public void Send(byte[] buffer)
{
try
{
Peer.LastTimestamp = (Peer.LastTimestamp == 0) ? RTSPSession.DateTimeToNptTimestamp32(DateTime.Now) : Peer.LastTimestamp + TIMESTAMP_SPACING;
for (int index = 0; index * RTP_MAX_PAYLOAD < buffer.Length; index++)
{
int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD);
int payloadLength = (offset + RTP_MAX_PAYLOAD < buffer.Length) ? RTP_MAX_PAYLOAD : buffer.Length - offset;
byte[] vp8HeaderBytes = (index == 0) ? new byte[] { 0x10 } : new byte[] { 0x00 };
RTPPacket rtpPacket = new RTPPacket(payloadLength + SRTP_AUTH_KEY_LENGTH + vp8HeaderBytes.Length);
rtpPacket.Header.SyncSource = Peer.SSRC;
rtpPacket.Header.SequenceNumber = Peer.SequenceNumber++;
rtpPacket.Header.Timestamp = Peer.LastTimestamp;
rtpPacket.Header.MarkerBit = ((offset + payloadLength) >= buffer.Length) ? 1 : 0; // Set marker bit for the last packet in the frame.
rtpPacket.Header.PayloadType = PAYLOAD_TYPE_ID;
Buffer.BlockCopy(vp8HeaderBytes, 0, rtpPacket.Payload, 0, vp8HeaderBytes.Length);
Buffer.BlockCopy(buffer, offset, rtpPacket.Payload, vp8HeaderBytes.Length, payloadLength);
var rtpBuffer = rtpPacket.GetBytes();
int rtperr = SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Warn("SRTP packet protection failed, result " + rtperr + ".");
}
else
{
var connectedIceCandidate = Peer.LocalIceCandidates.Where(y => y.RemoteRtpEndPoint != null).First();
connectedIceCandidate.LocalRtpSocket.SendTo(rtpBuffer, connectedIceCandidate.RemoteRtpEndPoint);
}
}
}
catch (Exception sendExcp)
{
// logger.Error("SendRTP exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
public void Send(byte[] buffer, uint samplePeriod)
{
if (m_remoteEndPoint == null)
{
logger.Warn("RTP packet could not be sent as remote end point has not yet been set.");
}
else
{
m_sendRTPHeader.SequenceNumber++;
m_sendRTPHeader.Timestamp += samplePeriod;
RTPPacket rtpPacket = new RTPPacket()
{
Header = m_sendRTPHeader,
Payload = buffer
};
logger.Debug("Sending RTP packet to " + m_remoteEndPoint + ", seq# " + rtpPacket.Header.SequenceNumber + ", timestamp " + rtpPacket.Header.Timestamp + ".");
byte[] rtpOut = rtpPacket.GetBytes();
m_rtpListener.Send(m_remoteEndPoint, rtpOut);
}
}
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.LogDebug("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.LogWarning("Discarding old frame for timestamp " + oldFrame.Timestamp + ".");
_frames.Remove(oldFrame);
}
if (OnFrameReady != null)
{
try
{
//if (frame.FramePackets.Count == 1)
//{
// // REMOVE.
// logger.LogWarning("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.LogError("Exception RTSPClient.ProcessRTPPackets OnFrameReady. " + frameReadyExcp);
}
}
}
}
}
}
if (DateTime.Now.Subtract(_lastRTPReceivedAt).TotalSeconds > RTP_TIMEOUT_SECONDS)
{
logger.LogWarning("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.LogError("Exception RTSPClient.ProcessRTPPackets. " + excp);
}
}
/// <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="payloadType">The payload type to set on the RTP packet.</param>
public void SendVP8Frame(byte[] frame, int payloadType)
{
try
{
if (_closed)
{
logger.Warn("SendVP8Frame cannot be called on a closed session.");
}
else if (_rtpSocketError != SocketError.Success)
{
logger.Warn("SendVP8Frame was called for an RTP socket in an error state of " + _rtpSocketError + ".");
}
else if (_remoteEndPoint == null)
{
logger.Warn("SendVP8Frame frame not sent as remote end point is not yet set.");
}
else
{
RecalculateTimestampStep();
_timestamp += _timestampStep;
//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) };
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 (RtpProtect != null)
{
rtpBytes = RtpProtect(rtpBytes);
}
//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, _remoteEndPoint);
//SocketAsyncEventArgs socketSendArgs = new SocketAsyncEventArgs();
//socketSendArgs.SetBuffer(rtpBytes, 0, rtpBytes.Length);
//socketSendArgs.RemoteEndPoint = _remoteEndPoint;
//_rtpSocket.SendToAsync(socketSendArgs);
_rtpSocket.BeginSendTo(rtpBytes, 0, rtpBytes.Length, SocketFlags.None, _remoteEndPoint, null, null);
//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 (!_closed)
{
logger.Warn("Exception RTSPSession.SendVP8Frame attempting to send to the RTP socket at " + _remoteEndPoint + ". " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected(_sessionID);
}
}
}
}
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
}
}
/// <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 (_isClosed)
{
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, _remoteEndPoint);
//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 (!_isClosed)
{
logger.Warn("Exception RTPChannel.SendVP8Frame attempting to send to the RTP socket at " + _remoteEndPoint + ". " + 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="payloadType">The payload type to set on the RTP packet.</param>
public void SendH264Frame(byte[] frame, int payloadType)
{
try
{
if (_closed)
{
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 if (_remoteEndPoint == null)
{
logger.Warn("SendH264Frame frame not sent as remote end point is not yet set.");
}
else
{
RecalculateTimestampStep();
_timestamp += _timestampStep;
//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, _remoteEndPoint);
//SocketAsyncEventArgs socketSendArgs = new SocketAsyncEventArgs();
//socketSendArgs.SetBuffer(rtpBytes, 0, rtpBytes.Length);
//socketSendArgs.RemoteEndPoint = _remoteEndPoint;
//_rtpSocket.SendToAsync(socketSendArgs);
_rtpSocket.BeginSendTo(rtpBytes, 0, rtpBytes.Length, SocketFlags.None, _remoteEndPoint, null, null);
//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 (!_closed)
{
logger.Warn("Exception RTSPSession.SendH264Frame attempting to send to the RTP socket at " + _remoteEndPoint + ". " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected(_sessionID);
}
}
}
}
private static void SendRTPFromRawRTPFile(string file)
{
try
{
StreamReader sr = new StreamReader(file);
List<string> samples = new List<string>();
while (!sr.EndOfStream)
{
samples.Add(sr.ReadLine());
}
sr.Close();
logger.Debug(samples.Count + " encoded samples loaded.");
//_newRTPReceiverSRTP = new SRTPManaged(Convert.FromBase64String(_sourceSRTPKey));
