// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void RtspMessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
RtspResponse message = e.Message as RtspResponse;
if (message.OriginalRequest == null)
{
return;
}
Logger.Info("Received " + message.OriginalRequest.ToString());
if (message.OriginalRequest is RtspRequestOptions)
{
ProcessOptionsRequest(message);
}
else if (message.OriginalRequest is RtspRequestDescribe)
{
ProcessDescribeRequest(message);
}
else if (message.OriginalRequest is RtspRequestSetup)
{
ProcessSetupRequest(message);
}
else if (message.OriginalRequest is RtspRequestPlay)
{
ProcessPlayRequest(message);
}
}
public void RtpDataReceived(object sender, Rtsp.RtspChunkEventArgs chunkEventArgs)
{
try
{
RtspData rtpData = chunkEventArgs.Message as RtspData;
if (IsRtcpGoodbye(rtpData))
{
HandleRtcpGoodbye();
}
// Check which channel the Data was received on.
// eg the Video Channel, the Video Control Channel (RTCP)
// In the future would also check the Audio Channel and Audio Control Channel
if (rtpData.Channel == videoRTCPChannel)
{
Logger.Info("Received a RTCP message on channel " + rtpData.Channel);
return;
}
if (rtpData.Channel == videoDataChannel)
{
ProcessRTPVideo(chunkEventArgs);
}
}
catch (Exception e)
{
Logger.Info(e.ToString());
rtspErrorSubject.OnNext(e.Message);
}
}
/// <summary>
/// Raises the <see cref="E:DataReceived"/> event.
/// </summary>
/// <param name="rtspChunkEventArgs">The <see cref="Rtsp.RtspChunkEventArgs"/> instance containing the event data.</param>
protected void OnDataReceived(Rtsp.RtspChunkEventArgs rtspChunkEventArgs)
{
EventHandler <Rtsp.RtspChunkEventArgs> handler = DataReceived;
if (handler != null)
{
handler(this, rtspChunkEventArgs);
}
}
private void ProcessRTPVideo(Rtsp.RtspChunkEventArgs e)
{
// Received some Video Data on the correct channel.
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
int rtpVersion = (e.Message.Data[0] >> 6);
int rtpPadding = (e.Message.Data[0] >> 5) & 0x01;
int rtpRxtension = (e.Message.Data[0] >> 4) & 0x01;
int rtpCSRCCount = (e.Message.Data[0] >> 0) & 0x0F;
int rtpMarker = (e.Message.Data[1] >> 7) & 0x01;
int rtpPayloadType = (e.Message.Data[1] >> 0) & 0x7F;
uint rtpSequenceNumber = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtpTimestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtpSSRC = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtpPayloadStart = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtpCSRCCount); // zero or more csrcs
if (rtpRxtension == 1)
{
uint rtpExtensionId = ((uint)e.Message.Data[rtpPayloadStart + 0] << 8) + (uint)(e.Message.Data[rtpPayloadStart + 1] << 0);
uint rtpExtensionSize = ((uint)e.Message.Data[rtpPayloadStart + 2] << 8) + (uint)(e.Message.Data[rtpPayloadStart + 3] << 0) * 4; // units of extension_size is 4-bytes
rtpPayloadStart += 4 + (int)rtpExtensionSize; // extension header and extension payload
}
//Logger.Info("RTP Data"
// + " V=" + rtpVersion
// + " P=" + rtpPadding
// + " X=" + rtpRxtension
// + " CC=" + rtpCSRCCount
// + " M=" + rtpMarker
// + " PT=" + rtpPayloadType
// + " Seq=" + rtpSequenceNumber
// + " Time (MS)=" + rtpTimestamp / 90 // convert from 90kHZ clock to ms
// + " SSRC=" + rtpSSRC
// + " Size=" + e.Message.Data.Length);
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (videoPayloadType > 0 && rtpPayloadType != videoPayloadType)
{
Logger.Info("Ignoring this RTP payload: " + rtpPayloadType);
return; // ignore this data
}
byte[] rtp_payload = new byte[e.Message.Data.Length - rtpPayloadStart]; // payload with RTP header removed
Array.Copy(e.Message.Data, rtpPayloadStart, rtp_payload, 0, rtp_payload.Length); // copy payload
chunkReadySubject.OnNext(rtp_payload);
}
public void RtpDataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
RtspData rtpData = e.Message as RtspData;
// Check which channel the Data was received on.
