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/**
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* Java RTP Library (jlibrtp)
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* Copyright (C) 2006 Arne Kepp
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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package jlibrtp;
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import java.net.DatagramSocket;
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import java.net.MulticastSocket;
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import java.net.DatagramPacket;
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import java.net.InetAddress;
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import java.net.InetSocketAddress;
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import java.util.Iterator;
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import java.util.concurrent.locks.*;
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import java.util.Random;
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import java.util.Enumeration;
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/**
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* The RTPSession object is the core of jlibrtp.
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*
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* One should be instantiated for every communication channel, i.e. if you send voice and video, you should create one for each.
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*
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* The instance holds a participant database, as well as other information about the session. When the application registers with the session, the necessary threads for receiving and processing RTP packets are spawned.
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*
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* RTP Packets are sent synchronously, all other operations are asynchronous.
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*
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* @author Arne Kepp
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*/
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public class RTPSession {
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/**
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* The debug level is final to avoid compilation of if-statements.</br>
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* 0 provides no debugging information, 20 provides everything </br>
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* Debug output is written to System.out</br>
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* Debug level for RTP related things.
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*/
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final static public int rtpDebugLevel = 0;
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/**
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* The debug level is final to avoid compilation of if-statements.</br>
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* 0 provides no debugging information, 20 provides everything </br>
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* Debug output is written to System.out</br>
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* Debug level for RTCP related things.
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*/
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final static public int rtcpDebugLevel = 0;
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/** RTP unicast socket */
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protected DatagramSocket rtpSock = null;
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/** RTP multicast socket */
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protected MulticastSocket rtpMCSock = null;
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/** RTP multicast group */
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protected InetAddress mcGroup = null;
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// Internal state
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/** Whether this session is a multicast session or not */
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protected boolean mcSession = false;
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/** Current payload type, can be changed by application */
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protected int payloadType = 0;
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/** SSRC of this session */
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protected long ssrc;
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/** The last timestamp when we sent something */
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protected long lastTimestamp = 0;
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/** Current sequence number */
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protected int seqNum = 0;
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/** Number of packets sent by this session */
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protected int sentPktCount = 0;
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/** Number of octets sent by this session */
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protected int sentOctetCount = 0;
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/** The random seed */
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protected Random random = null;
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/** Session bandwidth in BYTES per second */
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protected int bandwidth = 8000;
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/** By default we do not return packets from strangers in unicast mode */
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protected boolean naiveReception = false;
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/** Should the library attempt frame reconstruction? */
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protected boolean frameReconstruction = true;
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/** Maximum number of packets used for reordering */
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protected int pktBufBehavior = 3;
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/** Participant database */
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protected ParticipantDatabase partDb = new ParticipantDatabase(this);
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/** Handle to application interface for RTP */
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protected RTPAppIntf appIntf = null;
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/** Handle to application interface for RTCP (optional) */
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protected RTCPAppIntf rtcpAppIntf = null;
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/** Handle to application interface for AVPF, RFC 4585 (optional) */
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protected RTCPAVPFIntf rtcpAVPFIntf = null;
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/** Handle to application interface for debugging */
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protected DebugAppIntf debugAppIntf = null;
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/** The RTCP session associated with this RTP Session */
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protected RTCPSession rtcpSession = null;
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/** The thread for receiving RTP packets */
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protected RTPReceiverThread recvThrd = null;
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/** The thread for invoking callbacks for RTP packets */
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protected AppCallerThread appCallerThrd = null;
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/** Lock to protect the packet buffers */
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final protected Lock pktBufLock = new ReentrantLock();
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/** Condition variable, to tell the */
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final protected Condition pktBufDataReady = pktBufLock.newCondition();
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/** Enough is enough, set to true when you want to quit. */
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protected boolean endSession = false;
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/** Only one registered application, please */
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protected boolean registered = false;
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/** We're busy resolving a SSRC conflict, please try again later */
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protected boolean conflict = false;
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/** Number of conflicts observed, exessive number suggests loop in network */
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protected int conflictCount = 0;
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/** SDES CNAME */
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protected String cname = null;
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/** SDES The participant's real name */
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public String name = null;
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/** SDES The participant's email */
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public String email = null;
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/** SDES The participant's phone number */
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public String phone = null;
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/** SDES The participant's location*/
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public String loc = null;
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/** SDES The tool the participants is using */
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public String tool = null;
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/** SDES A note */
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public String note = null;
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/** SDES A priv string, loosely defined */
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public String priv = null;
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// RFC 4585 stuff. This should live on RTCPSession, but we need to have this
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// infromation ready by the time the RTCP Session starts
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// 0 = RFC 3550 , -1 = ACK , 1 = Immediate feedback, 2 = Early RTCP,
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protected int rtcpMode = 0;
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protected int fbEarlyThreshold = -1; // group size, immediate -> early transition point
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protected int fbRegularThreshold = -1; // group size, early -> regular transition point
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protected int minInterval = 5000; // minimum interval
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protected int fbMaxDelay = 1000; // how long the information is useful
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// RTCP bandwidth
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protected int rtcpBandwidth = -1;
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/**
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* Returns an instance of a <b>unicast</b> RTP session.
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* Following this you should adjust any settings and then register your application.
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*
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* The sockets should have external ip addresses, else your CNAME automatically
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* generated CNAMe will be bad.
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*
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* @param rtpSocket UDP socket to receive RTP communication on
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* @param rtcpSocket UDP socket to receive RTCP communication on, null if none.
