private void Listen()
{
try
{
UdpClient udpSvr = m_udpListener;
if (udpSvr == null)
{
logger.Error("The UDP server was not correctly initialised in the RTP sink when attempting to start the listener, the RTP stream has not been intialised.");
return;
}
else
{
logger.Debug("RTP Listener now listening on " + m_localEndPoint.Address + ":" + m_localEndPoint.Port + ".");
}
IPEndPoint remoteEndPoint = new IPEndPoint(IPAddress.Any, 0);
byte[] rcvdBytes = null;
m_startRTPReceiveTime = DateTime.MinValue;
m_lastRTPReceivedTime = DateTime.MinValue;
DateTime previousRTPReceiveTime = DateTime.MinValue;
uint previousTimestamp = 0;
UInt16 sequenceNumber = 0;
UInt16 previousSeqNum = 0;
uint senderSendSpacing = 0;
uint lastSenderSendSpacing = 0;
while (!StopListening)
{
rcvdBytes = null;
try
{
rcvdBytes = udpSvr.Receive(ref remoteEndPoint);
}
catch
{
//logger.Warn("Remote socket closed on receive. Last RTP received " + m_lastRTPReceivedTime.ToString("dd MMM yyyy HH:mm:ss") + ", last RTP successfull send " + m_lastRTPSentTime.ToString("dd MMM yyyy HH:mm:ss") + ".");
}
if (rcvdBytes != null && rcvdBytes.Length > 0)
{
// Check whether this is an RTCP report.
UInt16 firstWord = BitConverter.ToUInt16(rcvdBytes, 0);
if (BitConverter.IsLittleEndian)
{
firstWord = NetConvert.DoReverseEndian(firstWord);
}
ushort packetType = 0;
if (BitConverter.IsLittleEndian)
{
packetType = Convert.ToUInt16(firstWord & 0x00ff);
}
if (packetType == RTCPHeader.RTCP_PACKET_TYPE)
{
logger.Debug("RTP Listener received remote RTCP report from " + remoteEndPoint + ".");
try
{
RTCPPacket rtcpPacket = new RTCPPacket(rcvdBytes);
RTCPReportPacket rtcpReportPacket = new RTCPReportPacket(rtcpPacket.Reports);
if (RTCPReportReceived != null)
{
RTCPReportReceived(this, rtcpReportPacket);
}
}
catch (Exception rtcpExcp)
{
logger.Error("Exception processing remote RTCP report. " + rtcpExcp.Message);
}
continue;
}
// Channel statistics.
DateTime rtpReceiveTime = DateTime.Now;
if (m_startRTPReceiveTime == DateTime.MinValue)
{
m_startRTPReceiveTime = rtpReceiveTime;
//m_sampleStartTime = rtpReceiveTime;
}
previousRTPReceiveTime = new DateTime(m_lastRTPReceivedTime.Ticks);
m_lastRTPReceivedTime = rtpReceiveTime;
m_packetsReceived++;
m_bytesReceived += rcvdBytes.Length;
previousSeqNum = sequenceNumber;
// This stops the thread running the ListenerTimeout method from deciding the strema has recieved no RTP and therefore should be shutdown.
m_lastPacketReceived.Set();
// Let whoever has subscribed that an RTP packet has been received.
if (DataReceived != null)
{
try
{
DataReceived(m_streamId, rcvdBytes, remoteEndPoint);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink DataReceived. " + excp.Message);
}
}
if (m_packetsReceived % 500 == 0)
{
logger.Debug("Total packets received from " + remoteEndPoint.ToString() + " " + m_packetsReceived + ", bytes " + NumberFormatter.ToSIByteFormat(m_bytesReceived, 2) + ".");
}
try
{
RTPPacket rtpPacket = new RTPPacket(rcvdBytes);
uint syncSource = rtpPacket.Header.SyncSource;
uint timestamp = rtpPacket.Header.Timestamp;
sequenceNumber = rtpPacket.Header.SequenceNumber;
//logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", timestamp=" + timestamp);
if (previousRTPReceiveTime != DateTime.MinValue)
{
//uint senderSendSpacing = rtpPacket.Header.Timestamp - previousTimestamp;
// Need to cope with cases where the timestamp has looped, if this timestamp is < last timesatmp and there is a large difference in them then it's because the timestamp counter has looped.
