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;
}
}
public void SetSession()
{
RtspResponse testObject = new RtspResponse();
testObject.Session = "12345";
Assert.AreEqual("12345", testObject.Headers[RtspHeaderNames.Session]);
}
public void ReadSessionAndTimeout()
{
RtspResponse testObject = new RtspResponse();
testObject.Headers[RtspHeaderNames.Session] = "12345;timeout=33";
Assert.AreEqual("12345", testObject.Session);
Assert.AreEqual(33, testObject.Timeout);
}
/// <summary>
/// Gets the assiociate OK response with the request.
/// </summary>
/// <returns>
/// an Rtsp response corresponding to request.
/// </returns>
public override RtspResponse CreateResponse(ILogger logger)
{
RtspResponse response = base.CreateResponse(logger);
// Add genric suported operations.
response.Headers.Add(RtspHeaderNames.Public, "OPTIONS,DESCRIBE,SETUP,PLAY,PAUSE,TEARDOWN,GET_PARAMETER");
return(response);
}
public void SetSessionAndTimeout()
{
RtspResponse testObject = new RtspResponse();
testObject.Session = "12345";
testObject.Timeout = 10;
Assert.AreEqual("12345;timeout=10", testObject.Headers[RtspHeaderNames.Session]);
}
public void ChangeSession()
{
RtspResponse testObject = new RtspResponse();
testObject.Headers[RtspHeaderNames.Session] = "12345;timeout=33";
testObject.Session = "456";
Assert.AreEqual("456", testObject.Session);
Assert.AreEqual(33, testObject.Timeout);
Assert.AreEqual("456;timeout=33", testObject.Headers[RtspHeaderNames.Session]);
}
private void RTSP_ProcessTeardownRequest(RtspRequestTeardown message, RtspListener listener)
{
if (message.Session == _videoSessionId) // 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)
Rtsp.Messages.RtspResponse teardown_response = message.CreateResponse(_logger);
listener.SendMessage(teardown_response);
CloseConnection("teardown");
}
}
/// <summary>
/// Gets the assiociate OK response with the request.
/// </summary>
/// <returns>an Rtsp response correcponding to request.</returns>
public virtual RtspResponse CreateResponse(ILogger logger)
{
RtspResponse returnValue = new RtspResponse(logger);
returnValue.ReturnCode = 200;
returnValue.CSeq = this.CSeq;
if (this.Headers.ContainsKey(RtspHeaderNames.Session))
{
returnValue.Headers[RtspHeaderNames.Session] = this.Headers[RtspHeaderNames.Session];
}
return(returnValue);
}
private void RTSP_ProcessOptionsRequest(RtspRequestOptions message, RtspListener listener)
{
String requested_url = message.RtspUri.ToString();
_logger.Info($"Connection {listener.ConnectionId} requested for url: {requested_url}");
_videoSource = _requestUrlVideoSourceResolverStrategy.ResolveVideoSource(requested_url);
OnConnectionAdded?.Invoke(Id, _videoSource); //treat connection useful when VideoSource determined
// Create the reponse to OPTIONS
Rtsp.Messages.RtspResponse options_response = message.CreateResponse(_logger);
// Rtsp.Messages.RtspResponse options_response = OnRtspMessageReceived?.Invoke(message as Rtsp.Messages.RtspRequest,targetConnection);
listener.SendMessage(options_response);
}
private void RTSP_ProcessPauseRequest(RtspRequestPause message, RtspListener listener)
{
if (message.Session == _videoSessionId /* OR AUDIO SESSION ID */)
{
OnStop?.Invoke(Id);
// found the session
Play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO
}
// ToDo - only send back the OK response if the Session in the RTSP message was found
Rtsp.Messages.RtspResponse pause_response = message.CreateResponse(_logger);
listener.SendMessage(pause_response);
}
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);
}
}
/// <summary>
/// Create the good type of Rtsp Message from the header.
/// </summary>
/// <param name="aRequestLine">A request line.</param>
/// <returns>An Rtsp message</returns>
public static RtspMessage GetRtspMessage(string aRequestLine, ILogger logger)
{
// We can't determine the message
if (string.IsNullOrEmpty(aRequestLine))
{
return(new RtspMessage(logger));
}
string[] requestParts = aRequestLine.Split(new char[] { ' ' }, 3);
RtspMessage returnValue;
if (requestParts.Length == 3)
{
// A request is : Method SP Request-URI SP RTSP-Version
// A response is : RTSP-Version SP Status-Code SP Reason-Phrase
// RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
if (_rtspVersionTest.IsMatch(requestParts[2]))
{
returnValue = RtspRequest.GetRtspRequest(requestParts, logger);
}
else if (_rtspVersionTest.IsMatch(requestParts[0]))
{
returnValue = new RtspResponse(logger);
}
else
{
logger.Warn(CultureInfo.InvariantCulture, "Got a strange message {0}", aRequestLine);
returnValue = new RtspMessage(logger);
}
}
else
{
logger.Warn(CultureInfo.InvariantCulture, "Got a strange message {0}", aRequestLine);
returnValue = new RtspMessage(logger);
}
returnValue.Command = aRequestLine;
return(returnValue);
}
/// <summary>
/// Create the good type of Rtsp Message from the header.
