAcknowledged()) {
          channel.ackReceived();
          notifyChannelOpened(channel);
        } else {
          logger.warn("Redundant ACK received for SID: " + sid);
        }
      } else {
        logger.error("No channel exists on sid: " + sid);
      }
    } else if (messageType == MSG_OPEN_CHANNEL) {
      int channelType = /* 1 byte unsigned integer */ 0xFF & buffer.get();
      int priority = /* 2 bytes unsigned integer */ 0xFFFF & buffer.getShort();
      long reliability = /* 4 bytes unsigned integer */ 0xFFFFFFFFL & buffer.getInt();
      int labelLength = /* 2 bytes unsigned integer */ 0xFFFF & buffer.getShort();
      int protocolLength = /* 2 bytes unsigned integer */ 0xFFFF & buffer.getShort();
      String label;
      String protocol;

      if (labelLength == 0) {
        label = "";
      } else {
        byte[] labelBytes = new byte[labelLength];

        buffer.get(labelBytes);
        label = new String(labelBytes, "UTF-8");
      }
      if (protocolLength == 0) {
        protocol = "";
      } else {
        byte[] protocolBytes = new byte[protocolLength];

        buffer.get(protocolBytes);
        protocol = new String(protocolBytes, "UTF-8");
      }

      if (logger.isDebugEnabled()) {
        logger.debug(
            "!!! "
                + getEndpoint().getID()
                + " data channel open request on SID: "
                + sid
                + " type: "
                + channelType
                + " prio: "
                + priority
                + " reliab: "
                + reliability
                + " label: "
                + label
                + " proto: "
                + protocol);
      }

      if (channels.containsKey(sid)) {
        logger.error("Channel on sid: " + sid + " already exists");
      }

      WebRtcDataStream newChannel = new WebRtcDataStream(sctpSocket, sid, label, true);
      channels.put(sid, newChannel);

      sendOpenChannelAck(sid);

      notifyChannelOpened(newChannel);
    } else {
      logger.error("Unexpected ctrl msg type: " + messageType);
    }
  }
  /**
   * Maybe send a data channel command to the associated <tt>Endpoint</tt> to make it start
   * streaming its hq stream, if it's being watched by some receiver.
   */
  public void maybeSendStartHighQualityStreamCommand() {
    if (nativeSimulcast || !hasLayers()) {
      // In native simulcast the client adjusts its layers autonomously so
      // we don't need (nor we can) to control it with data channel
      // messages.
      return;
    }

    Endpoint newEndpoint = getSimulcastEngine().getVideoChannel().getEndpoint();
    SimulcastLayer[] newSimulcastLayers = getSimulcastLayers();

    SctpConnection sctpConnection;
    if (newSimulcastLayers == null
        || newSimulcastLayers.length <= 1
        /* newEndpoint != null is implied */
        || (sctpConnection = newEndpoint.getSctpConnection()) == null
        || !sctpConnection.isReady()
        || sctpConnection.isExpired()) {
      return;
    }

    // we have a new endpoint and it has an SCTP connection that is
    // ready and not expired. if somebody else is watching the new
    // endpoint, start its hq stream.

    boolean startHighQualityStream = false;

    for (Endpoint e :
        getSimulcastEngine().getVideoChannel().getContent().getConference().getEndpoints()) {
      // TODO(gp) need some synchronization here. What if the
      // selected endpoint changes while we're in the loop?

      if (e == newEndpoint) continue;

      Endpoint eSelectedEndpoint = e.getEffectivelySelectedEndpoint();

      if (newEndpoint == eSelectedEndpoint) {
        // somebody is watching the new endpoint or somebody has not
        // yet signaled its selected endpoint to the bridge, start
        // the hq stream.

        if (logger.isDebugEnabled()) {
          Map<String, Object> map = new HashMap<String, Object>(3);

          map.put("e", e);
          map.put("newEndpoint", newEndpoint);
          map.put("maybe", eSelectedEndpoint == null ? "(maybe) " : "");

          StringCompiler sc =
              new StringCompiler(map).c("{e.id} is {maybe} watching {newEndpoint.id}.");

          logDebug(sc.toString().replaceAll("\\s+", " "));
        }

        startHighQualityStream = true;
        break;
      }
    }

    if (startHighQualityStream) {
      // TODO(gp) this assumes only a single hq stream.

      logDebug(
          getSimulcastEngine().getVideoChannel().getEndpoint().getID()
              + " notifies "
              + newEndpoint.getID()
              + " to start its HQ stream.");

      SimulcastLayer hqLayer = newSimulcastLayers[newSimulcastLayers.length - 1];
      ;
      StartSimulcastLayerCommand command = new StartSimulcastLayerCommand(hqLayer);
      String json = mapper.toJson(command);

      try {
        newEndpoint.sendMessageOnDataChannel(json);
      } catch (IOException e) {
        logError(newEndpoint.getID() + " failed to send message on data channel.", e);
      }
    }
  }
    /**
     * 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;
        }
      }
    }