//_newRTPReceiverSRTP = new SRTPManaged();
int sampleIndex = 0;
while (true)
{
if (_webRTCClients.Count != 0)
{
var sampleItem = samples[sampleIndex];
string[] sampleFields = sampleItem.Split(',');
uint timestamp = Convert.ToUInt32(sampleFields[0]);
int markerBit = Convert.ToInt32(sampleFields[1]);
byte[] sample = Convert.FromBase64String(sampleFields[2]);
lock (_webRTCClients)
{
foreach (var client in _webRTCClients.Where(x => x.STUNExchangeComplete))
{
try
{
if (client.LastTimestamp == 0)
{
client.LastTimestamp = RTSPSession.DateTimeToNptTimestamp32(DateTime.Now);
}
//for (int index = 0; index * RTP_MAX_PAYLOAD < sample.Length; index++)
//{
// int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD) - 1;
// int payloadLength = (offset + RTP_MAX_PAYLOAD < sample.Length - 1) ? RTP_MAX_PAYLOAD : sample.Length - 1 - offset;
RTPPacket rtpPacket = new RTPPacket(sample.Length + SRTP_AUTH_KEY_LENGTH);
rtpPacket.Header.SyncSource = client.SSRC;
rtpPacket.Header.SequenceNumber = client.SequenceNumber++;
rtpPacket.Header.Timestamp = client.LastTimestamp;
rtpPacket.Header.MarkerBit = markerBit;
rtpPacket.Header.PayloadType = 100;
//if (offset + RTP_MAX_PAYLOAD > sample.Length - 1)
//{
// Last packet in the frame.
// rtpPacket.Header.MarkerBit = 1;
//}
Buffer.BlockCopy(sample, 0, rtpPacket.Payload, 0, sample.Length);
var rtpBuffer = rtpPacket.GetBytes();
int rtperr = client.SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Debug("New RTP packet protect result " + rtperr + ".");
}
//logger.Debug("Sending RTP " + sample.Length + " bytes to " + client.SocketAddress + ", timestamp " + client.LastTimestamp + ", marker " + rtpPacket.Header.MarkerBit + ".");
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, client.SocketAddress);
if (markerBit == 1)
{
client.LastTimestamp += TIMESTAMP_SPACING;
}
//}
}
catch (Exception sendExcp)
{
logger.Error("SendRTPFromFile exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
}
sampleIndex++;
if (sampleIndex >= samples.Count - 1)
{
sampleIndex = 0;
}
//Thread.Sleep(30);
}
}
}
catch (Exception excp)
{
Console.WriteLine("Exception SendRTPFromFile. " + excp);
}
}
private static void SendRTPFromRawRTPFileNewVP8Header(string file)
{
try
{
StreamReader sr = new StreamReader(file);
List<string> samples = new List<string>();
while (!sr.EndOfStream)
{
samples.Add(sr.ReadLine());
}
sr.Close();
logger.Debug(samples.Count + " encoded samples loaded.");
//_newRTPReceiverSRTP = new SRTPManaged(Convert.FromBase64String(_sourceSRTPKey));
//_newRTPReceiverSRTP = new SRTPManaged();
int sampleIndex = 0;
while (true)
{
if (_webRTCClients.Count != 0)
{
var sampleItem = samples[sampleIndex];
string[] sampleFields = sampleItem.Split(',');
//uint timestamp = Convert.ToUInt32(sampleFields[0]);
int markerBit = Convert.ToInt32(sampleFields[1]);
byte[] sample = Convert.FromBase64String(sampleFields[2]);
lock (_webRTCClients)
{
foreach (var client in _webRTCClients.Where(x => x.STUNExchangeComplete))
{
try
{
if (client.LastTimestamp == 0)
{
client.LastTimestamp = RTSPSession.DateTimeToNptTimestamp32(DateTime.Now);
}
RTPVP8Header origVP8Header = RTPVP8Header.GetVP8Header(sample);
if (origVP8Header.IsKeyFrame)
{
Console.WriteLine("Key frame");
}
RTPPacket rtpPacket = new RTPPacket(sample.Length + SRTP_AUTH_KEY_LENGTH);
rtpPacket.Header.SyncSource = client.SSRC;
rtpPacket.Header.SequenceNumber = client.SequenceNumber++;
rtpPacket.Header.Timestamp = client.LastTimestamp;
rtpPacket.Header.MarkerBit = markerBit;
rtpPacket.Header.PayloadType = 100;
if (origVP8Header.StartOfVP8Partition && markerBit == 1)
{
Console.WriteLine("My VP8 Header : " + BitConverter.ToString(origVP8Header.GetBytes()) + ".");
Console.WriteLine("Sample VP8 Header: " + BitConverter.ToString(sample, 0, 6) + ".");
Buffer.BlockCopy(origVP8Header.GetBytes(), 0, rtpPacket.Payload, 0, origVP8Header.Length);
Buffer.BlockCopy(sample, origVP8Header.Length, rtpPacket.Payload, origVP8Header.Length, sample.Length - origVP8Header.Length);
}
else
{
Buffer.BlockCopy(sample, 0, rtpPacket.Payload, 0, sample.Length);
}
var rtpBuffer = rtpPacket.GetBytes();
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH, _wiresharpEP);
int rtperr = client.SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Debug("New RTP packet protect result " + rtperr + ".");
}
logger.Debug("Sending RTP " + sample.Length + " bytes to " + client.SocketAddress + ", timestamp " + client.LastTimestamp + ", marker " + rtpPacket.Header.MarkerBit + ".");
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, client.SocketAddress);
if (markerBit == 1)
{
client.LastTimestamp += TIMESTAMP_SPACING;
}
//}
}
catch (Exception sendExcp)
{
logger.Error("SendRTPFromFile exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
}
sampleIndex++;
if (sampleIndex >= samples.Count - 1)
{
sampleIndex = 0;
}
//Thread.Sleep(30);
}
}
}
catch (Exception excp)
{
Console.WriteLine("Exception SendRTPFromFile. " + excp);
}
}
private static void RelayRTP(UdpClient rtpClient)
{
try
{
DateTime lastCleanup = DateTime.Now;
//_newRTPReceiverSRTP = new SRTPManaged(Convert.FromBase64String(_sourceSRTPKey));
//_newRTPReceiverSRTP = new SRTPManaged();
IPEndPoint remoteEndPoint = new IPEndPoint(IPAddress.Any, 0);
byte[] buffer = rtpClient.Receive(ref remoteEndPoint);
StreamWriter sw = new StreamWriter("rtpPackets.txt");
byte[] frame = new byte[1000000];
int framePosition = 0;
int sampleCount = 0;
DateTime lastReceiveTime = DateTime.Now;
while (buffer != null && buffer.Length > 0 && !m_exit)
{
int packetSpacingMilli = Convert.ToInt32(DateTime.Now.Subtract(lastReceiveTime).TotalMilliseconds);
Console.WriteLine("Packet spacing " + packetSpacingMilli + "ms.");
lastReceiveTime = DateTime.Now;
if (_webRTCClients.Count != 0)
{
RTPPacket triggerRTPPacket = new RTPPacket(buffer);
RTPVP8Header vp8Header = RTPVP8Header.GetVP8Header(triggerRTPPacket.Payload);
if (sampleCount < 1000)
{
sw.WriteLine(triggerRTPPacket.Header.Timestamp + "," + triggerRTPPacket.Header.MarkerBit + "," + Convert.ToBase64String(triggerRTPPacket.Payload));
//if (triggerRTPPacket.Header.MarkerBit == 1 && vp8Header.StartOfVP8Partition == true)
//{
// // This is a single packet frame.
