public void CloseConnection(string reason) { _logger.Info($"Connection {Id} closing. Reason: {reason}, _audioUdpPair.DataPort={_audioUdpPair?._dataPort}, _videoUdpPair.DataPort={_videoUdpPair?._dataPort}"); try { Play = false; // stop sending data if (_audioUdpPair != null) { ReleaseUDPSocket(_audioUdpPair); _audioUdpPair = null; } if (_videoUdpPair != null) { ReleaseUDPSocket(_videoUdpPair); _videoUdpPair = null; } _logger.Info($"Connection {Id} closed. Reason: {reason}"); _listener.MessageReceived -= RTSP_Message_Received; _listener.SocketExceptionRaised -= RTSP_SocketException_Raised; _listener.Disconnected -= RTSP_Disconnected; _listener.Stop(); _listener.Dispose(); } catch (Exception ex) { _logger.Warn($"{Id} error closing connection: {ex}"); } finally { var handler = OnConnectionRemoved; handler?.Invoke(Id, _videoSource); } }
// Process each RTSP message that is received private void RTSP_Message_Received(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); // Update the RTSP Keepalive Timeout // We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // found the connection connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; break; } } } #region Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse("DESCRIBE,SETUP,PLAY,TEARDOWN"); listener.SendMessage(options_response); } #endregion #region Handle DESCRIBE message if (message is Rtsp.Messages.RtspRequestDescribe) { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS String sps_str = Convert.ToBase64String(StreamInfo.SPS); String pps_str = Convert.ToBase64String(StreamInfo.PPS); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n"); sdp.Append($"s=SysDVR - https://github.com/exelix11/sysdvr - [PID {Process.GetCurrentProcess().Id}]\n"); if (video_source != null) { 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"); } if (audio_source != null) { sdp.Append("m=audio 0 RTP/AVP 97\n"); sdp.Append("a=rtpmap:97 L16/48000/2\n"); sdp.Append("a=control:trackID=1\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); } #endregion #region 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)) { var stream = int.Parse(setupMessage.RtspUri.LocalPath.Split('=')[1]); connection.sessions[stream].is_active = true; // found the connection // Add the transports to the connection connection.sessions[stream].client_transport = transport; connection.sessions[stream].transport_reply = transport_reply; // If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname connection.sessions[stream].udp_pair = udp_pair; connection.sessions[stream].session_id = session_handle.ToString(); session_handle++; // Copy the Session ID copy_of_session_id = connection.sessions[stream].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); } } #endregion #region Handle PLAY message // Must have 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 || message.Session == connection.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); } } } #endregion #region Handle TEARDOWN // Must have 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 { rtsp_list.Remove(connection); connection.Dispose(); listener.Dispose(); } } } #endregion }
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; } } }
private RtspTransport RTSP_ConstructReplyTransport(RtspTransport transport, out UDPSocket udp_pair) { udp_pair = null; RtspTransport transport_reply = new 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) { 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 try { _logger.Trace($"{Id} Start creating UDPSocket"); udp_pair = new Rtsp.UDPSocket(_ipAddress, 50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use udp_pair.DataReceived += UdpPair_DataReceived; udp_pair.ControlReceived += UdpPair_ControlReceived; udp_pair.Start(); // start listening for data on the UDP ports _logger.Trace($"{Id} End creating UDPSocket"); } catch { if (udp_pair != null) { ReleaseUDPSocket(udp_pair); throw; } } // 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._dataPort, udp_pair._controlPort); } 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; } return(transport_reply); }
// Process each RTSP message that is received private void RTSP_Message_Received(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); // Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce) if (auth != null) { bool authorized = false; if (message.Headers.ContainsKey("Authorization") == true) { // Check the message has the correct Authorization authorized = auth.IsValid(message); } if ((message.Headers.ContainsKey("Authorization") == false) || authorized == false) { // Send a 401 Authentication Required reply Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); return; } } // Update the RTSP Keepalive Timeout // We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // found the connection connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; break; } } } // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse(); listener.SendMessage(options_response); } // Handle DESCRIBE message if (message is Rtsp.Messages.RtspRequestDescribe) { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header