/// </summary>
/// <param name="aRequestLine">A request line.</param>
/// <returns>An Rtsp message</returns>
public static RtspMessage GetRtspMessage(string aRequestLine)
{
// We can't determine the message
if (string.IsNullOrEmpty(aRequestLine))
{
return(new RtspMessage());
}
string[] requestParts = aRequestLine.Split(new char[] { ' ' }, 3);
RtspMessage returnValue;
if (requestParts.Length == 3)
{
// A request is : Method SP Request-URI SP RTSP-Version
// A response is : RTSP-Version SP Status-Code SP Reason-Phrase
// RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
if (_rtspVersionTest.IsMatch(requestParts[2]))
{
returnValue = RtspRequest.GetRtspRequest(requestParts);
}
else if (_rtspVersionTest.IsMatch(requestParts[0]))
{
returnValue = new RtspResponse();
}
else
{
System.Diagnostics.Debug.WriteLine($"Got a strange message {aRequestLine}");
returnValue = new RtspMessage();
}
}
else
{
System.Diagnostics.Debug.WriteLine($"Got a strange message {aRequestLine}");
returnValue = new RtspMessage();
}
returnValue.Command = aRequestLine;
return(returnValue);
}
// 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();
}
}
}
}
}
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;
}
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header. If so we update the 'realm' and 'nonce'
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
// Parse www_authenticate
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
// Split on the = symbol and load in
string[] parts = item.Split(new char[] { '=' });
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
Console.WriteLine("WWW Authorize parsed for " + realm + " " + nonce);
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we will not have the authorization Nonce so we
// handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the request did not use Authorizarion.
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "DESCRIBE");
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (digest_authorization != null)
{
describe_message.Headers.Add(RtspHeaderNames.Authorization, digest_authorization);
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
int data_channel = 0;
int rtcp_channel = 0;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = 0;
video_rtcp_channel = 1;
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = 2;
audio_rtcp_channel = 3;
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(data_channel, rtcp_channel), // Eg Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(data_channel, rtcp_channel), // a Channel for data (video or audio). a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Add authorization (if there is a username and password)
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "SETUP");
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (digest_authorization != null)
{
setup_message.Headers.Add("Authorization", digest_authorization);
}
rtsp_client.SendMessage(setup_message);
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
/// <summary>
/// Create the good type of Rtsp Message from the header.
/// </summary>
/// <param name="aRequestLine">A request line.</param>
/// <returns>An Rtsp message</returns>
public static RtspMessage GetRtspMessage(string aRequestLine)
{
// We can't determine the message
if (string.IsNullOrEmpty(aRequestLine))
return new RtspMessage();
string[] requestParts = aRequestLine.Split(new char[] { ' ' }, 3);
RtspMessage returnValue;
if (requestParts.Length == 3)
{
// A request is : Method SP Request-URI SP RTSP-Version
// A response is : RTSP-Version SP Status-Code SP Reason-Phrase
// RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
if (_rtspVersionTest.IsMatch(requestParts[2]))
returnValue = RtspRequest.GetRtspRequest(requestParts);
else if (_rtspVersionTest.IsMatch(requestParts[0]))
returnValue = new RtspResponse();
else
{
_logger.Warn(CultureInfo.InvariantCulture, "Got a strange message {0}", aRequestLine);
returnValue = new RtspMessage();
}
}
else
{
_logger.Warn(CultureInfo.InvariantCulture, "Got a strange message {0}", aRequestLine);
returnValue = new RtspMessage();
}
returnValue.Command = aRequestLine;
return returnValue;
}
/// <summary>
/// Handles the response to a setup message.
/// </summary>
/// <param name="message">A response message.</param>
private void HandleResponseToSetup(RtspResponse message)
{
RtspRequest original = message.OriginalRequest;
string setupKey = original.SourcePort.RemoteAdress + "SEQ" + message.CSeq.ToString(CultureInfo.InvariantCulture);
if (message.IsOk)
{
Forwarder forwarder = ConfigureTransportAndForwarder(message, _setupForwarder[setupKey]);
RtspSession newSession;
string sessionKey = RtspSession.GetSessionName(original.RtspUri, message.Session);
if (_activesSession.ContainsKey(sessionKey))
{
newSession = _activesSession[sessionKey];
_logger.Info("There was an already a session with ths ID {0}", newSession.Name);
}
else
{
_logger.Info("Create a new session with the ID {0}", sessionKey);
newSession = new RtspSession();
newSession.Name = message.Session;
newSession.Destination = original.RtspUri.Authority;
_activesSession.Add(sessionKey, newSession);
}
newSession.AddForwarder(original.RtspUri, forwarder);
newSession.Timeout = message.Timeout;
}
// No needed here anymore.
