private void ContinueAfterPlay(RTSPClient rtspClient, int resultCode, string resultString)
{
do {
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (resultCode != 0) {
Console.Error.WriteLine (rtspClient + "Failed to start playing session: " + resultString);
break;
}
if (scs.duration > 0) {
const uint delaySlop = 2;
// number of seconds extra to delay, after the stream's expected duration. (This is optional.)
scs.duration += delaySlop;
var uSecsToDelay = (long)(scs.duration * 1000000);
scs.streamTimerTask = env.TaskScheduler.ScheduleDelayedTask (uSecsToDelay, streamTimerHandler, rtspClient.Native.Native);
}
Console.Error.Write (rtspClient + "Started playing session");
if (scs.duration > 0) {
Console.Error.Write (" (for up to " + scs.duration + " seconds)");
}
Console.Error.WriteLine ("...");
return;
} while (false);
// An unrecoverable error occurred with this stream.
ShutdownStream (rtspClient);
}
private void ContinueAfterPlay(RTSPClient rtspClient, int resultCode, string resultString)
{
do
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (resultCode != 0)
{
Console.Error.WriteLine(rtspClient + "Failed to start playing session: " + resultString);
break;
}
if (scs.duration > 0)
{
const uint delaySlop = 2;
// number of seconds extra to delay, after the stream's expected duration. (This is optional.)
scs.duration += delaySlop;
var uSecsToDelay = (long)(scs.duration * 1000000);
scs.streamTimerTask = env.TaskScheduler.ScheduleDelayedTask(uSecsToDelay, streamTimerHandler, rtspClient.Native.Native);
}
Console.Error.Write(rtspClient + "Started playing session");
if (scs.duration > 0)
{
Console.Error.Write(" (for up to " + scs.duration + " seconds)");
}
Console.Error.WriteLine("...");
return;
} while (false);
// An unrecoverable error occurred with this stream.
ShutdownStream(rtspClient);
}
private void ContinueAfterDescribe(RTSPClient rtspClient, int resultCode, string resultString)
{
while (true) {
var scs = ((OurRtspClient)rtspClient).scs;
if (resultCode != 0) {
Console.Error.WriteLine ("Failed to get a SDP description: " + resultString);
break;
}
var description = resultString;
Console.Error.WriteLine ("Got a SDP description:");
Console.Error.Write (description);
scs.session = MediaSession.CreateNew (env, description);
if (scs.session == null) {
Console.Error.WriteLine ("Failed to create a MediaSession object from the SDP description: " + env.GetResultMsg);
break;
}
if (scs.session.HasSubsessions == 0) {
Console.Error.WriteLine ("This session has no media subsessions (i.e., no \"m=\" lines)");
break;
}
scs.iter = new MediaSubsessionIterator (scs.session);
SetupNextSubsession (rtspClient);
return;
}
ShutdownStream (rtspClient);
}
private void SetupNextSubsession(RTSPClient rtspClient)
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
scs.subsession = scs.iter.Next();
if (scs.subsession != null)
{
if (scs.subsession.Initiate(0) == 0)
{
Console.Error.WriteLine("Failed to initiate the \"" + scs.subsession + "\" subsession: " + env.GetResultMsg);
SetupNextSubsession(rtspClient); // give up on this subsession; go to the next one
}
else
{
Console.Error.WriteLine("Initiated the \"" + scs.subsession +
"\" subsession (client ports " + scs.subsession.ClientPortNum + "-" + (scs.subsession.ClientPortNum + 1) + ")");
// Continue setting up this subsession, by sending a RTSP "SETUP" command:
rtspClient.SendSetupCommand(scs.subsession, ContinueAfterSetup, 0, 0, 0, null);
}
return;
}
// We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming:
scs.duration = scs.session.PlayEndTime() - scs.session.PlayStartTime();
rtspClient.SendPlayCommand(scs.session, ContinueAfterPlay, 0, 0, 0, null);
}
private void ShutdownStream(RTSPClient rtspClient, int exitCode = 1)
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (scs.session != null)
{
var someSubsessionsWereActive = false;
var iter = new MediaSubsessionIterator(scs.session);
MediaSubsession subsession;
while ((subsession = iter.Next()) != null)
{
if (subsession.sink != null)
{
Medium.Close(subsession.sink);
subsession.sink = null;
if (subsession.RtcpInstance() != null)
{
subsession.RtcpInstance().SetByeHandler(null, IntPtr.Zero, 1);
// in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = true;
}
}
if (someSubsessionsWereActive)
{
// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
// Don't bother handling the response to the "TEARDOWN".
rtspClient.SendTeardownCommand(scs.session, null, null);
}
}
Console.Error.WriteLine(rtspClient + "Closing the stream.");
Medium.Close(rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
if (--rtspClientCount == 0)
{
// The final stream has ended, so exit the application now.
