public void AddTransport(RtspTransport newTransport)
{
string actualTransport = string.Empty;
if(Headers.ContainsKey(RtspHeaderNames.Transport))
actualTransport = Headers[RtspHeaderNames.Transport] + ",";
Headers[RtspHeaderNames.Transport] = actualTransport + newTransport.ToString();
}
private static void ReadTransport(RtspTransport returnValue, string[] transportProtocolPart)
{
TransportType transport;
if (!Enum.TryParse <TransportType>(transportProtocolPart[0], out transport))
{
throw new ArgumentException("Transport type invalid", "aTransportString");
}
returnValue.Transport = transport;
}
private static void ReadProfile(RtspTransport returnValue, string[] transportProtocolPart)
{
ProfileType profile;
if (transportProtocolPart.Length < 2 || !Enum.TryParse <ProfileType>(transportProtocolPart[1], out profile))
{
throw new ArgumentException("Transport profile type invalid", "aTransportString");
}
returnValue.Profile = profile;
}
public void AddTransport(RtspTransport newTransport)
{
string actualTransport = string.Empty;
if (Headers.ContainsKey(RtspHeaderNames.Transport))
{
actualTransport = Headers[RtspHeaderNames.Transport] + ",";
}
Headers[RtspHeaderNames.Transport] = actualTransport + newTransport.ToString();
}
/// <summary>
/// Gets the transports associate with the request.
/// </summary>
/// <value>The transport.</value>
public RtspTransport[] GetTransports()
{
if (!Headers.ContainsKey(RtspHeaderNames.Transport))
{
return new RtspTransport[] { new RtspTransport() }
}
;
string[] items = Headers[RtspHeaderNames.Transport].Split(',');
return(items.Select(o => RtspTransport.Parse(o)).ToArray());
}
private static void ReadLowerTransport(RtspTransport returnValue, string[] transportProtocolPart)
{
if (transportProtocolPart.Length == 3)
{
LowerTransportType lowerTransport;
if (!Enum.TryParse <LowerTransportType>(transportProtocolPart[2], out lowerTransport))
{
throw new ArgumentException("Lower transport type invalid", "aTransportString");
}
returnValue.LowerTransport = lowerTransport;
}
}
// 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();
}
}
}
}
}
/// <summary>
/// Parses the specified transport string.
/// </summary>
/// <param name="aTransportString">A transport string.</param>
/// <returns>The transport class.</returns>
/// <exception cref="ArgumentNullException"><paramref name="aTransportString"/> is null.</exception>
public static RtspTransport Parse(string aTransportString)
{
if (aTransportString == null)
{
throw new ArgumentNullException("aTransportString");
}
Contract.EndContractBlock();
RtspTransport returnValue = new RtspTransport();
string[] transportPart = aTransportString.Split(';');
string[] transportProtocolPart = transportPart[0].Split('/');
ReadTransport(returnValue, transportProtocolPart);
ReadProfile(returnValue, transportProtocolPart);
ReadLowerTransport(returnValue, transportProtocolPart);
foreach (string part in transportPart)
{
string[] subPart = part.Split('=');
switch (subPart[0].ToUpperInvariant())
{
case "UNICAST":
returnValue.IsMulticast = false;
break;
case "MULTICAST":
returnValue.IsMulticast = true;
break;
case "DESTINATION":
if (subPart.Length == 2)
{
returnValue.Destination = subPart[1];
}
break;
case "SOURCE":
if (subPart.Length == 2)
{
returnValue.Source = subPart[1];
}
break;
case "INTERLEAVED":
returnValue.IsMulticast = false;
if (subPart.Length < 2)
{
throw new ArgumentException("interleaved value invalid", "aTransportString");
}
returnValue.Interleaved = PortCouple.Parse(subPart[1]);
break;
case "APPEND":
returnValue.IsAppend = true;
break;
case "TTL":
int ttl = 0;
if (subPart.Length < 2 || !int.TryParse(subPart[1], out ttl))
{
throw new ArgumentException("TTL value invalid", "aTransportString");
}
returnValue.TTL = ttl;
break;
case "LAYERS":
int layers = 0;
if (subPart.Length < 2 || !int.TryParse(subPart[1], out layers))
{
throw new ArgumentException("Layers value invalid", "aTransportString");
}
returnValue.TTL = layers;
break;
case "PORT":
if (subPart.Length < 2)
{
throw new ArgumentException("Port value invalid", "aTransportString");
}
returnValue.Port = PortCouple.Parse(subPart[1]);
break;
case "CLIENT_PORT":
if (subPart.Length < 2)
{
throw new ArgumentException("client_port value invalid", "aTransportString");
}
returnValue.ClientPort = PortCouple.Parse(subPart[1]);
break;
case "SERVER_PORT":
if (subPart.Length < 2)
{
throw new ArgumentException("server_port value invalid", "aTransportString");
}
returnValue.ServerPort = PortCouple.Parse(subPart[1]);
break;
case "SSRC":
if (subPart.Length < 2)
{
throw new ArgumentException("ssrc value invalid", "aTransportString");
}
returnValue.SSrc = subPart[1];
break;
case "MODE":
if (subPart.Length < 2)
{
throw new ArgumentException("mode value invalid", "aTransportString");
}
returnValue.Mode = subPart[1];
break;
default:
// TODO log invalid part
break;
}
}
return(returnValue);
}
/// <summary>
/// Parses the specified transport string.
