/// <summary>
/// Accepts the connection.
/// </summary>
private void AcceptConnection()
{
try
{
Profiler.BeginThreadProfiling("RTSP", "RTSPServer.AcceptConnection");
while (!_Stopping.WaitOne(0))
{
// Wait for an incoming TCP Connection
TcpClient oneClient = _RTSPServerListener.AcceptTcpClient();
Profiler.BeginSample("Add connection");
//Debug.Log("Connection from " + oneClient.Client.RemoteEndPoint.ToString());
// Hand the incoming TCP connection over to the RTSP classes
var rtsp_socket = new RtspTcpTransport(oneClient);
RtspListener newListener = new RtspListener(rtsp_socket);
newListener.MessageReceived += RTSP_Message_Received;
//RTSPDispatcher.Instance.AddListener(newListener);
// Add the RtspListener to the RTSPConnections List
RTSPConnection new_connection = new RTSPConnection();
new_connection.listener = newListener;
new_connection.client_hostname = newListener.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;
lock (rtsp_list)
rtsp_list.Add(new_connection);
newListener.Start();
Profiler.EndSample();
}
}
catch (SocketException /*error*/)
{
// _logger.Warn("Got an error listening, I have to handle the stopping which also throw an error", error);
}
catch (Exception /*error*/)
{
// _logger.Error("Got an error listening...", error);
throw;
}
}
// 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
RtspListener listener = sender as RtspListener;
Messages.RtspMessage message = e.Message as Messages.RtspMessage;
//Debug.Log("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)
{
// 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
Messages.RtspResponse authorization_response = (e.Message as 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;
}
}
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
Messages.RtspResponse authorization_response = (e.Message as 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 Messages.RtspRequestOptions)
{
// Create the reponse to OPTIONS
Messages.RtspResponse options_response = (e.Message as Messages.RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
}
// Handle DESCRIBE message
if (message is Messages.RtspRequestDescribe)
{
String requested_url = (message as Messages.RtspRequestDescribe).RtspUri.ToString();
//Debug.Log("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
String sps_str = Convert.ToBase64String(h264_encoder.GetRawSPS());
String pps_str = Convert.ToBase64String(h264_encoder.GetRawPPS());
//Debug.LogFormat("sps: {0}, pps: {1}", sps_str, pps_str);
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)
Messages.RtspResponse describe_response = (e.Message as 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 Messages.RtspRequestSetup)
{
//
var setupMessage = message as 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
Messages.RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Messages.RtspTransport transport_reply = new Messages.RtspTransport();
transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters
if (transport.LowerTransport == Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
UDPSocket udp_pair = null;
if (transport.LowerTransport == 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 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 = Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ServerPort = new Messages.PortCouple(udp_pair.data_port, udp_pair.control_port);
}
else
{
transport_reply = null;
}
}
if (transport.LowerTransport == 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 = Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new 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;
}
}
}
Messages.RtspResponse setup_response = setupMessage.CreateResponse();
setup_response.Headers[Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = copy_of_session_id;
listener.SendMessage(setup_response);
}
else
{
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 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=" + ((Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp;
// Send the reply
Messages.RtspResponse play_response = (e.Message as 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
Messages.RtspResponse play_failed_response = (e.Message as 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 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
Messages.RtspResponse pause_response = (e.Message as Messages.RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is Messages.RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
Messages.RtspResponse getparameter_response = (e.Message as Messages.RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN (sent with a Session ID)
if (message is 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();
}
}
}
}
}