lastSenderSendSpacing = senderSendSpacing;
senderSendSpacing = (Math.Abs(timestamp - previousTimestamp) > (uint.MaxValue / 2)) ? timestamp + uint.MaxValue - previousTimestamp : timestamp - previousTimestamp;
if (previousTimestamp > timestamp)
{
logger.Error("BUG: Listener previous timestamp (" + previousTimestamp + ") > timestamp (" + timestamp + "), last seq num=" + previousSeqNum + ", seqnum=" + sequenceNumber + ".");
// Cover for this bug until it's nailed down.
senderSendSpacing = lastSenderSendSpacing;
}
double senderSpacingMilliseconds = (double)senderSendSpacing / (double)TIMESTAMP_FACTOR;
double interarrivalReceiveTime = m_lastRTPReceivedTime.Subtract(previousRTPReceiveTime).TotalMilliseconds;
#region RTCP reporting.
if (m_rtcpSampler == null)
{
//resultsLogger.Info("First Packet: " + rtpPacket.Header.SequenceNumber + "," + m_arrivalTime.ToString("HH:mm:fff"));
m_rtcpSampler = new RTCPReportSampler(m_streamId, syncSource, remoteEndPoint, rtpPacket.Header.SequenceNumber, m_lastRTPReceivedTime, rcvdBytes.Length);
m_rtcpSampler.RTCPReportReady += new RTCPSampleReadyDelegate(m_rtcpSampler_RTCPReportReady);
m_rtcpSampler.StartSampling();
}
else
{
//m_receiverReports[syncSource].RecordRTPReceive(rtpPacket.Header.SequenceNumber, sendTime, rtpReceiveTime, rcvdBytes.Length);
// Transit time is calculated by knowing that the sender sent a packet at a certain time after the last send and the receiver received a pakcet a certain time after the last receive.
// The difference in these two times is the jitter present. The transit time can change with each transimission and as this methid relies on two sends two packet
// arrivals to calculate the transit time it's not going to be perfect (you'd need synchronised NTP clocks at each end to be able to be accurate).
// However if used tor an average calculation it should be pretty close.
//double transitTime = Math.Abs(interarrivalReceiveTime - senderSpacingMilliseconds);
uint jitter = (interarrivalReceiveTime - senderSpacingMilliseconds > 0) ? Convert.ToUInt32(interarrivalReceiveTime - senderSpacingMilliseconds) : 0;
if (jitter > 75)
{
logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", timestmap=" + timestamp + ", ts-prev=" + previousTimestamp + ", receive spacing=" + interarrivalReceiveTime + ", send spacing=" + senderSpacingMilliseconds + ", jitter=" + jitter);
}
else
{
//logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", receive spacing=" + interarrivalReceiveTime + ", timestamp=" + timestamp + ", transit time=" + transitTime);
}
m_rtcpSampler.RecordRTPReceive(m_lastRTPReceivedTime, rtpPacket.Header.SequenceNumber, rcvdBytes.Length, jitter);
}
#endregion
}
else
{
logger.Debug("RTPSink Listen SyncSource=" + rtpPacket.Header.SyncSource + ".");
}
previousTimestamp = timestamp;
}
catch (Exception excp)
{
logger.Error("Received data was not a valid RTP packet. " + excp.Message);
}
#region Switching endpoint if required to cope with NAT.
// If a packet is recieved from an endpoint that wasn't expected treat the stream as being NATted and switch the endpoint to the socket on the NAT server.
try
{
if (m_streamEndPoint != null && m_streamEndPoint.Address != null && remoteEndPoint != null && remoteEndPoint.Address != null && (m_streamEndPoint.Address.ToString() != remoteEndPoint.Address.ToString() || m_streamEndPoint.Port != remoteEndPoint.Port))
{
logger.Debug("Expecting RTP on " + IPSocket.GetSocketString(m_streamEndPoint) + " but received on " + IPSocket.GetSocketString(remoteEndPoint) + ", now sending to " + IPSocket.GetSocketString(remoteEndPoint) + ".");
m_streamEndPoint = remoteEndPoint;
if (RemoteEndPointChanged != null)
{
try
{
RemoteEndPointChanged(m_streamId, remoteEndPoint);
}
catch (Exception changeExcp)
{
logger.Error("Exception RTPListener Changing Remote EndPoint. " + changeExcp.Message);
}
}
}
}
catch (Exception setSendExcp)
{
logger.Error("Exception RTPListener setting SendTo Socket. " + setSendExcp.Message);
}
#endregion
}
else if (!StopListening) // Empty packet was received possibly indicating connection closure so check for timeout.
