public void RtpDataReceived(object sender, RtspChunkEventArgs chunkEventArgs) { try { RtspData rtpData = chunkEventArgs.Message as RtspData; if (IsRtcpGoodbye(rtpData)) { ProcessTeardownResponse(); return; } // Check which channel the Data was received on. // eg the Video Channel, the Video Control Channel (RTCP) // In the future would also check the Audio Channel and Audio Control Channel if (rtpData.Channel == videoRTCPChannel) { Logger.Info("Received a RTCP message on channel " + rtpData.Channel); return; } if (rtpData.Channel == videoDataChannel) { ProcessRTPVideo(chunkEventArgs); } } catch (Exception e) { Logger.Error(e); rtspErrorSubject.OnNext(e.Message); } }
/// <summary> /// Handles the data receive. /// </summary> /// <param name="sender">The sender.</param> /// <param name="e">The <see cref="RTSP.RTSPChunkEventArgs"/> instance containing the event data.</param> public void HandleDataReceive(object sender, RtspChunkEventArgs e) { if (e == null) { throw new ArgumentNullException("e"); } Contract.EndContractBlock(); try { RtspData data = e.Message as RtspData; if (data != null) { byte[] frame = data.Data; if (data.Channel == this.SourceInterleavedVideo) { ForwardVUdpPort.BeginSend(frame, frame.Length, new AsyncCallback(EndSendVideo), frame); } } } catch (Exception error) { _logger.Warn("Error during frame forwarding", error); } }
private void UdpPair_ControlReceived(object local_sender, RtspChunkEventArgs local_e) { _timeSinceLastRtcpKeepAlive = DateTime.UtcNow; // RTCP data received //_logger.Debug($"{listener.ConnectionId} RTCP data received " + local_sender.ToString() + " " + local_e.ToString()); }
private void UdpPair_DataReceived(object local_sender, RtspChunkEventArgs local_e) { // RTP data received //_logger.Debug($"{listener.ConnectionId} RTP data received " + local_sender.ToString() + " " + local_e.ToString()); }
// Process each RTSP message that is received private void RTSP_Message_Received(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); // Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce) if (auth != null) { bool authorized = false; if (message.Headers.ContainsKey("Authorization") == true) { // Check the message has the correct Authorization authorized = auth.IsValid(message); } if ((message.Headers.ContainsKey("Authorization") == false) || authorized == false) { // Send a 401 Authentication Required reply Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); return; } } // Update the RTSP Keepalive Timeout // We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // found the connection connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; break; } } } // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse(); listener.SendMessage(options_response); } // Handle DESCRIBE message if (message is Rtsp.Messages.RtspRequestDescribe) { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header String sps_str = Convert.ToBase64String(raw_sps); String pps_str = Convert.ToBase64String(raw_pps); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n"); sdp.Append("s=SharpRTSP Test Camera\n"); sdp.Append("m=video 0 RTP/AVP 96\n"); sdp.Append("c=IN IP4 0.0.0.0\n"); sdp.Append("a=control:trackID=0\n"); sdp.Append("a=rtpmap:96 H264/90000\n"); sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n"); byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString()); // Create the reponse to DESCRIBE // This must include the Session Description Protocol (SDP) Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(); describe_response.AddHeader("Content-Base: " + requested_url); describe_response.AddHeader("Content-Type: application/sdp"); describe_response.Data = sdp_bytes; describe_response.AdjustContentLength(); listener.SendMessage(describe_response); } // Handle SETUP message if (message is Rtsp.Messages.RtspRequestSetup) { // var setupMessage = message as Rtsp.Messages.RtspRequestSetup; // Check the RTSP transport // If it is UDP or Multicast, create the sockets // If it is RTP over RTSP we send data via the RTSP Listener // FIXME client may send more than one possible transport. // very rare Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0]; // Construct the Transport: reply from the Server to the client Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport(); transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP) { // RTP over RTSP mode} transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP; transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second); } Rtsp.UDPSocket udp_pair = null; if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == false) { Boolean