private void Send()
{
try
{
int payloadSize = RTPPacketSendSize;
RTPPacket rtpPacket = new RTPPacket(RTPPacketSendSize);
byte[] rtpBytes = rtpPacket.GetBytes();
RTPHeader rtpHeader = new RTPHeader();
rtpHeader.SequenceNumber = (UInt16)65000; //Convert.ToUInt16(Crypto.GetRandomInt(0, UInt16.MaxValue));
uint sendTimestamp = uint.MaxValue - 5000;
uint lastSendTimestamp = sendTimestamp;
UInt16 lastSeqNum = 0;
logger.Debug("RTP send stream starting to " + IPSocket.GetSocketString(m_streamEndPoint) +
" with payload size " + payloadSize + " bytes.");
Sending = true;
m_startRTPSendTime = DateTime.MinValue;
m_lastRTPSentTime = DateTime.MinValue;
m_sampleStartSeqNo = rtpHeader.SequenceNumber;
DateTime lastRTPSendAttempt = DateTime.Now;
while (m_udpListener != null && !StopListening)
{
// This may be changed by the listener so it needs to be set each iteration.
IPEndPoint dstEndPoint = m_streamEndPoint;
//logger.Info("Sending RTP packet to " + dstEndPoint.Address + ":" + dstEndPoint.Port);
if (payloadSize != m_rtpPacketSendSize)
{
payloadSize = m_rtpPacketSendSize;
logger.Info("Changing RTP payload to " + payloadSize);
rtpPacket = new RTPPacket(RTP_HEADER_SIZE + m_rtpPacketSendSize);
rtpBytes = rtpPacket.GetBytes();
}
try
{
if (m_startRTPSendTime == DateTime.MinValue)
{
m_startRTPSendTime = DateTime.Now;
rtpHeader.MarkerBit = 0;
logger.Debug("RTPSink Send SyncSource=" + rtpPacket.Header.SyncSource + ".");
}
else
{
lastSendTimestamp = sendTimestamp;
double milliSinceLastSend = DateTime.Now.Subtract(m_lastRTPSentTime).TotalMilliseconds;
sendTimestamp =
Convert.ToUInt32((lastSendTimestamp + (milliSinceLastSend * TIMESTAMP_FACTOR)) %
uint.MaxValue);
if (lastSendTimestamp > sendTimestamp)
{
logger.Error("RTP Sender previous timestamp (" + lastSendTimestamp + ") > timestamp (" +
sendTimestamp + ") ms since last send=" + milliSinceLastSend +
", lastseqnum=" + lastSeqNum + ", seqnum=" + rtpHeader.SequenceNumber +
".");
}
if (DateTime.Now.Subtract(m_lastRTPSentTime).TotalMilliseconds > 75)
{
logger.Debug("delayed send: " + rtpHeader.SequenceNumber + ", time since last send " +
DateTime.Now.Subtract(m_lastRTPSentTime).TotalMilliseconds + "ms.");
}
}
rtpHeader.Timestamp = sendTimestamp;
byte[] rtpHeaderBytes = rtpHeader.GetBytes();
Array.Copy(rtpHeaderBytes, 0, rtpBytes, 0, rtpHeaderBytes.Length);
// Send RTP packets and any extra channels required to emulate mutliple calls.