{
double noRTPRcvdDuration = (m_lastRTPReceivedTime != DateTime.MinValue) ? DateTime.Now.Subtract(m_lastRTPReceivedTime).TotalSeconds : 0;
double noRTPSentDuration = (m_lastRTPSentTime != DateTime.MinValue) ? DateTime.Now.Subtract(m_lastRTPSentTime).TotalSeconds : 0;
//logger.Warn("Remote socket closed on receive on " + m_localEndPoint.Address.ToString() + ":" + + m_localEndPoint.Port + ", reinitialising. No rtp for " + noRTPRcvdDuration + "s. last rtp " + m_lastRTPReceivedTime.ToString("dd MMM yyyy HH:mm:ss") + ".");
// If this check is not done then the stream will never time out if it doesn't receive the first packet.
if (m_lastRTPReceivedTime == DateTime.MinValue)
{
m_lastRTPReceivedTime = DateTime.Now;
}
remoteEndPoint = new IPEndPoint(IPAddress.Any, 0);
if ((noRTPRcvdDuration > NO_RTP_TIMEOUT || noRTPSentDuration > NO_RTP_TIMEOUT) && StopIfNoData)
{
logger.Warn("Disconnecting RTP listener on " + m_localEndPoint.ToString() + " due to not being able to send or receive any RTP for " + NO_RTP_TIMEOUT + "s.");
Shutdown();
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception Listen RTPSink: " + excp.Message);
}
finally
{
#region Shut down socket.
Shutdown();
if (ListenerClosed != null)
{
try
{
ListenerClosed(m_streamId, m_callDescriptorId);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink ListenerClosed. " + excp.Message);
}
}
#endregion
}
}
private void Listen()
{
try
{
UdpClient udpSvr = m_udpListener;
if (udpSvr == null)
{
logger.Error("The UDP server was not correctly initialised in the RTP sink when attempting to start the listener, the RTP stream has not been intialised.");
return;
}
else
{
logger.Debug("RTP Listener now listening on " + m_localEndPoint.Address + ":" + m_localEndPoint.Port + ".");
}
IPEndPoint remoteEndPoint = new IPEndPoint(IPAddress.Any, 0);
byte[] rcvdBytes = null;
m_startRTPReceiveTime = DateTime.MinValue;
m_lastRTPReceivedTime = DateTime.MinValue;
DateTime previousRTPReceiveTime = DateTime.MinValue;
uint previousTimestamp = 0;
UInt16 sequenceNumber = 0;
UInt16 previousSeqNum = 0;
uint senderSendSpacing = 0;
uint lastSenderSendSpacing = 0;
while (!StopListening)
{
rcvdBytes = null;
try
{
rcvdBytes = udpSvr.Receive(ref remoteEndPoint);
}
catch
{
//logger.Warn("Remote socket closed on receive. Last RTP received " + m_lastRTPReceivedTime.ToString("dd MMM yyyy HH:mm:ss") + ", last RTP successfull send " + m_lastRTPSentTime.ToString("dd MMM yyyy HH:mm:ss") + ".");
}
if (rcvdBytes != null && rcvdBytes.Length > 0)
{
// Check whether this is an RTCP report.
UInt16 firstWord = BitConverter.ToUInt16(rcvdBytes, 0);
if (BitConverter.IsLittleEndian)
{
firstWord = NetConvert.DoReverseEndian(firstWord);
}
ushort packetType = 0;
if (BitConverter.IsLittleEndian)
{
packetType = Convert.ToUInt16(firstWord & 0x00ff);
}
if (packetType == RTCPHeader.RTCP_PACKET_TYPE)
{
logger.Debug("RTP Listener received remote RTCP report from " + remoteEndPoint + ".");
try
{
RTCPPacket rtcpPacket = new RTCPPacket(rcvdBytes);
RTCPReportPacket rtcpReportPacket = new RTCPReportPacket(rtcpPacket.Reports);
if (RTCPReportReceived != null)
{
RTCPReportReceived(this, rtcpReportPacket);
}
}
catch (Exception rtcpExcp)
{
logger.Error("Exception processing remote RTCP report. " + rtcpExcp.Message);
}
continue;
}
// Channel statistics.