_setupForwarder.Remove(setupKey);
}
// 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);
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header and what type it is
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
// Parse www_authenticate
// EG: WWW-Authenticate: Basic realm="xxxxxxx"
// EG: WWW-Authenticate: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string[] items = www_authenticate.Split(new char[] { ',', ' ' }); // split on Comma and Space
// Process the first item
if (items.Count() >= 1 && items[0].Equals("Basic"))
{
authentication = AUTHENTICATION.BASIC;
}
else if (items.Count() >= 1 && items[0].Equals("Digest"))
{
authentication = AUTHENTICATION.DIGEST;
}
// Process the remaining items
for (int i = 1; i < items.Count(); i++)
{
string[] parts = items[i].Split(new char[] { '=' }); // Split on Equals
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we do not add any authorization (and we could not add it even if we wanted to
// as we will not have the authorization Nonce value required for Digest mode
// So we have to handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the original request did not use Authorization so try again with Authorization added
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "DESCRIBE");
if (authorization_string != null)
{
describe_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
// If the attribute is for Video, then carry out a SETUP and a PLAY
// Only do this for the first Video attribute in case there is more than one in the SDP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video)
{
// We only want the first video sub-stream
if (video_payload == -1)
{
video_payload = sdp_data.Medias[x].PayloadType;
// search the attributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264)
Rtsp.Sdp.AttributRtpMap rtpmap = null; // custom payload (>=96) details
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (rtpmap != null && rtpmap.Value.Contains("H264") && fmtp != null)
{
video_codec = "H264";
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
Output_SPS_PPS(sps, pps); // output SPS and PPS
}
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0; // Used in DataReceived event handler
video_rtcp_channel = 1; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
video_data_channel = udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = udp_pair.control_port; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(video_data_channel, video_rtcp_channel), // a Channel for video. a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "SETUP");
if (authorization_string != null)
{
setup_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(setup_message);
}
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udp_pair = new UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
udp_pair.DataReceived += Rtp_DataReceived;
udp_pair.Start();
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "PLAY");
if (authorization_string != null)
{
play_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
private void RTSP_ProcessGetParameterRequest(RtspRequestGetParameter message, RtspListener listener)
{
// Create the reponse to GET_PARAMETER
Rtsp.Messages.RtspResponse getparameter_response = message.CreateResponse(_logger);
listener.SendMessage(getparameter_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 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();
}
}
}
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// If message has a 401 - Unauthorised Error, then we re-send the message with Authorization
// using the most recently received 'realm' and 'nonce'
if (message.IsOk == false)
{
Console.WriteLine("Got Error in RTSP Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// the authorization failed.
Stop();
return;
}
// Check if the Reply has an Authenticate header.
if (message.ReturnCode == 401 && message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
// Process the WWW-Authenticate header
// EG: Basic realm="AProxy"
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
if (item.ToLower().Equals("basic"))
{
auth_type = "Basic";
}
else if (item.ToLower().Equals("digest"))
{
auth_type = "Digest";
}
else
{
// Split on the = symbol and update the realm and nonce
string[] parts = item.Split(new char[] { '=' }, 2); // max 2 parts in the results array
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
}
Console.WriteLine("WWW Authorize parsed for " + auth_type + " " + realm + " " + nonce);
}
RtspMessage resend_message = message.OriginalRequest.Clone() as RtspMessage;
if (auth_type != null)
{
AddAuthorization(resend_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(resend_message);
return;
}
// If we get a reply to OPTIONS then start the Keepalive Timer and send DESCRIBE
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (keepalive_timer == null)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// Send DESCRIBE
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (auth_type != null)
{
AddAuthorization(describe_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// RTP and RTCP 'channels' are used in TCP Interleaved mode (RTP over RTSP)
int next_free_rtp_channel = 0;
int next_free_rtcp_channel = 1;
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU", "AMR" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = next_free_rtp_channel;