udp_supported = true; if (udp_supported) { // RTP over UDP mode // Create a pair of UDP sockets - One is for the Video, one is for the RTCP udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => { // RTCP data received Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString()); }; udp_pair.Start(); // start listening for data on the UDP ports // Pass the Port of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = false; transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port); } else { transport_reply = null; } } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == true) { // RTP over Multicast UDP mode} // Create a pair of UDP sockets in Multicast Mode // Pass the Ports of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = true; transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX // for now until implemented transport_reply = null; } if (transport_reply != null) { // Update the session with transport information String copy_of_session_id = ""; lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream // or a SETUP for an Audio Stream. // In the SDP the H264 video track is TrackID 0 // found the connection // Add the transports to the connection connection.video_client_transport = transport; connection.video_transport_reply = transport_reply; // If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname connection.video_udp_pair = udp_pair; connection.video_session_id = session_handle.ToString(); session_handle++; // Copy the Session ID copy_of_session_id = connection.video_session_id; break; } } } Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString(); setup_response.Session = copy_of_session_id; listener.SendMessage(setup_response); } else { Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); // unsuported transport setup_response.ReturnCode = 461; listener.SendMessage(setup_response); } } // Handle PLAY message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPlay) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state to "PLAY" bool session_found = false; foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */ { // found the session session_found = true; connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO) string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp; // Send the reply Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_response.AddHeader("Range: " + range); play_response.AddHeader("RTP-Info: " + rtp_info); listener.SendMessage(play_response); break; } } if (session_found == false) { // Session ID was not found in the list of Sessions. Send a 454 error Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_failed_response.ReturnCode = 454; // Session Not Found listener.SendMessage(play_failed_response); } } } // Handle PAUSE message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPause) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */) { // found the session connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse(); listener.SendMessage(pause_response); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { // Create the reponse to GET_PARAMETER Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse(); listener.SendMessage(getparameter_response); } // Handle TEARDOWN (sent with a Session ID) if (message is Rtsp.Messages.RtspRequestTeardown) { lock (rtsp_list) { // Search for the Session in the Sessions List. foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list { if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL { // If this is UDP, close the transport // For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection) if (connection.video_udp_pair != null) { connection.video_udp_pair.Stop(); connection.video_udp_pair = null; } rtsp_list.Remove(connection); // Close the RTSP socket listener.Dispose(); } } } } }
// Process each RTSP message that is received private async System.Threading.Tasks.Task RTSP_Message_ReceivedAsync(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); var deviceId = ""; var streamId = ""; var unixTimestamp = 0; var startTime = new DateTime(1970, 1, 1); if (message is RtspRequest) { var rtspParameters = HttpUtility.ParseQueryString(((RtspRequest)message).RtspUri.Query); deviceId = rtspParameters["deviceId"]; streamId = rtspParameters["streamId"]; int.TryParse(rtspParameters["unixTimestamp"], out unixTimestamp); startTime = startTime.AddSeconds(unixTimestamp); Console.WriteLine($"{rtspParameters["deviceId"]}, {rtspParameters["streamId"]},{rtspParameters["unixTimestamp"]}"); } if (String.IsNullOrEmpty(deviceId)) { _logger.Error("No