// sw.WriteLine(Convert.ToBase64String(vp8Header.GetBytes()) + "," + Convert.ToBase64String(triggerRTPPacket.Payload, vp8Header.Length, triggerRTPPacket.Payload.Length - vp8Header.Length));
//}
//else if (vp8Header.StartOfVP8Partition == true)
//{
// // This is a first packet in a multi-packet frame.
// sw.Write(Convert.ToBase64String(vp8Header.GetBytes()) + ",");
// Buffer.BlockCopy(triggerRTPPacket.Payload, vp8Header.Length, frame, 0, triggerRTPPacket.Payload.Length - vp8Header.Length);
// framePosition = triggerRTPPacket.Payload.Length - vp8Header.Length;
//}
//else if (triggerRTPPacket.Header.MarkerBit == 1)
//{
// // This is the last continuation frame.
// Buffer.BlockCopy(triggerRTPPacket.Payload, vp8Header.Length, frame, framePosition, triggerRTPPacket.Payload.Length - vp8Header.Length);
// framePosition += triggerRTPPacket.Payload.Length - vp8Header.Length;
// sw.WriteLine(Convert.ToBase64String(frame, 0, framePosition));
// framePosition = 0;
//}
//else
//{
// // This is a middle continuation packet
// Buffer.BlockCopy(triggerRTPPacket.Payload, vp8Header.Length, frame, framePosition, triggerRTPPacket.Payload.Length - vp8Header.Length);
// framePosition += triggerRTPPacket.Payload.Length - vp8Header.Length;
//}
sampleCount++;
if (sampleCount == 1000)
{
Console.WriteLine("Sample collection complete.");
sw.Close();
}
}
lock (_webRTCClients)
{
foreach (var client in _webRTCClients.Where(x => x.STUNExchangeComplete))
{
try
{
if (client.LastTimestamp == 0)
{
client.LastTimestamp = RTSPSession.DateTimeToNptTimestamp32(DateTime.Now);
}
else if (vp8Header.StartOfVP8Partition)
{
client.LastTimestamp += 11520;
}
RTPPacket rtpPacket = new RTPPacket(triggerRTPPacket.Payload.Length + SRTP_AUTH_KEY_LENGTH);
rtpPacket.Header.SyncSource = client.SSRC;
rtpPacket.Header.SequenceNumber = client.SequenceNumber++;
rtpPacket.Header.Timestamp = client.LastTimestamp; //triggerRTPPacket.Header.Timestamp; // client.LastTimestamp;
rtpPacket.Header.MarkerBit = triggerRTPPacket.Header.MarkerBit;
rtpPacket.Header.PayloadType = 100;
Buffer.BlockCopy(triggerRTPPacket.Payload, 0, rtpPacket.Payload, 0, triggerRTPPacket.Payload.Length);
var rtpBuffer = rtpPacket.GetBytes();
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH, _wiresharpEP);
if (vp8Header.IsKeyFrame)
{
Console.WriteLine("key frame.");
}
//int rtperr = _newRTPReceiverSRTP.ProtectRTP(rtpBuffer, rtpBuffer.Length - 10);
int rtperr = client.SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Debug("New RTP packet protect result " + rtperr + ".");
}
logger.Debug("Sending RTP " + rtpBuffer.Length + " bytes to " + client.SocketAddress + ", timestamp " + rtpPacket.Header.Timestamp + ", trigger timestamp " + triggerRTPPacket.Header.Timestamp + ", marker bit " + rtpPacket.Header.MarkerBit + ".");
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, client.SocketAddress);
}
catch (Exception sendExcp)
{
logger.Error("RelayRTP exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
}
}
buffer = rtpClient.Receive(ref remoteEndPoint);
}
}
catch (Exception excp)
{
logger.Error("Exception RelayRTP. " + excp);
}
}
private static void SendRTPFromCamera()
{
try
{
unsafe
{
SIPSorceryMedia.MFVideoSampler videoSampler = new SIPSorceryMedia.MFVideoSampler();
//List<VideoMode> webcamModes = new List<VideoMode>();
//int deviceCount = videoSampler.GetVideoDevices(ref webcamModes);
//foreach (var videoMode in webcamModes)
//{
// Console.WriteLine(videoMode.DeviceFriendlyName + " " + (videoMode.VideoSubTypeFriendlyName ?? videoMode.VideoSubType.ToString()) + " " + videoMode.Width + "x" + videoMode.Height + ".");
//}
videoSampler.Init(_webcamIndex, _webcamVideoSubType, _webcamWidth, _webcamHeight);
SIPSorceryMedia.VPXEncoder vpxEncoder = new VPXEncoder();
vpxEncoder.InitEncoder(_webcamWidth, _webcamHeight);
SIPSorceryMedia.ImageConvert colorConverter = new ImageConvert();
byte pictureID = 0x1;
byte[] sampleBuffer = null;
byte[] encodedBuffer = new byte[4096];
while (true)
{
if (_webRTCClients.Any(x => x.STUNExchangeComplete == true && x.IsDtlsNegotiationComplete == true))
{
int result = videoSampler.GetSample(ref sampleBuffer);
if (result != 0)
{
Console.WriteLine("Video sampler returned a null sample.");
}
else
{
//Console.WriteLine("Got managed sample " + sample.Buffer.Length + ", is key frame " + sample.IsKeyFrame + ".");
fixed (byte* p = sampleBuffer)
{
byte[] convertedFrame = null;
colorConverter.ConvertToI420(p, _webcamVideoSubType, Convert.ToInt32(_webcamWidth), Convert.ToInt32(_webcamHeight), ref convertedFrame);
//int encodeResult = vpxEncoder.Encode(p, sampleBuffer.Length, 1, ref encodedBuffer);
fixed (byte* q = convertedFrame)
{
int encodeResult = vpxEncoder.Encode(q, sampleBuffer.Length, 1, ref encodedBuffer);
if (encodeResult != 0)
{
Console.WriteLine("VPX encode of video sample failed.");
continue;
}
}
}
lock (_webRTCClients)
{
foreach (var client in _webRTCClients.Where(x => x.STUNExchangeComplete && x.IsDtlsNegotiationComplete == true))
{
try
{
//if (client.LastRtcpSenderReportSentAt == DateTime.MinValue)
//{
// logger.Debug("Sending RTCP report to " + client.SocketAddress + ".");
// // Send RTCP report.
