internal RtspResponse HandlePullSetup(RtspRequestSetup request)
{
var response = request.CreateResponse();
if (string.IsNullOrEmpty(response.Session))
{
// TODO Allocate a real session ID
response.Session = sessionGenerator.Next().ToString();
}
var pushUri = GetPushUri(request.RtspUri.AbsolutePath);
RtspPushDescription description;
if (PushDescriptions.TryGetValue(pushUri, out description))
{
//TODO get port and multicast address from description.
var forwarder = description.GetForwarderFor(pushUri);
var transport = new RtspTransport();
RtspTransport newTransport = new RtspTransport()
{
IsMulticast = true,
Destination = forwarder.ForwardHostVideo,
Port = new PortCouple(forwarder.ForwardPortVideo, forwarder.ListenCommandPort)
};
response.Headers[RtspHeaderNames.Transport] = newTransport.ToString();
}
else
{
response.ReturnCode = 404;
}
return(response);
}
/// <summary>
/// Selects the transport based on the configuration of the system..
/// </summary>
/// <param name="requestSetup">The request setup message.</param>
/// <returns>The selected transport</returns>
/// <remarks>
/// The transport is selected by taking the first supported transport
/// in order of appearence.
/// </remarks>
private static RtspTransport SelectTransport(RtspRequestSetup requestSetup)
{
RtspTransport selectedTransport = null;
const bool acceptTCP = false;
const bool acceptUDPUnicast = true;
const bool acceptUDPMulticast = true;
foreach (RtspTransport proposedTransport in requestSetup.GetTransports())
{
if (acceptTCP && proposedTransport.LowerTransport == RtspTransport.LowerTransportType.TCP)
{
selectedTransport = proposedTransport;
break;
}
if (acceptUDPMulticast && proposedTransport.LowerTransport == RtspTransport.LowerTransportType.UDP &&
proposedTransport.IsMulticast)
{
selectedTransport = proposedTransport;
break;
}
if (acceptUDPUnicast && proposedTransport.LowerTransport == RtspTransport.LowerTransportType.UDP &&
!proposedTransport.IsMulticast)
{
selectedTransport = proposedTransport;
break;
}
}
return(selectedTransport);
}
public void DefaultValue()
{
RtspTransport testValue = new RtspTransport();
Assert.IsTrue(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.UDP, testValue.LowerTransport);
Assert.AreEqual("PLAY", testValue.Mode);
}
private void ProcessSetupRequest(RtspResponse message)
{
// If we get a reply to SETUP (which was our third command), then process then send PLAY
// Got Reply to SETUP
Logger.Info("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 multicastAddress = transport.Destination;
videoDataChannel = transport.Port.First;
videoRTCPChannel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udpPair = new UDPSocketPair(multicastAddress, videoDataChannel, multicastAddress, videoRTCPChannel);
udpPair.DataReceived += RtpDataReceived;
udpPair.Start();
}
}
RtspRequest play_message = new RtspRequestPlay
{
RtspUri = new Uri(url),
Session = session
};
rtspListener.SendMessage(play_message);
}
public void DefaultValue()
{
RtspTransport testValue = new RtspTransport();
Assert.IsTrue(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.UDP, testValue.LowerTransport);
Assert.AreEqual("PLAY", testValue.Mode);
}
public void Parse1()
{
RtspTransport testValue = RtspTransport.Parse("RTP/AVP/UDP;destination");
Assert.IsTrue(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.UDP, testValue.LowerTransport);
Assert.AreEqual("PLAY", testValue.Mode);
}
public void Parse3()
{
RtspTransport testValue = RtspTransport.Parse("RTP/AVP/TCP;interleaved=3-4");
Assert.IsFalse(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.TCP, testValue.LowerTransport);
Assert.AreEqual(3, testValue.Interleaved.First);
Assert.IsTrue(testValue.Interleaved.IsSecondPortPresent);
Assert.AreEqual(4, testValue.Interleaved.Second);
}
public void ToStringTCP()
{
RtspTransport transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(0, 1),
};
Assert.AreEqual("RTP/AVP/TCP;interleaved=0-1", transport.ToString());
}
public void Parse2()
{
RtspTransport testValue = RtspTransport.Parse("RTP/AVP/TCP;multicast;destination=test.example.com;ttl=234;ssrc=cd3b20a5");
Assert.IsTrue(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.TCP, testValue.LowerTransport);
Assert.AreEqual("test.example.com", testValue.Destination);
Assert.AreEqual("cd3b20a5", testValue.SSrc);
Assert.AreEqual("PLAY", testValue.Mode);
}
/// <summary>
/// Asynchronously plays a stream.
/// </summary>
/// <param name="url">The url of a stream to play.</param>
/// <param name="connectionTimeout">The connection timeout.</param>
/// <exception cref="StreamPlayerException">Failed to play the stream.</exception>
/// <param name="transport">RTSP transport protocol.</param>
/// <param name="flags">RTSP flags.</param>
internal void StartPlay(String url, TimeSpan connectionTimeout,
RtspTransport transport, RtspFlags flags)
{
if (_startPlayDelegate(url,
Convert.ToInt32(connectionTimeout.TotalMilliseconds),
Convert.ToInt32(transport),
Convert.ToInt32(flags)) != 0)
{
throw new StreamPlayerException("Failed to play the stream.");
}
}
/// <summary>
/// Asynchronously plays a stream.
