// 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);
}
}
public void Connect(String url, RTP_TRANSPORT rtp_transport)
{
Rtsp.RtspUtils.RegisterUri();
Console.WriteLine("Connecting to " + url);
this.url = url;
// Use URI to extract username and password
// and to make a new URL without the username and password
try {
Uri uri = new Uri(this.url);
hostname = uri.Host;
port = uri.Port;
if (uri.UserInfo.Length > 0)
{
username = uri.UserInfo.Split(new char[] { ':' })[0];
password = uri.UserInfo.Split(new char[] { ':' })[1];
this.url = uri.GetComponents((UriComponents.AbsoluteUri & ~UriComponents.UserInfo),
UriFormat.UriEscaped);
}
} catch {
username = null;
password = null;
}
// Connect to a RTSP Server. The RTSP session is a TCP connection
rtsp_socket_status = RTSP_STATUS.Connecting;
try
{
rtsp_socket = new Rtsp.RtspTcpTransport(hostname, port);
}
catch
{
rtsp_socket_status = RTSP_STATUS.ConnectFailed;
Console.WriteLine("Error - did not connect");
return;
}
if (rtsp_socket.Connected == false)
{
rtsp_socket_status = RTSP_STATUS.ConnectFailed;
Console.WriteLine("Error - did not connect");
return;
}
rtsp_socket_status = RTSP_STATUS.Connected;
// Connect a RTSP Listener to the RTSP Socket (or other Stream) to send RTSP messages and listen for RTSP replies
rtsp_client = new Rtsp.RtspListener(rtsp_socket);
rtsp_client.AutoReconnect = false;
rtsp_client.MessageReceived += Rtsp_MessageReceived;
rtsp_client.DataReceived += Rtp_DataReceived;
rtsp_client.Start(); // start listening for messages from the server (messages fire the MessageReceived event)
// Check the RTP Transport
// If the RTP transport is TCP then we interleave the RTP packets in the RTSP stream
// If the RTP transport is UDP, we initialise two UDP sockets (one for video, one for RTCP status messages)
// If the RTP transport is MULTICAST, we have to wait for the SETUP message to get the Multicast Address from the RTSP server
this.rtp_transport = rtp_transport;
if (rtp_transport == RTP_TRANSPORT.UDP)
{
video_udp_pair = new Rtsp.UDPSocket(50000, 50020); // give a range of 10 pairs (20 addresses) to try incase some address are in use
video_udp_pair.DataReceived += Rtp_DataReceived;
video_udp_pair.Start(); // start listening for data on the UDP ports
audio_udp_pair = new Rtsp.UDPSocket(50000, 50020); // give a range of 10 pairs (20 addresses) to try incase some address are in use
audio_udp_pair.DataReceived += Rtp_DataReceived;
audio_udp_pair.Start(); // start listening for data on the UDP ports
}
if (rtp_transport == RTP_TRANSPORT.TCP)
{
// Nothing to do. Data will arrive in the RTSP Listener
}
if (rtp_transport == RTP_TRANSPORT.MULTICAST)
{
// Nothing to do. Will open Multicast UDP sockets after the SETUP command
}
// Send OPTIONS
// In the Received Message handler we will send DESCRIBE, SETUP and PLAY
Rtsp.Messages.RtspRequest options_message = new Rtsp.Messages.RtspRequestOptions();
options_message.RtspUri = new Uri(this.url);
rtsp_client.SendMessage(options_message);
}
// 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);
}
}
