private void RTSP_ProcessPlayRequest(RtspRequestPlay message, RtspListener listener)
{
OnPlay?.Invoke(Id);
Play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO)
_logger.Info($"Connection {Id} play started");
string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end
string rtp_info = "url=" + message.RtspUri + ";seq=" + _videoSequenceNumber; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp;
// Send the reply
Rtsp.Messages.RtspResponse play_response = message.CreateResponse(_logger);
play_response.AddHeader("Range: " + range);
play_response.AddHeader("RTP-Info: " + rtp_info);
listener.SendMessage(play_response);
//TODO: find a p[lace for this check]
// Session ID was not found in the list of Sessions. Send a 454 error
/* Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
* play_failed_response.ReturnCode = 454; // Session Not Found
* listener.SendMessage(play_failed_response);*/
}
private void RTSP_ProcessAuthorization(RtspRequest message, RtspListener listener)
{
bool authorized = false;
if (message.Headers.ContainsKey("Authorization") == true)
{
// The Header contained Authorization
// Check the message has the correct Authorization
// If it does not have the correct Authorization then close the RTSP connection
authorized = _auth.IsValid(message);
if (authorized == false)
{
// Send a 401 Authentication Failed reply, then close the RTSP Socket
Rtsp.Messages.RtspResponse authorization_response = message.CreateResponse(_logger);
authorization_response.AddHeader("WWW-Authenticate: " + _auth.GetHeader());
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
CloseConnection("unauthorized");
listener.Dispose();
return;
}
}
if ((message.Headers.ContainsKey("Authorization") == false))
{
// Send a 401 Authentication Failed with extra info in WWW-Authenticate
// to tell the Client if we are using Basic or Digest Authentication
Rtsp.Messages.RtspResponse authorization_response = message.CreateResponse(_logger);
authorization_response.AddHeader("WWW-Authenticate: " + _auth.GetHeader()); // 'Basic' or 'Digest'
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
return;
}
}
private void RTSP_ProcessDescribeRequest(RtspRequestDescribe message, RtspListener listener)
{
String requested_url = message.RtspUri.ToString();
Task <byte[]> sdpDataTask = _videoSource != null?
OnProvideSdpData?.Invoke(Id, _videoSource)
: Task.FromResult <byte[]>(null);
byte[] sdpData = sdpDataTask.Result;
if (sdpData != null)
{
Rtsp.Messages.RtspResponse describe_response = message.CreateResponse(_logger);
describe_response.AddHeader("Content-Base: " + requested_url);
describe_response.AddHeader("Content-Type: application/sdp");
describe_response.Data = sdpData;
describe_response.AdjustContentLength();
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
describe_response.Headers.TryGetValue(RtspHeaderNames.ContentBase, out contentBase);
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(describe_response.Data)))
{
_sdpFile = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
listener.SendMessage(describe_response);
}
else
{
Rtsp.Messages.RtspResponse describe_response = (message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(_logger);
//Method Not Valid In This State"
describe_response.ReturnCode = 455;
listener.SendMessage(describe_response);
}
}
// Process each RTSP message that is received
private void RTSP_Message_Received(object sender, RtspChunkEventArgs e)
{
// Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message
Rtsp.RtspListener listener = sender as Rtsp.RtspListener;
Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage;
Console.WriteLine("RTSP message received " + message);
// Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce)
if (auth != null)
{
bool authorized = false;
if (message.Headers.ContainsKey("Authorization") == true)
{
// Check the message has the correct Authorization
authorized = auth.IsValid(message);
}
if ((message.Headers.ContainsKey("Authorization") == false) ||
authorized == false)
{
// Send a 401 Authentication Required reply
Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse();
authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader());
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
return;
}
}
// Update the RTSP Keepalive Timeout
// We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message
lock (rtsp_list)
{
foreach (RTSPConnection connection in rtsp_list)
{
if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress))
{
// found the connection
connection.time_since_last_rtsp_keepalive = DateTime.UtcNow;
break;
}
}
}
// Handle OPTIONS message
if (message is Rtsp.Messages.RtspRequestOptions)
{
// Create the reponse to OPTIONS
Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
}
// Handle DESCRIBE message
if (message is Rtsp.Messages.RtspRequestDescribe)
{
String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString();
Console.WriteLine("Request for " + requested_url);
// TODO. Check the requsted_url is valid. In this example we accept any RTSP URL