/// </summary>
/// <param name="uri">The uri of a stream to play.</param>
/// <param name="connectionTimeout"></param>
/// <exception cref="Win32Exception">Failed to load the FFmpeg facade dll.</exception>
/// <exception cref="StreamPlayerException">Failed to play the stream.</exception>
/// <param name="transport">RTSP transport protocol.</param>
/// <param name="flags">RTSP flags.</param>
public void StartPlay(Uri uri, TimeSpan connectionTimeout,
RtspTransport transport, RtspFlags flags, FrameDelay framedelay)
{
if (IsPlaying)
{
Stop();
}
Player.StartPlay(uri.IsFile ? uri.LocalPath : uri.ToString(),
connectionTimeout, transport, flags, framedelay);
}
public void Parse4()
{
RtspTransport testValue = RtspTransport.Parse("RTP/AVP;unicast;destination=1.2.3.4;source=3.4.5.6;server_port=5000-5001;client_port=5003-5004");
Assert.IsFalse(testValue.IsMulticast);
Assert.AreEqual(RtspTransport.LowerTransportType.UDP, testValue.LowerTransport);
Assert.AreEqual("1.2.3.4", testValue.Destination);
Assert.AreEqual("3.4.5.6", testValue.Source);
Assert.AreEqual(5000, testValue.ServerPort.First);
Assert.AreEqual(5001, testValue.ServerPort.Second);
Assert.AreEqual(5003, testValue.ClientPort.First);
Assert.AreEqual(5004, testValue.ClientPort.Second);
}
public void ToStringUDPUnicast()
{
RtspTransport transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(5000, 5001),
ServerPort = new PortCouple(5002, 5003),
Destination = "1.2.3.4"
};
Assert.AreEqual("RTP/AVP/UDP;unicast;destination=1.2.3.4;client_port=5000-5001;server_port=5002-5003", transport.ToString());
}
public void StartReadStream()
{
try
{
LoadIni();
Uri u = new Uri(m_strUri);
TimeSpan ts = new TimeSpan(0, 0, 0, 0, m_nTimeOut);
RtspTransport tsport = (RtspTransport)m_nTransPort;
m_streamControl.StartPlay(u, ts, tsport, RtspFlags.None);
m_camStatus = CamStatus.Connecting;
}
catch (Exception)
{
}
}
public void StartReadStream(string uri, int timeout_ms, int transport = 1) //transport : 1-TCP, 2-UDP, 0-NONE
{
m_strUri = uri;
m_nTimeOut = timeout_ms;
m_nTransPort = transport;
Uri u = new Uri(uri);
TimeSpan ts = new TimeSpan(0, 0, 0, 0, timeout_ms);
RtspTransport tsport = (RtspTransport)transport;
IniFile ini = new IniFile();
ini.IniWriteValue(INISECT_RTSP, INIKEY_URI, uri, DefPath.VisionSetting);
ini.IniWriteValue(INISECT_RTSP, INIKEY_TIMEOUT, timeout_ms.ToString(), DefPath.VisionSetting);
ini.IniWriteValue(INISECT_RTSP, INIKEY_TRANSPORT, transport.ToString(), DefPath.VisionSetting);
m_streamControl.StartPlay(u, ts, tsport, RtspFlags.None);
m_camStatus = CamStatus.Connecting;
}
private void ProcessSetupResponse(RtspResponse message)
{
rtspSession = message.Session; // Session value used with Play, Pause, Teardown
Logger.Info($"RTSP session: {rtspSession}");
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
if (transport.IsMulticast)
{
string multicastAddress = transport.Destination;
videoDataChannel = transport.Port.First;
videoRTCPChannel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udpSocketPair = new UDPSocketPair(multicastAddress, videoDataChannel, multicastAddress, videoRTCPChannel);
udpSocketPair.DataReceived += RtpDataReceived;
udpSocketPair.Start();
}
}
if (_suspendTransfer)
{
// Transfer suspended. Set state to playing but do not issue play request.
// Play request will be sent upon resume.
_currentState = State.Playing;
return;
}
_currentState = State.Paused;
PostRequest(new RtspRequestPlay
{
RtspUri = new Uri(rtspUrl),
Session = rtspSession
});
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// Check if the Message has an Authenticate header. If so we update the 'realm' and 'nonce'
if (message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
// Parse www_authenticate
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
// Split on the = symbol and load in
string[] parts = item.Split(new char[] { '=' });
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
Console.WriteLine("WWW Authorize parsed for " + realm + " " + nonce);
}
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the DESCRIBE. First time around we have no WWW-Authorise
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// First time we send DESCRIBE we will not have the authorization Nonce so we
// handle the Unauthorized 401 error here and send a new DESCRIBE message
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == false))
{
// Error 401 - Unauthorized, but the request did not use Authorizarion.
if (username == null || password == null)
{
// we do nothave a username or password. Abort
return;
}
// Send a new DESCRIBE with authorization
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "DESCRIBE");
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (digest_authorization != null)
{
describe_message.Headers.Add(RtspHeaderNames.Authorization, digest_authorization);
}
rtsp_client.SendMessage(describe_message);
return;
}
else if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// Authorization failed
return;
}
else
{
// some other error
return;
}
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
int data_channel = 0;
int rtcp_channel = 0;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = 0;
video_rtcp_channel = 1;
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = 2;
audio_rtcp_channel = 3;
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(data_channel, rtcp_channel), // Eg Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
data_channel = video_data_channel;
rtcp_channel = video_rtcp_channel;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
data_channel = audio_data_channel;
rtcp_channel = audio_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(data_channel, rtcp_channel), // a Channel for data (video or audio). a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Add authorization (if there is a username and password)
String digest_authorization = GenerateDigestAuthorization(username, password,
realm, nonce, url, "SETUP");
// Send SETUP
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (digest_authorization != null)
{
setup_message.Headers.Add("Authorization", digest_authorization);
}
rtsp_client.SendMessage(setup_message);
}
}
}
// If we get a reply to SETUP (which was our third command), then process and then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
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);
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
void Rtsp_MessageReceived(object sender, RtspChunkEventArgs e)
{
var message = e.Message as RtspResponse;
//logger.Debug ("Received " + message.OriginalRequest.Method);
// If message has a 401 - Unauthorised Error, then we re-send the message with Authorization
// using the most recently received 'realm' and 'nonce'
if (!message.IsOk)
{
logger.Warn("Got Error in RTSP Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// the authorization failed.