// (Of course, if you're embedding this code into your own application, you might want to comment this out.)
Environment.Exit(exitCode);
// If you choose *not* to "exit()" the application (i.e., if you comment out the call to "exit()" above),
// and you don't intend to do anything more with this "TaskScheduler" and "UsageEnvironment",
// then you can also reclaim the (small) memory used by these objects by doing the following.
// (However, you must not do this until after you have left the event loop.)
/*
* TaskScheduler* scheduler = &(env.taskScheduler());
* env.reclaim();
* delete scheduler;
*/
}
}
private void subsessionByeHandler(IntPtr clientData)
{
var subsession = new MediaSubsession(clientData);
var rtspClient = new RTSPClient(subsession.miscPtr);
Console.Error.WriteLine(rtspClient + "Received RTCP \"BYE\" on \"" + subsession + "\" subsession");
// Now act as if the subsession had closed:
SubsessionAfterPlaying(subsession.Native.Native);
}
private void ContinueAfterSetup(RTSPClient rtspClient, int resultCode, string resultString)
{
do
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (resultCode != 0)
{
Console.Error.WriteLine(rtspClient + "Failed to set up the \"" + scs.subsession + "\" subsession: " +
env.GetResultMsg);
break;
}
Console.Error.WriteLine(rtspClient + "Set up the \"" + scs.subsession
+ "\" subsession (client ports " + scs.subsession.ClientPortNum + "-" +
(scs.subsession.ClientPortNum + 1) + ")");
// Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
// (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
// after we've sent a RTSP "PLAY" command.)
scs.subsession.sink = DummySink.CreateNew(env, scs.subsession, rtspClient.Url);
// perhaps use your own custom "MediaSink" subclass instead
if (scs.subsession.sink == null)
{
Console.Error.WriteLine(rtspClient + "Failed to create a data sink for the \"" + scs.subsession
+ "\" subsession: " + env.GetResultMsg);
break;
}
Console.Error.WriteLine(rtspClient + "Created a data sink for the \"" + scs.subsession + "\" subsession");
scs.subsession.miscPtr = rtspClient.Native.Native;
// a hack to let subsession handle functions get the "RTSPClient" from the subsession
scs.subsession.sink.StartPlaying(scs.subsession.ReadSource(), SubsessionAfterPlaying,
scs.subsession.Native.Native);
// Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
if (scs.subsession.RtcpInstance() != null)
{
scs.subsession.RtcpInstance().SetByeHandler(subsessionByeHandler, scs.subsession.Native.Native, 1);
}
} while (false);
// Set up the next subsession, if any:
SetupNextSubsession(rtspClient);
}
public void Start(string commandData)
{
try
{
m_rtspAppStruct = new RTSPAppStruct(commandData, false);
RTSPClient rtspClient = new RTSPClient();
string sdp = rtspClient.GetStreamDescription(m_rtspAppStruct.URL);
SIPResponse sipResponse = GetOkResponse(m_clientTransaction.TransactionRequest, sdp);
m_clientTransaction.SendFinalResponse(sipResponse);
//rtspClient.Start(m_rtspAppStruct.URL, m_clientTransaction.RemoteEndPoint.Address, m_rtpPort);
}
catch (Exception excp)
{
logger.Error("Exception RTSPCall StartCall. " + excp.Message);
}
}
private void ContinueAfterDescribe(RTSPClient rtspClient, int resultCode, string resultString)
{
while (true)
{
var scs = ((OurRtspClient)rtspClient).scs;
if (resultCode != 0)
{
Console.Error.WriteLine("Failed to get a SDP description: " + resultString);
break;
}
var description = resultString;
Console.Error.WriteLine("Got a SDP description:");
Console.Error.Write(description);
scs.session = MediaSession.CreateNew(env, description);
if (scs.session == null)
{