String sps_str = Convert.ToBase64String(raw_sps); String pps_str = Convert.ToBase64String(raw_pps); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n"); sdp.Append("s=SharpRTSP Test Camera\n"); sdp.Append("m=video 0 RTP/AVP 96\n"); sdp.Append("c=IN IP4 0.0.0.0\n"); sdp.Append("a=control:trackID=0\n"); sdp.Append("a=rtpmap:96 H264/90000\n"); sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n"); byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString()); // Create the reponse to DESCRIBE // This must include the Session Description Protocol (SDP) Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(); describe_response.AddHeader("Content-Base: " + requested_url); describe_response.AddHeader("Content-Type: application/sdp"); describe_response.Data = sdp_bytes; describe_response.AdjustContentLength(); listener.SendMessage(describe_response); } // Handle SETUP message if (message is Rtsp.Messages.RtspRequestSetup) { // var setupMessage = message as Rtsp.Messages.RtspRequestSetup; // Check the RTSP transport // If it is UDP or Multicast, create the sockets // If it is RTP over RTSP we send data via the RTSP Listener // FIXME client may send more than one possible transport. // very rare Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0]; // Construct the Transport: reply from the Server to the client Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport(); transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP) { // RTP over RTSP mode} transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP; transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second); } Rtsp.UDPSocket udp_pair = null; if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == false) { Boolean udp_supported = true; if (udp_supported) { // RTP over UDP mode // Create a pair of UDP sockets - One is for the Video, one is for the RTCP udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => { // RTCP data received Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString()); }; udp_pair.Start(); // start listening for data on the UDP ports // Pass the Port of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = false; transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port); } else { transport_reply = null; } } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == true) { // RTP over Multicast UDP mode} // Create a pair of UDP sockets in Multicast Mode // Pass the Ports of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = true; transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX // for now until implemented transport_reply = null; } if (transport_reply != null) { // Update the session with transport information String copy_of_session_id = ""; lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream // or a SETUP for an Audio Stream. // In the SDP the H264 video track is TrackID 0 // found the connection // Add the transports to the connection connection.video_client_transport = transport; connection.video_transport_reply = transport_reply; // If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname connection.video_udp_pair = udp_pair; connection.video_session_id = session_handle.ToString(); session_handle++; // Copy the Session ID copy_of_session_id = connection.video_session_id; break; } } } Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString(); setup_response.Session = copy_of_session_id; listener.SendMessage(setup_response); } else { Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); // unsuported transport setup_response.ReturnCode = 461; listener.SendMessage(setup_response); } } // Handle PLAY message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPlay) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state to "PLAY" bool session_found = false; foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */ { // found the session session_found = true; connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO) string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp; // Send the reply Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_response.AddHeader("Range: " + range); play_response.AddHeader("RTP-Info: " + rtp_info); listener.SendMessage(play_response); break; } } if (session_found == false) { // Session ID was not found in the list of Sessions. Send a 454 error Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_failed_response.ReturnCode = 454; // Session Not Found listener.SendMessage(play_failed_response); } } } // Handle PAUSE message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPause) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */) { // found the session connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse(); listener.SendMessage(pause_response); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { // Create the reponse to GET_PARAMETER Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse(); listener.SendMessage(getparameter_response); } // Handle TEARDOWN (sent with a Session ID) if (message is Rtsp.Messages.RtspRequestTeardown) { lock (rtsp_list) { // Search for the Session in the Sessions List. foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list { if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL { // If this is UDP, close the transport // For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection) if (connection.video_udp_pair != null) { connection.video_udp_pair.Stop(); connection.video_udp_pair = null; } rtsp_list.Remove(connection); // Close the RTSP socket listener.Dispose(); } } } } }