deviceId"); return; } List <RTSPConnection> rtsp_list = new List <RTSPConnection>(); rtsp_list = _rtspList.GetOrAdd(deviceId, rtsp_list); // Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce) bool authorized = false; Authentication authInfo = null; if (message.Headers.ContainsKey("Authorization") == true) { // The Header contained Authorization // Check the message has the correct Authorization // If it does not have the correct Authorization then close the RTSP connection authInfo = Authentication.GetAuthenticationInfo(message); URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password); string loginResponse = null; authorized = nvrCmd.Login(ref loginResponse); if (authorized == false) { // Send a 401 Authentication Failed reply, then close the RTSP Socket Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list.ToArray()) { if (connection.listener == listener) { rtsp_list.Remove(connection); } } } listener.Dispose(); return; } else { lock (rtsp_list) { if (!rtsp_list.Any(rtsp => rtsp.listener.RemoteAdress == listener.RemoteAdress)) { RTSPConnection new_connection = new RTSPConnection(); new_connection.listener = listener; new_connection.client_hostname = listener.RemoteAdress.Split(':')[0]; new_connection.ssrc = global_ssrc; new_connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; new_connection.video_time_since_last_rtcp_keepalive = DateTime.UtcNow; rtsp_list.Add(new_connection); } } _logger.Info($"Login NVR success:{loginResponse}"); } } else { Rtsp.Messages.RtspResponse authorization_response = (e.Message as Rtsp.Messages.RtspRequest).CreateResponse(); authorization_response.AddHeader("WWW-Authenticate: " + auth.GetHeader()); // 'Basic' or 'Digest' authorization_response.ReturnCode = 401; listener.SendMessage(authorization_response); return; } // Update the RTSP Keepalive Timeout // We could check that the message is GET_PARAMETER or OPTIONS for a keepalive but instead we will update the timer on any message lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // found the connection connection.time_since_last_rtsp_keepalive = DateTime.UtcNow; break; } } } // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse(); listener.SendMessage(options_response); // parse and get deviceId from url if (!_nvrPlayerList.ContainsKey(deviceId)) { URLCommand nvrCmd = new URLCommand(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password); string deviceConfigXml = null; //DeviceConfig deviceConfig = null; //nvrCmd.GetDeviceConfig(ref deviceConfigXml, deviceId); //XmlDocument xdoc = new XmlDocument(); try { //xdoc.LoadXml(deviceConfigXml); //XmlNodeReader reader = new XmlNodeReader(xdoc.DocumentElement); //XmlSerializer ser = new XmlSerializer(typeof(DeviceConfig)); //deviceConfig = (DeviceConfig)ser.Deserialize(reader); //var resolutionList = deviceConfig.Device.VideoQuality.Quality.Resolution1.Split('x'); //var widthStr = resolutionList[0].Replace("N", ""); //var heightStr = resolutionList[1]; //TODO get framerate instead of hardcode } catch (Exception ex) { _logger.Error(ex.ToString()); } //TODO open wmfplayer by session (device + stream or device + stream + IP&Port + playback time) WmfPlayer wmfPlayer = new WmfPlayer(new IntPtr(Int32.Parse(deviceId))); wmfPlayer.m_SelectID = deviceId; if (_nvrPlayerList.TryAdd(deviceId, wmfPlayer)) { wmfPlayer.ReceivedYUVFrame += video_source_ReceivedYUVFrame; await wmfPlayer.OpenVideoAsync(_nvrIp, uint.Parse(_nvrPort), authInfo.Username, authInfo.Password, startTime, 1, deviceId, int.Parse(streamId)); } } } else if (message is Rtsp.Messages.RtspRequestDescribe) // Handle DESCRIBE message { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header String sps_str = Convert.ToBase64String(raw_sps); String pps_str = Convert.ToBase64String(raw_pps); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append($"o={authInfo.Username} 0 0 IN IP4 0.0.0.0\n"); sdp.Append("s=ACTi NVR\n"); sdp.Append("m=video 0 RTP/AVP 96\n"); sdp.Append("c=IN IP4 0.0.0.0\n"); sdp.Append("a=control:*\n"); sdp.Append("a=rtpmap:96 H264/90000\n"); sdp.Append("a=fmtp:96 packetization-mode=1;profile-level-id=4D6028; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n"); byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString()); // Create the reponse to DESCRIBE // This must include the Session Description Protocol (SDP) Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(); describe_response.AddHeader("Content-Base: " + requested_url); describe_response.AddHeader("Content-Type: application/sdp"); describe_response.Data = sdp_bytes; describe_response.AdjustContentLength(); listener.SendMessage(describe_response); } else if (message is Rtsp.Messages.RtspRequestSetup)// Handle SETUP message { // var setupMessage = message as Rtsp.Messages.RtspRequestSetup; // Check the RTSP transport // If it is UDP or Multicast, create the sockets // If it is RTP over RTSP we send data via the RTSP Listener // FIXME client may send more than one possible transport. // very rare Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0]; // Construct the Transport: reply from the Server to the client Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport(); transport_reply.SSrc = global_ssrc.ToString("X8"); // Convert to Hex, padded to 8 characters if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP) { // RTP over RTSP mode} transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP; transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second); } Rtsp.UDPSocket udp_pair = null; if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == false) { Boolean udp_supported = true; if (udp_supported) { // RTP over UDP mode // Create a pair of UDP sockets - One is for the Video, one is for the RTCP udp_pair = new Rtsp.UDPSocket(50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use udp_pair.DataReceived += (object local_sender, RtspChunkEventArgs local_e) => { // RTCP data received Console.WriteLine("RTCP data received " + local_sender.ToString() + " " + local_e.ToString()); }; udp_pair.Start(); // start listening for data on the UDP ports // Pass the Port of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = false; transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair.data_port, udp_pair.control_port); } else { transport_reply = null; } } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == true) { // RTP over Multicast UDP mode} // Create a pair of UDP sockets in Multicast Mode // Pass the Ports of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = true; transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX // for now until implemented transport_reply = null; } if (transport_reply != null) { // Update the session with transport information String copy_of_session_id = ""; lock (rtsp_list) { foreach (RTSPConnection connection in rtsp_list) { if (connection.listener.RemoteAdress.Equals(listener.RemoteAdress)) { // ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream // or a SETUP for an Audio Stream. // In the SDP the H264 video track is TrackID 0 // found the connection // Add the transports to the connection connection.video_client_transport = transport; connection.video_transport_reply = transport_reply; // If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname connection.video_udp_pair = udp_pair; connection.video_session_id = session_handle.ToString(); session_handle++; // Copy the Session ID copy_of_session_id = connection.video_session_id; break; } } } Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString(); setup_response.Session = copy_of_session_id; listener.SendMessage(setup_response); } else { Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); // unsuported transport setup_response.ReturnCode = 461; listener.SendMessage(setup_response); } } else if (message is Rtsp.Messages.RtspRequestPlay)// Handle PLAY message (Sent with a Session ID) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state to "PLAY" bool session_found = false; foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id) /* OR AUDIO_SESSION_ID */ { // found the session session_found = true; connection.play = true; // ACTUALLY YOU COULD PAUSE JUST THE VIDEO (or JUST THE AUDIO) string range = "npt=0-"; // Playing the 'video' from 0 seconds until the end string rtp_info = "url=" + ((Rtsp.Messages.RtspRequestPlay)message).RtspUri + ";seq=" + connection.video_sequence_number; // TODO Add rtptime +";rtptime="+session.rtp_initial_timestamp; // Send the reply Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_response.AddHeader("Range: " + range); play_response.AddHeader("RTP-Info: " + rtp_info); listener.SendMessage(play_response); break; } } if (session_found == false) { // Session ID was not found in the list of Sessions. Send a 454 error Rtsp.Messages.RtspResponse play_failed_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); play_failed_response.ReturnCode = 454; // Session Not Found listener.SendMessage(play_failed_response); } } } else if (message is Rtsp.Messages.RtspRequestPause) // Handle PAUSE message (Sent with a Session ID) { lock (rtsp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTSPConnection connection in