Console.Error.WriteLine("Failed to create a MediaSession object from the SDP description: " + env.GetResultMsg);
break;
}
if (scs.session.HasSubsessions == 0)
{
Console.Error.WriteLine("This session has no media subsessions (i.e., no \"m=\" lines)");
break;
}
scs.iter = new MediaSubsessionIterator(scs.session);
SetupNextSubsession(rtspClient);
return;
}
ShutdownStream(rtspClient);
}
private void SubsessionAfterPlaying(IntPtr clientData)
{
var subsession = new MediaSubsession(clientData);
var rtspClient = new RTSPClient(subsession.miscPtr);
// Begin by closing this subsession's stream:
Medium.Close(subsession.sink);
subsession.sink = null;
// Next, check whether *all* subsessions' streams have now been closed:
var session = subsession.ParentSession();
var iter = new MediaSubsessionIterator(session);
while ((subsession = iter.Next()) != null)
{
if (subsession.sink != null)
{
return; // this subsession is still active
}
}
// All subsessions' streams have now been closed, so shutdown the client:
ShutdownStream(rtspClient);
}
public void Connect()
{
byte [] rbs = new byte[4 + 8 + 16];
RNGCryptoServiceProvider rng = new RNGCryptoServiceProvider();
rng.GetBytes(rbs);
string sid = String.Format("{0:D10}",
BitConverter.ToUInt32(rbs, 0));
string sci = String.Format("{0:X16}",
BitConverter.ToUInt64(rbs, 4));
string sac = Convert.ToBase64String(rbs, 12, 16);
string url = String.Format("rtsp://{0}/{1}", local, sid);
rc = new RTSPClient(host, 5000, url);
rc.UserAgent = "iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3)";
rc.AddHeaders.Set("Client-Instance", sci);
rc.Connect();
string key = Convert.ToBase64String(RSAEncrypt(alg.Key));
string iv = Convert.ToBase64String(alg.IV);
string sdp = String.Format(
"v=0\r\n" +
"o=iTunes {0} 0 IN IP4 {1}\r\n" +
"s=iTunes\r\n" +
"c=IN IP4 {2}\r\n" +
"t=0 0\r\n" +
"m=audio 0 RTP/AVP 96\r\n" +
"a=rtpmap:96 AppleLossless\r\n" +
"a=fmtp:96 4096 0 16 40 10 14 2 255 0 0 44100\r\n" +
"a=rsaaeskey:{3}\r\n" +
"a=aesiv:{4}\r\n",
sid, local, host,
key.Replace("=", ""),
iv.Replace("=", ""));
rc.AddHeaders.Set("Apple-Challenge", sac.Replace("=", ""));
rc.AnnounceSDP(sdp);
rc.AddHeaders.Remove("Apple-Challenge");
Hashtable ht = rc.Setup();
string aj = (string)ht["Audio-Jack-Status"];
if (aj == null)
{
throw new Exception("Audio-Jack-Status is missing");
}
string [] ptokens = aj.Split(new char[] { ';' });
for (int i = 0; i < ptokens.Length; i++)
{
string [] ctokens = ptokens[i].Split(new char[] { '=' });
for (int j = 0; j < ctokens.Length; j++)
{
if (ctokens.Length == 1 &&
ctokens[0].Trim().Equals("connected"))
{
ajstatus = JACK_STATUS_CONNECTED;
}
else if (ctokens.Length == 2 &&
ctokens[0].Trim().Equals("type"))
{
if (ctokens[1].Trim().Equals("digital"))
{
ajtype = JACK_TYPE_DIGITAL;
}
}
}
}
rc.Record();
UpdateVolume();
tcdata = new TcpClient();
tcdata.Connect(host, rc.ServerPort);
nsdata = tcdata.GetStream();
}
private void subsessionByeHandler(IntPtr clientData)
{
var subsession = new MediaSubsession (clientData);
var rtspClient = new RTSPClient (subsession.miscPtr);
Console.Error.WriteLine (rtspClient + "Received RTCP \"BYE\" on \"" + subsession + "\" subsession");
// Now act as if the subsession had closed:
SubsessionAfterPlaying (subsession.Native.Native);
}
private void SubsessionAfterPlaying(IntPtr clientData)
{
var subsession = new MediaSubsession (clientData);
var rtspClient = new RTSPClient (subsession.miscPtr);
// Begin by closing this subsession's stream:
Medium.Close (subsession.sink);
subsession.sink = null;
// Next, check whether *all* subsessions' streams have now been closed:
var session = subsession.ParentSession ();