// Make the Base64 SPS and PPS
byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
String sps_str = Convert.ToBase64String(raw_sps);
String pps_str = Convert.ToBase64String(raw_pps);
StringBuilder sdp = new StringBuilder();
// Generate the SDP
// The sprop-parameter-sets provide the SPS and PPS for H264 video
// The packetization-mode defines the H264 over RTP payloads used but is Optional
sdp.Append("v=0\n");
sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n");
sdp.Append("s=SharpRTSP Test Camera\n");
sdp.Append("m=video 0 RTP/AVP 96\n");
sdp.Append("c=IN IP4 0.0.0.0\n");
sdp.Append("a=control:trackID=0\n");
sdp.Append("a=rtpmap:96 H264/90000\n");
sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n");
byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString());
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse();
describe_response.AddHeader("Content-Base: " + requested_url);
describe_response.AddHeader("Content-Type: application/sdp");
describe_response.Data = sdp_bytes;
describe_response.AdjustContentLength();
listener.SendMessage(describe_response);
}
// Handle SETUP message
if (message is Rtsp.Messages.RtspRequestSetup)
{
//
var setupMessage = message as Rtsp.Messages.RtspRequestSetup;
// Check the RTSP transport
// If it is UDP or Multicast, create the sockets
// If it is RTP over RTSP we send data via the RTSP Listener
// FIXME client may send more than one possible transport.
// very rare
Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport();
transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
Rtsp.UDPSocket udp_pair = null;
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == false)
{
Boolean udp_supported = true;
if (udp_supported)
{
// RTP over UDP mode
// Create a pair of UDP sockets - One is for the Video, one is for the RTCP
udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use
udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => {
// RTCP data received
Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString());
};
udp_pair.Start(); // start listening for data on the UDP ports
// Pass the Port of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port);
}
else
{
transport_reply = null;
}
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == true)
{
// RTP over Multicast UDP mode}
// Create a pair of UDP sockets in Multicast Mode
// Pass the Ports of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
if (transport_reply != null)
{
// Update the session with transport information
String copy_of_session_id = "";
lock (rtsp_list)
{
foreach (RTSPConnection connection in rtsp_list)
{
if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress))
{
// ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream
// or a SETUP for an Audio Stream.
// In the SDP the H264 video track is TrackID 0
// found the connection
// Add the transports to the connection
connection.video_client_transport = transport;
connection.video_transport_reply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
connection.video_udp_pair = udp_pair;
connection.video_session_id = session_handle.ToString();
session_handle++;
// Copy the Session ID
copy_of_session_id = connection.video_session_id;
break;
}
}
}
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = copy_of_session_id;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
}
// Handle PLAY message (Sent with a Session ID)
if (message is Rtsp.Messages.RtspRequestPlay)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List. Change the state to "PLAY"
bool session_found = false;
foreach (RTSPConnection connection in rtsp_list)
{
if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */
{
// found the session
session_found = true;
connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO)
string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end
string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp;
// Send the reply
Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_response.AddHeader("Range: " + range);
play_response.AddHeader("RTP-Info: " + rtp_info);
listener.SendMessage(play_response);
break;
}
}
if (session_found == false)
{
// Session ID was not found in the list of Sessions. Send a 454 error
Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_failed_response.ReturnCode = 454; // Session Not Found
listener.SendMessage(play_failed_response);
}
}
}
// Handle PAUSE message (Sent with a Session ID)
if (message is Rtsp.Messages.RtspRequestPause)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (RTSPConnection connection in rtsp_list)
{
if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */)
{
// found the session
connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO
break;
}
}
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is Rtsp.Messages.RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN (sent with a Session ID)
if (message is Rtsp.Messages.RtspRequestTeardown)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List.
foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list
{
if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL
{
// If this is UDP, close the transport
// For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection)
if (connection.video_udp_pair != null)
{
connection.video_udp_pair.Stop();
connection.video_udp_pair = null;
}
rtsp_list.Remove(connection);
// Close the RTSP socket
listener.Dispose();
}
}
}
}
}
// Process each RTSP message that is received
private async System.Threading.Tasks.Task RTSP_Message_ReceivedAsync(object sender, RtspChunkEventArgs e)
{
// Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message
Rtsp.RtspListener listener = sender as Rtsp.RtspListener;
Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage;
Console.WriteLine("RTSP message received " + message);
var deviceId = "";
var streamId = "";
var unixTimestamp = 0;
var startTime = new DateTime(1970, 1, 1);
if (message is RtspRequest)
{
var rtspParameters = HttpUtility.ParseQueryString(((RtspRequest)message).RtspUri.Query);
deviceId = rtspParameters["deviceId"];
streamId = rtspParameters["streamId"];
int.TryParse(rtspParameters["unixTimestamp"], out unixTimestamp);
startTime = startTime.AddSeconds(unixTimestamp);
Console.WriteLine($"{rtspParameters["deviceId"]}, {rtspParameters["streamId"]},{rtspParameters["unixTimestamp"]}");
}
if (String.IsNullOrEmpty(deviceId))
{
_logger.Error("No deviceId");
return;
}
List <RTSPConnection> rtsp_list = new List <RTSPConnection>();
rtsp_list = _rtspList.GetOrAdd(deviceId, rtsp_list);
// Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce)
bool authorized = false;
Authentication authInfo = null;
if (message.Headers.ContainsKey("Authorization") == true)
{
// The Header contained Authorization
// Check the message has the correct Authorization
// If it does not have the correct Authorization then close the RTSP connection
authInfo = Authentication.GetAuthenticationInfo(message);
URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password);
string loginResponse = null;
authorized = nvrCmd.Login(ref loginResponse);
if (authorized == false)
{
// Send a 401 Authentication Failed reply, then close the RTSP Socket
Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse();
authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader());
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
lock (rtsp_list)
{
foreach (RTSPConnection connection in rtsp_list.ToArray())
{
if (connection.listener == listener)
{
rtsp_list.Remove(connection);
}
}
}
listener.Dispose();
return;
}
else
{
lock (rtsp_list)
{
if (!rtsp_list.Any(rtsp => rtsp.listener.RemoteAdress == listener.RemoteAdress))
{
RTSPConnection new_connection = new RTSPConnection();
new_connection.listener = listener;
new_connection.client_hostname = listener.RemoteAdress.Split(':')[0];
new_connection.ssrc = global_ssrc;
new_connection.time_since_last_rtsp_keepalive = DateTime.UtcNow;
new_connection.video_time_since_last_rtcp_keepalive = DateTime.UtcNow;
rtsp_list.Add(new_connection);
}
}
_logger.Info($"Login NVR success:{loginResponse}");
}
}
else
{
Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse();
authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); // 'Basic' or 'Digest'
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
return;
}
// Update the RTSP Keepalive Timeout
// We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message
lock (rtsp_list)
{
foreach (RTSPConnection connection in rtsp_list)
{
if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress))
{
// found the connection
connection.time_since_last_rtsp_keepalive = DateTime.UtcNow;
break;
}
}
}
// Handle OPTIONS message
if (message is Rtsp.Messages.RtspRequestOptions)
{
// Create the reponse to OPTIONS
Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
// parse and get deviceId from url
if (!_nvrPlayerList.ContainsKey(deviceId))
{
URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password);
string deviceConfigXml = null;
//DeviceConfig deviceConfig = null;
//nvrCmd.GetDeviceConfig(ref deviceConfigXml, deviceId);