/// </summary>
/// <param name="aTransportString">A transport string.</param>
/// <returns>The transport class.</returns>
/// <exception cref="ArgumentNullException"><paramref name="aTransportString"/> is null.</exception>
public static RtspTransport Parse(string aTransportString)
{
if (aTransportString == null)
throw new ArgumentNullException("aTransportString");
Contract.EndContractBlock();
RtspTransport returnValue = new RtspTransport();
string[] transportPart = aTransportString.Split(';');
string[] transportProtocolPart = transportPart[0].Split('/');
ReadTransport(returnValue, transportProtocolPart);
ReadProfile(returnValue, transportProtocolPart);
ReadLowerTransport(returnValue, transportProtocolPart);
foreach (string part in transportPart)
{
string[] subPart = part.Split('=');
switch (subPart[0].ToUpperInvariant())
{
case "UNICAST":
returnValue.IsMulticast = false;
break;
case "MULTICAST":
returnValue.IsMulticast = true;
break;
case "DESTINATION":
if (subPart.Length == 2)
returnValue.Destination = subPart[1];
break;
case "SOURCE":
if (subPart.Length == 2)
returnValue.Source = subPart[1];
break;
case "INTERLEAVED":
if (subPart.Length < 2)
throw new ArgumentException("interleaved value invalid", "aTransportString");
returnValue.Interleaved = PortCouple.Parse(subPart[1]);
break;
case "APPEND":
returnValue.IsAppend = true;
break;
case "TTL":
int ttl = 0;
if (subPart.Length < 2 || !int.TryParse(subPart[1], out ttl))
throw new ArgumentException("TTL value invalid", "aTransportString");
returnValue.TTL = ttl;
break;
case "LAYERS":
int layers = 0;
if (subPart.Length < 2 || !int.TryParse(subPart[1], out layers))
throw new ArgumentException("Layers value invalid", "aTransportString");
returnValue.TTL = layers;
break;
case "PORT":
if (subPart.Length < 2)
throw new ArgumentException("Port value invalid", "aTransportString");
returnValue.Port = PortCouple.Parse(subPart[1]);
break;
case "CLIENT_PORT":
if (subPart.Length < 2)
throw new ArgumentException("client_port value invalid", "aTransportString");
returnValue.ClientPort = PortCouple.Parse(subPart[1]);
break;
case "SERVER_PORT":
if (subPart.Length < 2)
throw new ArgumentException("server_port value invalid", "aTransportString");
returnValue.ServerPort = PortCouple.Parse(subPart[1]);
break;
case "SSRC":
if (subPart.Length < 2)
throw new ArgumentException("ssrc value invalid", "aTransportString");
returnValue.SSrc = subPart[1];
break;
case "MODE":
if (subPart.Length < 2)
throw new ArgumentException("mode value invalid", "aTransportString");
returnValue.SSrc = subPart[1];
break;
default:
// TODO log invalid part
break;
}
}
return returnValue;
}
/// <summary>
/// Configures the transport and forwarder.
/// </summary>
/// <param name="aMessage">A message.</param>
/// <param name="forwarder">The preset forwarder.</param>
/// <returns>The configured forwarder.</returns>
private static Forwarder ConfigureTransportAndForwarder(RtspMessage aMessage, UDPForwarder forwarder)
{
RtspTransport transport = RtspTransport.Parse(aMessage.Headers[RtspHeaderNames.Transport]);
Forwarder resultForwarder;
if (transport.LowerTransport == RtspTransport.LowerTransportType.UDP)
{
if (transport.ServerPort != null)
{
forwarder.SourcePortVideo = transport.ServerPort.First;
forwarder.ForwardPortCommand = transport.ServerPort.Second;
}
resultForwarder = forwarder;
}
else
{
TCPtoUDPForwader TCPForwarder = new TCPtoUDPForwader();
TCPForwarder.ForwardCommand = aMessage.SourcePort;
TCPForwarder.SourceInterleavedVideo = transport.Interleaved.First;
TCPForwarder.ForwardInterleavedCommand = transport.Interleaved.Second;
// we need to transfer already getted values
TCPForwarder.ForwardHostVideo = forwarder.ForwardHostVideo;
TCPForwarder.ForwardPortVideo = forwarder.ForwardPortVideo;
TCPForwarder.SourcePortCommand = forwarder.SourcePortCommand;
TCPForwarder.ToMulticast = forwarder.ToMulticast;
resultForwarder = TCPForwarder;
}
if (resultForwarder.ToMulticast)
{
// Setup port and destination multicast.