for (int channelCount = 0; channelCount < m_channels; channelCount++)
{
//logger.Debug("Send rtp getting wallclock timestamp for " + DateTime.Now.ToString("dd MMM yyyy HH:mm:ss:fff"));
//DateTime sendTime = DateTime.Now;
//rtpHeader.Timestamp = RTPHeader.GetWallclockUTCStamp(sendTime);
//logger.Debug(rtpHeader.SequenceNumber + "," + rtpHeader.Timestamp);
m_udpListener.Send(rtpBytes, rtpBytes.Length, dstEndPoint);
m_lastRTPSentTime = DateTime.Now;
m_packetsSent++;
m_bytesSent += rtpBytes.Length;
if (m_packetsSent % 500 == 0)
{
logger.Debug("Total packets sent to " + dstEndPoint.ToString() + " " + m_packetsSent +
", bytes " + NumberFormatter.ToSIByteFormat(m_bytesSent, 2) + ".");
}
//sendLogger.Info(m_lastRTPSentTime.ToString("dd MMM yyyy HH:mm:ss:fff") + "," + m_lastRTPSentTime.Subtract(m_startRTPSendTime).TotalMilliseconds.ToString("0") + "," + rtpHeader.SequenceNumber + "," + rtpBytes.Length);
//sendLogger.Info(rtpHeader.SequenceNumber + "," + DateTime.Now.ToString("dd MMM yyyy HH:mm:ss:fff"));
if (DataSent != null)
{
try
{
DataSent(m_streamId, rtpBytes, dstEndPoint);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink DataSent. " + excp.Message);
}
}
lastSeqNum = rtpHeader.SequenceNumber;
if (rtpHeader.SequenceNumber == UInt16.MaxValue)
{
//logger.Debug("RTPSink looping the sequence number in sample.");
rtpHeader.SequenceNumber = 0;
}
else
{
rtpHeader.SequenceNumber++;
}
}
}
catch (Exception excp)
{
logger.Error("Exception RTP Send. " + excp.GetType() + ". " + excp.Message);
if (excp.GetType() == typeof(SocketException))
{
logger.Error("socket exception errorcode=" + ((SocketException)excp).ErrorCode + ".");
}
logger.Warn("Remote socket closed on send. Last RTP recevied " +
m_lastRTPReceivedTime.ToString("dd MMM yyyy HH:mm:ss") +
", last RTP successfull send " +
m_lastRTPSentTime.ToString("dd MMM yyyy HH:mm:ss") + ".");
}
Thread.Sleep(RTPFrameSize);
#region Check for whether the stream has timed out on a send or receive and if so shut down the stream.
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;
double testDuration = DateTime.Now.Subtract(m_startRTPSendTime).TotalSeconds;
if ((
noRTPRcvdDuration > NO_RTP_TIMEOUT ||
noRTPSentDuration > NO_RTP_TIMEOUT ||
(m_lastRTPReceivedTime == DateTime.MinValue && testDuration > NO_RTP_TIMEOUT)
) && // If the test request comes from a private or unreachable IP address then no RTP will ever be received.
StopIfNoData)
{
logger.Warn("Disconnecting RTP stream on " + m_localEndPoint.Address.ToString() + ":" +
m_localEndPoint.Port + " due to not being able to send any RTP for " +
NO_RTP_TIMEOUT + "s.");
StopListening = true;
}
// Shutdown the socket even if there is still RTP but the stay alive limit has been exceeded.
if (RTPMaxStayAlive > 0 && DateTime.Now.Subtract(m_startRTPSendTime).TotalSeconds > RTPMaxStayAlive)
{
logger.Warn("Shutting down RTPSink due to passing RTPMaxStayAlive time.");
Shutdown();
StopListening = true;
}
#endregion
}
}
catch (Exception excp)
{
logger.Error("Exception Send RTPSink: " + excp.Message);
}
finally
{
#region Shut down socket.
Shutdown();
if (SenderClosed != null)
{
try
{
SenderClosed(m_streamId, m_callDescriptorId);
}
catch (Exception excp)
{
logger.Error("Exception RTPSink SenderClosed. " + 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;
ushort sequenceNumber = 0;
ushort 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
}
}
/// <summary>
/// Sends a dynamically sized frame. The RTP marker bit will be set for the last transmitted packet in the frame.
