static void Main(string[] args)
{
RtpParticipant part = new RtpParticipant("*****@*****.**", "SENDER");
RtpSession session = new RtpSession(ip, part, true, true);
session.PacketTransform = new EncryptionTransform("You are a big freak!");
//session.PacketTransform = new XorTransform();
RtpSender sender = session.CreateRtpSender("My sender", PayloadType.Test, null);
Stream fs = File.OpenRead("data.txt");
int length = (int)fs.Length;
Console.Out.WriteLine("Opening file of length: " + length);
byte[] buffer = new byte[length];
int bytesRead = 0;
while (bytesRead < length)
{
bytesRead += fs.Read(buffer, bytesRead, Math.Min(16384, (length - bytesRead)));
}
for (int i = 0; i < 5; i++)
{
Console.Out.WriteLine("Sending buffer to address: " + ip);
sender.Send(buffer);
Thread.Sleep(1000);
}
}
public RTPReceiveConnection(IPEndPoint ip, NetworkArchiver na)
{
RtpEvents.RtpStreamAdded += new MSR.LST.Net.Rtp.RtpEvents.RtpStreamAddedEventHandler(this.handleRtpStreamAdded);
RtpEvents.RtpStreamRemoved += new MSR.LST.Net.Rtp.RtpEvents.RtpStreamRemovedEventHandler(this.handleRtpStreamRemoved);
this.m_Participant = new RtpParticipant("Recorder", "Recorder");
this.m_Session = new RtpSession(ip, this.m_Participant, true, true);
this.m_archiver = na;
}
public RTPMessageSender(IPEndPoint ep, PresenterModel model, ClassroomModel classroom)
{
this.m_Model = model;
this.m_Classroom = classroom;
this.m_RtpLocalParticipant = new RtpLocalParticipant(this.m_Model.Participant);
using(Synchronizer.Lock(this)) {
// Register the stream event listeners first, since otherwise there would be a chance
// that streams could be added between creating the RtpSession (which connects immediately).
this.m_ParticipantManager = new ParticipantManager(this);
this.m_RtpSession = new RtpSession(ep, this.m_RtpLocalParticipant, true, true);
// TODO: Choose a meaningful value for the RtpSender name.
ushort fec;
short interpacketdelay;
using (Synchronizer.Lock(model.SyncRoot)) {
using (Synchronizer.Lock(model.ViewerState.SyncRoot)) {
fec = (ushort)model.ViewerState.ForwardErrorCorrection;
interpacketdelay = (short)model.ViewerState.InterPacketDelay;
}
}
if( fec == 0 )
this.m_RtpSender = this.m_RtpSession.CreateRtpSender( "Classroom Presenter", PayloadType.dynamicPresentation, null );
else
this.m_RtpSender = this.m_RtpSession.CreateRtpSenderFec( "Classroom Presenter", PayloadType.dynamicPresentation, null, 0, fec );
this.m_RtpSender.DelayBetweenPackets = interpacketdelay;
// Initialize the message chunking utilities.
this.m_Encoder = new Chunk.ChunkEncoder();
// TODO: Make the buffer size dynamic, ie., grow if we send a single very large message.
// Make the buffer store up to 5MB worth of data.
this.m_FrameBuffer = new FrameBuffer(5 * 1024 * 1024 / this.m_Encoder.MaximumChunkSize);
// Create the NackManager which is responsible for sending NACKs on behalf of
// our set of RTPMessageReceivers.
this.m_NackManager = new RTPNackManager(this, this.m_Classroom);
}
// Create network services outside of the "lock(this)" so they can lock their own objects
// without worrying about locking order.
// Create the PresenterNetworkService which will watch for changes to the model and send messages.
this.m_PresenterNetworkService = new PresenterNetworkService(this, this.m_Model);
// Create the StudentSubmissionsNetworkService which will watch for requests to submit and send messages.
this.m_StudentSubmissionNetworkService = new StudentSubmissionNetworkService(this, this.m_Model);
// Create the SynchronizationNetworkService which will watch for all synchronization messages.
this.m_SynchronizationNetworkService = new SynchronizationNetworkService( this, this.m_Model );
// Create the ScriptingNetworkService which will watch for all scripting messages.
this.m_ScriptingNetworkService = new ScriptingNetworkService( this, this.m_Model );
// Create the BeaconService which will broadcast periodic information about the presentation.