var iter = new MediaSubsessionIterator (session);
while ((subsession = iter.Next ()) != null) {
if (subsession.sink != null) return; // this subsession is still active
}
// All subsessions' streams have now been closed, so shutdown the client:
ShutdownStream (rtspClient);
}
private void SetupNextSubsession(RTSPClient rtspClient)
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
scs.subsession = scs.iter.Next ();
if (scs.subsession != null) {
if (scs.subsession.Initiate (0) == 0) {
Console.Error.WriteLine ("Failed to initiate the \"" + scs.subsession + "\" subsession: " + env.GetResultMsg);
SetupNextSubsession (rtspClient); // give up on this subsession; go to the next one
} else {
Console.Error.WriteLine ("Initiated the \"" + scs.subsession +
"\" subsession (client ports " + scs.subsession.ClientPortNum + "-" + (scs.subsession.ClientPortNum + 1) + ")");
// Continue setting up this subsession, by sending a RTSP "SETUP" command:
rtspClient.SendSetupCommand (scs.subsession, ContinueAfterSetup, 0, 0, 0, null);
}
return;
}
// We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming:
scs.duration = scs.session.PlayEndTime () - scs.session.PlayStartTime ();
rtspClient.SendPlayCommand (scs.session, ContinueAfterPlay, 0, 0, 0, null);
}
private void ContinueAfterSetup(RTSPClient rtspClient, int resultCode, string resultString)
{
do {
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (resultCode != 0) {
Console.Error.WriteLine (rtspClient + "Failed to set up the \"" + scs.subsession + "\" subsession: " +
env.GetResultMsg);
break;
}
Console.Error.WriteLine (rtspClient + "Set up the \"" + scs.subsession
+ "\" subsession (client ports " + scs.subsession.ClientPortNum + "-" +
(scs.subsession.ClientPortNum + 1) + ")");
// Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
// (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
// after we've sent a RTSP "PLAY" command.)
scs.subsession.sink = DummySink.CreateNew (env, scs.subsession, rtspClient.Url);
// perhaps use your own custom "MediaSink" subclass instead
if (scs.subsession.sink == null) {
Console.Error.WriteLine (rtspClient + "Failed to create a data sink for the \"" + scs.subsession
+ "\" subsession: " + env.GetResultMsg);
break;
}
Console.Error.WriteLine (rtspClient + "Created a data sink for the \"" + scs.subsession + "\" subsession");
scs.subsession.miscPtr = rtspClient.Native.Native;
// a hack to let subsession handle functions get the "RTSPClient" from the subsession
scs.subsession.sink.StartPlaying (scs.subsession.ReadSource (), SubsessionAfterPlaying,
scs.subsession.Native.Native);
// Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
if (scs.subsession.RtcpInstance () != null) {
scs.subsession.RtcpInstance ().SetByeHandler (subsessionByeHandler, scs.subsession.Native.Native, 1);
}
} while (false);
// Set up the next subsession, if any:
SetupNextSubsession (rtspClient);
}
public void Connect()
{
byte [] rbs = new byte[ 4 + 8 + 16 ];
RNGCryptoServiceProvider rng = new RNGCryptoServiceProvider();
rng.GetBytes( rbs );
string sid = String.Format( "{0:D10}",
BitConverter.ToUInt32( rbs, 0 ) );
string sci = String.Format( "{0:X16}",
BitConverter.ToUInt64( rbs, 4 ) );
string sac = Convert.ToBase64String( rbs, 12, 16 );
string url = String.Format( "rtsp://{0}/{1}", local, sid );
rc = new RTSPClient( host, 5000, url );
rc.UserAgent = "iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3)";
rc.AddHeaders.Set( "Client-Instance", sci );
rc.Connect();
string key = Convert.ToBase64String( RSAEncrypt( alg.Key ) );
string iv = Convert.ToBase64String( alg.IV );
string sdp = String.Format(
"v=0\r\n" +