// eg the Video Channel, the Video Control Channel (RTCP)
// In the future would also check the Audio Channel and Audio Control Channel
if (rtpData.Channel == videoRTCPChannel)
{
Logger.Info("Received a RTCP message on channel " + rtpData.Channel);
return;
}
if (rtpData.Channel == videoDataChannel)
{
ProcessRTPVideo(e);
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void RtspMessageReceived(object sender, Rtsp.RtspChunkEventArgs chunkEventArgs)
{
RtspResponse message = chunkEventArgs.Message as RtspResponse;
if (message?.OriginalRequest == null)
{
return;
}
Logger.Info("Received " + message.OriginalRequest);
try
{
if (message.OriginalRequest is RtspRequestOptions)
{
ProcessOptionsRequest(message);
}
else if (message.OriginalRequest is RtspRequestDescribe)
{
ProcessDescribeRequest(message);
}
else if (message.OriginalRequest is RtspRequestSetup)
{
ProcessSetupRequest(message);
}
else if (message.OriginalRequest is RtspRequestPlay)
{
ProcessPlayRequest(message);
}
}
catch (Exception e)
{
Logger.Info(e.ToString());
rtspErrorSubject.OnNext("RTSP error occured.");
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header. If so we update the 'realm' and 'nonce'
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
// Parse www_authenticate
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
// Split on the = symbol and load in
string[] parts = item.Split(new char[] { '=' });
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
Console.WriteLine("WWW Authorize parsed for " + realm + " " + nonce);
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we will not have the authorization Nonce so we
// handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the request did not use Authorizarion.
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "DESCRIBE");
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (digest_authorization != null)
{
describe_message.Headers.Add(RtspHeaderNames.Authorization, digest_authorization);
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
int data_channel = 0;
int rtcp_channel = 0;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = 0;
video_rtcp_channel = 1;
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = 2;
audio_rtcp_channel = 3;
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(data_channel, rtcp_channel), // Eg Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(data_channel, rtcp_channel), // a Channel for data (video or audio). a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Add authorization (if there is a username and password)
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "SETUP");
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (digest_authorization != null)
{
setup_message.Headers.Add("Authorization", digest_authorization);
}
rtsp_client.SendMessage(setup_message);
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
int rtp_count = 0; // used for statistics
// RTP packet (or RTCP packet) has been received.
public void Rtp_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspData data_received = e.Message as Rtsp.Messages.RtspData;
// Check which channel the Data was received on.
