private boolean isFirstPacket(RtpPacket rtpPacket) {
if (firstPacket) {
lastPacketReceived = System.currentTimeMillis();
firstPacket = false;
if (log.isDebugEnabled())
log.debug(
"First packet ["
+ rtpPacket.getPayloadType()
+ ", length="
+ rtpPacket.getPayloadLength()
+ "] seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
return true;
}
return false;
}
private boolean isMarkerPacket(RtpPacket rtpPacket) {
/*
* FreeSWITCH sends a marker packet at the beginning of the voice frame.
* If you stop talking and then start talking, a marker packet is received on start talking. (ralam sept 20, 2010).
*/
if (rtpPacket.hasMarker()) {
if (log.isDebugEnabled())
log.debug(
"Marked packet ["
+ rtpPacket.getPayloadType()
+ ", length="
+ rtpPacket.getPayloadLength()
+ "] seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
return true;
}
return false;
}
private boolean shouldDropDelayedPacket(RtpPacket rtpPacket) {
long now = System.currentTimeMillis();
if (now - lastPacketReceived > 100) {
if (log.isDebugEnabled())
log.debug(
"Delayed packet ["
+ rtpPacket.getRtcpPayloadType()
+ ","
+ rtpPacket.getPayloadType()
+ ", length="
+ rtpPacket.getPayloadLength()
+ "] seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
lastPacketReceived = now;
return true;
}
lastPacketReceived = now;
return false;
}
private boolean resetDueToSuccessiveDroppedPackets() {
/*
* I notice that Asterisk (1.6.2.5) sets the rtp sequence number to 12 every time it sends a marked rtp packet. This screws up our
* way of determining which packet to drop. To get around this, we detect if consecutive packets have been dropped then reset
* the sequence number to handle the next incoming packets (ralam sept. 20, 2010).
*/
if (lastPacketDropped && successivePacketDroppedCount > 3) {
if (log.isDebugEnabled())
log.debug(
"Resetting after successive dropped packets [successivePacketDroppedCount="
+ successivePacketDroppedCount
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
lastPacketDropped = false;
successivePacketDroppedCount = 0;
return true;
}
return false;
}
private boolean seqNumRolledOver(RtpPacket rtpPacket) {
/*
* Max sequence num is 65535 (16-bits). Let's use 65000 as check to take into account
* delayed packets.
*/
if (lastSequenceNumber - rtpPacket.getSeqNum() > 65000) {
if (log.isDebugEnabled())
log.debug(
"Packet rolling over seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
return true;
}
return false;
}
public void receiveRtpPackets() {
int packetReceivedCounter = 0;
int internalBufferLength = payloadLength + RTP_HEADER_SIZE;
byte[] internalBuffer;
RtpPacket rtpPacket;
while (receivePackets) {
try {
internalBuffer = new byte[internalBufferLength];
rtpPacket = new RtpPacket(internalBuffer, internalBufferLength);
rtpSocket.receive(rtpPacket);
packetReceivedCounter++;
// log.debug("Received packet [" + rtpPacket.getRtcpPayloadType() + "," +
// rtpPacket.getPayloadType() + ", length=" + rtpPacket.getPayloadLength() + "]
// seqNum[rtpSeqNum=" + rtpPacket.getSeqNum() + ",lastSeqNum=" + lastSequenceNumber
// + "][rtpTS=" + rtpPacket.getTimestamp() + ",lastTS=" + lastPacketTimestamp +
// "][port=" + rtpSocket.getDatagramSocket().getLocalPort() + "]");
if (shouldDropDelayedPacket(rtpPacket)) {
continue;
}
if (rtpPacket.isRtcpPacket()) {
/**
* Asterisk (1.6.2.5) send RTCP packets. We just ignore them (for now). It could be for
* KeepAlive (http://tools.ietf.org/html/draft-ietf-avt-app-rtp-keepalive-09)
*/
if (log.isDebugEnabled())
log.debug(
"RTCP packet ["
+ rtpPacket.getRtcpPayloadType()
+ ", length="
+ rtpPacket.getPayloadLength()
+ "] seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
} else {
if (shouldHandlePacket(rtpPacket)) {
// log.debug("Handling packet [" + rtpPacket.getRtcpPayloadType() + "," +
// rtpPacket.getPayloadType() + ", length=" + rtpPacket.getPayloadLength() + "]
// seqNum[rtpSeqNum=" + rtpPacket.getSeqNum() + ",lastSeqNum=" + lastSequenceNumber
// + "][rtpTS=" + rtpPacket.getTimestamp() + ",lastTS=" +
// lastPacketTimestamp + "][port=" + rtpSocket.getDatagramSocket().getLocalPort() +
// "]");
processRtpPacket(rtpPacket);
} else {
if (log.isDebugEnabled())
log.debug(
"Corrupt packet ["
+ rtpPacket.getRtcpPayloadType()
+ ","
+ rtpPacket.getPayloadType()
+ ", length="
+ rtpPacket.getPayloadLength()
+ "] seqNum[rtpSeqNum="
+ rtpPacket.getSeqNum()
+ ",lastSeqNum="
+ lastSequenceNumber
+ "][rtpTS="
+ rtpPacket.getTimestamp()
+ ",lastTS="
+ lastPacketTimestamp
+ "][port="
+ rtpSocket.getDatagramSocket().getLocalPort()
+ "]");
if (lastPacketDropped) successivePacketDroppedCount++;
else lastPacketDropped = true;
}
}
} catch (
IOException e) { // We get this when the socket closes when the call hangs up.
receivePackets = false;
}
}
log.debug("Rtp Receiver stopped. Packet Received = " + packetReceivedCounter + ".");
if (listener != null) listener.onStoppedReceiving();
}