Stop();
return;
}
// Check if the Reply has an Authenticate header.
if (message.ReturnCode == 401 && message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
// Process the WWW-Authenticate header
// EG: Basic realm="AProxy"
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
string www_authenticate = message.Headers [RtspHeaderNames.WWWAuthenticate];
string [] items = www_authenticate.Split(new char [] { ',', ' ' });
foreach (string item in items)
{
if (item.ToLower().Equals("basic"))
{
auth_type = "Basic";
}
else if (item.ToLower().Equals("digest"))
{
auth_type = "Digest";
}
else
{
// Split on the = symbol and update the realm and nonce
string [] parts = item.Split(new char [] { '=' }, 2); // max 2 parts in the results array
if (parts.Count() >= 2 && parts [0].Trim().Equals("realm"))
{
realm = parts [1].Trim(new char [] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts [0].Trim().Equals("nonce"))
{
nonce = parts [1].Trim(new char [] { ' ', '\"' }); // trim space and quotes
}
}
}
//logger.Debug ("WWW Authorize parsed for " + auth_type + " " + realm + " " + nonce);
}
var resend_message = message.OriginalRequest.Clone() as RtspMessage;
if (auth_type != null)
{
AddAuthorization(resend_message, Username, Password, auth_type, realm, nonce, Url);
}
rtsp_client.SendMessage(resend_message);
return;
}
// If we get a reply to OPTIONS then start the Keepalive Timer and send DESCRIBE
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestOptions)
{
string public_methods = message.Headers [RtspHeaderNames.Public];
SupportedMethods.Clear();
foreach (string method in public_methods.Split(','))
{
SupportedMethods.Add(method.Trim());
}
if (keepalive_timer == null)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 5 * 1000;
keepalive_timer.Enabled = true;
// Send DESCRIBE
var describe_message = new RtspRequestDescribe();
describe_message.RtspUri = new Uri(Url);
describe_message.AddAccept("application/sdp");
if (auth_type != null)
{
AddAuthorization(describe_message, Username, Password, auth_type, realm, nonce, Url);
}
rtsp_client.SendMessage(describe_message);
}
}
// 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 Mictlanix.DotNet.Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
if (!message.IsOk)
{
logger.Warn("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
// Examine the SDP
//logger.Debug (Encoding.UTF8.GetString (message.Data));
Sdp.SdpFile sdp_data;
using (var sdp_stream = new StreamReader(new MemoryStream(message.Data))) {
sdp_data = Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool video = (sdp_data.Medias [x].MediaType == Mictlanix.DotNet.Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
string control = ""; // the "track" or "stream id"
Sdp.FmtpAttribute fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (var attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
string sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://", StringComparison.Ordinal))
{
control = sdp_control; //absolute path
}
else if (message.Headers.ContainsKey(RtspHeaderNames.ContentBase))
{
control = message.Headers [RtspHeaderNames.ContentBase] + sdp_control; // relative path
}
else
{
control = Url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Sdp.FmtpAttribute;
}
if (attrib.Key.Equals("rtpmap"))
{
var rtpmap = attrib as Sdp.RtpMapAttribute;
// Check if the Codec Used (EncodingName) is one we support
string [] valid_video_codecs = { "H264" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias [x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = 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];
ParameterSetsReceived?.Invoke(sps, pps);
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
// Server interleaves the RTP packets over the RTSP connection
// TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0;
video_rtcp_channel = 1;
var transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Eg Channel 0 for video. Channel 1 for RTCP status reports
};
// Send SETUP
var setup_message = new RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (auth_type != null)
{
AddAuthorization(setup_message, Username, Password, auth_type, realm, nonce, Url);
}
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 RtspRequestSetup)
{
// Got Reply to SETUP
if (!message.IsOk)
{
logger.Warn("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
//logger.Debug ("Got reply from SETUP Session=" + message.Session);
RtspSession = message.Session; // Session value used with Play, Pause, Teardown
// Send PLAY
RtspRequest play_message = new RtspRequestPlay();
play_message.RtspUri = new Uri(Url);
play_message.Session = RtspSession;
if (auth_type != null)
{
AddAuthorization(play_message, Username, Password, auth_type, realm, nonce, Url);
}
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is RtspRequestPlay)
{
// Got Reply to PLAY
if (!message.IsOk)
{
logger.Warn("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
//logger.Debug ("Got reply from PLAY " + message.Command);
}
}
private void RTSP_ProcessSetupRequest(RtspRequestSetup message, RtspListener listener)
{
//
var setupMessage = message;
// Check the RTSP transport
// If it is UDP or Multicast, create the sockets
// If it is RTP over RTSP we send data via the RTSP Listener
// FIXME client may send more than one possible transport.
// very rare
RtspTransport transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
Rtsp.UDPSocket udp_pair;
RtspTransport transport_reply = RTSP_ConstructReplyTransport(transport, out udp_pair);
bool mediaTransportSet = false;
if (transport_reply != null)
{
// Update the session with transport information
String copy_of_session_id = "";
// ToDo - Check the Track ID to determine if this is a SETUP for the Video Stream
// or a SETUP for an Audio Stream.