// RTCPPacket rtcp = new RTCPPacket(client.SSRC, 0, 0, 0, 0);
// byte[] rtcpBuffer = rtcp.GetBytes();
// _webRTCReceiverClient.BeginSend(rtcpBuffer, rtcpBuffer.Length, client.SocketAddress, null, null);
// //int rtperr = client.SrtpContext.ProtectRTP(rtcpBuffer, rtcpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
//}
//Console.WriteLine("Sending VP8 frame of " + encodedBuffer.Length + " bytes to " + client.SocketAddress + ".");
client.LastTimestamp = (client.LastTimestamp == 0) ? RTSPSession.DateTimeToNptTimestamp32(DateTime.Now) : client.LastTimestamp + TIMESTAMP_SPACING;
for (int index = 0; index * RTP_MAX_PAYLOAD < encodedBuffer.Length; index++)
{
int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD);
int payloadLength = (offset + RTP_MAX_PAYLOAD < encodedBuffer.Length) ? RTP_MAX_PAYLOAD : encodedBuffer.Length - offset;
byte[] vp8HeaderBytes = (index == 0) ? new byte[] { 0x10 } : new byte[] { 0x00 };
RTPPacket rtpPacket = new RTPPacket(payloadLength + SRTP_AUTH_KEY_LENGTH + vp8HeaderBytes.Length);
rtpPacket.Header.SyncSource = client.SSRC;
rtpPacket.Header.SequenceNumber = client.SequenceNumber++;
rtpPacket.Header.Timestamp = client.LastTimestamp;
rtpPacket.Header.MarkerBit = ((offset + payloadLength) >= encodedBuffer.Length) ? 1 : 0; // Set marker bit for the last packet in the frame.
rtpPacket.Header.PayloadType = PAYLOAD_TYPE_ID;
Buffer.BlockCopy(vp8HeaderBytes, 0, rtpPacket.Payload, 0, vp8HeaderBytes.Length);
Buffer.BlockCopy(encodedBuffer, offset, rtpPacket.Payload, vp8HeaderBytes.Length, payloadLength);
var rtpBuffer = rtpPacket.GetBytes();
//_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, _wiresharpEP);
int rtperr = client.SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Warn("SRTP packet protection failed, result " + rtperr + ".");
}
else
{
//logger.Debug("Sending RTP, offset " + offset + ", frame bytes " + payloadLength + ", vp8 header bytes " + vp8HeaderBytes.Length + ", timestamp " + rtpPacket.Header.Timestamp + ", seq # " + rtpPacket.Header.SequenceNumber + " to " + client.SocketAddress + ".");
//_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, client.SocketAddress);
_webRTCReceiverClient.BeginSend(rtpBuffer, rtpBuffer.Length, client.SocketAddress, null, null);
}
}
}
catch (Exception sendExcp)
{
logger.Error("SendRTP exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
}
pictureID++;
if (pictureID > 127)
{
pictureID = 1;
}
encodedBuffer = null;
sampleBuffer = null;
}
}
}
}
}
catch (Exception excp)
{
Console.WriteLine("Exception SendRTP. " + excp);
}
}
private static void SendRTPFromVP8FramesFile(string file)
{
try
{
StreamReader sr = new StreamReader(file);
List<string> samples = new List<string>();
while (!sr.EndOfStream)
{
string sample = sr.ReadLine();
samples.Add(sample);
//Console.WriteLine(sample);
//string[] sampleFields = sample.Split(',');
//RTPVP8Header frameVP8Header = RTPVP8Header.GetVP8Header(Convert.FromBase64String(sampleFields[0]));
//byte[] rtpPaylaod = Convert.FromBase64String(sampleFields[1]);
//Console.WriteLine((frameVP8Header.IsKeyFrame) ? "K" : "." + " " + frameVP8Header.FirstPartitionSize + " " + rtpPaylaod.Length + ".");
}
sr.Close();
logger.Debug(samples.Count + " encoded samples loaded.");
//_newRTPReceiverSRTP = new SRTPManaged(Convert.FromBase64String(_sourceSRTPKey));
//_newRTPReceiverSRTP = new SRTPManaged();
int sampleIndex = 0;
while (true)
{
if (_webRTCClients.Count != 0)
{
var sampleItem = samples[sampleIndex];
string[] sampleFields = sampleItem.Split(',');
RTPVP8Header frameVP8Header = RTPVP8Header.GetVP8Header(Convert.FromBase64String(sampleFields[0]));
byte[] sample = Convert.FromBase64String(sampleFields[1]);
if (frameVP8Header.IsKeyFrame)
{
Console.WriteLine("Key frame.");
}
lock (_webRTCClients)
{
foreach (var client in _webRTCClients.Where(x => x.STUNExchangeComplete))
{
try
{
if (client.LastTimestamp == 0)
{
client.LastTimestamp = RTSPSession.DateTimeToNptTimestamp32(DateTime.Now);
}
for (int index = 0; index * RTP_MAX_PAYLOAD < sample.Length; index++)
{
int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD);
int payloadLength = (offset + RTP_MAX_PAYLOAD < sample.Length) ? RTP_MAX_PAYLOAD : sample.Length - offset;
RTPVP8Header packetVP8Header = new RTPVP8Header()
{
ExtendedControlBitsPresent = true,
IsPictureIDPresent = true,
ShowFrame = true,
};
if (index == 0)
{
packetVP8Header.StartOfVP8Partition = true;
//packetVP8Header.FirstPartitionSize = frameVP8Header.FirstPartitionSize;
packetVP8Header.IsKeyFrame = frameVP8Header.IsKeyFrame;
packetVP8Header.PictureID = (frameVP8Header.IsKeyFrame) ? (byte)0x00 : frameVP8Header.PictureID;
}
byte[] vp8HeaderBytes = packetVP8Header.GetBytes();
RTPPacket rtpPacket = new RTPPacket(packetVP8Header.Length + payloadLength + SRTP_AUTH_KEY_LENGTH);