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*/
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public RTPSession(DatagramSocket rtpSocket, DatagramSocket rtcpSocket) {
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mcSession = false;
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rtpSock = rtpSocket;
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this.generateCNAME();
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this.generateSsrc();
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this.rtcpSession = new RTCPSession(this,rtcpSocket);
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// The sockets are not always imediately available?
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try { Thread.sleep(1); } catch (InterruptedException e) { System.out.println("RTPSession sleep failed"); }
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}
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/**
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* Returns an instance of a <b>multicast</b> RTP session.
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* Following this you should register your application.
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*
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* The sockets should have external ip addresses, else your CNAME automatically
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* generated CNAMe will be bad.
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*
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* @param rtpSock a multicast socket to receive RTP communication on
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* @param rtcpSock a multicast socket to receive RTP communication on
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* @param multicastGroup the multicast group that we want to communicate with.
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*/
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public RTPSession(MulticastSocket rtpSock, MulticastSocket rtcpSock, InetAddress multicastGroup) throws Exception {
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mcSession = true;
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rtpMCSock =rtpSock;
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mcGroup = multicastGroup;
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rtpMCSock.joinGroup(mcGroup);
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rtcpSock.joinGroup(mcGroup);
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this.generateCNAME();
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this.generateSsrc();
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this.rtcpSession = new RTCPSession(this,rtcpSock,mcGroup);
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// The sockets are not always imediately available?
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try { Thread.sleep(1); } catch (InterruptedException e) { System.out.println("RTPSession sleep failed"); }
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}
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/**
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* Registers an application (RTPAppIntf) with the RTP session.
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* The session will call receiveData() on the supplied instance whenever data has been received.
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*
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* Following this you should set the payload type and add participants to the session.
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*
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* @param rtpApp an object that implements the RTPAppIntf-interface
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* @param rtcpApp an object that implements the RTCPAppIntf-interface (optional)
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* @return -1 if this RTPSession-instance already has an application registered.
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*/
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public int RTPSessionRegister(RTPAppIntf rtpApp, RTCPAppIntf rtcpApp, DebugAppIntf debugApp) {
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if(registered) {
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System.out.println("RTPSessionRegister(): Can\'t register another application!");
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return -1;
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} else {
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registered = true;
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generateSeqNum();
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if(RTPSession.rtpDebugLevel > 0) {
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System.out.println("-> RTPSessionRegister");
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}
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this.appIntf = rtpApp;
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this.rtcpAppIntf = rtcpApp;
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this.debugAppIntf = debugApp;
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recvThrd = new RTPReceiverThread(this);
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appCallerThrd = new AppCallerThread(this, rtpApp);
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recvThrd.start();
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appCallerThrd.start();
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rtcpSession.start();
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return 0;
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}
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}
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/**
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* Send data to all participants registered as receivers, using the current timeStamp,
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* dynamic sequence number and the current payload type specified for the session.
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*
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* @param buf A buffer of bytes, less than 1496 bytes
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* @return null if there was a problem, {RTP Timestamp, Sequence number} otherwise
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*/
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public long[] sendData(byte[] buf) {
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byte[][] tmp = {buf};
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long[][] ret = this.sendData(tmp, null, null, -1, null);
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if(ret != null)
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return ret[0];
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return null;
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}
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/**
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* Send data to all participants registered as receivers, using the specified timeStamp,
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* sequence number and the current payload type specified for the session.
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*
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* @param buf A buffer of bytes, less than 1496 bytes
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* @param rtpTimestamp the RTP timestamp to be used in the packet
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* @param seqNum the sequence number to be used in the packet
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* @return null if there was a problem, {RTP Timestamp, Sequence number} otherwise
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*/
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public long[] sendData(byte[] buf, long rtpTimestamp, long seqNum) {
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byte[][] tmp = {buf};
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long[][] ret = this.sendData(tmp, null, null, -1, null);
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if(ret != null)
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return ret[0];
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return null;
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}
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/**
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* Send data to all participants registered as receivers, using the current timeStamp and
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* payload type. The RTP timestamp will be the same for all the packets.
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*
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* @param buffers A buffer of bytes, should not bed padded and less than 1500 bytes on most networks.
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* @param csrcArray an array with the SSRCs of contributing sources
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* @param markers An array indicating what packets should be marked. Rarely anything but the first one
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* @param rtpTimestamp The RTP timestamp to be applied to all packets
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* @param seqNumbers An array with the sequence number associated with each byte[]
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* @return null if there was a problem sending the packets, 2-dim array with {RTP Timestamp, Sequence number}
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*/
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public long[][] sendData(byte[][] buffers, long[] csrcArray, boolean[] markers, long rtpTimestamp, long[] seqNumbers) {
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if(RTPSession.rtpDebugLevel > 5) {
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System.out.println("-> RTPSession.sendData(byte[])");
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}
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// Same RTP timestamp for all
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if(rtpTimestamp < 0)
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rtpTimestamp = System.currentTimeMillis();
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// Return values
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long[][] ret = new long[buffers.length][2];
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for(int i=0; i<buffers.length; i++) {
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byte[] buf = buffers[i];
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boolean marker = false;
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if(markers != null)
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marker = markers[i];
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if(buf.length > 1500) {
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System.out.println("RTPSession.sendData() called with buffer exceeding 1500 bytes ("+buf.length+")");
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}
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// Get the return values
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ret[i][0] = rtpTimestamp;
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if(seqNumbers == null) {
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ret[i][1] = getNextSeqNum();
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} else {
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ret[i][1] = seqNumbers[i];
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}
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// Create a new RTP Packet
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RtpPkt pkt = new RtpPkt(rtpTimestamp,this.ssrc,(int) ret[i][1],this.payloadType,buf);
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if(csrcArray != null)
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pkt.setCsrcs(csrcArray);
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pkt.setMarked(marker);
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// Creates a raw packet
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byte[] pktBytes = pkt.encode();
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//System.out.println(Integer.toString(StaticProcs.bytesToUIntInt(pktBytes, 2)));
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// Pre-flight check, are resolving an SSRC conflict?