// In the SDP the H264 video track is TrackID 0
// Add the transports to the connection
if (contentBase != null)
{
string controlTrack = setupMessage.RtspUri.AbsoluteUri.Replace(contentBase, string.Empty);
var requestMedia = _sdpFile.Medias.FirstOrDefault(media =>
media.Attributs.FirstOrDefault(a => a.Key == "control" &&
(a.Value == controlTrack || "/" + a.Value == controlTrack)) != null);
if (requestMedia != null)
{
if (requestMedia.MediaType == Media.MediaTypes.video)
{
_videoClientTransport = transport;
_videoTransportReply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
if (_videoUdpPair != null)
{
ReleaseUDPSocket(_videoUdpPair);
}
_videoUdpPair = udp_pair;
mediaTransportSet = true;
if (setupMessage.Session == null)
{
_videoSessionId = _sessionHandle.ToString();
_sessionHandle++;
}
else
{
_videoSessionId = setupMessage.Session;
}
// Copy the Session ID
copy_of_session_id = _videoSessionId;
}
if (requestMedia.MediaType == Media.MediaTypes.audio)
{
_audioClientTransport = transport;
_audioTransportReply = transport_reply;
// If we are sending in UDP mode, add the UDP Socket pair and the Client Hostname
if (_audioUdpPair != null)
{
ReleaseUDPSocket(_audioUdpPair);
}
_audioUdpPair = udp_pair;
mediaTransportSet = true;
if (setupMessage.Session == null)
{
_audioSessionId = _sessionHandle.ToString();
_sessionHandle++;
}
else
{
_audioSessionId = setupMessage.Session;
}
// Copy the Session ID
copy_of_session_id = _audioSessionId;
}
}
}
if (false == mediaTransportSet)
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
// unsuported mediatime
setup_response.ReturnCode = 415;
listener.SendMessage(setup_response);
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
setup_response.Headers[Rtsp.Messages.RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = copy_of_session_id;
setup_response.Timeout = timeout_in_seconds;
listener.SendMessage(setup_response);
}
}
else
{
Rtsp.Messages.RtspResponse setup_response = setupMessage.CreateResponse(_logger);
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
if (false == mediaTransportSet)
{
if (udp_pair != null)
{
ReleaseUDPSocket(udp_pair);
udp_pair = null;
}
}
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// If we get a reply to OPTIONS and CSEQ is 1 (which was our first command), then send the DESCRIBE
// If we fer a reply to OPTIONS and CSEQ is not 1, it must have been a keepalive command
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (message.CSeq == 1)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// send the Describe
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// Process each 'Media' Attribute in the SDP (each sub-stream)
// If the attribute is for Video, then carry out a SETUP and a PLAY
// Only do this for the first Video attribute in case there is more than one in the SDP
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
if (sdp_data.Medias[x].GetMediaType() == Rtsp.Sdp.Media.MediaType.video)
{
// We only want the first video sub-stream
if (video_payload == -1)
{
// seach the atributes for control, fmtp and rtpmap
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64
Rtsp.Sdp.AttributRtpMap rtpmap = null; // holds Payload format, eg 96 often used with H264 as first dynamic payload value
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
control = attrib.Value;
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
}
}
// Split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count > 0)
{
video_sps = sps_pps[0];
}
if (sps_pps.Count > 1)
{
video_pps = sps_pps[1];
}
Output_NAL(sps_pps); // output SPS and PPS
}
// Split the rtpmap to get the Payload Type
video_payload = 0;
if (rtpmap != null)
{
video_payload = rtpmap.PayloadNumber;
}
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(url + "/" + control);
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
video_data_channel = 0; // Used in DataReceived event handler
video_rtcp_channel = 1; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(video_data_channel, video_rtcp_channel), // Channel 0 for video. Channel 1 for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
video_data_channel = udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = udp_pair.control_port; // Used in DataReceived event handler
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(video_data_channel, video_rtcp_channel), // a Channel for video. a Channel for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
setup_message.AddTransport(transport);
rtsp_client.SendMessage(setup_message);
}
}
}
}
// If we get a reply to SETUP (which was our third command), then process then send PLAY
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
udp_pair = new UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
udp_pair.DataReceived += Rtp_DataReceived;
udp_pair.Start();
}
}
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
rtsp_client.SendMessage(play_message);
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
Console.WriteLine("Got reply from Play " + message.Command);
}
}
private RtspTransport RTSP_ConstructReplyTransport(RtspTransport transport, out UDPSocket udp_pair)
{
udp_pair = null;
RtspTransport transport_reply = new RtspTransport();
transport_reply.SSrc = GLOBAL_SSRC.ToString("X8"); // Convert to Hex, padded to 8 characters
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new Rtsp.Messages.PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == false)
{
Boolean udp_supported = true;
if (udp_supported)
{
// RTP over UDP mode
// Create a pair of UDP sockets - One is for the Video, one is for the RTCP
try
{
_logger.Trace($"{Id} Start creating UDPSocket");
udp_pair = new Rtsp.UDPSocket(_ipAddress, 50000, 51000); // give a range of 500 pairs (1000 addresses) to try incase some address are in use
udp_pair.DataReceived += UdpPair_DataReceived;
udp_pair.ControlReceived += UdpPair_ControlReceived;
udp_pair.Start(); // start listening for data on the UDP ports
_logger.Trace($"{Id} End creating UDPSocket");
}
catch
{
if (udp_pair != null)
{
ReleaseUDPSocket(udp_pair);
throw;
}
}
// Pass the Port of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = new Rtsp.Messages.PortCouple(udp_pair._dataPort, udp_pair._controlPort);
}
else
{
transport_reply = null;
}
}
if (transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast == true)
{
// RTP over Multicast UDP mode}
// Create a pair of UDP sockets in Multicast Mode
// Pass the Ports of the two sockets back in the reply
transport_reply.LowerTransport = Rtsp.Messages.RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new Rtsp.Messages.PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
return(transport_reply);
}
// RTSP Messages are OPTIONS, DESCRIBE, SETUP, PLAY etc
private void Rtsp_MessageReceived(object sender, Rtsp.RtspChunkEventArgs e)
{
Rtsp.Messages.RtspResponse message = e.Message as Rtsp.Messages.RtspResponse;
Console.WriteLine("Received " + message.OriginalRequest.ToString());
// If message has a 401 - Unauthorised Error, then we re-send the message with Authorization
// using the most recently received 'realm' and 'nonce'
if (message.IsOk == false)
{
Console.WriteLine("Got Error in RTSP Reply " + message.ReturnCode + " " + message.ReturnMessage);
if (message.ReturnCode == 401 && (message.OriginalRequest.Headers.ContainsKey(RtspHeaderNames.Authorization) == true))
{
// the authorization failed.
