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Sharp RTSP

Build Status Coverage Status

A C# library to handle RTSP connections and RTP data streams.

  • RTSP Client - will connect to a RTSP server, setup the stream and play the stream. UDP, TCP and Multicast are supported.
  • RTP Receiver - will recieve RTP and RTCP packets and pass them to a transport handler
  • RTSP Server - will accept RTSP connections and talk to clients
  • RTP Sender - will send RTP packets to clients
  • Transport Handler - A H264 transport handler is provided. Other video and audio formats are not supported at the current time.

⚠️ : This library do not handle the decoding of the video (eg converting H264 into a bitmap). SharpRTSP is limited to the transport layer. You mut use a library like FFMPEG or use Operating System APIs to do the decoding.

RTSP Client Example

  • STEP 1 - Open TCP Socket connection to the RTSP Server

              // Connect to a RTSP Server
              tcp_socket = new Rtsp.RtspTcpTransport(host,port);
    
              if (tcp_socket.Connected == false)
              {
                  Console.WriteLine("Error - did not connect");
                  return;
              }

    This opens a connection for a 'TCP' mode RTSP/RTP session where RTP packets are set in the RTSP socket.

  • STEP 2 - Create a RTSP Listener and attach it to the RTSP TCP Socket

              // Connect a RTSP Listener to the TCP Socket to send messages and listen for replies
              rtsp_client = new Rtsp.RtspListener(tcp_socket);
    
              rtsp_client.MessageReceived += Rtsp_client_MessageReceived;
              rtsp_client.DataReceived += Rtsp_client_DataReceived;
    
              rtsp_client.Start(); // start reading messages from the server

    The RTSP Listener class lets you SEND messages to the RTSP Server (see below).
    The RTSP Listner class has a worker thread that listens for replies from the RTSP Server.
    When replies are received the MessageReceived Event is fired.
    When RTP packets are received the DataReceived Event is fired.

  • STEP 3 - Send Messages to the RTSP Server

    The samples below show how to send messages.

    Send OPTIONS with this code :

              Rtsp.Messages.RtspRequest options_message = new Rtsp. Messages.RtspRequestOptions();
              options_message.RtspUri = new Uri(url);
              rtsp_client.SendMessage(options_message);

    Send DESCRIBE with this code :

              // send the Describe
              Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
              describe_message.RtspUri = new Uri(url);
              rtsp_client.SendMessage(describe_message);
              // The reply will include the SDP data

    Send SETUP with this code :

              // the value of 'control' comes from parsing the SDP for the desired video or audio sub-stream
              Rtsp.Messages.RtspRequest setup_message = new Rtsp.Messages.RtspRequestSetup();
              setup_message.RtspUri = new Uri(url + "/" + control);
              setup_message.AddHeader("Transport: RTP/AVP/TCP;interleaved=0");
              rtsp_client.SendMessage(setup_message);
              // The reply will include the Session

    Send PLAY with this code :

              // the value of 'session' comes from the reply of the SETUP command
              Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
              play_message.RtspUri = new Uri(url);
              play_message.Session = session;
              rtsp_client.SendMessage(play_message);
  • STEP 4 - Handle Replies when the MessageReceived event is fired

    This example assumes the main program sends an OPTIONS Command.
    It looks for a reply from the server for OPTIONS and then sends DESCRIBE.
    It looks for a reply from the server for DESCRIBE and then sends SETUP (for the video stream)
    It looks for a reply from the server for SETUP and then sends PLAY.
    Once PLAY has been sent the video, in the form of RTP packets, will be received.

          private void Rtsp_client_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
          {
              Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
    
              Console.WriteLine("Received " + message.OriginalRequest.ToString());
    
              if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
              {
                  // send the DESCRIBE
                  Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
                  describe_message.RtspUri = new Uri(url);
                  rtsp_client.SendMessage(describe_message);
              }
    
              if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
              {
                  // Got a reply for DESCRIBE
                  // Examine the SDP
                  Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
    
                  Rtsp.Sdp.SdpFile sdp_data;
                  using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
                  {
                      sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
                  }
    
