public void SendVp8Frame(uint timestamp, byte[] buffer, int streamID = 0)
{
if (m_isClosed || m_rtpEventInProgress || DestinationEndPoint == null)
{
return;
}
try
{
RTPSessionStream sessionStream = m_sessionStreams.Where(x => x.ID == streamID).Single();
for (int index = 0; index *RTP_MAX_PAYLOAD < buffer.Length; index++)
{
sessionStream.SeqNum = (ushort)(sessionStream.SeqNum % UInt16.MaxValue);
int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD);
int payloadLength = (offset + RTP_MAX_PAYLOAD < buffer.Length) ? RTP_MAX_PAYLOAD : buffer.Length - offset;
int srtpProtectionLength = (SrtpProtect != null) ? SRTP_AUTH_KEY_LENGTH : 0;
byte[] vp8HeaderBytes = (index == 0) ? new byte[] { 0x10 } : new byte[] { 0x00 };
RTPPacket rtpPacket = new RTPPacket(payloadLength + vp8HeaderBytes.Length + srtpProtectionLength);
rtpPacket.Header.SyncSource = sessionStream.Ssrc;
rtpPacket.Header.SequenceNumber = sessionStream.SeqNum++;
rtpPacket.Header.Timestamp = timestamp;
rtpPacket.Header.MarkerBit = ((offset + payloadLength) >= buffer.Length) ? 1 : 0; // Set marker bit for the last packet in the frame.
rtpPacket.Header.PayloadType = sessionStream.PayloadTypeID;
Buffer.BlockCopy(vp8HeaderBytes, 0, rtpPacket.Payload, 0, vp8HeaderBytes.Length);
Buffer.BlockCopy(buffer, offset, rtpPacket.Payload, vp8HeaderBytes.Length, payloadLength);
OnRtpPacketSent?.Invoke(rtpPacket);
var rtpBuffer = rtpPacket.GetBytes();
int rtperr = SrtpProtect == null ? 0 : SrtpProtect(rtpBuffer, rtpBuffer.Length - srtpProtectionLength);
if (rtperr != 0)
{
logger.LogError("SendVp8Frame SRTP packet protection failed, result " + rtperr + ".");
}
else
{
RtpChannel.SendAsync(RTPChannelSocketsEnum.RTP, DestinationEndPoint, rtpBuffer);
}
m_lastRtpTimestamp = timestamp;
}
}
catch (SocketException sockExcp)
{
logger.LogError("SocketException SendVp8Frame. " + sockExcp.Message);
}
}
public void SendAudioFrame(uint timestamp, byte[] buffer, int streamID = 0)
{
if (m_isClosed || m_rtpEventInProgress || DestinationEndPoint == null)
{
return;
}
try
{
RTPSessionStream sessionStream = m_sessionStreams.Where(x => x.ID == streamID).Single();
for (int index = 0; index *RTP_MAX_PAYLOAD < buffer.Length; index++)
{
sessionStream.SeqNum = (ushort)(sessionStream.SeqNum % UInt16.MaxValue);
int offset = (index == 0) ? 0 : (index * RTP_MAX_PAYLOAD);
int payloadLength = (offset + RTP_MAX_PAYLOAD < buffer.Length) ? RTP_MAX_PAYLOAD : buffer.Length - offset;
int srtpProtectionLength = (SrtpProtect != null) ? SRTP_AUTH_KEY_LENGTH : 0;
RTPPacket rtpPacket = new RTPPacket(payloadLength + srtpProtectionLength);
rtpPacket.Header.SyncSource = sessionStream.Ssrc;
rtpPacket.Header.SequenceNumber = sessionStream.SeqNum++;
rtpPacket.Header.Timestamp = timestamp;
// RFC3551 specifies that for audio the marker bit should always be 0 except for when returning
// from silence suppression. For video the marker bit DOES get set to 1 for the last packet
// in a frame.