"o=iTunes {0} 0 IN IP4 {1}\r\n" +
"s=iTunes\r\n" +
"c=IN IP4 {2}\r\n" +
"t=0 0\r\n" +
"m=audio 0 RTP/AVP 96\r\n" +
"a=rtpmap:96 AppleLossless\r\n" +
"a=fmtp:96 4096 0 16 40 10 14 2 255 0 0 44100\r\n" +
"a=rsaaeskey:{3}\r\n" +
"a=aesiv:{4}\r\n",
sid, local, host,
key.Replace( "=", "" ),
iv.Replace( "=", "" ) );
rc.AddHeaders.Set( "Apple-Challenge", sac.Replace( "=", "" ) );
rc.AnnounceSDP( sdp );
rc.AddHeaders.Remove( "Apple-Challenge" );
Hashtable ht = rc.Setup();
string aj = (string)ht[ "Audio-Jack-Status" ];
if( aj == null )
throw new Exception( "Audio-Jack-Status is missing" );
string [] ptokens = aj.Split( new char[] { ';' } );
for( int i = 0; i < ptokens.Length; i++ )
{
string [] ctokens = ptokens[ i ].Split( new char[] { '=' } );
for( int j = 0; j < ctokens.Length; j++ )
{
if( ctokens.Length == 1 &&
ctokens[ 0 ].Trim().Equals( "connected" ) )
{
ajstatus = JACK_STATUS_CONNECTED;
}
else if( ctokens.Length == 2 &&
ctokens[ 0 ].Trim().Equals( "type" ) )
{
if( ctokens[ 1 ].Trim().Equals( "digital" ) )
ajtype = JACK_TYPE_DIGITAL;
}
}
}
rc.Record();
UpdateVolume();
tcdata = new TcpClient();
tcdata.Connect( host, rc.ServerPort );
nsdata = tcdata.GetStream();
}
public async void rtspClientStart()
{
rtspCancel = new CancellationTokenSource();
var url = "rtsp://192.168.0.10:8554/H264Video";
String now = DateTime.Now.ToString("yyyyMMdd_HHmmss");
MemoryStream fs_vps = null; // used to write the video
MemoryStream fs_v = null; // used to write the video
MemoryStream fs_a = null; // used to write the audio
h264 = false;
h265 = false;
bool spsdone = false;
RTSPClient c = new RTSPClient();
// The SPS/PPS comes from the SDP data
// or it is the first SPS/PPS from the H264 video stream
c.Received_SPS_PPS += (byte[] sps, byte[] pps) => {
h264 = true;
if (fs_vps == null)
{
String filename = "rtsp_capture_" + now + ".264";
fs_vps = new MemoryStream();
}
if (fs_vps != null)
{
fs_vps.SetLength(0);
fs_vps.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_vps.Write(sps, 0, sps.Length);
fs_vps.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_vps.Write(pps, 0, pps.Length);
}
};
c.Received_VPS_SPS_PPS += (byte[] vps, byte[] sps, byte[] pps) => {
h265 = true;
if (fs_vps == null)
{
String filename = "rtsp_capture_" + now + ".265";
fs_vps = new MemoryStream();
}
if (fs_vps != null)
{
fs_vps.SetLength(0);
fs_vps.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_vps.Write(vps, 0, vps.Length); // Video parameter set
fs_vps.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_vps.Write(sps, 0, sps.Length); // Sequence Parameter Set
fs_vps.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_vps.Write(pps, 0, pps.Length); // Picture Parameter Set
}
};
// Video NALs. May also include the SPS and PPS in-band for H264
c.Received_NALs += (List <byte[]> nal_units) =>
{
foreach (byte[] nal_unit in nal_units)
{
// Output some H264 stream information
if (h264 && nal_unit.Length > 0)
{
int nal_ref_idc = (nal_unit[0] >> 5) & 0x03;
int nal_unit_type = nal_unit[0] & 0x1F;
String description = "";
if (nal_unit_type == 1)
{
description = "NON IDR NAL";
}
else if (nal_unit_type == 5)
{
description = "IDR NAL";
}
else if (nal_unit_type == 6)
{
description = "SEI NAL";
}
else if (nal_unit_type == 7)
{
description = "SPS NAL";
}
else if (nal_unit_type == 8)
{
description = "PPS NAL";
}
else if (nal_unit_type == 9)
{
description = "ACCESS UNIT DELIMITER NAL";
}
else
{
description = "OTHER NAL";
}
//Console.WriteLine("NAL Ref = " + nal_ref_idc + " NAL Type = " + nal_unit_type + " " + description);
}
// Output some H265 stream information
if (h265 && nal_unit.Length > 0)
{