Stop();
return;
}
// Check if the Reply has an Authenticate header.
if (message.ReturnCode == 401 && message.Headers.ContainsKey(RtspHeaderNames.WWWAuthenticate))
{
// Process the WWW-Authenticate header
// EG: Basic realm="AProxy"
// EG: Digest realm="AXIS_WS_ACCC8E3A0A8F", nonce="000057c3Y810622bff50b36005eb5efeae118626a161bf", stale=FALSE
String www_authenticate = message.Headers[RtspHeaderNames.WWWAuthenticate];
string[] items = www_authenticate.Split(new char[] { ',', ' ' });
foreach (string item in items)
{
if (item.ToLower().Equals("basic"))
{
auth_type = "Basic";
}
else if (item.ToLower().Equals("digest"))
{
auth_type = "Digest";
}
else
{
// Split on the = symbol and update the realm and nonce
string[] parts = item.Split(new char[] { '=' }, 2); // max 2 parts in the results array
if (parts.Count() >= 2 && parts[0].Trim().Equals("realm"))
{
realm = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
else if (parts.Count() >= 2 && parts[0].Trim().Equals("nonce"))
{
nonce = parts[1].Trim(new char[] { ' ', '\"' }); // trim space and quotes
}
}
}
Console.WriteLine("WWW Authorize parsed for " + auth_type + " " + realm + " " + nonce);
}
RtspMessage resend_message = message.OriginalRequest.Clone() as RtspMessage;
if (auth_type != null)
{
AddAuthorization(resend_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(resend_message);
return;
}
// If we get a reply to OPTIONS then start the Keepalive Timer and send DESCRIBE
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestOptions)
{
if (keepalive_timer == null)
{
// Start a Timer to send an OPTIONS command (for keepalive) every 20 seconds
keepalive_timer = new System.Timers.Timer();
keepalive_timer.Elapsed += Timer_Elapsed;
keepalive_timer.Interval = 20 * 1000;
keepalive_timer.Enabled = true;
// Send DESCRIBE
Rtsp.Messages.RtspRequest describe_message = new Rtsp.Messages.RtspRequestDescribe();
describe_message.RtspUri = new Uri(url);
if (auth_type != null)
{
AddAuthorization(describe_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(describe_message);
}
else
{
// do nothing
}
}
// If we get a reply to DESCRIBE (which was our second command), then prosess SDP and send the SETUP
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestDescribe)
{
// Got a reply for DESCRIBE
if (message.IsOk == false)
{
Console.WriteLine("Got Error in DESCRIBE Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
// Examine the SDP
Console.Write(System.Text.Encoding.UTF8.GetString(message.Data));
Rtsp.Sdp.SdpFile sdp_data;
using (StreamReader sdp_stream = new StreamReader(new MemoryStream(message.Data)))
{
sdp_data = Rtsp.Sdp.SdpFile.Read(sdp_stream);
}
// RTP and RTCP 'channels' are used in TCP Interleaved mode (RTP over RTSP)
int next_free_rtp_channel = 0;
int next_free_rtcp_channel = 1;
// Process each 'Media' Attribute in the SDP (each sub-stream)
for (int x = 0; x < sdp_data.Medias.Count; x++)
{
bool audio = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.audio);
bool video = (sdp_data.Medias[x].MediaType == Rtsp.Sdp.Media.MediaTypes.video);
if (video && video_payload != -1)
{
continue; // have already matched an video payload
}
if (audio && audio_payload != -1)
{
continue; // have already matched an audio payload
}
if (audio || video)
{
// search the attributes for control, rtpmap and fmtp
// (fmtp only applies to video)
String control = ""; // the "track" or "stream id"
Rtsp.Sdp.AttributFmtp fmtp = null; // holds SPS and PPS in base64 (h264 video)
foreach (Rtsp.Sdp.Attribut attrib in sdp_data.Medias[x].Attributs)
{
if (attrib.Key.Equals("control"))
{
String sdp_control = attrib.Value;
if (sdp_control.ToLower().StartsWith("rtsp://"))
{
control = sdp_control; //absolute path
}
else
{
control = url + "/" + sdp_control; // relative path
}
}
if (attrib.Key.Equals("fmtp"))
{
fmtp = attrib as Rtsp.Sdp.AttributFmtp;
}
if (attrib.Key.Equals("rtpmap"))
{
Rtsp.Sdp.AttributRtpMap rtpmap = attrib as Rtsp.Sdp.AttributRtpMap;
// Check if the Codec Used (EncodingName) is one we support
String[] valid_video_codecs = { "H264" };
String[] valid_audio_codecs = { "PCMA", "PCMU", "AMR" };
if (video && Array.IndexOf(valid_video_codecs, rtpmap.EncodingName) >= 0)
{
// found a valid codec
video_codec = rtpmap.EncodingName;
video_payload = sdp_data.Medias[x].PayloadType;
}
if (audio && Array.IndexOf(valid_audio_codecs, rtpmap.EncodingName) >= 0)
{
audio_codec = rtpmap.EncodingName;
audio_payload = sdp_data.Medias[x].PayloadType;
}
}
}
// If the rtpmap contains H264 then split the fmtp to get the sprop-parameter-sets which hold the SPS and PPS in base64
if (video && video_codec.Contains("H264") && fmtp != null)
{
var param = Rtsp.Sdp.H264Parameters.Parse(fmtp.FormatParameter);
var sps_pps = param.SpropParameterSets;
if (sps_pps.Count() >= 2)
{
byte[] sps = sps_pps[0];
byte[] pps = sps_pps[1];
if (Received_SPS_PPS != null)
{
Received_SPS_PPS(sps, pps);
}
}
}
// Send the SETUP RTSP command if we have a matching Payload Decoder
if (video && video_payload == -1)
{
continue;
}
if (audio && audio_payload == -1)
{
continue;
}
RtspTransport transport = null;
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Server interleaves the RTP packets over the RTSP connection