/// </summary>
/// <param name="frame">The frame to transmit.</param>
/// <param name="frameSpacing">The increment to add to the RTP timestamp for each new frame.</param>
/// <param name="payloadType">The payload type to set on the RTP packet.</param>
public void SendVP8Frame(byte[] frame, uint frameSpacing, int payloadType)
{
try
{
if (!IsRunning)
{
logger.Warn("SendVP8Frame cannot be called on a closed RTP channel.");
}
else if (_rtpSocketError != SocketError.Success)
{
logger.Warn("SendVP8Frame was called for an RTP socket in an error state of " + _rtpSocketError +
".");
}
else
{
_timestamp = (_timestamp == 0)
? DateTimeToNptTimestamp32(DateTime.Now)
: (_timestamp + frameSpacing) % UInt32.MaxValue;
//System.Diagnostics.Debug.WriteLine("Sending " + frame.Length + " encoded bytes to client, timestamp " + _timestamp + ", starting sequence number " + _sequenceNumber + ".");
for (int index = 0; index *RTP_MAX_PAYLOAD < frame.Length; index++)
{
//byte[] vp8HeaderBytes = (index == 0) ? new byte[VP8_RTP_HEADER_LENGTH] { 0x90, 0x80, (byte)(_sequenceNumber % 128) } : new byte[VP8_RTP_HEADER_LENGTH] { 0x80, 0x80, (byte)(_sequenceNumber % 128) };
byte[] vp8HeaderBytes = (index == 0)
? new byte[VP8_RTP_HEADER_LENGTH] {
0x10
}
: new byte[VP8_RTP_HEADER_LENGTH] {
0x00
};
int offset = index * RTP_MAX_PAYLOAD;
int payloadLength = ((index + 1) * RTP_MAX_PAYLOAD < frame.Length)
? RTP_MAX_PAYLOAD
: frame.Length - index * RTP_MAX_PAYLOAD;
// RTPPacket rtpPacket = new RTPPacket(payloadLength + VP8_RTP_HEADER_LENGTH + ((_srtp != null) ? SRTP_SIGNATURE_LENGTH : 0));
RTPPacket rtpPacket = new RTPPacket(payloadLength + VP8_RTP_HEADER_LENGTH);
rtpPacket.Header.SyncSource = _syncSource;
rtpPacket.Header.SequenceNumber = _sequenceNumber++;
rtpPacket.Header.Timestamp = _timestamp;
rtpPacket.Header.MarkerBit = (offset + payloadLength >= frame.Length) ? 1 : 0;
rtpPacket.Header.PayloadType = payloadType;
Buffer.BlockCopy(vp8HeaderBytes, 0, rtpPacket.Payload, 0, vp8HeaderBytes.Length);
Buffer.BlockCopy(frame, offset, rtpPacket.Payload, vp8HeaderBytes.Length, payloadLength);
byte[] rtpBytes = rtpPacket.GetBytes();
//if (_srtp != null)
//{
// int rtperr = _srtp.ProtectRTP(rtpBytes, rtpBytes.Length - SRTP_SIGNATURE_LENGTH);
// if (rtperr != 0)
// {
// logger.Warn("An error was returned attempting to sign an SRTP packet for " + _remoteEndPoint + ", error code " + rtperr + ".");
// }
//}
//System.Diagnostics.Debug.WriteLine(" offset " + (index * RTP_MAX_PAYLOAD) + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header .MarkerBit + ".");
//Stopwatch sw = new Stopwatch();
//sw.Start();
_rtpSocket.SendTo(rtpBytes, rtpBytes.Length, SocketFlags.None, _remoteEP);
//sw.Stop();
//if (sw.ElapsedMilliseconds > 15)
//{
// logger.Warn(" SendVP8Frame offset " + offset + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header.MarkerBit + ", took " + sw.ElapsedMilliseconds + "ms.");
//}
}
}
}
catch (Exception excp)
{
if (IsRunning)
{
logger.Warn("Exception RTPChannel.SendVP8Frame attempting to send to the RTP socket at " +
_remoteEP + ". " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
}
}
}
/// <summary>
/// H264 frames need a two byte header when transmitted over RTP.