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if(this.conflict) {
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System.out.println("RTPSession.sendData() called while trying to resolve conflict.");
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return null;
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}
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if(this.mcSession) {
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DatagramPacket packet = null;
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try {
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packet = new DatagramPacket(pktBytes,pktBytes.length,this.mcGroup,this.rtpMCSock.getPort());
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} catch (Exception e) {
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System.out.println("RTPSession.sendData() packet creation failed.");
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e.printStackTrace();
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return null;
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}
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try {
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rtpMCSock.send(packet);
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//Debug
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if(this.debugAppIntf != null) {
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this.debugAppIntf.packetSent(1, (InetSocketAddress) packet.getSocketAddress(),
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new String("Sent multicast RTP packet of size " + packet.getLength() +
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" to " + packet.getSocketAddress().toString() + " via "
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+ rtpMCSock.getLocalSocketAddress().toString()));
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}
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} catch (Exception e) {
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System.out.println("RTPSession.sendData() multicast failed.");
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e.printStackTrace();
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return null;
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}
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} else {
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// Loop over recipients
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Iterator<Participant> iter = partDb.getUnicastReceivers();
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while(iter.hasNext()) {
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InetSocketAddress receiver = iter.next().rtpAddress;
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DatagramPacket packet = null;
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if(RTPSession.rtpDebugLevel > 15) {
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System.out.println(" Sending to " + receiver.toString());
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}
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try {
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packet = new DatagramPacket(pktBytes,pktBytes.length,receiver);
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} catch (Exception e) {
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System.out.println("RTPSession.sendData() packet creation failed.");
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e.printStackTrace();
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return null;
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}
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//Actually send the packet
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try {
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rtpSock.send(packet);
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//Debug
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if(this.debugAppIntf != null) {
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this.debugAppIntf.packetSent(0, (InetSocketAddress) packet.getSocketAddress(),
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new String("Sent unicast RTP packet of size " + packet.getLength() +
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" to " + packet.getSocketAddress().toString() + " via "
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+ rtpSock.getLocalSocketAddress().toString()));
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}
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} catch (Exception e) {
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System.out.println("RTPSession.sendData() unicast failed.");
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e.printStackTrace();
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return null;
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}
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}
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}
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//Update our stats
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this.sentPktCount++;
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this.sentOctetCount++;
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400 |
if(RTPSession.rtpDebugLevel > 5) {
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System.out.println("<- RTPSession.sendData(byte[]) " + pkt.getSeqNumber());
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|
402 |
}
|
|
403 |
}
|
|
404 |
|
|
405 |
return ret;
|
|
406 |
}
|
|
407 |
|
|
408 |
/**
|
|
409 |
* Send RTCP App packet to receiver specified by ssrc
|
|
410 |
*
|
|
411 |
*
|
|
412 |
*
|
|
413 |
* Return values:
|
|
414 |
* 0 okay
|
|
415 |
* -1 no RTCP session established
|
|
416 |
* -2 name is not byte[4];
|
|
417 |
* -3 data is not byte[x], where x = 4*y for syme y
|
|
418 |
* -4 type is not a 5 bit unsigned integer
|
|
419 |
*
|
|
420 |
* Note that a return value of 0 does not guarantee delivery.
|
|
421 |
* The participant must also exist in the participant database,
|
|
422 |
* otherwise the message will eventually be deleted.
|
|
423 |
*
|
|
424 |
* @param ssrc of the participant you want to reach
|
|
425 |
* @param type the RTCP App packet subtype, default 0
|
|
426 |
* @param name the ASCII (in byte[4]) representation
|
|
427 |
* @param data the data itself
|
|
428 |
* @return 0 if okay, negative value otherwise (see above)
|
|
429 |
*/
|
|
430 |
|
|
431 |
public int sendRTCPAppPacket(long ssrc, int type, byte[] name, byte[] data) {
|
|
432 |
if(this.rtcpSession == null)
|
|
433 |
return -1;
|
|
434 |
|
|
435 |
if(name.length != 4)
|
|
436 |
return -2;
|
|
437 |
|
|
438 |
if(data.length % 4 != 0)
|
|
439 |
return -3;
|
|
440 |
|
|
441 |
if(type > 63 || type < 0 )
|
|
442 |
return -4;
|
|
443 |
|
|
444 |
RtcpPktAPP pkt = new RtcpPktAPP(ssrc, type, name, data);
|
|
445 |
this.rtcpSession.addToAppQueue(ssrc, pkt);
|
|
446 |
|
|
447 |
return 0;
|
|
448 |
}
|
|
449 |
/**
|
|
450 |
* Add a participant object to the participant database.
|
|
451 |
*
|
|
452 |
* If packets have already been received from this user, we will try to update the automatically inserted participant with the information provided here.
|
|
453 |
*
|
|
454 |
* @param p A participant.
|
|
455 |
*/
|
|
456 |
public int addParticipant(Participant p) {
|
|
457 |
//For now we make all participants added this way persistent
|
|
458 |
p.unexpected = false;
|
|
459 |
return this.partDb.addParticipant(0, p);
|
|
460 |
}
|
|
461 |
|
|
462 |
/**
|
|
463 |
* Remove a participant from the database. All buffered packets will be destroyed.
|
|
464 |
*
|
|
465 |
* @param p A participant.