this.m_BeaconService = new Beacons.DefaultBeaconService(this, this.m_Model);
}
public SimpleTest()
{
// Create participant
RtpParticipant rtpParticipant = null;
//rtpParticipant.SetTool(true);
// Create session with Participant and Rtp data
rtpSession = new RtpSession(new IPEndPoint(IPAddress.Parse(addr), rtpPort),
rtpParticipant, false, true);
}
static void Main(string[] args)
{
RtpEvents.RtpStreamAdded += new RtpEvents.RtpStreamAddedEventHandler(OnNewRtpStream);
RtpParticipant part = new RtpParticipant("*****@*****.**", "Receiver");
session = new RtpSession(ip, part, true, true);
//session.PacketTransform = new XorTransform();
session.PacketTransform = new EncryptionTransform("You are a big freak!");
// make sure this thing doesn't terminate
while (true)
{
Thread.Sleep(10000);
}
}
/// <summary>
/// get details of the stream...raw data and start tiume etc
/// set up the stream senders associated with us. we have n, one active and the rest acting as buffers
/// </summary>
public StreamPlayer( RtpSession session, int newStreamID, ConferencePlayerPC cppc)
{
int maxFrameSize, maxFrameCount, maxBufferSize;
// This occasionally throws due to bad data. Let ConferencePlayer handle it.
DBHelper.GetStreamStatistics( newStreamID, out this.firstStreamTicks, out maxFrameSize, out maxFrameCount, out maxBufferSize);
streamID = newStreamID;
totalFramesSent = 0;
totalBytesSent = 0;
totalLateness = 0;
buffers = new BufferPlayer[Constants.BuffersPerStream];
perfCounter = cppc;
string payload;
DBHelper.GetStreamAndParticipantDetails( streamID, out name, out payload, out cname, out privExtns );
streamPayload = (PayloadType)Enum.Parse( typeof(PayloadType), payload, true );
// for delayed buffers (late joiners), we create the rtpSender later
this.rtpSession = session;
Trace.WriteLine("Playing back stream: "+streamID+"; payload: "+streamPayload+"; name: "+ (cname + " : " +name));
// buffer n buffers worth of data
long startingTick = this.firstStreamTicks;
for ( int i = 0; i < buffers.Length; i++)
{
buffers[i] = new BufferPlayer( streamID, maxFrameSize, maxFrameCount, maxBufferSize);
buffers[i].Populate( startingTick );
startingTick = buffers[i].LastTick + 1;
perfCounter.AddInstanceForCollection(buffers[i]);
}
// first stream is initially active
activeBufferIndex = 0;
activeBuffer = buffers[activeBufferIndex];
}
private void InitializeNetwork()
{
RtpEvents.ReceiverReport += new RtpEvents.ReceiverReportEventHandler(RtpReceiverReport);
RtpEvents.DuplicateCNameDetected += new RtpEvents.DuplicateCNameDetectedEventHandler(DuplicateCNameDetected);
// Create participant
rtpParticipant = new RtpParticipant(cName, name);
rtpParticipant.SetTool(true);
// Create session with Participant and Rtp data
rtpSession = new RtpSession(new IPEndPoint(IPAddress.Parse(ipAddress), port), rtpParticipant, true, true);
// Create RtpSender
rtpSender = rtpSession.CreateRtpSender(HostName, PayloadType.PipecleanerSignal, null, null);
}
public void StopPlaying()
{
if( playing == false )
return;
if( avPlayer != null )
{
avPlayer.Stop();
avPlayer = null;
}
if( otherPlayer != null )
{
otherPlayer.Stop();
otherPlayer = null;
}
if( this.perfCounter != null )
{
perfCounter.Dispose();
}
lock(this)
{
if( rtpSession != null )
{
rtpSession.Dispose();
rtpSession = null;
}
playing = false;
}
}
public void Play( IPEndPoint venue, int[] streams, bool receiveData )
{
if( this.playing )
throw new InvalidOperationException("ConferencePlayer::Play called when the player is already playing.");
if( streams == null )
throw new ArgumentNullException("streams cannot be null");
venueIPE = venue;
// Create an RtpSession
RtpParticipant me = new RtpParticipant(Constants.PersistenceCName, Constants.PersistenceName + " (Playing)");
rtpSession = new RtpSession(venue, me, true, receiveData);
// Hook the static stream-ended event
StreamPlayer.EndOfStreamReached += new EventHandler(EndOfStreamReached);
// Create a new perf counter for this ConferencePlayer instance
this.perfCounter = new ConferencePlayerPC(venue.ToString());
// Keep track of the streamPlayers
ArrayList avStreams = new ArrayList();
ArrayList otherStreams = new ArrayList();
// Find the first stored ticks of all the streams while creating them...
this.firstStoredTick = long.MaxValue;
// Create all of our StreamPlayers (one per stream...)
for ( int i = 0; i < streams.Length; i++)
{
StreamPlayer newStream = null;
try
{
newStream = new StreamPlayer(rtpSession, streams[i], perfCounter);
}
catch( Exception ex )
{
eventLog.WriteEntry(String.Format("Stream with bad data reached. Continuing playback of all other streams."