DateTime rtpReceiveTime = DateTime.Now;
if (m_startRTPReceiveTime == DateTime.MinValue)
{
m_startRTPReceiveTime = rtpReceiveTime;
//m_sampleStartTime = rtpReceiveTime;
}
previousRTPReceiveTime = new DateTime(m_lastRTPReceivedTime.Ticks);
m_lastRTPReceivedTime = rtpReceiveTime;
m_packetsReceived++;
m_bytesReceived += rcvdBytes.Length;
previousSeqNum = sequenceNumber;
// This stops the thread running the ListenerTimeout method from deciding the strema has recieved no RTP and therefore should be shutdown.
m_lastPacketReceived.Set();
// Let whoever has subscribed that an RTP packet has been received.
if (DataReceived != null)
{
try
{
DataReceived(m_streamId, rcvdBytes, remoteEndPoint);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink DataReceived. " + excp.Message);
}
}
if (m_packetsReceived % 500 == 0)
{
logger.Debug("Total packets received from " + remoteEndPoint.ToString() + " " + m_packetsReceived + ", bytes " + NumberFormatter.ToSIByteFormat(m_bytesReceived, 2) + ".");
}
try
{
RTPPacket rtpPacket = new RTPPacket(rcvdBytes);
uint syncSource = rtpPacket.Header.SyncSource;
uint timestamp = rtpPacket.Header.Timestamp;
sequenceNumber = rtpPacket.Header.SequenceNumber;
//logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", timestamp=" + timestamp);
if (previousRTPReceiveTime != DateTime.MinValue)
{
//uint senderSendSpacing = rtpPacket.Header.Timestamp - previousTimestamp;
// Need to cope with cases where the timestamp has looped, if this timestamp is < last timesatmp and there is a large difference in them then it's because the timestamp counter has looped.
lastSenderSendSpacing = senderSendSpacing;
senderSendSpacing = (Math.Abs(timestamp - previousTimestamp) > (uint.MaxValue / 2)) ? timestamp + uint.MaxValue - previousTimestamp : timestamp - previousTimestamp;
if (previousTimestamp > timestamp)
{
logger.Error("BUG: Listener previous timestamp (" + previousTimestamp + ") > timestamp (" + timestamp + "), last seq num=" + previousSeqNum + ", seqnum=" + sequenceNumber + ".");
// Cover for this bug until it's nailed down.
senderSendSpacing = lastSenderSendSpacing;
}
double senderSpacingMilliseconds = (double)senderSendSpacing / (double)TIMESTAMP_FACTOR;
double interarrivalReceiveTime = m_lastRTPReceivedTime.Subtract(previousRTPReceiveTime).TotalMilliseconds;
#region RTCP reporting.
if (m_rtcpSampler == null)
{
//resultsLogger.Info("First Packet: " + rtpPacket.Header.SequenceNumber + "," + m_arrivalTime.ToString("HH:mm:fff"));
m_rtcpSampler = new RTCPReportSampler(m_streamId, syncSource, remoteEndPoint, rtpPacket.Header.SequenceNumber, m_lastRTPReceivedTime, rcvdBytes.Length);
m_rtcpSampler.RTCPReportReady += new RTCPSampleReadyDelegate(m_rtcpSampler_RTCPReportReady);
m_rtcpSampler.StartSampling();
}
else
{
//m_receiverReports[syncSource].RecordRTPReceive(rtpPacket.Header.SequenceNumber, sendTime, rtpReceiveTime, rcvdBytes.Length);
// Transit time is calculated by knowing that the sender sent a packet at a certain time after the last send and the receiver received a pakcet a certain time after the last receive.
// The difference in these two times is the jitter present. The transit time can change with each transimission and as this methid relies on two sends two packet
// arrivals to calculate the transit time it's not going to be perfect (you'd need synchronised NTP clocks at each end to be able to be accurate).
// However if used tor an average calculation it should be pretty close.