// Process each RTSP message that is received private async System.Threading.Tasks.Task RTSP_Message_ReceivedAsync(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); var deviceId = ""; var streamId = ""; var unixTimestamp = 0; var startTime = new DateTime(1970, 1, 1); if (message is RtspRequest) { var rtspParameters = HttpUtility.ParseQueryString(((RtspRequest)message).RtspUri.Query); deviceId = rtspParameters["deviceId"]; streamId = rtspParameters["streamId"]; int.TryParse(rtspParameters["unixTimestamp"], out unixTimestamp); startTime = startTime.AddSeconds(unixTimestamp); Console.WriteLine($"{rtspParameters["deviceId"]}, {rtspParameters["streamId"]},{rtspParameters["unixTimestamp"]}"); } if (String.IsNullOrEmpty(deviceId)) { _logger.Error("No deviceId"); return; } List <RTSPConnection> rtsp_list = new List <RTSPConnection>(); rtsp_list = _rtspList.GetOrAdd(deviceId, rtsp_list); // Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce) bool authorized = false; Authentication authInfo = null; if (message.Headers.ContainsKey("Authorization") == true) { // The Header contained Authorization // Check the message has the correct Authorization // If it does not have the correct Authorization then close the RTSP connection authInfo = Authentication.GetAuthenticationInfo(message); URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password); string loginResponse = null; authorized = nvrCmd.Login(ref loginResponse); if (authorized == false) { // Send a 401 Authentication Failed reply, then close the RTSP Socket Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list.ToArray()) { if (connection.listener == listener) { rtsp_list.Remove(connection); } } } listener.Dispose(); return; } else { lock (rtsp_list) { if (!rtsp_list.Any(rtsp => rtsp.listener.RemoteAdress == listener.RemoteAdress)) { RTSPConnection new_connection = new RTSPConnection(); new_connection.listener = listener; new_connection.client_hostname = listener.RemoteAdress.Split(':')[0]; new_connection.ssrc = global_ssrc; new_connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; new_connection.video_time_since_last_rtcp_keepalive = DateTime.UtcNow; rtsp_list.Add(new_connection); } } _logger.Info($"Login NVR success:{loginResponse}"); } } else { Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); // 'Basic' or 'Digest' authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); return; } // Update the RTSP Keepalive Timeout // We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // found the connection connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; break; } } } // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse(); listener.SendMessage(options_response); // parse and get deviceId from url if (!_nvrPlayerList.ContainsKey(deviceId)) { URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password); string deviceConfigXml = null; //DeviceConfig deviceConfig = null; //nvrCmd.GetDeviceConfig(ref deviceConfigXml, deviceId); //XmlDocument xdoc = new XmlDocument(); try { //xdoc.LoadXml(deviceConfigXml); //XmlNodeReader reader = new XmlNodeReader(xdoc.DocumentElement); //XmlSerializer ser = new XmlSerializer(typeof(DeviceConfig)); //deviceConfig = (DeviceConfig)ser.Deserialize(reader); //var resolutionList = deviceConfig.Device.VideoQuality.Quality.Resolution1.Split('x'); //var widthStr = resolutionList[0].Replace("N", ""); //var heightStr = resolutionList[1]; //TODO get framerate instead of hardcode } catch (Exception ex) { _logger.Error(ex.ToString()); } //TODO open wmfplayer by session (device + stream or device + stream + IP&Port + playback time) WmfPlayer wmfPlayer = new WmfPlayer(new IntPtr(Int32.Parse(deviceId))); wmfPlayer.m_SelectID = deviceId; if (_nvrPlayerList.TryAdd(deviceId, wmfPlayer)) { wmfPlayer.ReceivedYUVFrame += video_source_ReceivedYUVFrame; await wmfPlayer.OpenVideoAsync(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password, startTime, 1, deviceId, int.Parse(streamId)); } } } else if (message is Rtsp.Messages.RtspRequestDescribe) // Handle DESCRIBE message { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header String sps_str = Convert.ToBase64String(raw_sps); String pps_str = Convert.ToBase64String(raw_pps); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append($"o={authInfo.Username} 0 0 IN IP4 0.0.0.0\n"); sdp.Append("s=ACTi NVR\n"); sdp.Append("m=video 0 RTP/AVP 96\n"); sdp.Append("c=IN IP4 0.0.0.0\n"); sdp.Append("a=control:*\n"); sdp.Append("a=rtpmap:96 H264/90000\n"); sdp.Append("a=fmtp:96 packetization-mode=1;profile-level-id=4D6028; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n"); byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString()); // Create