|
|
466 |
*/
|
|
467 |
public void removeParticipant(Participant p) {
|
|
468 |
partDb.removeParticipant(p);
|
|
469 |
}
|
|
470 |
|
|
471 |
public Iterator<Participant> getUnicastReceivers() {
|
|
472 |
return partDb.getUnicastReceivers();
|
|
473 |
}
|
|
474 |
|
|
475 |
public Enumeration<Participant> getParticipants() {
|
|
476 |
return partDb.getParticipants();
|
|
477 |
}
|
|
478 |
|
|
479 |
/**
|
|
480 |
* End the RTP Session. This will halt all threads and send bye-messages to other participants.
|
|
481 |
*
|
|
482 |
* RTCP related threads may require several seconds to wake up and terminate.
|
|
483 |
*/
|
|
484 |
public void endSession() {
|
|
485 |
this.endSession = true;
|
|
486 |
|
|
487 |
// No more RTP packets, please
|
|
488 |
if(this.mcSession) {
|
|
489 |
this.rtpMCSock.close();
|
|
490 |
} else {
|
|
491 |
this.rtpSock.close();
|
|
492 |
}
|
|
493 |
|
|
494 |
// Signal the thread that pushes data to application
|
|
495 |
this.pktBufLock.lock();
|
|
496 |
try { this.pktBufDataReady.signalAll(); } finally {
|
|
497 |
this.pktBufLock.unlock();
|
|
498 |
}
|
|
499 |
// Interrupt what may be sleeping
|
|
500 |
this.rtcpSession.senderThrd.interrupt();
|
|
501 |
|
|
502 |
// Give things a chance to cool down.
|
|
503 |
try { Thread.sleep(50); } catch (Exception e){ };
|
|
504 |
|
|
505 |
this.appCallerThrd.interrupt();
|
|
506 |
|
|
507 |
// Give things a chance to cool down.
|
|
508 |
try { Thread.sleep(50); } catch (Exception e){ };
|
|
509 |
|
|
510 |
if(this.rtcpSession != null) {
|
|
511 |
// No more RTP packets, please
|
|
512 |
if(this.mcSession) {
|
|
513 |
this.rtcpSession.rtcpMCSock.close();
|
|
514 |
} else {
|
|
515 |
this.rtcpSession.rtcpSock.close();
|
|
516 |
}
|
|
517 |
}
|
|
518 |
}
|
|
519 |
|
|
520 |
|
|
521 |
/**
|
|
522 |
* Check whether this session is ending.
|
|
523 |
*
|
|
524 |
* @return true if session and associated threads are terminating.
|
|
525 |
*/
|
|
526 |
boolean isEnding() {
|
|
527 |
return this.endSession;
|
|
528 |
}
|
|
529 |
|
|
530 |
/**
|
|
531 |
* Overrides CNAME, used for outgoing RTCP packets.
|
|
532 |
*
|
|
533 |
* @param cname a string, e.g. username@hostname. Must be unique for session.
|
|
534 |
*/
|
|
535 |
public void CNAME(String cname) {
|
|
536 |
this.cname = cname;
|
|
537 |
}
|
|
538 |
|
|
539 |
/**
|
|
540 |
* Get the current CNAME, used for outgoing SDES packets
|
|
541 |
*/
|
|
542 |
public String CNAME() {
|
|
543 |
return this.cname;
|
|
544 |
}
|
|
545 |
|
|
546 |
public long getSsrc() {
|
|
547 |
return this.ssrc;
|
|
548 |
}
|
|
549 |
|
|
550 |
private void generateCNAME() {
|
|
551 |
String hostname;
|
|
552 |
|
|
553 |
if(this.mcSession) {
|
|
554 |
hostname = this.rtpMCSock.getLocalAddress().getCanonicalHostName();
|
|
555 |
} else {
|
|
556 |
hostname = this.rtpSock.getLocalAddress().getCanonicalHostName();
|
|
557 |
}
|
|
558 |
|
|
559 |
//if(hostname.equals("0.0.0.0") && System.getenv("HOSTNAME") != null) {
|
|
560 |
// hostname = System.getenv("HOSTNAME");
|
|
561 |
//}
|
|
562 |
|
|
563 |
cname = System.getProperty("user.name") + "@" + hostname;
|
|
564 |
}
|
|
565 |
|
|
566 |
/**
|
|
567 |
* Change the RTP socket of the session.
|
|
568 |
* Peers must be notified through SIP or other signalling protocol.
|
|
569 |
* Only valid if this is a unicast session to begin with.
|
|
570 |
*
|
|
571 |
* @param newSock integer for new port number, check it is free first.
|
|
572 |
*/
|
|
573 |
public int updateRTPSock(DatagramSocket newSock) {
|
|
574 |
if(!mcSession) {
|
|
575 |
rtpSock = newSock;
|
|
576 |
return 0;
|
|
577 |
} else {
|
|
578 |
System.out.println("Can't switch from multicast to unicast.");
|
|
579 |
return -1;
|
|
580 |
}
|
|
581 |
}
|
|
582 |
|
|
583 |
/**
|
|
584 |
* Change the RTCP socket of the session.
|
|
585 |
* Peers must be notified through SIP or other signalling protocol.
|
|
586 |
* Only valid if this is a unicast session to begin with.
|
|
587 |
*
|
|
588 |
* @param newSock the new unicast socket for RTP communication.