//XmlDocument xdoc = new XmlDocument();
try
{
//xdoc.LoadXml(deviceConfigXml);
//XmlNodeReader reader = new XmlNodeReader(xdoc.DocumentElement);
//XmlSerializer ser = new XmlSerializer(typeof(DeviceConfig));
//deviceConfig = (DeviceConfig)ser.Deserialize(reader);
//var resolutionList = deviceConfig.Device.VideoQuality.Quality.Resolution1.Split('x');
//var widthStr = resolutionList[0].Replace("N", "");
//var heightStr = resolutionList[1];
//TODO get framerate instead of hardcode
}
catch (Exception ex)
{
_logger.Error(ex.ToString());
}
//TODO open wmfplayer by session (device + stream or device + stream + IP&Port + playback time)
WmfPlayer wmfPlayer = new WmfPlayer(new IntPtr(Int32.Parse(deviceId)));
wmfPlayer.m_SelectID = deviceId;
if (_nvrPlayerList.TryAdd(deviceId, wmfPlayer))
{
wmfPlayer.ReceivedYUVFrame += video_source_ReceivedYUVFrame;
await wmfPlayer.OpenVideoAsync(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password, startTime, 1, deviceId, int.Parse(streamId));
}
}
}
else if (message is Rtsp.Messages.RtspRequestDescribe) // Handle DESCRIBE message
{
String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString();
Console.WriteLine("Request for " + requested_url);
// TODO. Check the requsted_url is valid. In this example we accept any RTSP URL
// Make the Base64 SPS and PPS
raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
String sps_str = Convert.ToBase64String(raw_sps);
String pps_str = Convert.ToBase64String(raw_pps);
StringBuilder sdp = new StringBuilder();
// Generate the SDP
// The sprop-parameter-sets provide the SPS and PPS for H264 video
// The packetization-mode defines the H264 over RTP payloads used but is Optional
sdp.Append("v=0\n");
sdp.Append($"o={authInfo.Username} 0 0 IN IP4 0.0.0.0\n");
sdp.Append("s=ACTi NVR\n");
sdp.Append("m=video 0 RTP/AVP 96\n");
sdp.Append("c=IN IP4 0.0.0.0\n");
sdp.Append("a=control:*\n");
sdp.Append("a=rtpmap:96 H264/90000\n");
sdp.Append("a=fmtp:96 packetization-mode=1;profile-level-id=4D6028; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n");
byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString());
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse();
describe_response.AddHeader("Content-Base: " + requested_url);
describe_response.AddHeader("Content-Type: application/sdp");
describe_response.Data = sdp_bytes;
describe_response.AdjustContentLength();
listener.SendMessage(describe_response);
}
else if (message is Rtsp.Messages.RtspRequestSetup)// Handle SETUP message
{
//
var setupMessage = message as Rtsp.Messages.RtspRequestSetup;
// Check the RTSP transport
// If it is UDP or Multicast, create the sockets
// If it is RTP over RTSP we send data via the RTSP Listener
// FIXME client may send more than one possible transport.
// very rare
Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport();
transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
Rtsp.UDPSocket udp_pair = null;
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == false)
{
Boolean udp_supported = true;
if (udp_supported)
{
// RTP over UDP mode
// Create a pair of UDP sockets - One is for the Video, one is for the RTCP
udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use
udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => {
// RTCP data received
Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString());
};
udp_pair.Start(); // start listening for data on the UDP ports
// Pass the Port of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port);
}
else
{
transport_reply = null;
}
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == true)
{
// RTP over Multicast UDP mode}
// Create a pair of UDP sockets in Multicast Mode
// Pass the Ports of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
if (transport_reply != null)
{
// Update the session with transport information
String copy_of_session_id = "";
lock (rtsp_list)
{
foreach (RTSPConnection connection in rtsp_list)
{
if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress))
{
// ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream
// or a SETUP for an Audio Stream.
