/// <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)
{
EventHandler <RtspChunkEventArgs> handler = DataReceived;
if (handler != null)
{
handler(this, rtspChunkEventArgs);
}
}
/// <summary>
/// Raises the <see cref="E:MessageReceived"/> event.
/// </summary>
/// <param name="e">The <see cref="Rtsp.RtspChunkEventArgs"/> instance containing the event data.</param>
protected void OnMessageReceived(RtspChunkEventArgs e)
{
EventHandler <RtspChunkEventArgs> handler = MessageReceived;
if (handler != null)
{
handler(this, e);
}
}
private void ProcessRTPVideo(Rtsp.RtspChunkEventArgs e)
{
// Received some Video 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 rtpVersion = (e.Message.Data[0] >> 6);
int rtpPadding = (e.Message.Data[0] >> 5) & 0x01;
int rtpRxtension = (e.Message.Data[0] >> 4) & 0x01;
int rtpCSRCCount = (e.Message.Data[0] >> 0) & 0x0F;
int rtpMarker = (e.Message.Data[1] >> 7) & 0x01;
int rtpPayloadType = (e.Message.Data[1] >> 0) & 0x7F;
uint rtpSequenceNumber = ((uint)e.Message.Data[2] << 8) + (uint)(e.Message.Data[3]);
uint rtpTimestamp = ((uint)e.Message.Data[4] << 24) + (uint)(e.Message.Data[5] << 16) + (uint)(e.Message.Data[6] << 8) + (uint)(e.Message.Data[7]);
uint rtpSSRC = ((uint)e.Message.Data[8] << 24) + (uint)(e.Message.Data[9] << 16) + (uint)(e.Message.Data[10] << 8) + (uint)(e.Message.Data[11]);
int rtpPayloadStart = 4 // V,P,M,SEQ
+ 4 // time stamp
+ 4 // ssrc
+ (4 * rtpCSRCCount); // zero or more csrcs
if (rtpRxtension == 1)
{
uint rtpExtensionId = ((uint)e.Message.Data[rtpPayloadStart + 0] << 8) + (uint)(e.Message.Data[rtpPayloadStart + 1] << 0);
uint rtpExtensionSize = ((uint)e.Message.Data[rtpPayloadStart + 2] << 8) + (uint)(e.Message.Data[rtpPayloadStart + 3] << 0) * 4; // units of extension_size is 4-bytes
rtpPayloadStart += 4 + (int)rtpExtensionSize; // extension header and extension payload
}
//Logger.Info("RTP Data"
// + " V=" + rtpVersion
// + " P=" + rtpPadding
// + " X=" + rtpRxtension
// + " CC=" + rtpCSRCCount
// + " M=" + rtpMarker
// + " PT=" + rtpPayloadType
// + " Seq=" + rtpSequenceNumber
// + " Time (MS)=" + rtpTimestamp / 90 // convert from 90kHZ clock to ms
// + " SSRC=" + rtpSSRC
// + " Size=" + e.Message.Data.Length);
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (videoPayloadType > 0 && rtpPayloadType != videoPayloadType)
{
Logger.Info("Ignoring this RTP payload: " + rtpPayloadType);
return; // ignore this data
}
byte[] rtp_payload = new byte[e.Message.Data.Length - rtpPayloadStart]; // payload with RTP header removed
Array.Copy(e.Message.Data, rtpPayloadStart, rtp_payload, 0, rtp_payload.Length); // copy payload
PushChunk(rtp_payload);
}
/// <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);
}
}
/// <summary>
/// Handles the MessageReceived event of the Listener control.
/// </summary>
/// <param name="sender">The source of the event.</param>
/// <param name="e">The <see cref="RTSP.RTSPChunkEventArgs"/> instance containing the event data.</param>
private void Listener_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Contract.Requires(e.Message != null);
this.Enqueue(e.Message as RtspMessage);
}
/// <summary>
/// Raises the <see cref="E:MessageReceived"/> event.