|
|
589 |
*/
|
|
590 |
public int updateRTCPSock(DatagramSocket newSock) {
|
|
591 |
if(!mcSession) {
|
|
592 |
this.rtcpSession.rtcpSock = newSock;
|
|
593 |
return 0;
|
|
594 |
} else {
|
|
595 |
System.out.println("Can't switch from multicast to unicast.");
|
|
596 |
return -1;
|
|
597 |
}
|
|
598 |
}
|
|
599 |
|
|
600 |
/**
|
|
601 |
* Change the RTP multicast socket of the session.
|
|
602 |
* Peers must be notified through SIP or other signalling protocol.
|
|
603 |
* Only valid if this is a multicast session to begin with.
|
|
604 |
*
|
|
605 |
* @param newSock the new multicast socket for RTP communication.
|
|
606 |
*/
|
|
607 |
public int updateRTPSock(MulticastSocket newSock) {
|
|
608 |
if(mcSession) {
|
|
609 |
this.rtpMCSock = newSock;
|
|
610 |
return 0;
|
|
611 |
} else {
|
|
612 |
System.out.println("Can't switch from unicast to multicast.");
|
|
613 |
return -1;
|
|
614 |
}
|
|
615 |
}
|
|
616 |
|
|
617 |
/**
|
|
618 |
* Change the RTCP multicast socket of the session.
|
|
619 |
* Peers must be notified through SIP or other signalling protocol.
|
|
620 |
* Only valid if this is a multicast session to begin with.
|
|
621 |
*
|
|
622 |
* @param newSock the new multicast socket for RTCP communication.
|
|
623 |
*/
|
|
624 |
public int updateRTCPSock(MulticastSocket newSock) {
|
|
625 |
if(mcSession) {
|
|
626 |
this.rtcpSession.rtcpMCSock = newSock;
|
|
627 |
return 0;
|
|
628 |
} else {
|
|
629 |
System.out.println("Can't switch from unicast to multicast.");
|
|
630 |
return -1;
|
|
631 |
}
|
|
632 |
}
|
|
633 |
|
|
634 |
/**
|
|
635 |
* Update the payload type used for the session. It is represented as a 7 bit integer, whose meaning must be negotiated elsewhere (see IETF RFCs <a href="http://www.ietf.org/rfc/rfc3550.txt">3550</a> and <a href="http://www.ietf.org/rfc/rfc3550.txt">3551</a>)
|
|
636 |
*
|
|
637 |
* @param payloadT an integer representing the payload type of any subsequent packets that are sent.
|
|
638 |
*/
|
|
639 |
public int payloadType(int payloadT) {
|
|
640 |
if(payloadT > 128 || payloadT < 0) {
|
|
641 |
return -1;
|
|
642 |
} else {
|
|
643 |
this.payloadType = payloadT;
|
|
644 |
return this.payloadType;
|
|
645 |
}
|
|
646 |
}
|
|
647 |
|
|
648 |
/**
|
|
649 |
* Get the payload type that is currently used for outgoing RTP packets.
|
|
650 |
*
|
|
651 |
* @return payload type as integer
|
|
652 |
*/
|
|
653 |
public int payloadType() {
|
|
654 |
return this.payloadType;
|
|
655 |
}
|
|
656 |
|
|
657 |
/**
|
|
658 |
* Should packets from unknown participants be returned to the application? This can be dangerous.
|
|
659 |
*
|
|
660 |
* @param doAccept packets from participants not added by the application.
|
|
661 |
*/
|
|
662 |
public void naivePktReception(boolean doAccept) {
|
|
663 |
naiveReception = doAccept;
|
|
664 |
}
|
|
665 |
|
|
666 |
/**
|
|
667 |
* Are packets from unknown participants returned to the application?
|
|
668 |
*
|
|
669 |
* @return whether we accept packets from participants not added by the application.
|
|
670 |
*/
|
|
671 |
public boolean naivePktReception() {
|
|
672 |
return naiveReception;
|
|
673 |
}
|
|
674 |
|
|
675 |
/**
|
|
676 |
* Set the number of RTP packets that should be buffered when a packet is
|
|
677 |
* missing or received out of order. Setting this number high increases
|
|
678 |
* the chance of correctly reordering packets, but increases latency when
|
|
679 |
* a packet is dropped by the network.
|
|
680 |
*
|
|
681 |
* Packets that arrive in order are not affected, they are passed straight
|
|
682 |
* to the application.
|
|
683 |
*
|
|
684 |
* The maximum delay is numberofPackets * packet rate , where the packet rate
|
|
685 |
* depends on the codec and profile used by the sender.
|
|
686 |
*
|
|
687 |
* Valid values:
|
|
688 |
* >0 - The maximum number of packets (based on RTP Timestamp) that may accumulate
|
|
689 |
* 0 - All valid packets received in order will be given to the application
|
|
690 |
* -1 - All valid packets will be given to the application
|
|
691 |
*
|
|
692 |
* @param behavior the be
|
|
693 |
* @return the behavior set, unchanged in the case of a erroneous value
|
|
694 |
*/
|
|
695 |
public int packetBufferBehavior(int behavior) {
|
|
696 |
if(behavior > -2) {
|
|
697 |
this.pktBufBehavior = behavior;
|
|
698 |
// Signal the thread that pushes data to application
|
|
699 |
this.pktBufLock.lock();
|
|
700 |
try { this.pktBufDataReady.signalAll(); } finally {
|
|
701 |
this.pktBufLock.unlock();
|
|
702 |
}
|
|
703 |
return this.pktBufBehavior;
|
|
704 |
} else {
|
|
705 |
return this.pktBufBehavior;
|
|
706 |
}
|
|
707 |
}
|
|
708 |
|
|
709 |
/**
|
|
710 |
* The number of RTP packets that should be buffered when a packet is
|
|
711 |
* missing or received out of order. A high number increases the chance
|
|
712 |
* of correctly reordering packets, but increases latency when a packet is
|
|
713 |
* dropped by the network.