video_rtcp_channel = next_free_rtcp_channel;
}
if (audio)
{
audio_data_channel = next_free_rtp_channel;
audio_rtcp_channel = next_free_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(next_free_rtp_channel, next_free_rtcp_channel), // Eg Channel 0 for RTP video data. Channel 1 for RTCP status reports
};
next_free_rtp_channel += 2;
next_free_rtcp_channel += 2;
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
int rtp_port = 0;
int rtcp_port = 0;
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = video_udp_pair.data_port;
rtcp_port = video_udp_pair.control_port;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = audio_udp_pair.data_port;
rtcp_port = audio_udp_pair.control_port;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(rtp_port, rtcp_port), // a UDP Port for data (video or audio). a UDP Port for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Generate SETUP messages
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (auth_type != null)
{
AddAuthorization(setup_message, username, password, auth_type, realm, nonce, url);
}
// Add SETUP message to list of mesages to send
setup_messages.Add(setup_message);
}
}
// Send the FIRST SETUP message and remove it from the list of Setup Messages
rtsp_client.SendMessage(setup_messages[0]);
setup_messages.RemoveAt(0);
}
// If we get a reply to SETUP (which was our third command), then we
// (i) check if we have any more SETUP commands to send out (eg if we are doing SETUP for Video and Audio)
// (ii) send a PLAY command if all the SETUP command have been sent
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown and and additional Setups
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Check if we have another SETUP command to send, then remote it from the list
if (setup_messages.Count > 0)
{
// send the next SETUP message, after adding in the 'session'
Rtsp.Messages.RtspRequestSetup next_setup = setup_messages[0];
next_setup.Session = session;
rtsp_client.SendMessage(next_setup);
setup_messages.RemoveAt(0);
}
else
{
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
if (auth_type != null)
{
AddAuthorization(play_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(play_message);
}
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
/// <summary>
/// Handles the response.
/// </summary>
/// <param name="message">A message.</param>
private void HandleResponse(RtspResponse message)
{
Contract.Requires(message != null);
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestSetup)
{
HandleResponseToSetup(message);
}
UpdateSessionState(message);
//TODO rewrite instead of remove
if (message.Headers.ContainsKey(RtspHeaderNames.ContentBase))
message.Headers.Remove(RtspHeaderNames.ContentBase);
if (message.Headers.ContainsKey(RtspHeaderNames.ContentType) &&
message.Headers[RtspHeaderNames.ContentType] == "application/sdp")
{
RewriteSDPMessage(message);
}
}
// 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>
/// Updates the state of the session.
/// </summary>
/// <param name="message">A response message.</param>
private void UpdateSessionState(RtspResponse message)
{
// if no session can be found
if (message.OriginalRequest == null ||
message.Session == null ||
message.OriginalRequest.RtspUri == null)
return;
//Update session state and handle special message
string sessionKey = RtspSession.GetSessionName(message.OriginalRequest.RtspUri, message.Session);
if (_activesSession.ContainsKey(sessionKey))
{
if (message.ReturnCode >= 300 && message.ReturnCode < 400)
_activesSession[sessionKey].State = RtspSession.SessionState.Init;
else if (message.ReturnCode < 300)
{
switch (message.OriginalRequest.RequestTyped)
{
case RtspRequest.RequestType.SETUP:
if (_activesSession[sessionKey].State == RtspSession.SessionState.Init)
_activesSession[sessionKey].State = RtspSession.SessionState.Ready;
break;
case RtspRequest.RequestType.PLAY:
if (_activesSession[sessionKey].State == RtspSession.SessionState.Ready)
_activesSession[sessionKey].State = RtspSession.SessionState.Playing;
break;
case RtspRequest.RequestType.RECORD:
if (_activesSession[sessionKey].State == RtspSession.SessionState.Ready)
_activesSession[sessionKey].State = RtspSession.SessionState.Recording;
break;
case RtspRequest.RequestType.PAUSE:
if (_activesSession[sessionKey].State == RtspSession.SessionState.Playing ||
_activesSession[sessionKey].State == RtspSession.SessionState.Recording)
_activesSession[sessionKey].State = RtspSession.SessionState.Ready;
break;
case RtspRequest.RequestType.TEARDOWN:
_activesSession[sessionKey].State = RtspSession.SessionState.Init;
break;
}
}
}
else
{
_logger.Warn("Command {0} for session {1} which was not found", message.OriginalRequest.RequestTyped, sessionKey);
}
}
/// <summary>
/// Gets the assiociate OK response with the request.
/// </summary>
/// <returns>an Rtsp response correcponding to request.</returns>
public virtual RtspResponse CreateResponse()
{
RtspResponse returnValue = new RtspResponse();
returnValue.ReturnCode = 200;
returnValue.CSeq = this.CSeq;
if (this.Headers.ContainsKey(RtspHeaderNames.Session))
{
returnValue.Headers[RtspHeaderNames.Session] = this.Headers[RtspHeaderNames.Session];
}
return returnValue;
}