// eg the Video Channel, the Video Control Channel (RTCP)
// the Audio Channel or the Audio Control Channel (RTCP)
if (data_received.Channel == video_rtcp_channel || data_received.Channel == audio_rtcp_channel)
{
Console.WriteLine("Received a RTCP message on channel " + data_received.Channel);
// RTCP Packet
// - Version, Padding and Receiver Report Count
// - Packet Type
// - Length
// - SSRC
// - payload
// There can be multiple RTCP packets transmitted together. Loop ever each one
long packetIndex = 0;
while (packetIndex < e.Message.Data.Length)
{
int rtcp_version = (e.Message.Data[packetIndex + 0] >> 6);
int rtcp_padding = (e.Message.Data[packetIndex + 0] >> 5) & 0x01;
int rtcp_reception_report_count = (e.Message.Data[packetIndex + 0] & 0x1F);
byte rtcp_packet_type = e.Message.Data[packetIndex + 1]; // Values from 200 to 207
uint rtcp_length = ((uint)e.Message.Data[packetIndex + 2] << 8) + (uint)(e.Message.Data[packetIndex + 3]); // number of 32 bit words
uint rtcp_ssrc = ((uint)e.Message.Data[packetIndex + 4] << 24) + (uint)(e.Message.Data[packetIndex + 5] << 16)
+ (uint)(e.Message.Data[packetIndex + 6] << 8) + (uint)(e.Message.Data[packetIndex + 7]);
// 200 = SR = Sender Report
// 201 = RR = Receiver Report
// 202 = SDES = Source Description
// 203 = Bye = Goodbye
// 204 = APP = Application Specific Method
// 207 = XR = Extended Reports
Console.WriteLine("RTCP Data. PacketType=" + rtcp_packet_type
+ " SSRC=" + rtcp_ssrc);
if (rtcp_packet_type == 200)
{
// Send a Receiver Report
try
{
byte[] rtcp_receiver_report = new byte[8];
int version = 2;
int paddingBit = 0;
int reportCount = 0; // an empty report
int packetType = 201; // Receiver Report
int length = (rtcp_receiver_report.Length / 4) - 1; // num 32 bit words minus 1
rtcp_receiver_report[0] = (byte)((version << 6) + (paddingBit << 5) + reportCount);
rtcp_receiver_report[1] = (byte)(packetType);
rtcp_receiver_report[2] = (byte)((length >> 8) & 0xFF);
rtcp_receiver_report[3] = (byte)((length >> 0) & 0XFF);
rtcp_receiver_report[4] = (byte)((ssrc >> 24) & 0xFF);
rtcp_receiver_report[5] = (byte)((ssrc >> 16) & 0xFF);
rtcp_receiver_report[6] = (byte)((ssrc >> 8) & 0xFF);
rtcp_receiver_report[7] = (byte)((ssrc >> 0) & 0xFF);
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Send it over via the RTSP connection
rtsp_client.SendData(video_rtcp_channel, rtcp_receiver_report);
}
if (rtp_transport == RTP_TRANSPORT.UDP || rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Send it via a UDP Packet
Console.WriteLine("TODO - Need to implement RTCP over UDP");
}
}
catch
{
Console.WriteLine("Error writing RTCP packet");
}
}
packetIndex = packetIndex + ((rtcp_length + 1) * 4);
}
return;
}
if (data_received.Channel == video_data_channel || data_received.Channel == audio_data_channel)
{
// Received some Video or Audio Data on the correct channel.
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
int rtp_version = (e.Message.Data[0] >> 6);
int rtp_padding = (e.Message.Data[0] >> 5) & 0x01;
int rtp_extension = (e.Message.Data[0] >> 4) & 0x01;
int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F;
int rtp_marker = (e.Message.Data[1] >> 7) & 0x01;
int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F;
uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtp_payload_start = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtp_csrc_count); // zero or more csrcs
uint rtp_extension_id = 0;
uint rtp_extension_size = 0;
if (rtp_extension == 1)
{
rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0);
rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0) * 4; // units of extension_size is 4-bytes
rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload
}
Console.WriteLine("RTP Data"
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
+ " Seq=" + rtp_sequence_number
+ " Time (MS)=" + rtp_timestamp / 90 // convert from 90kHZ clock to ms
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (data_received.Channel == video_data_channel && rtp_payload_type != video_payload)
{
Console.WriteLine("Ignoring this Video RTP payload");
return; // ignore this data
}
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
else if (data_received.Channel == audio_data_channel && rtp_payload_type != audio_payload)
{
Console.WriteLine("Ignoring this Audio RTP payload");
return; // ignore this data
}
else if (data_received.Channel == video_data_channel &&
rtp_payload_type >= 96 && rtp_payload_type <= 127 &&
video_codec.Equals("H264"))
{
// H264 RTP Packet
// If rtp_marker is '1' then this is the final transmission for this packet.