resultForwarder.ForwardHostVideo = CreateNextMulticastAddress();
resultForwarder.ForwardPortVideo = forwarder.FromForwardVideoPort;
RtspTransport newTransport = new RtspTransport()
{
IsMulticast = true,
Destination = resultForwarder.ForwardHostVideo,
Port = new PortCouple(resultForwarder.ForwardPortVideo, resultForwarder.ListenCommandPort)
};
if ((resultForwarder is UDPForwarder && forwarder.ForwardPortCommand == 0)
|| (resultForwarder is TCPtoUDPForwader && (resultForwarder as TCPtoUDPForwader).ForwardInterleavedCommand == 0))
{
newTransport.Port = null;
}
aMessage.Headers[RtspHeaderNames.Transport] = newTransport.ToString();
}
else
{
RtspTransport newTransport = new RtspTransport()
{
IsMulticast = false,
Destination = forwarder.ForwardHostVideo,
ClientPort = new PortCouple(resultForwarder.ForwardPortVideo, resultForwarder.SourcePortCommand),
ServerPort = new PortCouple(resultForwarder.FromForwardVideoPort, resultForwarder.ListenCommandPort)
};
if ((resultForwarder is UDPForwarder && forwarder.ForwardPortCommand == 0)
|| (resultForwarder is TCPtoUDPForwader && (resultForwarder as TCPtoUDPForwader).ForwardInterleavedCommand == 0))
{
newTransport.ServerPort = null;
}
aMessage.Headers[RtspHeaderNames.Transport] = newTransport.ToString();
}
return resultForwarder;
}
// 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];
// 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
RTPSession new_session = new RTPSession();
new_session.session_id = session_count.ToString();
new_session.listener = listener;
new_session.sequence_number = (UInt16)rnd.Next(65535); // start with a random 16 bit sequence number
new_session.ssrc = 1;
// 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)
{
// Add the transports to the Session
new_session.client_transport = transport;
new_session.transport_reply = transport_reply;
// 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);
}
}
private static void ReadTransport(RtspTransport returnValue, string[] transportProtocolPart)
{
TransportType transport;
if (!Enum.TryParse<TransportType>(transportProtocolPart[0], out transport))
throw new ArgumentException("Transport type invalid", "aTransportString");
returnValue.Transport = transport;
}
private static void ReadProfile(RtspTransport returnValue, string[] transportProtocolPart)
{
ProfileType profile;
if (transportProtocolPart.Length < 2 || !Enum.TryParse<ProfileType>(transportProtocolPart[1], out profile))
throw new ArgumentException("Transport profile type invalid", "aTransportString");
returnValue.Profile = profile;
}
private static void ReadLowerTransport(RtspTransport returnValue, string[] transportProtocolPart)
{
if (transportProtocolPart.Length == 3)
{
LowerTransportType lowerTransport;
if (!Enum.TryParse<LowerTransportType>(transportProtocolPart[2], out lowerTransport))
throw new ArgumentException("Lower transport type invalid", "aTransportString");
returnValue.LowerTransport = lowerTransport;
}
}
// 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();
}
}
}
}
}
// 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);
}
}
private void RTSP_ProcessSetupRequest(RtspRequestSetup message, RtspListener listener)
{
//
var setupMessage = message;
// 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
RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.UDPSocket udp_pair;
RtspTransport transport_reply = RTSP_ConstructReplyTransport(transport, out udp_pair);
bool mediaTransportSet = false;
if (transport_reply != null)
{
// Update the session with transport information
String copy_of_session_id = "";
// 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
// Add the transports to the connection
if (contentBase != null)
{
string controlTrack = setupMessage.RtspUri.AbsoluteUri.Replace(contentBase, string.Empty);
var requestMedia = _sdpFile.Medias.FirstOrDefault(media =>
media.Attributs.FirstOrDefault(a => a.Key == "control" &&
(a.Value == controlTrack || "/" + a.Value == controlTrack)) != null);
if (requestMedia != null)
{
if (requestMedia.MediaType == Media.MediaTypes.video)
{
_videoClientTransport = transport;
_videoTransportReply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
if (_videoUdpPair != null)
{
ReleaseUDPSocket(_videoUdpPair);
}
_videoUdpPair = udp_pair;
mediaTransportSet = true;
if (setupMessage.Session == null)
{
_videoSessionId = _sessionHandle.ToString();
_sessionHandle++;
}
else
{
_videoSessionId = setupMessage.Session;
}
// Copy the Session ID
copy_of_session_id = _videoSessionId;
}
if (requestMedia.MediaType == Media.MediaTypes.audio)
{
_audioClientTransport = transport;
_audioTransportReply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
if (_audioUdpPair != null)
{
ReleaseUDPSocket(_audioUdpPair);
}
_audioUdpPair = udp_pair;
mediaTransportSet = true;
if (setupMessage.Session == null)
{
_audioSessionId = _sessionHandle.ToString();
_sessionHandle++;
}
else
{
_audioSessionId = setupMessage.Session;
}
// Copy the Session ID
copy_of_session_id = _audioSessionId;
}
}
}
if (false == mediaTransportSet)
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
// unsuported mediatime
setup_response.ReturnCode = 415;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = copy_of_session_id;
setup_response.Timeout = timeout_in_seconds;
listener.SendMessage(setup_response);
}
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
if (false == mediaTransportSet)
{
if (udp_pair != null)
{
ReleaseUDPSocket(udp_pair);
udp_pair = null;
}
}
}
// 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();
}
}
}
}
}
/// <summary>
/// Handles a request setup.