|
|
714 |
*
|
|
715 |
* A negative value disables the buffering, out of order packets will simply be dropped.
|
|
716 |
*
|
|
717 |
* @return the maximum number of packets that can accumulate before the first is returned
|
|
718 |
*/
|
|
719 |
public int packetBufferBehavior() {
|
|
720 |
return this.pktBufBehavior;
|
|
721 |
}
|
|
722 |
|
|
723 |
/**
|
|
724 |
* Set whether the stack should operate in RFC 4585 mode.
|
|
725 |
*
|
|
726 |
* This will automatically call adjustPacketBufferBehavior(-1),
|
|
727 |
* i.e. disable all RTP packet buffering in jlibrtp,
|
|
728 |
* and disable frame reconstruction
|
|
729 |
*
|
|
730 |
* @param rtcpAVPFIntf the in
|
|
731 |
*/
|
|
732 |
public int registerAVPFIntf(RTCPAVPFIntf rtcpAVPFIntf, int maxDelay, int earlyThreshold, int regularThreshold ) {
|
|
733 |
if(this.rtcpSession != null) {
|
|
734 |
this.packetBufferBehavior(-1);
|
|
735 |
this.frameReconstruction = false;
|
|
736 |
this.rtcpAVPFIntf = rtcpAVPFIntf;
|
|
737 |
this.fbEarlyThreshold = earlyThreshold;
|
|
738 |
this.fbRegularThreshold = regularThreshold;
|
|
739 |
return 0;
|
|
740 |
} else {
|
|
741 |
return -1;
|
|
742 |
}
|
|
743 |
}
|
|
744 |
|
|
745 |
/**
|
|
746 |
* Unregisters the RTCP AVPF interface, thereby going from
|
|
747 |
* RFC 4585 mode to RFC 3550
|
|
748 |
*
|
|
749 |
* You still have to adjust packetBufferBehavior() and
|
|
750 |
* frameReconstruction.
|
|
751 |
*
|
|
752 |
*/
|
|
753 |
public void unregisterAVPFIntf() {
|
|
754 |
this.fbEarlyThreshold = -1;
|
|
755 |
this.fbRegularThreshold = -1;
|
|
756 |
this.rtcpAVPFIntf = null;
|
|
757 |
}
|
|
758 |
|
|
759 |
/**
|
|
760 |
* Enable / disable frame reconstruction in the packet buffers.
|
|
761 |
* This is only relevant if getPacketBufferBehavior > 0;
|
|
762 |
*
|
|
763 |
* Default is true.
|
|
764 |
*/
|
|
765 |
public void frameReconstruction(boolean toggle) {
|
|
766 |
this.frameReconstruction = toggle;
|
|
767 |
}
|
|
768 |
|
|
769 |
/**
|
|
770 |
* Whether the packet buffer will attempt to reconstruct
|
|
771 |
* packet automatically.
|
|
772 |
*
|
|
773 |
* @return the status
|
|
774 |
*/
|
|
775 |
public boolean frameReconstruction() {
|
|
776 |
return this.frameReconstruction;
|
|
777 |
}
|
|
778 |
|
|
779 |
/**
|
|
780 |
* The bandwidth currently allocated to the session,
|
|
781 |
* in bytes per second. The default is 8000.
|
|
782 |
*
|
|
783 |
* This value is not enforced and currently only
|
|
784 |
* used to calculate the RTCP interval to ensure the
|
|
785 |
* control messages do not exceed 5% of the total bandwidth
|
|
786 |
* described here.
|
|
787 |
*
|
|
788 |
* Since the actual value may change a conservative
|
|
789 |
* estimate should be used to avoid RTCP flooding.
|
|
790 |
*
|
|
791 |
* see rtcpBandwidth(void)
|
|
792 |
*
|
|
793 |
* @return current bandwidth setting
|
|
794 |
*/
|
|
795 |
public int sessionBandwidth() {
|
|
796 |
return this.bandwidth;
|
|
797 |
}
|
|
798 |
|
|
799 |
/**
|
|
800 |
* Set the bandwidth of the session.
|
|
801 |
*
|
|
802 |
* See sessionBandwidth(void) for details.
|
|
803 |
*
|
|
804 |
* @param bandwidth the new value requested, in bytes per second
|
|
805 |
* @return the actual value set
|
|
806 |
*/
|
|
807 |
public int sessionBandwidth(int bandwidth) {
|
|
808 |
if(bandwidth < 1) {
|
|
809 |
this.bandwidth = 8000;
|
|
810 |
} else {
|
|
811 |
this.bandwidth = bandwidth;
|
|
812 |
}
|
|
813 |
return this.bandwidth;
|
|
814 |
}
|
|
815 |
|
|
816 |
|
|
817 |
/**
|
|
818 |
* RFC 3550 dictates that 5% of the total bandwidth,
|
|
819 |
* as set by sessionBandwidth, should be dedicated
|
|
820 |
* to RTCP traffic. This
|
|
821 |
*
|
|
822 |
* This should normally not be done, but is permissible in
|
|
823 |
* conjunction with feedback (RFC 4585) and possibly
|
|
824 |
* other profiles.
|
|
825 |
*
|
|
826 |
* Also see sessionBandwidth(void)
|
|
827 |
*
|
|
828 |
* @return current RTCP bandwidth setting, -1 means not in use
|
|
829 |
*/
|
|
830 |
public int rtcpBandwidth() {
|
|
831 |
return this.rtcpBandwidth;
|
|
832 |
}
|
|
833 |
|
|
834 |
/**
|
|
835 |
* Set the RTCP bandwidth, see rtcpBandwidth(void) for details.