int nal_unit_type = (nal_unit[0] >> 1) & 0x3F;
String description = "";
if (nal_unit_type == 1)
{
description = "NON IDR NAL";
}
else if (nal_unit_type == 19)
{
description = "IDR NAL";
}
else if (nal_unit_type == 32)
{
description = "VPS NAL";
}
else if (nal_unit_type == 33)
{
description = "SPS NAL";
}
else if (nal_unit_type == 34)
{
description = "PPS NAL";
}
else if (nal_unit_type == 39)
{
description = "SEI NAL";
}
else
{
description = "OTHER NAL";
}
//Console.WriteLine("NAL Type = " + nal_unit_type + " " + description);
}
// we need sps... first
if (!h264 && !h265)
{
return;
}
if (!spsdone)
{
if (callbacks == null || callbacks.buffers.Count == 0)
{
return;
}
var index = callbacks.buffers.Pop();
var buffer = codec.GetInputBuffer(index);
buffer.Clear();
buffer.Put(fs_vps.ToArray());
codec.QueueInputBuffer(index, 0, (int)fs_vps.Length, 0, MediaCodecBufferFlags.CodecConfig);
spsdone = true;
fs_v = new MemoryStream();
}
if (fs_v != null)
{
fs_v.Write(new byte[] { 0x00, 0x00, 0x00, 0x01 }, 0, 4); // Write Start Code
fs_v.Write(nal_unit, 0, nal_unit.Length); // Write NAL
}
if (callbacks == null || fs_v == null || callbacks.buffers.Count == 0)
{
return;
}
try
{
var index = callbacks.buffers.Pop();
var buffer = codec.GetInputBuffer(index);
buffer.Clear();
buffer.Put(fs_v.ToArray());
codec.QueueInputBuffer(index, 0, (int)fs_v.Length, 0, MediaCodecBufferFlags.None);
fs_v.SetLength(0);
}
catch
{
}
}
};
// seperate and stay running
Task.Run(() =>
{
while (rtspCancel != null && true)
{
try
{
if (rtspCancel.Token.IsCancellationRequested)
{
return;
}
c.Connect(url, RTSPClient.RTP_TRANSPORT.UDP);
var lastrtp = 0;
int cnt = 0;
while (!c.StreamingFinished())
{
rtsprunning = true;
Thread.Sleep(500);
// existing
if (rtspCancel.Token.IsCancellationRequested)
{
c.Stop();
return;
}
// no rtp in .5 sec
if (lastrtp == c.rtp_count && cnt++ > 5)
{
c.Stop();
rtspCancel = null;
return;
}
lastrtp = c.rtp_count;
}
rtsprunning = false;
}
catch
{
}
}
});
}
private void ShutdownStream(RTSPClient rtspClient, int exitCode = 1)
{
var scs = ((OurRtspClient)rtspClient).scs; // alias
if (scs.session != null) {
var someSubsessionsWereActive = false;
var iter = new MediaSubsessionIterator (scs.session);
MediaSubsession subsession;
while ((subsession = iter.Next ()) != null) {
if (subsession.sink != null) {
Medium.Close (subsession.sink);
subsession.sink = null;
if (subsession.RtcpInstance () != null) {
subsession.RtcpInstance ().SetByeHandler (null, IntPtr.Zero, 1);
// in case the server sends a RTCP "BYE" while handling "TEARDOWN"
}
someSubsessionsWereActive = true;
}
}
if (someSubsessionsWereActive) {
// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
// Don't bother handling the response to the "TEARDOWN".
rtspClient.SendTeardownCommand (scs.session, null, null);
}
}
Console.Error.WriteLine (rtspClient + "Closing the stream.");
Medium.Close (rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
if (--rtspClientCount == 0) {
// The final stream has ended, so exit the application now.
// (Of course, if you're embedding this code into your own application, you might want to comment this out.)
Environment.Exit (exitCode);
// If you choose *not* to "exit()" the application (i.e., if you comment out the call to "exit()" above),
// and you don't intend to do anything more with this "TaskScheduler" and "UsageEnvironment",
// then you can also reclaim the (small) memory used by these objects by doing the following.
// (However, you must not do this until after you have left the event loop.)
/*
TaskScheduler* scheduler = &(env.taskScheduler());
env.reclaim();
delete scheduler;
*/
}
}