// If rtp_marker is '0' we need to accumulate data with the same timestamp
// ToDo - Check Timestamp
// Add the RTP packet to the tempoary_rtp list until we have a complete 'Frame'
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
List <byte[]> nal_units = h264Payload.Process_H264_RTP_Packet(rtp_payload, rtp_marker); // this will cache the Packets until there is a Frame
if (nal_units == null)
{
// we have not passed in enough RTP packets to make a Frame of video
}
else
{
// we have a frame of NAL Units. Write them to the file
if (Received_NALs != null)
{
Received_NALs(nal_units);
}
}
}
else if (data_received.Channel == audio_data_channel && (rtp_payload_type == 0 || rtp_payload_type == 8 || audio_codec.Equals("PCMA") || audio_codec.Equals("PCMU")))
{
// G711 PCMA or G711 PCMU
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
List <byte[]> audio_frames = g711Payload.Process_G711_RTP_Packet(rtp_payload, rtp_marker);
if (audio_frames == null)
{
// some error
}
else
{
// Write the audio frames to the file
if (Received_G711 != null)
{
Received_G711(audio_codec, audio_frames);
}
}
}
else if (data_received.Channel == video_data_channel && rtp_payload_type == 26)
{
Console.WriteLine("No parser has been written for JPEG RTP packets. Please help write one");
return; // ignore this data
}
else
{
Console.WriteLine("No parser for RTP payload " + rtp_payload_type);
}
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// If message has a 401 - Unauthorised Error, then we re-send the message with Authorization
// using the most recently received 'realm' and 'nonce'
if (message.IsOk == false)
{
Console.WriteLine("Got Error in RTSP Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// the authorization failed.
Stop();
return;
}
// Check if the Reply has an Authenticate header.
if (message.ReturnCode == 401 && message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
// Process the WWW-Authenticate header
// EG: Basic realm="AProxy"
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
if (item.ToLower().Equals("basic"))
{
auth_type = "Basic";
}
else if (item.ToLower().Equals("digest"))
{
auth_type = "Digest";
}
else
{
// Split on the = symbol and update the realm and nonce
string[] parts = item.Split(new char[] { '=' }, 2); // max 2 parts in the results array
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
}
Console.WriteLine("WWW Authorize parsed for " + auth_type + " " + realm + " " + nonce);
}
RtspMessage resend_message = message.OriginalRequest.Clone() as RtspMessage;
if (auth_type != null)
{
AddAuthorization(resend_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(resend_message);
return;
}
// If we get a reply to OPTIONS then start the Keepalive Timer and send DESCRIBE
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (keepalive_timer == null)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// Send DESCRIBE
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (auth_type != null)
{
AddAuthorization(describe_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// RTP and RTCP 'channels' are used in TCP Interleaved mode (RTP over RTSP)
int next_free_rtp_channel = 0;
int next_free_rtcp_channel = 1;
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU", "AMR" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = next_free_rtp_channel;
video_rtcp_channel = next_free_rtcp_channel;
}
if (audio)
{
audio_data_channel = next_free_rtp_channel;
audio_rtcp_channel = next_free_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(next_free_rtp_channel, next_free_rtcp_channel), // Eg Channel 0 for RTP video data. Channel 1 for RTCP status reports
};
next_free_rtp_channel += 2;
next_free_rtcp_channel += 2;
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
int rtp_port = 0;
int rtcp_port = 0;
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = video_udp_pair.data_port;
rtcp_port = video_udp_pair.control_port;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = audio_udp_pair.data_port;
rtcp_port = audio_udp_pair.control_port;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(rtp_port, rtcp_port), // a UDP Port for data (video or audio). a UDP Port for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Generate SETUP messages
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (auth_type != null)
{
AddAuthorization(setup_message, username, password, auth_type, realm, nonce, url);
}
// Add SETUP message to list of mesages to send
setup_messages.Add(setup_message);
}
}
// Send the FIRST SETUP message and remove it from the list of Setup Messages
rtsp_client.SendMessage(setup_messages[0]);
setup_messages.RemoveAt(0);
}
// If we get a reply to SETUP (which was our third command), then we
// (i) check if we have any more SETUP commands to send out (eg if we are doing SETUP for Video and Audio)
// (ii) send a PLAY command if all the SETUP command have been sent
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown and and additional Setups
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Check if we have another SETUP command to send, then remote it from the list
if (setup_messages.Count > 0)
{
// send the next SETUP message, after adding in the 'session'
Rtsp.Messages.RtspRequestSetup next_setup = setup_messages[0];
next_setup.Session = session;
rtsp_client.SendMessage(next_setup);
setup_messages.RemoveAt(0);
}
else
{
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
if (auth_type != null)
{
AddAuthorization(play_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(play_message);
}
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