the reponse to DESCRIBE // This must include the Session Description Protocol (SDP) Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(); describe_response.AddHeader("Content-Base: " + requested_url); describe_response.AddHeader("Content-Type: application/sdp"); describe_response.Data = sdp_bytes; describe_response.AdjustContentLength(); listener.SendMessage(describe_response); } else if (message is Rtsp.Messages.RtspRequestSetup)// Handle SETUP message { // var setupMessage = message as Rtsp.Messages.RtspRequestSetup; // Check the RTSP transport // If it is UDP or Multicast, create the sockets // If it is RTP over RTSP we send data via the RTSP Listener // FIXME client may send more than one possible transport. // very rare Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0]; // Construct the Transport: reply from the Server to the client Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport(); transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP) { // RTP over RTSP mode} transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP; transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second); } Rtsp.UDPSocket udp_pair = null; if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == false) { Boolean udp_supported = true; if (udp_supported) { // RTP over UDP mode // Create a pair of UDP sockets - One is for the Video, one is for the RTCP udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => { // RTCP data received Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString()); }; udp_pair.Start(); // start listening for data on the UDP ports // Pass the Port of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = false; transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port); } else { transport_reply = null; } } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == true) { // RTP over Multicast UDP mode} // Create a pair of UDP sockets in Multicast Mode // Pass the Ports of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = true; transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX // for now until implemented transport_reply = null; } if (transport_reply != null) { // Update the session with transport information String copy_of_session_id = ""; lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream // or a SETUP for an Audio Stream. // In the SDP the H264 video track is TrackID 0 // found the connection // Add the transports to the connection connection.video_client_transport = transport; connection.video_transport_reply = transport_reply; // If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname connection.video_udp_pair = udp_pair; connection.video_session_id = session_handle.ToString(); session_handle++; // Copy the Session ID copy_of_session_id = connection.video_session_id; break; } } } Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString(); setup_response.Session = copy_of_session_id; listener.SendMessage(setup_response); } else { Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); // unsuported transport setup_response.ReturnCode = 461; listener.SendMessage(setup_response); } } else if (message is Rtsp.Messages.RtspRequestPlay)// Handle PLAY message (Sent with a Session ID) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state to "PLAY" bool session_found = false; foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */ { // found the session session_found = true; connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO) string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp; // Send the reply Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_response.AddHeader("Range: " + range); play_response.AddHeader("RTP-Info: " + rtp_info); listener.SendMessage(play_response); break; } } if (session_found == false) { // Session ID was not found in the list of Sessions. Send a 454 error Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_failed_response.ReturnCode = 454; // Session Not Found listener.SendMessage(play_failed_response); } } } else if (message is Rtsp.Messages.RtspRequestPause) // Handle PAUSE message (Sent with a Session ID) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */) { // found the session connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse(); listener.SendMessage(pause_response); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { // Create the reponse to GET_PARAMETER Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse(); listener.SendMessage(getparameter_response); } // Handle TEARDOWN (sent with a Session ID) if (message is Rtsp.Messages.RtspRequestTeardown) { lock (rtsp_list) { // Search for the Session in the Sessions List. foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list { if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL { // If this is UDP, close the transport // For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection) if (connection.video_udp_pair != null) { connection.video_udp_pair.Stop(); connection.video_udp_pair = null; } rtsp_list.Remove(connection); // Close the RTSP socket listener.Dispose(); } } } } }