rtpPacket.Header.SyncSource = client.SSRC;
rtpPacket.Header.SequenceNumber = client.SequenceNumber++;
rtpPacket.Header.Timestamp = client.LastTimestamp;
rtpPacket.Header.MarkerBit = ((offset + payloadLength) >= sample.Length) ? 1 : 0;
rtpPacket.Header.PayloadType = 100;
Buffer.BlockCopy(vp8HeaderBytes, 0, rtpPacket.Payload, 0, packetVP8Header.Length);
Buffer.BlockCopy(sample, offset, rtpPacket.Payload, packetVP8Header.Length, payloadLength);
var rtpBuffer = rtpPacket.GetBytes();
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH, _wiresharpEP);
int rtperr = client.SrtpContext.ProtectRTP(rtpBuffer, rtpBuffer.Length - SRTP_AUTH_KEY_LENGTH);
if (rtperr != 0)
{
logger.Debug("New RTP packet protect result " + rtperr + ".");
}
logger.Debug("Sending RTP " + sample.Length + " bytes to " + client.SocketAddress + ", timestamp " + client.LastTimestamp + ", marker " + rtpPacket.Header.MarkerBit + ".");
_webRTCReceiverClient.Send(rtpBuffer, rtpBuffer.Length, client.SocketAddress);
}
client.LastTimestamp += TIMESTAMP_SPACING;
}
catch (Exception sendExcp)
{
logger.Error("SendRTPFromVP8FramesFile exception sending to " + client.SocketAddress + ". " + sendExcp.Message);
}
}
}
sampleIndex++;
if (sampleIndex >= samples.Count - 1)
{
sampleIndex = 0;
}
Thread.Sleep(30);
}
}
}
catch (Exception excp)
{
Console.WriteLine("Exception SendRTPFromVP8FramesFile. " + excp);
}
}
/// <summary>
/// Audio frames are generally contained within a single RTP packet. This method is a shortcut
/// to construct a frame from a single RTP packet.
/// </summary>
public static RTPFrame MakeSinglePacketFrame(RTPPacket rtpPacket)
{
RTPFrame frame = new RTPFrame();
frame.AddRTPPacket(rtpPacket);
frame.Timestamp = rtpPacket.Header.Timestamp;
return frame;
}
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 SendRTPPacket(string sourceSocket, string destinationSocket)
{
try
{
//logger.Debug("Attempting to send RTP packet from " + sourceSocket + " to " + destinationSocket + ".");
Log("Attempting to send RTP packet from " + sourceSocket + " to " + destinationSocket + ".");
IPEndPoint sourceEP = IPSocket.GetIPEndPoint(sourceSocket);
IPEndPoint destEP = IPSocket.GetIPEndPoint(destinationSocket);
RTPPacket rtpPacket = new RTPPacket(80);
rtpPacket.Header.SequenceNumber = (UInt16)6500;
rtpPacket.Header.Timestamp = 100000;
UDPPacket udpPacket = new UDPPacket(sourceEP.Port, destEP.Port, rtpPacket.GetBytes());
IPv4Header ipHeader = new IPv4Header(ProtocolType.Udp, Crypto.GetRandomInt(6), sourceEP.Address, destEP.Address);
IPv4Packet ipPacket = new IPv4Packet(ipHeader, udpPacket.GetBytes());
byte[] data = ipPacket.GetBytes();
Socket rawSocket = new Socket(AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.IP);
rawSocket.SetSocketOption(SocketOptionLevel.IP, SocketOptionName.HeaderIncluded, 1);
rawSocket.SendTo(data, destEP);
}
catch (Exception excp)
{
logger.Error("Exception SendRTPPacket. " + excp.Message);
}
}
private void Listen()
{
try
{
UdpClient udpSvr = m_udpListener;
if (udpSvr == null)
{
logger.LogError("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.LogDebug("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.LogWarning("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.LogDebug("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.LogError("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.LogError("Exception RTPSink DataReceived. " + excp.Message);
}
}
if (m_packetsReceived % 500 == 0)
{
logger.LogDebug("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.LogDebug("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.LogError("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.LogInformation("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.LogDebug("seqno=" + rtpPacket.Header.SequenceNumber + ", timestmap=" + timestamp + ", ts-prev=" + previousTimestamp + ", receive spacing=" + interarrivalReceiveTime + ", send spacing=" + senderSpacingMilliseconds + ", jitter=" + jitter);
}
else
{
//logger.LogDebug("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.LogDebug("RTPSink Listen SyncSource=" + rtpPacket.Header.SyncSource + ".");
}
previousTimestamp = timestamp;
}
catch (Exception excp)
{
logger.LogError("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.LogDebug("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.LogError("Exception RTPListener Changing Remote EndPoint. " + changeExcp.Message);
}
}
}
}
catch (Exception setSendExcp)
{
logger.LogError("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.LogWarning("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.LogWarning("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.LogError("Exception Listen RTPSink: " + excp.Message);
}
finally
{
#region Shut down socket.