/// </summary>
/// <param name="e">The <see cref="Rtsp.RtspChunkEventArgs"/> instance containing the event data.</param>
protected void OnMessageReceived(RtspChunkEventArgs e)
{
EventHandler<RtspChunkEventArgs> handler = MessageReceived;
if (handler != null)
handler(this, e);
}
/// <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)
{
EventHandler<RtspChunkEventArgs> handler = DataReceived;
if (handler != null)
handler(this, rtspChunkEventArgs);
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header and what type it is
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
// Parse www_authenticate
// EG: WWW-Authenticate: Basic realm="xxxxxxx"
// EG: WWW-Authenticate: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string[] items = www_authenticate.Split(new char[] { ',', ' ' }); // split on Comma and Space
// Process the first item
if (items.Count() >= 1 && items[0].Equals("Basic"))
{
authentication = AUTHENTICATION.BASIC;
}
else if (items.Count() >= 1 && items[0].Equals("Digest"))
{
authentication = AUTHENTICATION.DIGEST;
}
// Process the remaining items
for (int i = 1; i < items.Count(); i++)
{
string[] parts = items[i].Split(new char[] { '=' }); // Split on Equals
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we do not add any authorization (and we could not add it even if we wanted to
// as we will not have the authorization Nonce value required for Digest mode
// So we have to handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the original request did not use Authorization so try again with Authorization added
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "DESCRIBE");
if (authorization_string != null)
{
describe_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
// If the attribute is for Video, then carry out a SETUP and a PLAY
// Only do this for the first Video attribute in case there is more than one in the SDP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video)
{
// We only want the first video sub-stream
if (video_payload == -1)
{
video_payload = sdp_data.Medias[x].PayloadType;
// search the attributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264)
Rtsp.Sdp.AttributRtpMap rtpmap = null; // custom payload (>=96) details
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (rtpmap != null && rtpmap.Value.Contains("H264") && fmtp != null)
{
video_codec = "H264";
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
Output_SPS_PPS(sps, pps); // output SPS and PPS
}
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0; // Used in DataReceived event handler
video_rtcp_channel = 1; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
video_data_channel = udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = udp_pair.control_port; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(video_data_channel, video_rtcp_channel), // a Channel for video. a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "SETUP");
if (authorization_string != null)
{
setup_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(setup_message);
}
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udp_pair = new UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
udp_pair.DataReceived += Rtp_DataReceived;
udp_pair.Start();
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
if (authentication != AUTHENTICATION.NONE)
{
String authorization_string = GenerateAuthorization(username, password, authentication,
realm, nonce, url, "PLAY");
if (authorization_string != null)
{
play_message.Headers.Add("Authorization", authorization_string);
}
}
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
int rtp_count = 0; // used for statistics
// RTP packet (or RTCP packet) has been received.
private void Rtp_DataReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspData data_received = e.Message as Rtsp.Messages.RtspData;
// 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 (data_received.Channel == video_rtcp_channel)
{
Console.WriteLine("Received a RTCP message on channel " + data_received.Channel);
return;
}
if (data_received.Channel == video_data_channel)
{
// Received some Video 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
}
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 (MS)=" + rtp_timestamp / 90 // convert from 90kHZ clock to ms
+ " SSRC=" + rtp_ssrc
+ " Size=" + e.Message.Data.Length);
String msg = "RTP Data " + rtp_count++
+ " 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 (fs2 != null)
{
fs2.WriteLine(msg);
}
if (fs2 != null)
{
fs2.Flush();
}
// Check the payload type in the RTP packet matches the Payload Type value from the SDP
if (rtp_payload_type != video_payload)
{
Console.WriteLine("Ignoring this RTP payload");
return; // ignore this data
}
if (rtp_payload_type >= 96 && rtp_payload_type <= 127 && video_codec.Equals("H264"))
{
// H264 RTP Packet
// 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
// Add the RTP packet to the tempoary_rtp list until we have a complete 'Frame'
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
List <byte[]> nal_units = h264Payload.Process_H264_RTP_Packet(rtp_payload, rtp_marker); // this will cache the Packets until there is a Frame
if (nal_units == null)
{
// we have not passed in enough RTP packets to make a Frame of video
}
else
{
// we have a frame of NAL Units. Write them to the file
Output_NAL(nal_units);
}
}
else if (rtp_payload_type == 26)
{
Console.WriteLine("No parser for JPEG RTP packets");
}
else
{
Console.WriteLine("No parser for this RTP payload");
}
}
}
// 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];
// 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
RTPSession new_session = new RTPSession();
new_session.session_id = session_count.ToString();
new_session.listener = listener;
new_session.sequence_number = (UInt16)rnd.Next(65535); // start with a random 16 bit sequence number
new_session.ssrc = 1;
// 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)
{
// Add the transports to the Session
new_session.client_transport = transport;
new_session.transport_reply = transport_reply;
// 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);
}
}
/// <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);
}
/// <summary>
/// Raises the <see cref="E:MessageReceived"/> event.
/// </summary>
/// <param name="e">The <see cref="Rtsp.RtspChunkEventArgs"/> instance containing the event data.</param>
protected void OnMessageReceived(RtspChunkEventArgs e)
{
MessageReceived?.Invoke(this, e);
}