/// </summary>
/// <param name="frame">The H264 encoded frame to transmit.</param>
/// <param name="frameSpacing">The increment to add to the RTP timestamp for each new frame.</param>
/// <param name="payloadType">The payload type to set on the RTP packet.</param>
public void SendH264Frame(byte[] frame, uint frameSpacing, int payloadType)
{
try
{
if (!IsRunning)
{
logger.Warn("SendH264Frame cannot be called on a closed session.");
}
else if (_rtpSocketError != SocketError.Success)
{
logger.Warn("SendH264Frame was called for an RTP socket in an error state of " + _rtpSocketError +
".");
}
else
{
_timestamp = (_timestamp == 0)
? DateTimeToNptTimestamp32(DateTime.Now)
: (_timestamp + frameSpacing) % UInt32.MaxValue;
//System.Diagnostics.Debug.WriteLine("Sending " + frame.Length + " H264 encoded bytes to client, timestamp " + _timestamp + ", starting sequence number " + _sequenceNumber + ".");
for (int index = 0; index *RTP_MAX_PAYLOAD < frame.Length; index++)
{
uint offset = Convert.ToUInt32(index * RTP_MAX_PAYLOAD);
int payloadLength = ((index + 1) * RTP_MAX_PAYLOAD < frame.Length)
? RTP_MAX_PAYLOAD
: frame.Length - index * RTP_MAX_PAYLOAD;
RTPPacket rtpPacket = new RTPPacket(payloadLength + H264_RTP_HEADER_LENGTH);
rtpPacket.Header.SyncSource = _syncSource;
rtpPacket.Header.SequenceNumber = _sequenceNumber++;
rtpPacket.Header.Timestamp = _timestamp;
rtpPacket.Header.MarkerBit = 0;
rtpPacket.Header.PayloadType = payloadType;
// Start RTP packet in frame 0x1c 0x89
// Middle RTP packet in frame 0x1c 0x09
// Last RTP packet in frame 0x1c 0x49
byte[] h264Header = new byte[] { 0x1c, 0x09 };
if (index == 0 && frame.Length < RTP_MAX_PAYLOAD)
{
// First and last RTP packet in the frame.
h264Header = new byte[] { 0x1c, 0x49 };
rtpPacket.Header.MarkerBit = 1;
}
else if (index == 0)
{
h264Header = new byte[] { 0x1c, 0x89 };
}
else if ((index + 1) * RTP_MAX_PAYLOAD > frame.Length)
{
h264Header = new byte[] { 0x1c, 0x49 };
rtpPacket.Header.MarkerBit = 1;
}
var h264Stream = frame.Skip(index * RTP_MAX_PAYLOAD).Take(payloadLength).ToList();
h264Stream.InsertRange(0, h264Header);
rtpPacket.Payload = h264Stream.ToArray();
byte[] rtpBytes = rtpPacket.GetBytes();
//System.Diagnostics.Debug.WriteLine(" offset " + (index * RTP_MAX_PAYLOAD) + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header .MarkerBit + ".");
//Stopwatch sw = new Stopwatch();
//sw.Start();
_rtpSocket.SendTo(rtpBytes, rtpBytes.Length, SocketFlags.None, _remoteEP);
//sw.Stop();
//if (sw.ElapsedMilliseconds > 15)
//{
// logger.Warn(" SendH264Frame offset " + offset + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header.MarkerBit + ", took " + sw.ElapsedMilliseconds + "ms.");
//}
}
}
}
catch (Exception excp)
{
if (IsRunning)
{
logger.Warn("Exception RTPChannel.SendH264Frame attempting to send to the RTP socket at " +
_remoteEP + ". " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
}
}
}
/// <summary>
/// Helper method to send a low quality JPEG image over RTP. This method supports a very abbreviated version of RFC 2435 "RTP Payload Format for JPEG-compressed Video".
/// It's intended as a quick convenient way to send something like a test pattern image over an RTSP connection. More than likely it won't be suitable when a high
/// quality image is required since the header used in this method does not support quantization tables.