// In the SDP the H264 video track is TrackID 0
// found the connection
// Add the transports to the connection
connection.video_client_transport = transport;
connection.video_transport_reply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
connection.video_udp_pair = udp_pair;
connection.video_session_id = session_handle.ToString();
session_handle++;
// Copy the Session ID
copy_of_session_id = connection.video_session_id;
break;
}
}
}
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = copy_of_session_id;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
}
else if (message is Rtsp.Messages.RtspRequestPlay)// Handle PLAY message (Sent with a Session ID)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List. Change the state to "PLAY"
bool session_found = false;
foreach (RTSPConnection connection in rtsp_list)
{
if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */
{
// found the session
session_found = true;
connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO)
string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end
string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp;
// Send the reply
Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_response.AddHeader("Range: " + range);
play_response.AddHeader("RTP-Info: " + rtp_info);
listener.SendMessage(play_response);
break;
}
}
if (session_found == false)
{
// Session ID was not found in the list of Sessions. Send a 454 error
Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_failed_response.ReturnCode = 454; // Session Not Found
listener.SendMessage(play_failed_response);
}
}
}
else if (message is Rtsp.Messages.RtspRequestPause) // Handle PAUSE message (Sent with a Session ID)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (RTSPConnection connection in rtsp_list)
{
if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */)
{
// found the session
connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO
break;
}
}
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is Rtsp.Messages.RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN (sent with a Session ID)
if (message is Rtsp.Messages.RtspRequestTeardown)
{
lock (rtsp_list)
{
// Search for the Session in the Sessions List.
foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list
{
if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL
{
// If this is UDP, close the transport
// For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection)
if (connection.video_udp_pair != null)
{
connection.video_udp_pair.Stop();
connection.video_udp_pair = null;
}
rtsp_list.Remove(connection);
// Close the RTSP socket
listener.Dispose();
}
}
}
}
}
// Process each RTSP message that is received
private void RTSP_Message_Received(object sender, RtspChunkEventArgs e)
{
// Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message
Rtsp.RtspListener listener = sender as Rtsp.RtspListener;
Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage;
Console.WriteLine("RTSP message received " + message);
// Handle OPTIONS message
if (message is Rtsp.Messages.RtspRequestOptions)
{
// Create the reponse to OPTIONS
Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
}
// Handle DESCRIBE message
if (message is Rtsp.Messages.RtspRequestDescribe)
{
String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString();
Console.WriteLine("Request for " + requested_url);
// TODO. Check the requsted_url is valid. In this example we accept any RTSP URL
// Make the Base64 SPS and PPS
byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
String sps_str = Convert.ToBase64String(raw_sps);
String pps_str = Convert.ToBase64String(raw_pps);
StringBuilder sdp = new StringBuilder();
// Generate the SDP
// The sprop-parameter-sets provide the SPS and PPS for H264 video
// The packetization-mode defines the H264 over RTP payloads used but is Optional
sdp.Append("v=0\n");
sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n");
sdp.Append("s=SharpRTSP Test Camera\n");
sdp.Append("m=video 0 RTP/AVP 96\n");
sdp.Append("c=IN IP4 0.0.0.0\n");
sdp.Append("a=control:trackID=0\n");
sdp.Append("a=rtpmap:96 H264/90000\n");
sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n");
byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString());
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse();
describe_response.AddHeader("Content-Base: " + requested_url);
describe_response.AddHeader("Content-Type: application/sdp");
describe_response.Data = sdp_bytes;
describe_response.AdjustContentLength();
listener.SendMessage(describe_response);
}
// Handle SETUP message
if (message is Rtsp.Messages.RtspRequestSetup)
{
//
var setupMessage = message as Rtsp.Messages.RtspRequestSetup;
// Check the RTSP transport
// If it is UDP or Multicast, create the sockets
// If it is RTP over RTSP we send data via the RTSP Listener
// FIXME client may send more than one possible transport.