private void Rtsp_client_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
int rtp_version = (e.Message.Data[0] >> 6);
int rtp_padding = (e.Message.Data[0] >> 5) & 0x01;
int rtp_extension = (e.Message.Data[0] >> 4) & 0x01;
int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F;
int rtp_marker = (e.Message.Data[1] >> 7) & 0x01;
int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F;
uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtp_payload_start = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtp_csrc_count); // zero or more csrcs
uint rtp_extension_id = 0;
uint rtp_extension_size = 0;
if (rtp_extension == 1)
{
rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0);
rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0);
rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload
}
Console.WriteLine("RTP Data"
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
+ " Seq=" + rtp_sequence_number
+ " Time=" + rtp_timestamp
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
if (rtp_payload_type != video_payload)
{
Console.WriteLine("Ignoring this RTP payload");
return; // ignore this data
}
// If rtp_marker is '1' then this is the final transmission for this packet.
// If rtp_marker is '0' we need to accumulate data with the same timestamp
// ToDo - Check Timestamp
// ToDo - could avoid a copy if there is only one RTP frame for the data (temp list is zero)
// Add to the tempoary_rtp List
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
temporary_rtp_payloads.Add(rtp_payload);
if (rtp_marker == 1)
{
// Process the RTP frame
Process_RTP_Frame(temporary_rtp_payloads);
temporary_rtp_payloads.Clear();
}
}
private void Rtsp_client_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Console.WriteLine("Message Received " + e.ToString());
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
// send the Describe
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP.
// If the attribute is for Video, then carry out a SETUP and a PLAY
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].GetMediaType() == Rtsp.Sdp.Media.MediaType.video)
{
// seach the atributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
String fmtp = ""; // holds SPS and PPS
String rtpmap = ""; // holds Payload format, eg 96 often used with H264 as first dynamic payload value
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
control = attrib.Value;
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib.Value;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib.Value;
}
}
String[] split_rtpmap = rtpmap.Split(' ');
video_payload = 0;
bool result = Int32.TryParse(split_rtpmap[0], out video_payload);
// Transport: RTP/AVP;unicast;client_port=8000-8001
// Transport: RTP/AVP/TCP;interleaved=0-1
Rtsp.Messages.RtspRequest setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(url + "/" + control);
setup_message.AddHeader("Transport: RTP/AVP/TCP;interleaved=0");
rtsp_client.SendMessage(setup_message);
}
}
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
String session = message.Session; // Session value used with Play, Pause, Teardown
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
// play_message.Timeout = 65;
rtsp_client.SendMessage(play_message);
}
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAU
Console.WriteLine("Got reply from Play " + message.Command);
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the Describe
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
// If the attribute is for Video, then carry out a SETUP and a PLAY
// Only do this for the first Video attribute in case there is more than one in the SDP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].GetMediaType() == Rtsp.Sdp.Media.MediaType.video)
{
// We only want the first video sub-stream
if (video_payload == -1)
{
// seach the atributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64
Rtsp.Sdp.AttributRtpMap rtpmap = null; // holds Payload format, eg 96 often used with H264 as first dynamic payload value
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
control = attrib.Value;
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
}
}
// Split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count > 0)
{
video_sps = sps_pps[0];
}
if (sps_pps.Count > 1)
{
video_pps = sps_pps[1];
}
Output_NAL(sps_pps); // output SPS and PPS
}
// Split the rtpmap to get the Payload Type
video_payload = 0;
if (rtpmap != null)
{
video_payload = rtpmap.PayloadNumber;
}
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(url + "/" + control);
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0; // Used in DataReceived event handler
video_rtcp_channel = 1; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
video_data_channel = udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = udp_pair.control_port; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(video_data_channel, video_rtcp_channel), // a Channel for video. a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
setup_message.AddTransport(transport);
rtsp_client.SendMessage(setup_message);
}
}
}
}
// If we get a reply to SETUP (which was our third command), then process then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udp_pair = new UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
udp_pair.DataReceived += Rtp_DataReceived;
udp_pair.Start();
}
}
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
Console.WriteLine("Got reply from Play " + message.Command);
}
}
int rtp_count = 0; // used for statistics
// RTP packet (or RTCP packet) has been received.