Shutdown();
if (ListenerClosed != null)
{
try
{
ListenerClosed(m_streamId, m_callDescriptorId);
}
catch (Exception excp)
{
logger.LogError("Exception RTPSink ListenerClosed. " + excp.Message);
}
}
#endregion
}
}
/// <summary>
/// Sends an audio frame where the payload size is less than the maximum RTP packet payload size.
/// </summary>
/// <param name="payload">The audio payload 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 SendAudioFrame(byte[] payload, uint frameSpacing, int payloadType)
{
try
{
if (_isClosed)
{
logger.Warn("SendAudioFrame cannot be called on a closed RTP channel.");
}
else if (_rtpSocketError != SocketError.Success)
{
logger.Warn("SendAudioFrame was called for an RTP socket in an error state of " + _rtpSocketError + ".");
}
else
{
_timestamp = (_timestamp == 0) ? DateTimeToNptTimestamp32(DateTime.Now) : (_timestamp + frameSpacing) % UInt32.MaxValue;
RTPPacket rtpPacket = new RTPPacket(payload.Length);
rtpPacket.Header.SyncSource = _syncSource;
rtpPacket.Header.SequenceNumber = _sequenceNumber++;
rtpPacket.Header.Timestamp = _timestamp;
rtpPacket.Header.MarkerBit = 1;
rtpPacket.Header.PayloadType = payloadType;
Buffer.BlockCopy(payload, 0, rtpPacket.Payload, 0, payload.Length);
byte[] rtpBytes = rtpPacket.GetBytes();
//Stopwatch sw = new Stopwatch();
//sw.Start();
_rtpSocket.SendTo(rtpBytes, rtpBytes.Length, SocketFlags.None, _remoteEndPoint);
//sw.Stop();
//if (sw.ElapsedMilliseconds > 15)
//{
// logger.Warn(" SendAudioFrame offset " + offset + ", payload length " + payloadLength + ", sequence number " + rtpPacket.Header.SequenceNumber + ", marker " + rtpPacket.Header.MarkerBit + ", took " + sw.ElapsedMilliseconds + "ms.");
//}
}
}
catch (Exception excp)
{
if (!_isClosed)
{
logger.Warn("Exception RTPChannel.SendAudioFrame attempting to send to the RTP socket at " + _remoteEndPoint + ". " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
}
}
}
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.LogDebug("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.LogInformation("Sending RTP packet to " + dstEndPoint.Address + ":" + dstEndPoint.Port);
if (payloadSize != m_rtpPacketSendSize)
{
payloadSize = m_rtpPacketSendSize;
logger.LogInformation("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.LogDebug("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.LogError("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.LogDebug("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.LogDebug("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.LogDebug(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.LogDebug("Total packets sent to " + dstEndPoint.ToString() + " " + m_packetsSent + ", bytes " + NumberFormatter.ToSIByteFormat(m_bytesSent, 2) + ".");
}
//sendlogger.LogInformation(m_lastRTPSentTime.ToString("dd MMM yyyy HH:mm:ss:fff") + "," + m_lastRTPSentTime.Subtract(m_startRTPSendTime).TotalMilliseconds.ToString("0") + "," + rtpHeader.SequenceNumber + "," + rtpBytes.Length);
//sendlogger.LogInformation(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.LogError("Exception RTPSink DataSent. " + excp.Message);
}
}
lastSeqNum = rtpHeader.SequenceNumber;
if (rtpHeader.SequenceNumber == UInt16.MaxValue)
{
//logger.LogDebug("RTPSink looping the sequence number in sample.");
rtpHeader.SequenceNumber = 0;
}
else
{
rtpHeader.SequenceNumber++;
}
}
}
catch (Exception excp)
{
logger.LogError("Exception RTP Send. " + excp.GetType() + ". " + excp.Message);
if (excp.GetType() == typeof(SocketException))
{
logger.LogError("socket exception errorcode=" + ((SocketException)excp).ErrorCode + ".");
}
logger.LogWarning("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.LogWarning("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.LogWarning("Shutting down RTPSink due to passing RTPMaxStayAlive time.");
Shutdown();
StopListening = true;
}
#endregion
}
}
catch (Exception excp)
{
logger.LogError("Exception Send RTPSink: " + excp.Message);
}
finally
{
#region Shut down socket.
Shutdown();
if (SenderClosed != null)
{
try
{
SenderClosed(m_streamId, m_callDescriptorId);
}
catch (Exception excp)
{
logger.LogError("Exception RTPSink SenderClosed. " + excp.Message);
}
}
#endregion
}
}
private void RTPReceive()
{
try
{
Thread.CurrentThread.Name = "rtpchanrecv-" + _rtpPort;
byte[] buffer = new byte[2048];
while (!_isClosed)
{
try
{
int bytesRead = _rtpSocket.Receive(buffer);
if (bytesRead > 0)
{
_rtpSocket.SendTo(buffer, bytesRead, SocketFlags.None, _wiresharkEP);
_rtpLastActivityAt = DateTime.Now;
if (bytesRead > RTPHeader.MIN_HEADER_LEN)
{
if ((buffer[0] & 0x80) == 0)
{
#region STUN Packet.