|
|
836 |
*
|
|
837 |
* This function must be
|
|
838 |
*
|
|
839 |
* @param bandwidth the new value requested, in bytes per second or -1 to disable
|
|
840 |
* @return the actual value set
|
|
841 |
*/
|
|
842 |
public int rtcpBandwidth(int bandwidth) {
|
|
843 |
if(bandwidth < -1) {
|
|
844 |
this.rtcpBandwidth = -1;
|
|
845 |
} else {
|
|
846 |
this.rtcpBandwidth = bandwidth;
|
|
847 |
}
|
|
848 |
return this.rtcpBandwidth;
|
|
849 |
}
|
|
850 |
|
|
851 |
/********************************************* Feedback message stuff ***************************************/
|
|
852 |
|
|
853 |
/**
|
|
854 |
* Adds a Picture Loss Indication to the feedback queue
|
|
855 |
*
|
|
856 |
* @param ssrcMediaSource
|
|
857 |
* @return 0 if packet was queued, -1 if no feedback support, 1 if redundant
|
|
858 |
*/
|
|
859 |
public int fbPictureLossIndication(long ssrcMediaSource) {
|
|
860 |
int ret = 0;
|
|
861 |
|
|
862 |
if(this.rtcpAVPFIntf == null)
|
|
863 |
return -1;
|
|
864 |
|
|
865 |
RtcpPktPSFB pkt = new RtcpPktPSFB(this.ssrc, ssrcMediaSource);
|
|
866 |
pkt.makePictureLossIndication();
|
|
867 |
ret = this.rtcpSession.addToFbQueue(ssrcMediaSource, pkt);
|
|
868 |
if(ret == 0)
|
|
869 |
this.rtcpSession.wakeSenderThread(ssrcMediaSource);
|
|
870 |
return ret;
|
|
871 |
}
|
|
872 |
|
|
873 |
/**
|
|
874 |
* Adds a Slice Loss Indication to the feedback queue
|
|
875 |
*
|
|
876 |
* @param ssrcMediaSource
|
|
877 |
* @param sliFirst macroblock (MB) address of the first lost macroblock
|
|
878 |
* @param sliNumber number of lost macroblocks
|
|
879 |
* @param sliPictureId six least significant bits of the codec-specific identif
|
|
880 |
* @return 0 if packet was queued, -1 if no feedback support, 1 if redundant
|
|
881 |
*/
|
|
882 |
public int fbSlicLossIndication(long ssrcMediaSource, int[] sliFirst, int[] sliNumber, int[] sliPictureId) {
|
|
883 |
int ret = 0;
|
|
884 |
if(this.rtcpAVPFIntf == null)
|
|
885 |
return -1;
|
|
886 |
|
|
887 |
RtcpPktPSFB pkt = new RtcpPktPSFB(this.ssrc, ssrcMediaSource);
|
|
888 |
pkt.makeSliceLossIndication(sliFirst, sliNumber, sliPictureId);
|
|
889 |
|
|
890 |
ret = this.rtcpSession.addToFbQueue(ssrcMediaSource, pkt);
|
|
891 |
if(ret == 0)
|
|
892 |
this.rtcpSession.wakeSenderThread(ssrcMediaSource);
|
|
893 |
return ret;
|
|
894 |
}
|
|
895 |
|
|
896 |
/**
|
|
897 |
* Adds a Reference Picture Selection Indication to the feedback queue
|
|
898 |
*
|
|
899 |
* @param ssrcMediaSource
|
|
900 |
* @param bitPadding number of padded bits at end of bitString
|
|
901 |
* @param payloadType RTP payload type for codec
|
|
902 |
* @param bitString RPSI information as natively defined by the video codec
|
|
903 |
* @return 0 if packet was queued, -1 if no feedback support, 1 if redundant
|
|
904 |
*/
|
|
905 |
public int fbRefPictureSelIndic(long ssrcMediaSource, int bitPadding, int payloadType, byte[] bitString) {
|
|
906 |
int ret = 0;
|
|
907 |
|
|
908 |
if(this.rtcpAVPFIntf == null)
|
|
909 |
return -1;
|
|
910 |
|
|
911 |
RtcpPktPSFB pkt = new RtcpPktPSFB(this.ssrc, ssrcMediaSource);
|
|
912 |
pkt.makeRefPictureSelIndic(bitPadding, payloadType, bitString);
|
|
913 |
ret = this.rtcpSession.addToFbQueue(ssrcMediaSource, pkt);
|
|
914 |
if(ret == 0)
|
|
915 |
this.rtcpSession.wakeSenderThread(ssrcMediaSource);
|
|
916 |
return ret;
|
|
917 |
}
|
|
918 |
|
|
919 |
/**
|
|
920 |
* Adds a Picture Loss Indication to the feedback queue
|
|
921 |
*
|
|
922 |
* @param ssrcMediaSource
|
|
923 |
* @param bitString the original application message
|
|
924 |
* @return 0 if packet was queued, -1 if no feedback support, 1 if redundant
|
|
925 |
*/
|
|
926 |
public int fbAppLayerFeedback(long ssrcMediaSource, byte[] bitString) {
|
|
927 |
int ret = 0;
|
|
928 |
|
|
929 |
if(this.rtcpAVPFIntf == null)
|
|
930 |
return -1;
|
|
931 |
|
|
932 |
RtcpPktPSFB pkt = new RtcpPktPSFB(this.ssrc, ssrcMediaSource);
|
|
933 |
pkt.makeAppLayerFeedback(bitString);
|
|
934 |
ret = this.rtcpSession.addToFbQueue(ssrcMediaSource, pkt);
|
|
935 |
if(ret == 0)
|
|
936 |
this.rtcpSession.wakeSenderThread(ssrcMediaSource);
|
|
937 |
return ret;
|
|
938 |
}
|
|
939 |
|
|
940 |
|
|
941 |
/**
|
|
942 |
* Adds a RTP Feedback packet to the feedback queue.