// very rare
Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport();
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == false)
{
// RTP over UDP mode}
// Create a pair of UDP sockets
// Pass the Port of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = transport.ClientPort; // FIX
// for now until implemented
transport_reply = null;
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == true)
{
// RTP over Multicast UDP mode}
// Create a pair of UDP sockets in Multicast Mode
// Pass the Ports of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
if (transport_reply != null)
{
RTPSession new_session = new RTPSession();
new_session.listener = listener;
new_session.sequence_number = (UInt16)rnd.Next(65535); // start with a random 16 bit sequence number
new_session.ssrc = 1;
// Add the transports to the Session
new_session.client_transport = transport;
new_session.transport_reply = transport_reply;
lock (rtp_list)
{
// Create a 'Session' and add it to the Session List
// ToDo - Check the Track ID. In the SDP the H264 video track is TrackID 0
// Place Lock() here so the Session Count and the addition to the list is locked
new_session.session_id = session_count.ToString();
// Add the new session to the Sessions List
rtp_list.Add(new_session);
session_count++;
}
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = new_session.session_id;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
}
// Handle PLAY message
if (message is Rtsp.Messages.RtspRequestPlay)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (RTPSession session in rtp_list)
{
if (session.session_id.Equals(message.Session))
{
// found the session
session.play = true;
break;
}
}
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
listener.SendMessage(play_response);
}
// Handle PLAUSE message
if (message is Rtsp.Messages.RtspRequestPause)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (RTPSession session in rtp_list)
{
if (session.session_id.Equals(message.Session))
{
// found the session
session.play = false;
break;
}
}
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is Rtsp.Messages.RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN
if (message is Rtsp.Messages.RtspRequestTeardown)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List.
foreach (RTPSession session in rtp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list
{
if (session.session_id.Equals(message.Session))
{
// TODO - Close UDP or Multicast transport
// For TCP there is no transport to close
rtp_list.Remove(session);
// Close the RTSP socket
listener.Dispose();
}
}
}
}
}
// Process each RTSP message that is received
private void RTSP_Message_Received(object sender, RtspChunkEventArgs e)
{
// Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message
Rtsp.RtspListener listener = sender as Rtsp.RtspListener;
Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage;
Console.WriteLine("RTSP message received " + message);
// Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce)
if (auth != null)
{
bool authorized = false;
if (message.Headers.ContainsKey("Authorization") == true)
{
// Check the message has the correct Authorization
authorized = auth.IsValid(message);
}
if ((message.Headers.ContainsKey("Authorization") == false) ||
authorized == false)
{
// Send a 401 Authentication Required reply
Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse();
authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader());
authorization_response.ReturnCode = 401;
listener.SendMessage(authorization_response);
return;
}
}
// Handle OPTIONS message
if (message is Rtsp.Messages.RtspRequestOptions)
{
// Create the reponse to OPTIONS
Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
}
// Handle DESCRIBE message
if (message is Rtsp.Messages.RtspRequestDescribe)
{
String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString();
Console.WriteLine("Request for " + requested_url);
// TODO. Check the requsted_url is valid. In this example we accept any RTSP URL
//// Make the Base64 SPS and PPS
//byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
//byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
//String sps_str = Convert.ToBase64String(raw_sps);
//String pps_str = Convert.ToBase64String(raw_pps);
StringBuilder sdp = new StringBuilder();
// Generate the SDP
// The sprop-parameter-sets provide the SPS and PPS for H264 video
// The packetization-mode defines the H264 over RTP payloads used but is Optional
sdp.Append("v=0\n");
sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n");
sdp.Append("s=WPFRtspServer\n");
sdp.Append("c=IN IP4 0.0.0.0\n");
if (mStreams != null)
{
foreach (var item in mStreams)
{
switch (item.Item1)
{
case StreamType.Video:
{
sdp.Append(string.Format("m=video 0 RTP/AVP {0}\n", (uint)item.Item1));
sdp.Append(string.Format("a=rtpmap:{0} {1}/90000\n", (uint)item.Item1, item.Item3));
sdp.Append(string.Format("a=fmtp:{0}\n", (uint)item.Item1));
sdp.Append(string.Format("a=control:trackID={0}\n", item.Item2));
//sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n");
}
break;
case StreamType.Audio:
{
sdp.Append(string.Format("m=audio 0 RTP/AVP {0}\n", (uint)item.Item1));
sdp.Append(string.Format("a=rtpmap:{0} mpeg4-generic/90000/2\n", (uint)item.Item1));
sdp.Append(string.Format("a=fmtp:{0}\n", (uint)item.Item1));
sdp.Append(string.Format("a=control:trackID={0}\n", item.Item2));
//sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n");
}
break;
case StreamType.None:
default:
break;
}
}
}
byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString());
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse();
describe_response.AddHeader("Content-Base: " + requested_url);
describe_response.AddHeader("Content-Type: application/sdp");
describe_response.Data = sdp_bytes;
describe_response.AdjustContentLength();
listener.SendMessage(describe_response);
}
// Handle SETUP message
if (message is Rtsp.Messages.RtspRequestSetup)
{
//
var setupMessage = message as Rtsp.Messages.RtspRequestSetup;
// Check the RTSP transport
// If it is UDP or Multicast, create the sockets
// If it is RTP over RTSP we send data via the RTSP Listener
// FIXME client may send more than one possible transport.