rtpPacket.Header.MarkerBit = 0;
rtpPacket.Header.PayloadType = sessionStream.PayloadTypeID;
Buffer.BlockCopy(buffer, offset, rtpPacket.Payload, 0, payloadLength);
OnRtpPacketSent?.Invoke(rtpPacket);
var rtpBuffer = rtpPacket.GetBytes();
int rtperr = SrtpProtect == null ? 0 : SrtpProtect(rtpBuffer, rtpBuffer.Length - srtpProtectionLength);
if (rtperr != 0)
{
logger.LogError("SendAudioFrame SRTP packet protection failed, result " + rtperr + ".");
}
else
{
RtpChannel.SendAsync(RTPChannelSocketsEnum.RTP, DestinationEndPoint, rtpBuffer);
}
m_lastRtpTimestamp = timestamp;
}
}
catch (SocketException sockExcp)
{
logger.LogError("SocketException SendAudioFrame. " + sockExcp.Message);
}
}
/// <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>
/// <param name="framesPerSecond">The rate at which the JPEG frames are being transmitted at. used to calculate the timestamp.</param>
public void SendJpegFrame(uint timestamp, byte[] jpegBytes, int jpegQuality, int jpegWidth, int jpegHeight, int streamID = 0)
{
if (m_isClosed || m_rtpEventInProgress || DestinationEndPoint == null)
{
return;
}
try
{
//System.Diagnostics.Debug.WriteLine("Sending " + jpegBytes.Length + " encoded bytes to client, timestamp " + _timestamp + ", starting sequence number " + _sequenceNumber + ", image dimensions " + jpegWidth + " x " + jpegHeight + ".");
RTPSessionStream sessionStream = m_sessionStreams.Where(x => x.ID == streamID).Single();
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 = sessionStream.Ssrc;
rtpPacket.Header.SequenceNumber = sessionStream.SeqNum++;
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();
OnRtpPacketSent?.Invoke(rtpPacket);
byte[] rtpBytes = rtpPacket.GetBytes();
RtpChannel.SendAsync(RTPChannelSocketsEnum.RTP, DestinationEndPoint, rtpBytes);
}
}
catch (SocketException sockExcp)
{
logger.LogError("SocketException SendJpegFrame. " + sockExcp.Message);
}
}
/// <summary>
/// Sends the RTCP report to the remote call party.
/// </summary>
/// <param name="report">RTCP report to send.</param>
private void SendRtcpReport(SDPMediaTypesEnum mediaType, RTCPCompoundPacket report)
{
if (IsSecure && !IsSecureContextReady)
{
logger.LogWarning("SendRtcpReport cannot be called on a secure session before calling SetSecurityContext.");
}
else if (ControlDestinationEndPoint != null)
{
var reportBytes = report.GetBytes();
//logger.LogDebug($"SendRtcpReport: {reportBytes.HexStr()}");
var sendOnSocket = (m_isRtcpMultiplexed) ? RTPChannelSocketsEnum.RTP : RTPChannelSocketsEnum.Control;
if (m_srtcpControlProtect == null)
{
RtpChannel.SendAsync(sendOnSocket, ControlDestinationEndPoint, reportBytes);
}
else
{
byte[] sendBuffer = new byte[reportBytes.Length + SRTP_MAX_PREFIX_LENGTH];
Buffer.BlockCopy(reportBytes, 0, sendBuffer, 0, reportBytes.Length);
int outBufLen = 0;
int rtperr = m_srtcpControlProtect(sendBuffer, sendBuffer.Length - SRTP_MAX_PREFIX_LENGTH, out outBufLen);
if (rtperr != 0)
{
logger.LogWarning("SRTP RTCP packet protection failed, result " + rtperr + ".");
}
else
{
RtpChannel.SendAsync(sendOnSocket, ControlDestinationEndPoint, sendBuffer.Take(outBufLen).ToArray());
}
}
OnSendReport?.Invoke(mediaType, report);
}
}
/// <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(uint timestamp, byte[] frame, uint frameSpacing, int payloadType)
{
if (m_rtpEventInProgress || DestinationEndPoint == null)
{
return;
}
try
{
//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 = Ssrc;
rtpPacket.Header.SequenceNumber = SeqNum++;
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();
OnRtpPacketSent?.Invoke(rtpPacket);
byte[] rtpBytes = rtpPacket.GetBytes();
RtpChannel.SendAsync(RTPChannelSocketsEnum.RTP, DestinationEndPoint, rtpBytes);
}
}
catch (SocketException sockExcp)
{
logger.LogError("SocketException SendH264Frame. " + sockExcp.Message);
}
}