+ " Stream ID: {0}. Exception: \n {1}", streams[i], ex.ToString()),
EventLogEntryType.Warning, ArchiveServiceEventLog.ID.BadStreamInDB);
}
if( newStream != null )
{
perfCounter.AddInstanceForCollection( newStream );
if( newStream.FirstStreamsTicks < this.firstStoredTick )
this.firstStoredTick = newStream.FirstStreamsTicks;
// Add the new StreamPlayer to the right collection
// Pri3: Consider other methods here. Maybe a smarter detection of large frame sizes,
// or initializing the stream so it has enough packets ahead of time?
if( newStream.Payload == PayloadType.dynamicAudio || newStream.Payload == PayloadType.dynamicVideo )
avStreams.Add(newStream);
else // RTDocs stream or other large-payload stream, most likely
otherStreams.Add(newStream);
}
}
// Start the StreamsGroupPlayers
// Pri2: Change this to be compatable with use of the "playback speed" feature.
long startTime = DateTime.Now.Ticks + 1500*Constants.TicksPerMs; // we'll start in 1.5 seconds (for init time)
// Create the StreamsGroupPlayer(s)
avPlayer = new StreamsGroupPlayer(avStreams, this.firstStoredTick, startTime);
otherPlayer = new StreamsGroupPlayer(otherStreams, this.firstStoredTick, startTime);
this.playing = true;
}
/// <summary>
/// Join the Venue and start participating. AutoSend default devices if AutoSend == true
/// </summary>
public static void JoinVenue(IVenue venue)
{
lock(venue)
{
if (Conference.ActiveVenue != null)
{
throw new InvalidOperationException("Cannot join a venue when already joined to a venue.");
}
if (venue.VenueData.VenueType == VenueType.Invalid)
{
throw new ArgumentException("Can not join a venue that is of type Invalid");
}
// (pfb) Why are we checking this here? When would it happen?
if (rtpSession == null)
{
// Do this first, or else the locks in the eventHandlers will throw, due to null argument
Conference.activeVenue = venue;
HookRtpEvents();
#region Start the RtpSession
try
{
// Pri3: Add casting of a Participant to an RtpParticipant
Participant p = Conference.LocalParticipant;
RtpParticipant rp = new RtpParticipant(p.Identifier, p.Name);
rp.Email = p.Email;
rp.Location = Conference.Location;
rp.Phone = p.Phone;
if (Conference.ReflectorEnabled && MSR.LST.Net.Utility.IsMulticast(venue.EndPoint))
{
IPEndPoint refEP;
IPAddress reflectorIP;
int unicastPort;
// Resolve the reflector address
try
{
reflectorIP = System.Net.Dns.GetHostEntry(
Conference.ReflectorAddress).AddressList[0];
}
catch(Exception e)
{
throw new Exception("Reflector address not found - "
+ Conference.ReflectorAddress, e);
}
// Send a join request to the reflector and receive the RTP unicast port back
regClient = new RegistrarClient(reflectorIP, Conference.ReflectorJoinPort);
try
{
unicastPort = regClient.join(venue.EndPoint.Address, venue.EndPoint.Port);
}
catch(Exception e)
{
throw new Exception("Reflector join request failed", e);
}
refEP = new IPEndPoint(reflectorIP, unicastPort);
rtpSession = new RtpSession(venue.EndPoint, rp, true, true, refEP);
}
else
{
rtpSession = new RtpSession(venue.EndPoint, rp, true, true);
}
}
catch (System.Net.Sockets.SocketException se)
{
eventLog.WriteEntry(se.ToString(), EventLogEntryType.Error, 99);
LeaveVenue();
if (se.ErrorCode == 10013)
{
throw new Exception("A security error occurred connecting to the network "
+ "(WSAEACCES). Common causes of this error are; no network "
+ "connection can be found, a change to the .NET Framework or "
+ "Winsock without a reboot, or recently disconnecting from a "
+ "VPN session which leaves Winsock in an indeterminate state. "
+ "Check your network connectivity or reboot your computer and try "
+ "again.");
}
else
{
throw;
}
}
catch (Exception e)
{
eventLog.WriteEntry(e.ToString(), EventLogEntryType.Error, 99);
LeaveVenue();
throw;
}
#endregion
}
}
}
/// <summary>
/// Leave the Venue and do all the cleanup work.