// very rare
Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport();
transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == false)
{
// RTP over UDP mode}
// Create a pair of UDP sockets
// Pass the Port of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = transport.ClientPort; // FIX
// for now until implemented
transport_reply = null;
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == true)
{
// RTP over Multicast UDP mode}
// Create a pair of UDP sockets in Multicast Mode
// Pass the Ports of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
if (transport_reply != null)
{
RTPSession new_session = new RTPSession();
new_session.listener = listener;
new_session.sequence_number = (UInt16)rnd.Next(65535); // start with a random 16 bit sequence number
new_session.ssrc = global_ssrc;
// Add the transports to the Session
new_session.client_transport = transport;
new_session.transport_reply = transport_reply;
lock (rtp_list)
{
if (setupMessage.RtspUri.Segments != null && setupMessage.RtspUri.Segments.Length == 2)
{
var split = setupMessage.RtspUri.Segments[1].Split(new char[] { '=' });
if (split != null && split.Length == 2)
{
int.TryParse(split[1], out new_session.trackID);
}
}
//setupMessage.SourcePort.RemoteAdress
// Create a 'Session' and add it to the Session List
// ToDo - Check the Track ID. In the SDP the H264 video track is TrackID 0
// Place Lock() here so the Session Count and the addition to the list is locked
new_session.session_id = session_count.ToString();
// Add the new session to the Sessions List
rtp_list.Add(new_session);
session_count++;
}
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = new_session.session_id;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse();
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
}
// Handle PLAY message
if (message is Rtsp.Messages.RtspRequestPlay)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
bool session_found = false;
foreach (RTPSession session in rtp_list)
{
//if (session.session_id.Equals(message.Session))
{
// found the session
session_found = true;
session.play = true;
string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end
string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + session.sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp;
// Send the reply
Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_response.AddHeader("Range: " + range);
play_response.AddHeader("RTP-Info: " + rtp_info);
listener.SendMessage(play_response);
}
}
if (session_found == false)
{
// Session ID was not found in the list of Sessions. Send a 454 error
Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse();
play_failed_response.ReturnCode = 454; // Session Not Found
listener.SendMessage(play_failed_response);
}
}
}
// Handle PLAUSE message
if (message is Rtsp.Messages.RtspRequestPause)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (RTPSession session in rtp_list)
{
if (session.session_id.Equals(message.Session))
{
// found the session
session.play = false;
break;
}
}
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is Rtsp.Messages.RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN
if (message is Rtsp.Messages.RtspRequestTeardown)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List.
foreach (RTPSession session in rtp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list
{
if (session.session_id.Equals(message.Session))
{
// TODO - Close UDP or Multicast transport
// For TCP there is no transport to close
rtp_list.Remove(session);
// Close the RTSP socket
listener.Dispose();
}
}
}
}
}