public void Rtp_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspData data_received = e.Message as Rtsp.Messages.RtspData;
// Check which channel the Data was received on.
// eg the Video Channel, the Video Control Channel (RTCP)
// In the future would also check the Audio Channel and Audio Control Channel
if (data_received.Channel == video_rtcp_channel)
{
Console.WriteLine("Received a RTCP message on channel " + data_received.Channel);
return;
}
if (data_received.Channel == video_data_channel)
{
// Received some Video Data on the correct channel.
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
int rtp_version = (e.Message.Data[0] >> 6);
int rtp_padding = (e.Message.Data[0] >> 5) & 0x01;
int rtp_extension = (e.Message.Data[0] >> 4) & 0x01;
int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F;
int rtp_marker = (e.Message.Data[1] >> 7) & 0x01;
int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F;
uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtp_payload_start = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtp_csrc_count); // zero or more csrcs
uint rtp_extension_id = 0;
uint rtp_extension_size = 0;
if (rtp_extension == 1)
{
rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0);
rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0) * 4; // units of extension_size is 4-bytes
rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload
}
Console.WriteLine("RTP Data"
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
+ " Seq=" + rtp_sequence_number
+ " Time (MS)=" + rtp_timestamp / 90 // convert from 90kHZ clock to ms
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
String msg = "RTP Data " + rtp_count++
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
// + " Seq=" + rtp_sequence_number
// + " Time=" + rtp_timestamp
// + " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length;
fs2.WriteLine(msg);
fs2.Flush();
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (rtp_payload_type != video_payload)
{
Console.WriteLine("Ignoring this RTP payload");
return; // ignore this data
}
// If rtp_marker is '1' then this is the final transmission for this packet.
// If rtp_marker is '0' we need to accumulate data with the same timestamp
// ToDo - Check Timestamp
// ToDo - Could avoid a copy if there is only one RTP frame for the data (temp list is zero)
// Add the RTP packet to the tempoary_rtp list
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
temporary_rtp_payloads.Add(rtp_payload);
if (rtp_marker == 1)
{
// End Marker is set. Process the RTP frame
Process_RTP_Frame(temporary_rtp_payloads);
temporary_rtp_payloads.Clear();
}
}
}
int rtp_count = 0; // used for statistics
// RTP packet (or RTCP packet) has been received.
public void Rtp_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspData data_received = e.Message as Rtsp.Messages.RtspData;
// Check which channel the Data was received on.
// eg the Video Channel, the Video Control Channel (RTCP)
// In the future would also check the Audio Channel and Audio Control Channel
if (data_received.Channel == video_rtcp_channel)
{
Console.WriteLine("Received a RTCP message on channel " + data_received.Channel);
return;
}
if (data_received.Channel == video_data_channel)
{
// Received some Video Data on the correct channel.