rtsp_list) { if (message.Session == connection.video_session_id /* OR AUDIO SESSION ID */) { // found the session connection.play = false; // COULD HAVE PLAY/PAUSE FOR VIDEO AND AUDIO break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse(); listener.SendMessage(pause_response); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { // Create the reponse to GET_PARAMETER Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse(); listener.SendMessage(getparameter_response); } // Handle TEARDOWN (sent with a Session ID) if (message is Rtsp.Messages.RtspRequestTeardown) { lock (rtsp_list) { // Search for the Session in the Sessions List. foreach (RTSPConnection connection in rtsp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list { if (message.Session == connection.video_session_id) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL { // If this is UDP, close the transport // For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection) if (connection.video_udp_pair != null) { connection.video_udp_pair.Stop(); connection.video_udp_pair = null; } rtsp_list.Remove(connection); // Close the RTSP socket listener.Dispose(); } } } } }
// Process each RTSP message that is received private void RTSP_Message_Received(object sender, RtspChunkEventArgs e) { // Cast the 'sender' and 'e' into the RTSP Listener (the Socket) and the RTSP Message Rtsp.RtspListener listener = sender as Rtsp.RtspListener; Rtsp.Messages.RtspMessage message = e.Message as Rtsp.Messages.RtspMessage; Console.WriteLine("RTSP message received " + message); // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { // Create the reponse to OPTIONS Rtsp.Messages.RtspResponse options_response = (e.Message as Rtsp.Messages.RtspRequestOptions).CreateResponse(); listener.SendMessage(options_response); } // Handle DESCRIBE message if (message is Rtsp.Messages.RtspRequestDescribe) { String requested_url = (message as Rtsp.Messages.RtspRequestDescribe).RtspUri.ToString(); Console.WriteLine("Request for " + requested_url); // TODO. Check the requsted_url is valid. In this example we accept any RTSP URL // Make the Base64 SPS and PPS byte[] raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header byte[] raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header String sps_str = Convert.ToBase64String(raw_sps); String pps_str = Convert.ToBase64String(raw_pps); StringBuilder sdp = new StringBuilder(); // Generate the SDP // The sprop-parameter-sets provide the SPS and PPS for H264 video // The packetization-mode defines the H264 over RTP payloads used but is Optional sdp.Append("v=0\n"); sdp.Append("o=user 123 0 IN IP4 0.0.0.0\n"); sdp.Append("s=SharpRTSP Test Camera\n"); sdp.Append("m=video 0 RTP/AVP 96\n"); sdp.Append("c=IN IP4 0.0.0.0\n"); sdp.Append("a=control:trackID=0\n"); sdp.Append("a=rtpmap:96 H264/90000\n"); sdp.Append("a=fmtp:96 profile-level-id=42A01E; sprop-parameter-sets=" + sps_str + "," + pps_str + ";\n"); byte[] sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString()); // Create the reponse to DESCRIBE // This must include the Session Description Protocol (SDP) Rtsp.Messages.RtspResponse describe_response = (e.Message as Rtsp.Messages.RtspRequestDescribe).CreateResponse(); describe_response.AddHeader("Content-Base: " + requested_url); describe_response.AddHeader("Content-Type: application/sdp"); describe_response.Data = sdp_bytes; describe_response.AdjustContentLength(); listener.SendMessage(describe_response); } // Handle SETUP message if (message is Rtsp.Messages.RtspRequestSetup) { // var setupMessage = message as Rtsp.Messages.RtspRequestSetup; // Check the RTSP transport // If it is UDP or Multicast, create the sockets // If it is RTP over RTSP we send data via the RTSP Listener // FIXME client may send more than one possible transport. // very rare Rtsp.Messages.RtspTransport transport = setupMessage.GetTransports()[0]; // Construct the Transport: reply from the Server to the client Rtsp.Messages.RtspTransport transport_reply = new Rtsp.Messages.RtspTransport(); if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP) { // RTP over RTSP mode} transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP; transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second); } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == false) { // RTP over UDP mode} // Create a pair of UDP sockets // Pass the Port of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = false; transport_reply.ClientPort = transport.ClientPort; // FIX // for now until implemented transport_reply = null; } if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && transport.IsMulticast == true) { // RTP over Multicast UDP mode} // Create a pair of UDP sockets in Multicast Mode // Pass the Ports of the two sockets back in the reply transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP; transport_reply.IsMulticast = true; transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX // for now until implemented transport_reply = null; } if (transport_reply != null) { RTPSession new_session = new RTPSession(); new_session.listener = listener; new_session.sequence_number = (UInt16)rnd.Next(65535); // start with a random 16 bit sequence number new_session.ssrc = 1; // Add the transports to the Session new_session.client_transport = transport; new_session.transport_reply = transport_reply; lock (rtp_list) { // Create a 'Session' and add it to the Session List // ToDo - Check the Track ID. In the SDP the H264 video track is TrackID 0 // Place Lock() here so the Session Count and the addition to the list is locked new_session.session_id = session_count.ToString(); // Add the new session to the Sessions List rtp_list.Add(new_session); session_count++; } Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString(); setup_response.Session = new_session.session_id; listener.SendMessage(setup_response); } else { Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(); // unsuported transport setup_response.ReturnCode = 461; listener.SendMessage(setup_response); } } // Handle PLAY message if (message is Rtsp.Messages.RtspRequestPlay) { lock (rtp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTPSession session in rtp_list) { if (session.session_id.Equals(message.Session)) { // found the session session.play = true; break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse play_response = (e.Message as Rtsp.Messages.RtspRequestPlay).CreateResponse(); listener.SendMessage(play_response); } // Handle PLAUSE message if (message is Rtsp.Messages.RtspRequestPause) { lock (rtp_list) { // Search for the Session in the Sessions List. Change the state of "PLAY" foreach (RTPSession session in rtp_list) { if (session.session_id.Equals(message.Session)) { // found the session session.play = false; break; } } } // ToDo - only send back the OK response if the Session in the RTSP message was found Rtsp.Messages.RtspResponse pause_response = (e.Message as Rtsp.Messages.RtspRequestPause).CreateResponse(); listener.SendMessage(pause_response); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { // Create the reponse to GET_PARAMETER Rtsp.Messages.RtspResponse getparameter_response = (e.Message as Rtsp.Messages.RtspRequestGetParameter).CreateResponse(); listener.SendMessage(getparameter_response); } // Handle TEARDOWN if (message is Rtsp.Messages.RtspRequestTeardown) { lock (rtp_list) { // Search for the Session in the Sessions List. foreach (RTPSession session in rtp_list.ToArray()) // Convert to ToArray so we can delete from the rtp_list { if (session.session_id.Equals(message.Session)) { // TODO - Close UDP or Multicast transport // For TCP there is no transport to close rtp_list.Remove(session); // Close the RTSP socket listener.Dispose(); } } } } }
void DataReceived(object sender, RtspChunkEventArgs e) { _receivedData.Add(e.Message); }
void MessageReceived(object sender, RtspChunkEventArgs e) { _receivedMessage.Add(e.Message); }
/// <summary> /// Raises the <see cref="E:DataReceived"/> event. /// </summary> /// <param name="rtspChunkEventArgs">The <see cref="Rtsp.RtspChunkEventArgs"/> instance containing the event data.</param> protected void OnDataReceived(RtspChunkEventArgs rtspChunkEventArgs) { DataReceived?.Invoke(this, rtspChunkEventArgs); }
// 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 MessageReceived(object sender, RtspChunkEventArgs e) { RtspResponse message = e.Message as RtspResponse; // If we get a reply to OPTIONS then start the Keepalive Timer and send DESCRIBE if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestOptions) { // Start a Timer to send an Keepalive RTSP command every 20 seconds _keepaliveTimer = new Timer(); _keepaliveTimer.Elapsed += SendKeepalive; _keepaliveTimer.Interval = 20 * 1000; _keepaliveTimer.Enabled = true; // Send DESCRIBE RtspRequest describe_message = new RtspRequestDescribe(); describe_message.RtspUri = new Uri(_url); _client.SendMessage(describe_message); } if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestDescribe) { Rtsp.Sdp.SdpFile sdp_data; using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data))) { sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream); } Uri