//double transitTime = Math.Abs(interarrivalReceiveTime - senderSpacingMilliseconds);
uint jitter = (interarrivalReceiveTime - senderSpacingMilliseconds > 0) ? Convert.ToUInt32(interarrivalReceiveTime - senderSpacingMilliseconds) : 0;
if(jitter > 75)
{
logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", timestmap=" + timestamp + ", ts-prev=" + previousTimestamp + ", receive spacing=" + interarrivalReceiveTime + ", send spacing=" + senderSpacingMilliseconds + ", jitter=" + jitter);
}
else
{
//logger.Debug("seqno=" + rtpPacket.Header.SequenceNumber + ", receive spacing=" + interarrivalReceiveTime + ", timestamp=" + timestamp + ", transit time=" + transitTime);
}
m_rtcpSampler.RecordRTPReceive(m_lastRTPReceivedTime, rtpPacket.Header.SequenceNumber, rcvdBytes.Length, jitter);
}
#endregion
}
else
{
logger.Debug("RTPSink Listen SyncSource=" + rtpPacket.Header.SyncSource + ".");
}
previousTimestamp = timestamp;
}
catch (Exception excp)
{
logger.Error("Received data was not a valid RTP packet. " + excp.Message);
}
#region Switching endpoint if required to cope with NAT.
// If a packet is recieved from an endpoint that wasn't expected treat the stream as being NATted and switch the endpoint to the socket on the NAT server.
try
{
if (m_streamEndPoint != null && m_streamEndPoint.Address != null && remoteEndPoint != null && remoteEndPoint.Address != null && (m_streamEndPoint.Address.ToString() != remoteEndPoint.Address.ToString() || m_streamEndPoint.Port != remoteEndPoint.Port))
{
logger.Debug("Expecting RTP on " + IPSocket.GetSocketString(m_streamEndPoint) + " but received on " + IPSocket.GetSocketString(remoteEndPoint) + ", now sending to " + IPSocket.GetSocketString(remoteEndPoint) + ".");
m_streamEndPoint = remoteEndPoint;
if (RemoteEndPointChanged != null)
{
try
{
RemoteEndPointChanged(m_streamId, remoteEndPoint);
}
catch (Exception changeExcp)
{
logger.Error("Exception RTPListener Changing Remote EndPoint. " + changeExcp.Message);
}
}
}
}
catch (Exception setSendExcp)
{
logger.Error("Exception RTPListener setting SendTo Socket. " + setSendExcp.Message);
}
#endregion
}
else if (!StopListening) // Empty packet was received possibly indicating connection closure so check for timeout.
{
double noRTPRcvdDuration = (m_lastRTPReceivedTime != DateTime.MinValue) ? DateTime.Now.Subtract(m_lastRTPReceivedTime).TotalSeconds : 0;
double noRTPSentDuration = (m_lastRTPSentTime != DateTime.MinValue) ? DateTime.Now.Subtract(m_lastRTPSentTime).TotalSeconds : 0;
//logger.Warn("Remote socket closed on receive on " + m_localEndPoint.Address.ToString() + ":" + + m_localEndPoint.Port + ", reinitialising. No rtp for " + noRTPRcvdDuration + "s. last rtp " + m_lastRTPReceivedTime.ToString("dd MMM yyyy HH:mm:ss") + ".");
// If this check is not done then the stream will never time out if it doesn't receive the first packet.
if (m_lastRTPReceivedTime == DateTime.MinValue)
{
m_lastRTPReceivedTime = DateTime.Now;
}
remoteEndPoint = new IPEndPoint(IPAddress.Any, 0);
if ((noRTPRcvdDuration > NO_RTP_TIMEOUT || noRTPSentDuration > NO_RTP_TIMEOUT) && StopIfNoData)
{
logger.Warn("Disconnecting RTP listener on " + m_localEndPoint.ToString() + " due to not being able to send or receive any RTP for " + NO_RTP_TIMEOUT + "s.");
Shutdown();
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception Listen RTPSink: " + excp.Message);
}
finally
{
#region Shut down socket.
Shutdown();
if (ListenerClosed != null)
{
try
{
ListenerClosed(m_streamId, m_callDescriptorId);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink ListenerClosed. " + excp.Message);
}
}
#endregion
}
}