/// </summary>
/// <param name="jpegBytes">The raw encoded bytes of teh JPEG image to transmit.</param>
/// <param name="jpegQuality">The encoder quality of the JPEG image.</param>
/// <param name="jpegWidth">The width of the JPEG image.</param>
/// <param name="jpegHeight">The height of the JPEG image.</param>
/// <param name="framesPerSecond">The rate at which the JPEG frames are being transmitted at. used to calculate the timestamp.</param>
public void SendJpegFrame(byte[] jpegBytes, int jpegQuality, int jpegWidth, int jpegHeight, int framesPerSecond)
{
try
{
if (!IsRunning)
{
logger.Warn("SendJpegFrame cannot be called on a closed session.");
}
else if (_rtpSocketError != SocketError.Success)
{
logger.Warn("SendJpegFrame was called for an RTP socket in an error state of " + _rtpSocketError +
".");
}
else
{
_timestamp = (_timestamp == 0)
? DateTimeToNptTimestamp32(DateTime.Now)
: (_timestamp + (uint)(RFC_2435_FREQUENCY_BASELINE / framesPerSecond)) % UInt32.MaxValue;
//System.Diagnostics.Debug.WriteLine("Sending " + jpegBytes.Length + " encoded bytes to client, timestamp " + _timestamp + ", starting sequence number " + _sequenceNumber + ", image dimensions " + jpegWidth + " x " + jpegHeight + ".");
for (int index = 0; index *RTP_MAX_PAYLOAD < jpegBytes.Length; index++)
{
uint offset = Convert.ToUInt32(index * RTP_MAX_PAYLOAD);
int payloadLength = ((index + 1) * RTP_MAX_PAYLOAD < jpegBytes.Length)
? RTP_MAX_PAYLOAD
: jpegBytes.Length - index * RTP_MAX_PAYLOAD;
byte[] jpegHeader = CreateLowQualityRtpJpegHeader(offset, jpegQuality, jpegWidth, jpegHeight);
List <byte> packetPayload = new List <byte>();
packetPayload.AddRange(jpegHeader);
packetPayload.AddRange(jpegBytes.Skip(index * RTP_MAX_PAYLOAD).Take(payloadLength));
RTPPacket rtpPacket = new RTPPacket(packetPayload.Count);
rtpPacket.Header.SyncSource = _syncSource;
rtpPacket.Header.SequenceNumber = _sequenceNumber++;
rtpPacket.Header.Timestamp = _timestamp;
rtpPacket.Header.MarkerBit = ((index + 1) * RTP_MAX_PAYLOAD < jpegBytes.Length) ? 0 : 1;
rtpPacket.Header.PayloadType = (int)SDPMediaFormatsEnum.JPEG;
rtpPacket.Payload = packetPayload.ToArray();
byte[] rtpBytes = rtpPacket.GetBytes();
//System.Diagnostics.Debug.WriteLine(" offset " + offset + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header.MarkerBit + ".");
//Stopwatch sw = new Stopwatch();
//sw.Start();
_rtpSocket.SendTo(rtpBytes, _remoteEP);
//sw.Stop();
//if (sw.ElapsedMilliseconds > 15)
//{
// logger.Warn(" SendJpegFrame offset " + offset + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header.MarkerBit + ", took " + sw.ElapsedMilliseconds + "ms.");
//}
}
//sw.Stop();
//System.Diagnostics.Debug.WriteLine("SendJpegFrame took " + sw.ElapsedMilliseconds + ".");
}
}
catch (Exception excp)
{
if (IsRunning)
{
logger.Warn("Exception RTPChannel.SendJpegFrame attempting to send to the RTP socket at " +
_remoteEP + ". " + excp);
//_rtpSocketError = SocketError.SocketError;
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
}
}
}
private void ProcessRTPPackets()
{
try
{
Thread.CurrentThread.Name = "rtspclient-rtp";
_lastRTPReceivedAt = DateTime.Now;
_lastBWCalcAt = DateTime.Now;
while (!_isClosed)
{
while (_rtspSession.HasRTPPacket())
{
RTPPacket rtpPacket = _rtspSession.GetNextRTPPacket();
if (rtpPacket != null)
{
_lastRTPReceivedAt = DateTime.Now;
_bytesSinceLastBWCalc += RTPHeader.MIN_HEADER_LEN + rtpPacket.Payload.Length;
if (_rtpTrackingAction != null)
{
double bwCalcSeconds = DateTime.Now.Subtract(_lastBWCalcAt).TotalSeconds;
if (bwCalcSeconds > BANDWIDTH_CALCULATION_SECONDS)