// Example for TCP mode (RTP over RTSP) Transport: RTP/AVP/TCP;interleaved=0-1
if (video)
{
video_data_channel = next_free_rtp_channel;
video_rtcp_channel = next_free_rtcp_channel;
}
if (audio)
{
audio_data_channel = next_free_rtp_channel;
audio_rtcp_channel = next_free_rtcp_channel;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.TCP,
Interleaved = new PortCouple(next_free_rtp_channel, next_free_rtcp_channel), // Eg Channel 0 for RTP video data. Channel 1 for RTCP status reports
};
next_free_rtp_channel += 2;
next_free_rtcp_channel += 2;
}
if (rtp_transport == RTP_TRANSPORT.UDP)
{
int rtp_port = 0;
int rtcp_port = 0;
// Server sends the RTP packets to a Pair of UDP Ports (one for data, one for rtcp control messages)
// Example for UDP mode Transport: RTP/AVP;unicast;client_port=8000-8001
if (video)
{
video_data_channel = video_udp_pair.data_port; // Used in DataReceived event handler
video_rtcp_channel = video_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = video_udp_pair.data_port;
rtcp_port = video_udp_pair.control_port;
}
if (audio)
{
audio_data_channel = audio_udp_pair.data_port; // Used in DataReceived event handler
audio_rtcp_channel = audio_udp_pair.control_port; // Used in DataReceived event handler
rtp_port = audio_udp_pair.data_port;
rtcp_port = audio_udp_pair.control_port;
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = false,
ClientPort = new PortCouple(rtp_port, rtcp_port), // a UDP Port for data (video or audio). a UDP Port for RTCP status reports
};
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Server sends the RTP packets to a Pair of UDP ports (one for data, one for rtcp control messages)
// using Multicast Address and Ports that are in the reply to the SETUP message
// Example for MULTICAST mode Transport: RTP/AVP;multicast
if (video)
{
video_data_channel = 0; // we get this information in the SETUP message reply
video_rtcp_channel = 0; // we get this information in the SETUP message reply
}
if (audio)
{
audio_data_channel = 0; // we get this information in the SETUP message reply
audio_rtcp_channel = 0; // we get this information in the SETUP message reply
}
transport = new RtspTransport()
{
LowerTransport = RtspTransport.LowerTransportType.UDP,
IsMulticast = true
};
}
// Generate SETUP messages
Rtsp.Messages.RtspRequestSetup setup_message = new Rtsp.Messages.RtspRequestSetup();
setup_message.RtspUri = new Uri(control);
setup_message.AddTransport(transport);
if (auth_type != null)
{
AddAuthorization(setup_message, username, password, auth_type, realm, nonce, url);
}
// Add SETUP message to list of mesages to send
setup_messages.Add(setup_message);
}
}
// Send the FIRST SETUP message and remove it from the list of Setup Messages
rtsp_client.SendMessage(setup_messages[0]);
setup_messages.RemoveAt(0);
}
// If we get a reply to SETUP (which was our third command), then we
// (i) check if we have any more SETUP commands to send out (eg if we are doing SETUP for Video and Audio)
// (ii) send a PLAY command if all the SETUP command have been sent
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestSetup)
{
// Got Reply to SETUP
if (message.IsOk == false)
{
Console.WriteLine("Got Error in SETUP Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Setup. Session is " + message.Session);
session = message.Session; // Session value used with Play, Pause, Teardown and and additional Setups
// Check the Transport header
if (message.Headers.ContainsKey(RtspHeaderNames.Transport))
{
RtspTransport transport = RtspTransport.Parse(message.Headers[RtspHeaderNames.Transport]);
// Check if Transport header includes Multicast
if (transport.IsMulticast)
{
String multicast_address = transport.Destination;
video_data_channel = transport.Port.First;
video_rtcp_channel = transport.Port.Second;
// Create the Pair of UDP Sockets in Multicast mode
video_udp_pair = new Rtsp.UDPSocket(multicast_address, video_data_channel, multicast_address, video_rtcp_channel);
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start();
// TODO - Need to set audio_udp_pair
}
}
// Check if we have another SETUP command to send, then remote it from the list
if (setup_messages.Count > 0)
{
// send the next SETUP message, after adding in the 'session'
Rtsp.Messages.RtspRequestSetup next_setup = setup_messages[0];
next_setup.Session = session;
rtsp_client.SendMessage(next_setup);
setup_messages.RemoveAt(0);
}
else
{
// Send PLAY
Rtsp.Messages.RtspRequest play_message = new Rtsp.Messages.RtspRequestPlay();
play_message.RtspUri = new Uri(url);
play_message.Session = session;
if (auth_type != null)
{
AddAuthorization(play_message, username, password, auth_type, realm, nonce, url);
}
rtsp_client.SendMessage(play_message);
}
}
// If we get a reply to PLAY (which was our fourth command), then we should have video being received
if (message.OriginalRequest != null && message.OriginalRequest is Rtsp.Messages.RtspRequestPlay)
{
// Got Reply to PLAY
if (message.IsOk == false)
{
Console.WriteLine("Got Error in PLAY Reply " + message.ReturnCode + " " + message.ReturnMessage);
return;
}
Console.WriteLine("Got reply from Play " + message.Command);
}
}
/// <summary>
/// Configures the transport and forwarder.