|
|
943 |
*
|
|
944 |
* These are mostly used for NACKs.
|
|
945 |
*
|
|
946 |
* @param ssrcMediaSource
|
|
947 |
* @param FMT the Feedback Message Subtype
|
|
948 |
* @param PID RTP sequence numbers of lost packets
|
|
949 |
* @param BLP bitmask of following lost packets, shared index with PID
|
|
950 |
* @return 0 if packet was queued, -1 if no feedback support, 1 if redundant
|
|
951 |
*/
|
|
952 |
public int fbPictureLossIndication(long ssrcMediaSource, int FMT, int[] PID, int[] BLP) {
|
|
953 |
int ret = 0;
|
|
954 |
|
|
955 |
if(this.rtcpAVPFIntf == null)
|
|
956 |
return -1;
|
|
957 |
|
|
958 |
RtcpPktRTPFB pkt = new RtcpPktRTPFB(this.ssrc, ssrcMediaSource, FMT, PID, BLP);
|
|
959 |
ret = this.rtcpSession.addToFbQueue(ssrcMediaSource, pkt);
|
|
960 |
if(ret == 0)
|
|
961 |
this.rtcpSession.wakeSenderThread(ssrcMediaSource);
|
|
962 |
return ret;
|
|
963 |
}
|
|
964 |
|
|
965 |
/**
|
|
966 |
* Fetches the next sequence number for RTP packets.
|
|
967 |
* @return the next sequence number
|
|
968 |
*/
|
|
969 |
private int getNextSeqNum() {
|
|
970 |
seqNum++;
|
|
971 |
// 16 bit number
|
|
972 |
if(seqNum > 65536) {
|
|
973 |
seqNum = 0;
|
|
974 |
}
|
|
975 |
return seqNum;
|
|
976 |
}
|
|
977 |
|
|
978 |
/**
|
|
979 |
* Initializes a random variable
|
|
980 |
*
|
|
981 |
*/
|
|
982 |
private void createRandom() {
|
|
983 |
this.random = new Random(System.currentTimeMillis() + Thread.currentThread().getId()
|
|
984 |
- Thread.currentThread().hashCode() + this.cname.hashCode());
|
|
985 |
}
|
|
986 |
|
|
987 |
|
|
988 |
/**
|
|
989 |
* Generates a random sequence number
|
|
990 |
*/
|
|
991 |
private void generateSeqNum() {
|
|
992 |
if(this.random == null)
|
|
993 |
createRandom();
|
|
994 |
|
|
995 |
seqNum = this.random.nextInt();
|
|
996 |
if(seqNum < 0)
|
|
997 |
seqNum = -seqNum;
|
|
998 |
while(seqNum > 65535) {
|
|
999 |
seqNum = seqNum / 10;
|
|
1000 |
}
|
|
1001 |
}
|
|
1002 |
|
|
1003 |
/**
|
|
1004 |
* Generates a random SSRC
|
|
1005 |
*/
|
|
1006 |
private void generateSsrc() {
|
|
1007 |
if(this.random == null)
|
|
1008 |
createRandom();
|
|
1009 |
|
|
1010 |
// Set an SSRC
|
|
1011 |
this.ssrc = this.random.nextInt();
|
|
1012 |
if(this.ssrc < 0) {
|
|
1013 |
this.ssrc = this.ssrc * -1;
|
|
1014 |
}
|
|
1015 |
}
|
|
1016 |
|
|
1017 |
/**
|
|
1018 |
* Resolve an SSRC conflict.
|
|
1019 |
*
|
|
1020 |
* Also increments the SSRC conflict counter, after 5 conflicts
|
|
1021 |
* it is assumed there is a loop somewhere and the session will
|
|
1022 |
* terminate.
|
|
1023 |
*
|
|
1024 |
*/
|
|
1025 |
protected void resolveSsrcConflict() {
|
|
1026 |
System.out.println("!!!!!!! Beginning SSRC conflict resolution !!!!!!!!!");
|
|
1027 |
this.conflictCount++;
|
|
1028 |
|
|
1029 |
if(this.conflictCount < 5) {
|
|
1030 |
//Don't send any more regular packets out until we have this sorted out.
|
|
1031 |
this.conflict = true;
|
|
1032 |
|
|
1033 |
//Send byes
|
|
1034 |
rtcpSession.sendByes();
|
|
1035 |
|
|
1036 |
//Calculate the next delay
|
|
1037 |
rtcpSession.calculateDelay();
|
|
1038 |
|
|
1039 |
//Generate a new Ssrc for ourselves
|
|
1040 |
generateSsrc();
|
|
1041 |
|
|
1042 |
//Get the SDES packets out faster
|
|
1043 |
rtcpSession.initial = true;
|
|
1044 |
|
|
1045 |
this.conflict = false;
|
|
1046 |
System.out.println("SSRC conflict resolution complete");
|
|
1047 |
|
|
1048 |
} else {
|
|
1049 |
System.out.println("Too many conflicts. There is probably a loop in the network.");
|
|
1050 |
this.endSession();
|
|
1051 |
}
|
|
1052 |
}
|
|
1053 |
}
|