                  // Process each 'Media' Attribute in the SDP.
                  // If the attribute is for Video, then send a SETUP
                  for (int x = 0; x < sdp_data.Medias.Count; x++)
                  {
                      if (sdp_data.Medias[x].GetMediaType() == Rtsp.Sdp.Media.MediaType.video)
                      {
                          // seach the atributes for control, fmtp and rtpmap
                          String control = "";  // the "track" or "stream id"
                          String fmtp = ""; // holds SPS and PPS
                          String rtpmap = ""; // holds the Payload format, 96 is often used with H264
                          foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
                          {
                              if (attrib.Key.Equals("control")) control = attrib.Value;
                              if (attrib.Key.Equals("fmtp")) fmtp = attrib.Value;
                              if (attrib.Key.Equals("rtpmap")) rtpmap = attrib.Value;
                          }
                          
                          // Get the Payload format number for the Video Stream
                          String[] split_rtpmap = rtpmap.Split(' ');
                          video_payload = 0;
                          bool result = Int32.TryParse(split_rtpmap[0], out video_payload);
    
                          // Send SETUP for the Video Stream
                          // using Interleaved mode (RTP frames over the RTSP socket)
                          Rtsp.Messages.RtspRequest setup_message = new Rtsp.Messages.RtspRequestSetup();
                          setup_message.RtspUri = new Uri(url + "/" + control);
                          setup_message.AddHeader("Transport: RTP/AVP/TCP;interleaved=0");
                          rtsp_client.SendMessage(setup_message);
                      }
                  }
              }
    
              if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
              {
                  // Got Reply to SETUP
                  Console.WriteLine("Got reply from Setup. Session is " + message.Session);
    
                  String session = message.Session; // Session value used with Play, Pause, Teardown
                  
                  // Send PLAY
                  Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
                  play_message.RtspUri = new Uri(url);
                  play_message.Session = session;
                  rtsp_client.SendMessage(play_message);
              }
    
              if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
              {
                  // Got Reply to PLAY
                  Console.WriteLine("Got reply from Play  " + message.Command);
              }
          }
  • STEP 5 - Handle RTP Video

    This code handles each incoming RTP packet, combining RTP packets that are all part of the same frame of vdeo (using the Marker Bit). Once a full frame is received it can be passed to a De-packetiser to get the compressed video data

          List<byte[]> temporary_rtp_payloads = new List<byte[]>();
    
          private void Rtsp_client_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
          {
              // 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);
                  rtp_payload_start += 4 + (int)rtp_extension_size;  // extension header and extension payload
              }
    
              Console.WriteLine("RTP Data"
                                 + " V=" + rtp_version
                                 + " P=" + rtp_padding
                                 + " X=" + rtp_extension
                                 + " CC=" + rtp_csrc_count
                                 + " M=" + rtp_marker
                                 + " PT=" + rtp_payload_type
                                 + " Seq=" + rtp_sequence_number
                                 + " Time=" + rtp_timestamp
                                 + " SSRC=" + rtp_ssrc
                                 + " Size=" + e.Message.Data.Length);
    
    
              if (rtp_payload_type != video_payload)
              {
                  Console.WriteLine("Ignoring this RTP payload");
                  return; // ignore this data
              }
    
    
              // If rtp_marker is '1' then this is the final transmission for this packet.
              // If rtp_marker is '0' we need to accumulate data with the same timestamp
    
              // ToDo - Check Timestamp matches
    
              // Add to the tempoary_rtp List
              byte[] rtp_payload = new byte[e.Message.Data.Length - rtp_payload_start]; // payload with RTP header removed
              System.Array.Copy(e.Message.Data, rtp_payload_start, rtp_payload, 0, rtp_payload.Length); // copy payload
              temporary_rtp_payloads.Add(rtp_payload);
    
              if (rtp_marker == 1)
              {
                  // Process the RTP frame
                  Process_RTP_Frame(temporary_rtp_payloads);
                  temporary_rtp_payloads.Clear();
              }
          }
  • STEP 6 - Process RTP frame

    An RTP frame consists of 1 or more RTP packets
    H264 video is packed into one or more RTP packets and this sample extracts Normal Packing and Fragmented Unit type A packing (the common two)
    This example writes the video to a .264 file which can be played with FFPLAY

          FileStream fs = null;
          byte[] nal_header = new byte[]{ 0x00, 0x00, 0x00, 0x01 };
          int norm, fu_a, fu_b, stap_a, stap_b, mtap16, mtap24 = 0; // stats counters
    
          public void Process_RTP_Frame(List<byte[]>rtp_payloads)
          {
              Console.WriteLine("RTP Data comprised of " + rtp_payloads.Count + " rtp packets");
    
              if (fs == null)
              {
                  // Create the file
                  String filename = "rtsp_capture_" + DateTime.Now.ToString("yyyyMMdd_HHmmss") + ".h264";
                  fs = new FileStream(filename, FileMode.Create);
                  