/// </summary>
/// <param name="aMessage">A message.</param>
/// <param name="forwarder">The preset forwarder.</param>
/// <returns>The configured forwarder.</returns>
private static Forwarder ConfigureTransportAndForwarder(RtspMessage aMessage, UDPForwarder forwarder)
{
RtspTransport transport = RtspTransport.Parse(aMessage.Headers[RtspHeaderNames.Transport]);
Forwarder resultForwarder;
if (transport.LowerTransport == RtspTransport.LowerTransportType.UDP)
{
if (transport.ServerPort != null)
{
forwarder.SourcePortVideo = transport.ServerPort.First;
forwarder.ForwardPortCommand = transport.ServerPort.Second;
}
resultForwarder = forwarder;
}
else
{
TCPtoUDPForwader TCPForwarder = new TCPtoUDPForwader();
TCPForwarder.ForwardCommand = aMessage.SourcePort;
TCPForwarder.SourceInterleavedVideo = transport.Interleaved.First;
TCPForwarder.ForwardInterleavedCommand = transport.Interleaved.Second;
// we need to transfer already getted values
TCPForwarder.ForwardHostVideo = forwarder.ForwardHostVideo;
TCPForwarder.ForwardPortVideo = forwarder.ForwardPortVideo;
TCPForwarder.SourcePortCommand = forwarder.SourcePortCommand;
TCPForwarder.ToMulticast = forwarder.ToMulticast;
resultForwarder = TCPForwarder;
}
if (resultForwarder.ToMulticast)
{
// Setup port and destination multicast.
resultForwarder.ForwardHostVideo = CreateNextMulticastAddress();
resultForwarder.ForwardPortVideo = forwarder.FromForwardVideoPort;
RtspTransport newTransport = new RtspTransport()
{
IsMulticast = true,
Destination = resultForwarder.ForwardHostVideo,
Port = new PortCouple(resultForwarder.ForwardPortVideo, resultForwarder.ListenCommandPort)
};
if ((resultForwarder is UDPForwarder && forwarder.ForwardPortCommand == 0) ||
(resultForwarder is TCPtoUDPForwader && (resultForwarder as TCPtoUDPForwader).ForwardInterleavedCommand == 0))
{
newTransport.Port = null;
}
aMessage.Headers[RtspHeaderNames.Transport] = newTransport.ToString();
}
else
{
RtspTransport newTransport = new RtspTransport()
{
IsMulticast = false,
Destination = forwarder.ForwardHostVideo,
ClientPort = new PortCouple(resultForwarder.ForwardPortVideo, resultForwarder.SourcePortCommand),
ServerPort = new PortCouple(resultForwarder.FromForwardVideoPort, resultForwarder.ListenCommandPort)
};
if ((resultForwarder is UDPForwarder && forwarder.ForwardPortCommand == 0) ||
(resultForwarder is TCPtoUDPForwader && (resultForwarder as TCPtoUDPForwader).ForwardInterleavedCommand == 0))
{
newTransport.ServerPort = null;
}
aMessage.Headers[RtspHeaderNames.Transport] = newTransport.ToString();
}
return(resultForwarder);
}
// 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
var listener = sender as RtspListener;
var message = e.Message as RtspMessage;
Console.WriteLine("RTSP message received " + message);
// Handle OPTIONS message
if (message is RtspRequestOptions)
{
// Create the reponse to OPTIONS
var options_response = (e.Message as RtspRequestOptions).CreateResponse();
listener.SendMessage(options_response);
}
// Handle DESCRIBE message
if (message is RtspRequestDescribe)
{
var requested_url = (message as 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
var raw_sps = h264_encoder.GetRawSPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
var raw_pps = h264_encoder.GetRawPPS(); // no 0x00 0x00 0x00 0x01 or 32 bit size header
var sps_str = Convert.ToBase64String(raw_sps);
var pps_str = Convert.ToBase64String(raw_pps);
var 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");
var sdp_bytes = Encoding.ASCII.GetBytes(sdp.ToString());
// Create the reponse to DESCRIBE
// This must include the Session Description Protocol (SDP)
var describe_response = (e.Message as 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 RtspRequestSetup)
{
//
var setupMessage = message as 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
var transport = setupMessage.GetTransports()[0];
// Construct the Transport: reply from the Server to the client
var transport_reply = new RtspTransport();
if (transport.LowerTransport == RtspTransport.LowerTransportType.TCP)
{
// RTP over RTSP mode}
transport_reply.LowerTransport = RtspTransport.LowerTransportType.TCP;
transport_reply.Interleaved = new PortCouple(transport.Interleaved.First, transport.Interleaved.Second);
}
if (transport.LowerTransport == 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 = RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = false;
transport_reply.ClientPort = transport.ClientPort; // FIX
// for now until implemented
transport_reply = null;
}
if (transport.LowerTransport == RtspTransport.LowerTransportType.UDP &&
transport.IsMulticast)
{
// 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 = RtspTransport.LowerTransportType.UDP;
transport_reply.IsMulticast = true;
transport_reply.Port = new PortCouple(7000, 7001); // FIX
// for now until implemented
transport_reply = null;
}
if (transport_reply != null)
{
var 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++;
}
var setup_response = setupMessage.CreateResponse();
setup_response.Headers[RtspHeaderNames.Transport] = transport_reply.ToString();
setup_response.Session = new_session.session_id;
listener.SendMessage(setup_response);
}
else
{
var setup_response = setupMessage.CreateResponse();
// unsuported transport
setup_response.ReturnCode = 461;
listener.SendMessage(setup_response);
}
}
// Handle PLAY message
if (message is RtspRequestPlay)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (var 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
var play_response = (e.Message as RtspRequestPlay).CreateResponse();
listener.SendMessage(play_response);
}
// Handle PLAUSE message
if (message is RtspRequestPause)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List. Change the state of "PLAY"
foreach (var 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
var pause_response = (e.Message as RtspRequestPause).CreateResponse();
listener.SendMessage(pause_response);
}
// Handle GET_PARAMETER message, often used as a Keep Alive
if (message is RtspRequestGetParameter)
{
// Create the reponse to GET_PARAMETER
var getparameter_response = (e.Message as RtspRequestGetParameter).CreateResponse();
listener.SendMessage(getparameter_response);
}
// Handle TEARDOWN
if (message is RtspRequestTeardown)
{
lock (rtp_list)
{
// Search for the Session in the Sessions List.