// RTP Packet Header
// 0 - Version, P, X, CC, M, PT and Sequence Number
//32 - Timestamp
//64 - SSRC
//96 - CSRCs (optional)
//nn - Extension ID and Length
//nn - Extension header
int rtp_version = (e.Message.Data[0] >> 6);
int rtp_padding = (e.Message.Data[0] >> 5) & 0x01;
int rtp_extension = (e.Message.Data[0] >> 4) & 0x01;
int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F;
int rtp_marker = (e.Message.Data[1] >> 7) & 0x01;
int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F;
uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtp_payload_start = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtp_csrc_count); // zero or more csrcs
uint rtp_extension_id = 0;
uint rtp_extension_size = 0;
if (rtp_extension == 1)
{
rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0);
rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0) * 4; // units of extension_size is 4-bytes
rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload
}
Console.WriteLine("RTP Data"
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
+ " Seq=" + rtp_sequence_number
+ " Time (MS)=" + rtp_timestamp / 90 // convert from 90kHZ clock to ms
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
String msg = "RTP Data " + rtp_count++
+ " V=" + rtp_version
+ " P=" + rtp_padding
+ " X=" + rtp_extension
+ " CC=" + rtp_csrc_count
+ " M=" + rtp_marker
+ " PT=" + rtp_payload_type
// + " Seq=" + rtp_sequence_number
// + " Time=" + rtp_timestamp
// + " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length;
if (fs2 != null)
{
fs2.WriteLine(msg);
}
if (fs2 != null)
{
fs2.Flush();
}
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (rtp_payload_type != video_payload)
{
Console.WriteLine("Ignoring this RTP payload");
return; // ignore this data
}
if (rtp_payload_type >= 96 && rtp_payload_type <= 127 && video_codec.Equals("H264"))
{
// H264 RTP Packet
// If rtp_marker is '1' then this is the final transmission for this packet.
// If rtp_marker is '0' we need to accumulate data with the same timestamp
// ToDo - Check Timestamp
// Add the RTP packet to the tempoary_rtp list until we have a complete 'Frame'
byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
List <byte[]> nal_units = h264Payload.Process_H264_RTP_Packet(rtp_payload, rtp_marker); // this will cache the Packets until there is a Frame
if (nal_units == null)
{
// we have not passed in enough RTP packets to make a Frame of video
}
else
{
// we have a frame of NAL Units. Write them to the file
Output_NAL(nal_units);
}
}
else if (rtp_payload_type == 26)
{
Console.WriteLine("No parser for JPEG RTP packets");
}
else
{
Console.WriteLine("No parser for this RTP payload");
}
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header. If so we update the 'realm' and 'nonce'
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
var authHeader = AuthenticationHeaderValue.Parse(www_authenticate);
authorizationScheme = authHeader.Scheme;
var authParams = authHeader.Parameter
.Split(',')
.Select(p => p.Split('='))
.ToDictionary(p => p[0].Trim(' ', '"'), p => p[1].Trim(' ', '"'));
// For now, assuming auth scheme is either Basic or Digest
realm = authParams["realm"];
if (authorizationScheme == "Digest")
{
nonce = authParams["nonce"];
}
Console.WriteLine("WWW Authorize parsed for " + realm + " " + nonce ?? "<NULL>");
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we will not have the authorization Nonce so we
// handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the request did not use Authorizarion.
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
String authorization = GenerateAuthorization("DESCRIBE");
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (authorization != null)
{
describe_message.Headers.Add(RtspHeaderNames.Authorization, authorization);
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
// If the attribute is for Video, then carry out a SETUP and a PLAY
// Only do this for the first Video attribute in case there is more than one in the SDP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video)
{
// We only want the first video sub-stream
if (video_payload == -1)
{
video_payload = sdp_data.Medias[x].PayloadType;
// search the attributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264)
Rtsp.Sdp.AttributRtpMap rtpmap = null; // custom payload (>=96) details
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (rtpmap != null && rtpmap.Value.Contains("H264") && fmtp != null)
{
video_codec = "H264";
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
Output_SPS_PPS(sps, pps); // output SPS and PPS
}
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0; // Used in DataReceived event handler
video_rtcp_channel = 1; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
video_data_channel = udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = udp_pair.control_port; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(video_data_channel, video_rtcp_channel), // a Channel for video. a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Add authorization (if there is a username and password)
String digest_authorization = GenerateAuthorization("SETUP");
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (digest_authorization != null)
{
setup_message.Headers.Add("Authorization", digest_authorization);
}
rtsp_client.SendMessage(setup_message);
}
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udp_pair = new UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
udp_pair.DataReceived += Rtp_DataReceived;
udp_pair.Start();
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