if (_iceState != null)
{
try
{
STUNv2Message stunMessage = STUNv2Message.ParseSTUNMessage(buffer, bytesRead);
//logger.Debug("STUN message received from Receiver Client @ " + stunMessage.Header.MessageType + ".");
if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingRequest)
{
//logger.Debug("Sending STUN response to Receiver Client @ " + remoteEndPoint + ".");
STUNv2Message stunResponse = new STUNv2Message(STUNv2MessageTypesEnum.BindingSuccessResponse);
stunResponse.Header.TransactionId = stunMessage.Header.TransactionId;
stunResponse.AddXORMappedAddressAttribute(_remoteEndPoint.Address, _remoteEndPoint.Port);
byte[] stunRespBytes = stunResponse.ToByteBuffer(_iceState.SenderPassword, true);
_rtpSocket.SendTo(stunRespBytes, _remoteEndPoint);
//logger.Debug("Sending Binding request to Receiver Client @ " + remoteEndPoint + ".");
STUNv2Message stunRequest = new STUNv2Message(STUNv2MessageTypesEnum.BindingRequest);
stunRequest.Header.TransactionId = Guid.NewGuid().ToByteArray().Take(12).ToArray();
stunRequest.AddUsernameAttribute(_iceState.ReceiverUser + ":" + _iceState.SenderUser);
stunRequest.Attributes.Add(new STUNv2Attribute(STUNv2AttributeTypesEnum.Priority, new byte[] { 0x6e, 0x7f, 0x1e, 0xff }));
byte[] stunReqBytes = stunRequest.ToByteBuffer(_iceState.ReceiverPassword, true);
_rtpSocket.SendTo(stunReqBytes, _remoteEndPoint);
_iceState.LastSTUNMessageReceivedAt = DateTime.Now;
}
else if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingSuccessResponse)
{
if (!_iceState.IsSTUNExchangeComplete)
{
_iceState.IsSTUNExchangeComplete = true;
logger.Debug("WebRTC client STUN exchange complete for " + _remoteEndPoint.ToString() + ".");
}
}
else if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingErrorResponse)
{
logger.Warn("A STUN binding error response was received from " + _remoteEndPoint + ".");
}
else
{
logger.Warn("An unrecognised STUN request was received from " + _remoteEndPoint + ".");
}
}
catch (SocketException sockExcp)
{
logger.Debug("RTPChannel.RTPReceive STUN processing (" + _remoteEndPoint + "). " + sockExcp.Message);
continue;
}
catch (Exception stunExcp)
{
logger.Warn("Exception RTPChannel.RTPReceive STUN processing (" + _remoteEndPoint + "). " + stunExcp);
continue;
}
}
else
{
logger.Warn("A STUN reponse was received on RTP socket from " + _remoteEndPoint + " but no ICE state was set.");
}
#endregion
}
else
{
RTPPacket rtpPacket = new RTPPacket(buffer.Take(bytesRead).ToArray());
//System.Diagnostics.Debug.WriteLine("RTPReceive ssrc " + rtpPacket.Header.SyncSource + ", seq num " + rtpPacket.Header.SequenceNumber + ", timestamp " + rtpPacket.Header.Timestamp + ", marker " + rtpPacket.Header.MarkerBit + ".");
lock (_packets)
{
if (_packets.Count > RTP_PACKETS_MAX_QUEUE_LENGTH)
{
System.Diagnostics.Debug.WriteLine("RTPChannel.RTPReceive packets queue full, clearing.");
logger.Warn("RTPChannel.RTPReceive packets queue full, clearing.");
_packets.Clear();
if (OnRTPQueueFull != null)
{
OnRTPQueueFull();
}
}
else
{
_packets.Enqueue(rtpPacket);
}
}
}
}
}
else
{
logger.Warn("Zero bytes read from RTPChannel RTP socket connected to " + _remoteEndPoint + ".");
//break;
}
}
catch (SocketException sockExcp)
{
if (!_isClosed)
{
_rtpSocketError = sockExcp.SocketErrorCode;
if (_rtpSocketError == SocketError.Interrupted)
{
// If the receive has been interrupted it means the socket has been closed.
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
break;
}
else
{
throw;
}
}
}
catch (Exception excp)
{
if (!_isClosed)
{
logger.Error("Exception RTPChannel.RTPReceive receiving. " + excp);
}
}
}
}
catch (Exception excp)
{
if (!_isClosed)
{
logger.Error("Exception RTPChannel.RTPReceive. " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected();
}
}
}
}
/// <summary>
/// Audio frames are generally contained within a single RTP packet. This method is a shortcut
/// to construct a frame from a single RTP pakcet.
/// </summary>
public static RTPFrame MakeSinglePacketFrame(RTPPacket rtpPacket)
{
RTPFrame frame = new RTPFrame();
//frame.FramePayload = rtpPacket.Payload;
frame.Timestamp = rtpPacket.Header.Timestamp;
return frame;
}
/// <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 the 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>
public void SendJpegFrame(byte[] jpegBytes, int jpegQuality, int jpegWidth, int jpegHeight)
{
try
{
if (_closed)
{
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 if (_remoteEndPoint == null)
{
logger.Warn("SendJpegFrame frame not sent as remote end point is not yet set.");
}
else
{
//_timestamp = (_timestamp == 0) ? DateTimeToNptTimestamp32(DateTime.Now) : (_timestamp + (uint)(RFC_2435_FREQUENCY_BASELINE / DEFAULT_INITAL_FRAME_RATE)) % UInt32.MaxValue;
RecalculateTimestampStep();
_timestamp += _timestampStep;
//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, _remoteEndPoint);
//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 (!_closed)
{
logger.Warn("Exception RTPSession.SendJpegFrame attempting to send to the RTP socket at " + _remoteEndPoint + ". " + excp);
//_rtpSocketError = SocketError.SocketError;
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected(_sessionID);
}
}
}
}
public void AddRTPPacket(RTPPacket rtpPacket)
{
_packets.Add(rtpPacket);
//if (HasMarker && FramePayload == null)
//{
// FramePayload = IsComplete(_packets, payloadHeaderLength);
//}
}
private void RTPReceive()
{
try
{
Thread.CurrentThread.Name = "rtspsess-rtprecv";
byte[] buffer = new byte[2048];
while (!_closed)
{
try
{
EndPoint remoteEP = (EndPoint)new IPEndPoint(IPAddress.Any, 0);
//int bytesRead = _rtpSocket.Receive(buffer);
int bytesRead = _rtpSocket.ReceiveFrom(buffer, ref remoteEP);
IPEndPoint remoteIPEndPoint = remoteEP as IPEndPoint;
//logger.Debug("RTPReceive from " + remoteEP + ".");
//if (((IPEndPoint)remoteEP).Address.ToString() != _remoteEndPoint.Address.ToString())
//{
// var oldEndPoint = _remoteEndPoint;
// _remoteEndPoint = remoteEP as IPEndPoint;
// logger.Warn("RtspSession " + _sessionID + " switched to new remote endpoint at " + _remoteEndPoint + " (old end point " + oldEndPoint + ").");
//}
if (bytesRead > 0)
{
_rtpLastActivityAt = DateTime.Now;
if (bytesRead > RTPHeader.MIN_HEADER_LEN)
{
if ((buffer[0] >= 20) && (buffer[0] <= 64))
{
// DTLS.