                  // TODO. Get SPS and PPS from the SDP Attributes (the fmtp attribute) and write to the file
                  // for IP cameras that only out the SPS and PPS out-of-band
              }
    
              for (int payload_index = 0; payload_index < rtp_payloads.Count; payload_index++) {
                  // Examine the first rtp_payload and the first byte (the NAL header)
                  int nal_header_f_bit = (rtp_payloads[payload_index][0] >> 7) & 0x01;
                  int nal_header_nri = (rtp_payloads[payload_index][0] >> 5) & 0x03;
                  int nal_header_type = (rtp_payloads[payload_index][0] >> 0) & 0x1F;
    
                  // If the NAL Header Type is in the range 1..23 this is a normal NAL (not fragmented)
                  // So write the NAL to the file
                  if (nal_header_type >= 1 && nal_header_type <= 23)
                  {
                      Console.WriteLine("Normal NAL");
                      norm++;
                      fs.Write(nal_header, 0, nal_header.Length);
                      fs.Write(rtp_payloads[payload_index], 0, rtp_payloads[payload_index].Length);
                  }
                  else if (nal_header_type == 24)
                  {
                      // There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
                      Console.WriteLine("Agg STAP-A not supported");
                      stap_a++;
                  }
                  else if (nal_header_type == 25)
                  {
                      // There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
                      Console.WriteLine("Agg STAP-B not supported");
                      stap_b++;
                  }
                  else if (nal_header_type == 26)
                  {
                      // There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
                      Console.WriteLine("Agg MTAP16 not supported");
                      mtap16++;
                  }
                  else if (nal_header_type == 27)
                  {
                      // There are 4 types of Aggregation Packet (multiple NALs in one RTP packet)
                      Console.WriteLine("Agg MTAP24 not supported");
                      mtap24++;
                  }
                  else if (nal_header_type == 28)
                  {
                      Console.WriteLine("Fragmented Packet Type FU-A");
                      fu_a++;
    
                      // Parse Fragmentation Unit Header
                      int fu_header_s = (rtp_payloads[payload_index][1] >> 7) & 0x01;  // start marker
                      int fu_header_e = (rtp_payloads[payload_index][1] >> 6) & 0x01;  // end marker
                      int fu_header_r = (rtp_payloads[payload_index][1] >> 5) & 0x01;  // reserved. should be 0
                      int fu_header_type = (rtp_payloads[payload_index][1] >> 0) & 0x1F; // Original NAL unit header
    
                      Console.WriteLine("Frag FU-A s="+fu_header_s + "e="+fu_header_e);
    
                      // Start Flag set
                      if (fu_header_s == 1)
                      {
                          // Write 00 00 00 01 header
                          fs.Write(nal_header, 0, nal_header.Length); // 0x00 0x00 0x00 0x01
    
                          // Modify the NAL Header that was at the start of the RTP packet
                          // Keep the F and NRI flags but substitute the type field with the fu_header_type
                          byte reconstructed_nal_type = (byte)((nal_header_nri << 5) + fu_header_type);
                          fs.WriteByte(reconstructed_nal_type); // NAL Unit Type
                          fs.Write(rtp_payloads[payload_index], 2, rtp_payloads[payload_index].Length - 2); // start after NAL Unit Type and FU Header byte
                      }
    
                      if (fu_header_s == 0)
                      {
                          // append this payload to the output NAL stream
                          // Data starts after the NAL Unit Type byte and the FU Header byte
    
                          fs.Write(rtp_payloads[payload_index], 2, rtp_payloads[payload_index].Length-2); // start after NAL Unit Type and FU Header byte
                      }
                  }
    
                  else if (nal_header_type == 29)
                  {
                      Console.WriteLine("Fragmented Packet  FU-B not supported");
                      fu_b++;
                  }
                  else
                  {
                      Console.WriteLine("Unknown NAL header " + nal_header_type);
                  }
    
              }
              // ensure video is written to disk
              fs.Flush(true);
              
              // Print totals
              Console.WriteLine("Norm=" + norm + " ST-A=" + stap_a + " ST-B=" + stap_b + " M16=" + mtap16 + " M24=" + mtap24 + " FU-A=" + fu_a + " FU-B=" + fu_b);
          }

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