{
_lastBWCalc = _bytesSinceLastBWCalc * 8 / bwCalcSeconds;
_lastFrameRate = _framesSinceLastCalc / bwCalcSeconds;
_bytesSinceLastBWCalc = 0;
_framesSinceLastCalc = 0;
_lastBWCalcAt = DateTime.Now;
}
var abbrevURL = (_url.Length <= 50) ? _url : _url.Substring(0, 50);
string rtpTrackingText = String.Format("Url: {0}\r\nRcvd At: {1}\r\nSeq Num: {2}\r\nTS: {3}\r\nPayoad: {4}\r\nFrame Size: {5}\r\nBW: {6}\r\nFrame Rate: {7}", abbrevURL, DateTime.Now.ToString("HH:mm:ss:fff"), rtpPacket.Header.SequenceNumber, rtpPacket.Header.Timestamp, ((SDPMediaFormatsEnum)rtpPacket.Header.PayloadType).ToString(), _lastFrameSize + " bytes", _lastBWCalc.ToString("0.#") + "bps", _lastFrameRate.ToString("0.##") + "fps");
_rtpTrackingAction(rtpTrackingText);
}
if (rtpPacket.Header.Timestamp < _lastCompleteFrameTimestamp)
{
System.Diagnostics.Debug.WriteLine("Ignoring RTP packet with timestamp " + rtpPacket.Header.Timestamp + " as it's earlier than the last complete frame.");
}
else
{
while (_frames.Count > MAX_FRAMES_QUEUE_LENGTH)
{
var oldestFrame = _frames.OrderBy(x => x.Timestamp).First();
_frames.Remove(oldestFrame);
System.Diagnostics.Debug.WriteLine("Receive queue full, dropping oldest frame with timestamp " + oldestFrame.Timestamp + ".");
}
var frame = _frames.Where(x => x.Timestamp == rtpPacket.Header.Timestamp).SingleOrDefault();
if (frame == null)
{
frame = new RTPFrame() { Timestamp = rtpPacket.Header.Timestamp, HasMarker = rtpPacket.Header.MarkerBit == 1 };
frame.AddRTPPacket(rtpPacket);
_frames.Add(frame);
}
else
{
frame.HasMarker = (rtpPacket.Header.MarkerBit == 1);
frame.AddRTPPacket(rtpPacket);
}
if (frame.IsComplete)
{
// The frame is ready for handing over to the UI.
byte[] imageBytes = frame.GetFramePayload();
_lastFrameSize = imageBytes.Length;
_framesSinceLastCalc++;
_lastCompleteFrameTimestamp = rtpPacket.Header.Timestamp;
//System.Diagnostics.Debug.WriteLine("Frame ready " + frame.Timestamp + ", sequence numbers " + frame.StartSequenceNumber + " to " + frame.EndSequenceNumber + ", payload length " + imageBytes.Length + ".");
//logger.Debug("Frame ready " + frame.Timestamp + ", sequence numbers " + frame.StartSequenceNumber + " to " + frame.EndSequenceNumber + ", payload length " + imageBytes.Length + ".");
_frames.Remove(frame);
// Also remove any earlier frames as we don't care about anything that's earlier than the current complete frame.
foreach (var oldFrame in _frames.Where(x => x.Timestamp <= rtpPacket.Header.Timestamp).ToList())
{
System.Diagnostics.Debug.WriteLine("Discarding old frame for timestamp " + oldFrame.Timestamp + ".");
logger.Warn("Discarding old frame for timestamp " + oldFrame.Timestamp + ".");
_frames.Remove(oldFrame);
}
if (OnFrameReady != null)
{
try
{
//if (frame.FramePackets.Count == 1)
//{
// // REMOVE.
// logger.Warn("Discarding frame as there should have been more than 1 RTP packets.");
//}
//else
//{
//System.Diagnostics.Debug.WriteLine("RTP frame ready for timestamp " + frame.Timestamp + ".");
OnFrameReady(this, frame);
//}
}
catch (Exception frameReadyExcp)
{
logger.Error("Exception RTSPClient.ProcessRTPPackets OnFrameReady. " + frameReadyExcp);
}
}
}
}
}
}
if (DateTime.Now.Subtract(_lastRTPReceivedAt).TotalSeconds > RTP_TIMEOUT_SECONDS)
{
logger.Warn("No RTP packets were received on RTSP session " + _rtspSession.SessionID + " for " + RTP_TIMEOUT_SECONDS + ". The session will now be closed.");
Close();
}
else
{
Thread.Sleep(1);
}
}
}
catch (Exception excp)
{
logger.Error("Exception RTSPClient.ProcessRTPPackets. " + excp);
}
}