if(OnDtlsReceive != null)
{
try
{
OnDtlsReceive(buffer, bytesRead, SendRTPRaw);
}
catch(Exception dtlsExcp)
{
logger.Error("Exception RTSPSession.RTPReceive DTLS. " + dtlsExcp);
}
}
else
{
logger.Warn("RTSPSession.RTPReceive received a DTLS packet from " + _remoteEndPoint + "but bo DTLS handler has been set.");
}
}
else if ((buffer[0] == 0) || (buffer[0] == 1))
{
// STUN.
if (_iceState != null)
{
try
{
STUNv2Message stunMessage = STUNv2Message.ParseSTUNMessage(buffer, bytesRead);
//logger.Debug("STUN message received from Receiver Client @ " + stunMessage.Header.MessageType + ".");
if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingRequest)
{
//logger.Debug("Sending STUN response to Receiver Client @ " + remoteEndPoint + ".");
STUNv2Message stunResponse = new STUNv2Message(STUNv2MessageTypesEnum.BindingSuccessResponse);
stunResponse.Header.TransactionId = stunMessage.Header.TransactionId;
stunResponse.AddXORMappedAddressAttribute(remoteIPEndPoint.Address, remoteIPEndPoint.Port);
byte[] stunRespBytes = stunResponse.ToByteBuffer(_iceState.SenderPassword, true);
_rtpSocket.SendTo(stunRespBytes, remoteIPEndPoint);
//logger.Debug("Sending Binding request to Receiver Client @ " + remoteEndPoint + ".");
STUNv2Message stunRequest = new STUNv2Message(STUNv2MessageTypesEnum.BindingRequest);
stunRequest.Header.TransactionId = Guid.NewGuid().ToByteArray().Take(12).ToArray();
stunRequest.AddUsernameAttribute(_iceState.ReceiverUser + ":" + _iceState.SenderUser);
stunRequest.Attributes.Add(new STUNv2Attribute(STUNv2AttributeTypesEnum.Priority, new byte[] { 0x6e, 0x7f, 0x1e, 0xff }));
stunRequest.Attributes.Add(new STUNv2Attribute(STUNv2AttributeTypesEnum.UseCandidate, null)); // Must send this to get DTLS started.
byte[] stunReqBytes = stunRequest.ToByteBuffer(_iceState.ReceiverPassword, true);
_rtpSocket.SendTo(stunReqBytes, remoteIPEndPoint);
_iceState.LastSTUNMessageReceivedAt = DateTime.Now;
}
else if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingSuccessResponse)
{
if (!_iceState.IsSTUNExchangeComplete)
{
_iceState.IsSTUNExchangeComplete = true;
logger.Debug("WebRTC client STUN exchange complete for " + remoteIPEndPoint.ToString() + " and ICE ufrag " + _iceState.ReceiverUser + ".");
_remoteEndPoint = remoteIPEndPoint;
}
}
else if (stunMessage.Header.MessageType == STUNv2MessageTypesEnum.BindingErrorResponse)
{
//logger.Warn("A STUN binding error response was received from " + remoteIPEndPoint + ".");
}
else
{
//logger.Warn("An unrecognised STUN request was received from " + remoteIPEndPoint + ".");
}
}
catch (SocketException sockExcp)
{
logger.Debug("RTPSession.RTPReceive STUN processing (" + remoteIPEndPoint + "). " + sockExcp.Message);
continue;
}
catch (Exception stunExcp)
{
logger.Warn("Exception RTPSession.RTPReceive STUN processing (" + remoteIPEndPoint + "). " + stunExcp);
continue;
}
}
else
{
//logger.Warn("A STUN reponse was received on RTP socket from " + remoteIPEndPoint + " but no ICE state was set.");
}
}
else if ((buffer[0] >= 128) && (buffer[0] <= 191))
{
if (buffer[1] == 0xC8 /* RTCP SR */ || buffer[1] == 0xC9 /* RTCP RR */)
{
// RTCP packet.
}
else
{
// RTP Packet.
RTPPacket rtpPacket = new RTPPacket(buffer.Take(bytesRead).ToArray());
//System.Diagnostics.Debug.WriteLine("RTPReceive ssrc " + rtpPacket.Header.SyncSource + ", seq num " + rtpPacket.Header.SequenceNumber + ", timestamp " + rtpPacket.Header.Timestamp + ", marker " + rtpPacket.Header.MarkerBit + ".");
//logger.Debug("RTPReceive remote " + remoteIPEndPoint + ", ssrc " + rtpPacket.Header.SyncSource + ", seq num " + rtpPacket.Header.SequenceNumber + ", timestamp " + rtpPacket.Header.Timestamp + ", bytes " + bytesRead + ", marker " + rtpPacket.Header.MarkerBit + ".");
lock (_packets)
{
if (_packets.Count > RTP_PACKETS_MAX_QUEUE_LENGTH)
{
System.Diagnostics.Debug.WriteLine("RTSPSession.RTPReceive packets queue full, clearing.");
logger.Warn("RTSPSession.RTPReceive packets queue full, clearing.");
_packets.Clear();
if (OnRTPQueueFull != null)
{
OnRTPQueueFull();
}
}
else
{
_packets.Enqueue(rtpPacket);
}
}
}
}
}
else
{
logger.Warn("RTSPSession.RTPReceive an unrecognised packet was received for session ID " + SessionID + " and " + remoteIPEndPoint + ".");
}
}
else
{
logger.Warn("Zero bytes read from RTSPSession RTP socket for session ID " + SessionID + " and " + remoteIPEndPoint + ".");
break;
}
}
catch (SocketException sockExcp)
{
if (!_closed)
{
_rtpSocketError = sockExcp.SocketErrorCode;
if (_rtpSocketError == SocketError.Interrupted)
{
// If the receive has been interrupted it means the socket has been closed most likely as a result of an RTSP TEARDOWN request.
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected(_sessionID);
}
break;
}
else
{
throw;
}
}
}
catch (Exception excp)
{
if (!_closed)
{
logger.Error("Exception RTSPSession.RTPReceive receiving. " + excp);
}
}
}
}
catch (Exception excp)
{
if (!_closed)
{
logger.Error("Exception RTSPSession.RTPReceive. " + excp);
if (OnRTPSocketDisconnected != null)
{
OnRTPSocketDisconnected(_sessionID);
}
}
}
}
public void AddRTPPacket(RTPPacket rtpPacket)
{
_packets.Add(rtpPacket);
}