foreach (var 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();
}
}
}
}
}
/// <summary>
/// Handles a request setup.
/// </summary>
/// <param name="destination">The destination.</param>
/// <param name="requestSetup">The request setup.</param>
/// <returns>The rewritten message</returns>
/// <remarks>
/// The destination can be modified.
/// </remarks>
private RtspMessage HandleRequestSetup(ref RtspListener destination, RtspRequestSetup requestSetup)
{
Contract.Requires(requestSetup != null);
Contract.Requires(destination != null);
Contract.Ensures(Contract.Result <RtspMessage>() != null);
Contract.Ensures(Contract.ValueAtReturn(out destination) != null);
// look if we already have a multicast streaming playing for this URI.
foreach (var session in _activesSession.Values)
{
if (session.State == RtspSession.SessionState.Playing && session.ListOfForwader.ContainsKey(requestSetup.RtspUri))
{
Forwarder existingForwarder = session.ListOfForwader[requestSetup.RtspUri];
if (existingForwarder != null && existingForwarder.ToMulticast)
{
RtspResponse returnValue = requestSetup.CreateResponse();
returnValue.Headers[RtspHeaderNames.Transport] = new RtspTransport()
{
IsMulticast = true,
Destination = existingForwarder.ForwardHostVideo,
Port = new PortCouple(existingForwarder.ForwardPortVideo, existingForwarder.ListenCommandPort),
}.ToString();
returnValue.Session = session.Name;
destination = requestSetup.SourcePort;
return(returnValue);
}
}
}
string setupKey = requestSetup.SourcePort.RemoteAdress + "SEQ" + requestSetup.CSeq.ToString(CultureInfo.InvariantCulture);
RtspTransport selectedTransport = SelectTransport(requestSetup);
// We do not handle asked transport so return directly.
if (selectedTransport == null)
{
_logger.Info("No transport asked are supported, sorry");
RtspResponse returnValue = requestSetup.CreateResponse();
// Unsupported transport;
returnValue.ReturnCode = 461;
destination = requestSetup.SourcePort;
return(returnValue);
}
UDPForwarder forwarder = new UDPForwarder();
forwarder.ToMulticast = selectedTransport.IsMulticast;
// this part of config is only valid in unicast.
if (!selectedTransport.IsMulticast)
{
forwarder.ForwardPortVideo = selectedTransport.ClientPort.First;
forwarder.SourcePortCommand = selectedTransport.ClientPort.Second;
// If the client did not set the destination.. get it from TCP source
if (!string.IsNullOrEmpty(selectedTransport.Destination))
{
forwarder.ForwardHostVideo = selectedTransport.Destination;
}
else
{
forwarder.ForwardHostVideo = requestSetup.SourcePort.RemoteAdress.Split(':')[0];
_logger.Debug("Destination get from TCP port {0}", forwarder.ForwardHostVideo);
}
}
// Configured the transport asked.
forwarder.ForwardHostCommand = destination.RemoteAdress.Split(':')[0];
RtspTransport firstNewTransport = new RtspTransport()
{
IsMulticast = false,
ClientPort = new PortCouple(forwarder.ListenVideoPort, forwarder.FromForwardCommandPort),
};
RtspTransport secondTransport = new RtspTransport()
{
IsMulticast = false,
LowerTransport = RtspTransport.LowerTransportType.TCP,
};
requestSetup.Headers[RtspHeaderNames.Transport] = firstNewTransport.ToString() + ", " + secondTransport.ToString();
requestSetup.Headers[RtspHeaderNames.Transport] = firstNewTransport.ToString();// +", " + secondTransport.ToString();
_setupForwarder.Add(setupKey, forwarder);
return(requestSetup);
}
/// <summary>
/// Constructs a stream.
/// </summary>
/// <param name="uri">The uri of a stream to play.</param>
/// <param name="connectionTimeout">The connection timeout.</param>
/// <param name="streamTimeout">The stream timeout.</param>
/// <param name="transport">RTSP transport protocol.</param>
/// <param name="flags">RTSP flags.</param>
public static Stream FromUri(Uri uri,
TimeSpan connectionTimeout, TimeSpan streamTimeout, RtspTransport transport, RtspFlags flags)
{
var type = Load();
var proxy = type.GetMethod("FromUri").Invoke(null,
new Object[] { uri, connectionTimeout, streamTimeout, transport, flags });
return(new Stream(proxy));
}
// 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);
}
}