video_uri = null; foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[0].Attributs) { if (attrib.Key.Equals("control")) { video_uri = new Uri(attrib.Value); } } RtspTransport transport = new RtspTransport() { LowerTransport = RtspTransport.LowerTransportType.TCP, Interleaved = new PortCouple(0, 1), }; // Generate SETUP messages RtspRequestSetup setup_message = new RtspRequestSetup(); setup_message.RtspUri = video_uri; setup_message.AddTransport(transport); _client.SendMessage(setup_message); } if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestSetup) { if (message.Timeout > 0 && message.Timeout > _keepaliveTimer.Interval / 1000) { _keepaliveTimer.Interval = message.Timeout * 1000 / 2; } // Send PLAY RtspRequest play_message = new RtspRequestPlay(); play_message.RtspUri = new Uri(_url); play_message.Session = message.Session; _client.SendMessage(play_message); } }
private void DataReceived(object sender, RtspChunkEventArgs e) { RtspData data_received = e.Message as RtspData; if (data_received.Channel == _videoDataChannel) { // Received some Video or Audio Data on the correct channel. // RTP Packet Header // 0 - Version, P, X, CC, M, PT and Sequence Number //32 - Timestamp //64 - SSRC //96 - CSRCs (optional) //nn - Extension ID and Length //nn - Extension header int rtp_version = (e.Message.Data[0] >> 6); int rtp_padding = (e.Message.Data[0] >> 5) & 0x01; int rtp_extension = (e.Message.Data[0] >> 4) & 0x01; int rtp_csrc_count = (e.Message.Data[0] >> 0) & 0x0F; int rtp_marker = (e.Message.Data[1] >> 7) & 0x01; int rtp_payload_type = (e.Message.Data[1] >> 0) & 0x7F; uint rtp_sequence_number = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]); uint rtp_timestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]); uint rtp_ssrc = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]); int rtp_payload_start = 4 // V,P,M,SEQ + 4 // time stamp + 4 // ssrc + (4 * rtp_csrc_count); // zero or more csrcs uint rtp_extension_id = 0; uint rtp_extension_size = 0; if (rtp_extension == 1) { rtp_extension_id = ((uint)e.Message.Data[rtp_payload_start + 0] << 8) + (uint)(e.Message.Data[rtp_payload_start + 1] << 0); rtp_extension_size = ((uint)e.Message.Data[rtp_payload_start + 2] << 8) + (uint)(e.Message.Data[rtp_payload_start + 3] << 0) * 4; // units of extension_size is 4-bytes rtp_payload_start += 4 + (int)rtp_extension_size; // extension header and extension payload } if (data_received.Channel == _videoDataChannel && rtp_payload_type == 96) { byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); List <byte[]> nal_units = _h264Payload.Process_H264_RTP_Packet(rtp_payload, rtp_marker); if (nal_units != null) { // we have passed in enough RTP packets to make a Frame of video try { if (_decoder == null) { _decoder = new H264Decoder(nal_units); } _currentBitmap = _decoder.Update(nal_units); NewBitmapAvailable.Set(); } catch (Exception error) { Console.WriteLine(error.Message); } } } } }
// 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; _logger.Debug($"{listener.ConnectionId} RTSP message received " + message); // Check if the RTSP Message has valid authentication (validating against username,password,realm and nonce) if (_auth != null) { RTSP_ProcessAuthorization(message as RtspRequest, listener); } // 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 _timeSinceLastRtcpKeepAlive = DateTime.UtcNow; // Handle OPTIONS message if (message is Rtsp.Messages.RtspRequestOptions) { RTSP_ProcessOptionsRequest(message as RtspRequestOptions, listener); } // Handle DESCRIBE message if (message is Rtsp.Messages.RtspRequestDescribe) { RTSP_ProcessDescribeRequest(message as RtspRequestDescribe, listener); } // Handle SETUP message if (message is Rtsp.Messages.RtspRequestSetup) { RTSP_ProcessSetupRequest(message as RtspRequestSetup, listener); } // Handle PLAY message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPlay) { RTSP_ProcessPlayRequest(message as RtspRequestPlay, listener); } // Handle PAUSE message (Sent with a Session ID) if (message is Rtsp.Messages.RtspRequestPause) { RTSP_ProcessPauseRequest(message as RtspRequestPause, listener); } // Handle GET_PARAMETER message, often used as a Keep Alive if (message is Rtsp.Messages.RtspRequestGetParameter) { RTSP_ProcessGetParameterRequest(message as RtspRequestGetParameter, listener); } // Handle TEARDOWN (sent with a Session ID) if (message is Rtsp.Messages.RtspRequestTeardown) { RTSP_ProcessTeardownRequest(message as RtspRequestTeardown, listener); } }
// 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 }