/// </summary>
/// <param name="destination">The destination.</param>
/// <param name="requestSetup">The request setup.</param>
/// <returns>The rewritten message</returns>
/// <remarks>
/// The destination can be modified.
/// </remarks>
private RtspMessage HandleRequestSetup(ref RtspListener destination, RtspRequestSetup requestSetup)
{
Contract.Requires(requestSetup != null);
Contract.Requires(destination != null);
Contract.Ensures(Contract.Result<RtspMessage>() != null);
Contract.Ensures(Contract.ValueAtReturn(out destination) != null);
// look if we already have a multicast streaming playing for this URI.
foreach (var session in _activesSession.Values)
{
if (session.State == RtspSession.SessionState.Playing && session.ListOfForwader.ContainsKey(requestSetup.RtspUri))
{
Forwarder existingForwarder = session.ListOfForwader[requestSetup.RtspUri];
if (existingForwarder != null && existingForwarder.ToMulticast)
{
RtspResponse returnValue = requestSetup.CreateResponse();
returnValue.Headers[RtspHeaderNames.Transport] = new RtspTransport()
{
IsMulticast = true,
Destination = existingForwarder.ForwardHostVideo,
Port = new PortCouple(existingForwarder.ForwardPortVideo, existingForwarder.ListenCommandPort),
}.ToString();
returnValue.Session = session.Name;
destination = requestSetup.SourcePort;
return returnValue;
}
}
}
string setupKey = requestSetup.SourcePort.RemoteAdress + "SEQ" + requestSetup.CSeq.ToString(CultureInfo.InvariantCulture);
RtspTransport selectedTransport = SelectTransport(requestSetup);
// We do not handle asked transport so return directly.
if (selectedTransport == null)
{
_logger.Info("No transport asked are supported, sorry");
RtspResponse returnValue = requestSetup.CreateResponse();
// Unsupported transport;
returnValue.ReturnCode = 461;
destination = requestSetup.SourcePort;
return returnValue;
}
UDPForwarder forwarder = new UDPForwarder();
forwarder.ToMulticast = selectedTransport.IsMulticast;
// this part of config is only valid in unicast.
if (!selectedTransport.IsMulticast)
{
forwarder.ForwardPortVideo = selectedTransport.ClientPort.First;
forwarder.SourcePortCommand = selectedTransport.ClientPort.Second;
// If the client did not set the destination.. get it from TCP source
if (!string.IsNullOrEmpty(selectedTransport.Destination))
{
forwarder.ForwardHostVideo = selectedTransport.Destination;
}
else
{
forwarder.ForwardHostVideo = requestSetup.SourcePort.RemoteAdress.Split(':')[0];
_logger.Debug("Destination get from TCP port {0}", forwarder.ForwardHostVideo);
}
}
// Configured the transport asked.
forwarder.ForwardHostCommand = destination.RemoteAdress.Split(':')[0];
RtspTransport firstNewTransport = new RtspTransport()
{
IsMulticast = false,
ClientPort = new PortCouple(forwarder.ListenVideoPort, forwarder.FromForwardCommandPort),
};
RtspTransport secondTransport = new RtspTransport()
{
IsMulticast = false,
LowerTransport = RtspTransport.LowerTransportType.TCP,
};
requestSetup.Headers[RtspHeaderNames.Transport] = firstNewTransport.ToString() + ", " + secondTransport.ToString();
_setupForwarder.Add(setupKey, forwarder);
return requestSetup;
}
// 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();
}
}
}
}
}