/// </summary>
public static void LeaveVenue()
{
if( Conference.ActiveVenue == null )
throw new InvalidOperationException("Cannot exit a venue when not in a venue!");
lock(Conference.ActiveVenue)
{
// So nothing new is created from the network
UnhookRtpEvents();
// Clean up participants + owned capabilities
foreach(Participant p in new ArrayList(participants.Values))
{
// A dependency on the local participant dictates that we should always have the local participant
// in the venue.
if( p != Conference.localParticipant )
RemoveParticipant(p);
}
Debug.Assert(participants.Count == 1);
// Dispose remaining capabilities (channels)
foreach(ICapability ic in new ArrayList(capabilities.Values))
{
DisposeCapability(ic);
}
Debug.Assert(capabilities.Count == 0);
Debug.Assert(capabilityViewers.Count == 0);
Debug.Assert(capabilitySenders.Count == 0);
// Clear collection of form positions
capabilityForms.Clear();
// Leave the network
if (rtpSession != null)
{
rtpSession.Dispose();
rtpSession = null;
}
// Send a leave request to the reflector if the reflector was enabled when joining
// the venue. Do this after leaving the Session, so that BYE packets have been sent
// through reflector
if (Conference.ReflectorEnabled && regClient != null)
{
regClient.leave();
}
Conference.activeVenue = null;
}
}
/// <summary>
/// Stops recording the current conference
/// </summary>
public void StopRecording()
{
if( stopTimer != null )
{
stopTimer.Dispose();
stopTimer = null;
}
if( this.perfCounter != null )
{
this.perfCounter.Dispose();
}
if( this.recording )
{
perfCounter.Dispose();
// Stop all the streams
foreach ( StreamRecorder sm in streams.Values)
{
sm.StopListening();
}
streams = null;
participants = null;
// Stop the rtpSession
RtpEvents.RtpParticipantAdded -= new RtpEvents.RtpParticipantAddedEventHandler(this.OnNewRtpParticipant);
RtpEvents.RtpStreamAdded -= new RtpEvents.RtpStreamAddedEventHandler(this.OnNewRtpStream);
RtpEvents.RtpStreamRemoved -= new RtpEvents.RtpStreamRemovedEventHandler(this.OnRtpStreamRemoved);
if( rtpSession != null )
{
rtpSession.Dispose();
rtpSession = null;
}
DBHelper.RecordConferenceEndTime( conferenceID, System.DateTime.Now);
this.recording = false;
}
}
/// <summary>
/// Starts recording a conference. Sets up the conference data and then records all streams received until told to stop.
/// </summary>
public void RecordConference( string conferenceDescription, string venueIdentifier, IPEndPoint venue )
{
if( conferenceDescription.Length >= Constants.MaxDBStringSize || venueIdentifier.Length >= Constants.MaxDBStringSize )
throw new ArgumentException("String longer than accepted by database.");
recording = true;
venueIPE = venue;
streams = new Hashtable(Constants.InitialStreams);
participants = new Hashtable();
conferenceID = DBHelper.CreateConference( conferenceDescription, venueIdentifier, DateTime.Now);
// Store info about this conference to the instance, for debugging reference (mostly)
this.conferenceDescription = conferenceDescription;
this.venueIdentifier = venueIdentifier;
this.venue = venue;
// Create our performance counter
perfCounter = new ConferenceRecorderPC( venueIdentifier + " : " + conferenceDescription );
// Set up RTCP properties
RtpEvents.RtpParticipantAdded += new MSR.LST.Net.Rtp.RtpEvents.RtpParticipantAddedEventHandler(this.OnNewRtpParticipant);
RtpEvents.RtpStreamAdded += new MSR.LST.Net.Rtp.RtpEvents.RtpStreamAddedEventHandler(this.OnNewRtpStream);
RtpEvents.RtpStreamRemoved += new MSR.LST.Net.Rtp.RtpEvents.RtpStreamRemovedEventHandler(this.OnRtpStreamRemoved);
// Start listening
RtpParticipant rtpMe = new RtpParticipant(Constants.PersistenceCName, Constants.PersistenceName + " (RECORDING)");
rtpSession = new RtpSession(venue, rtpMe, true, true);
}
public void JoinRTPSession(bool receive)
{
if (receive)
{
RegisterRtpUserEvents();
RegisterRtpStreamEvents();
}
rtpSession = new RtpSession(network.GetChannel(channel), new RtpParticipant(Environment.MachineName, Environment.MachineName), true, receive);
rtpSender = rtpSession.CreateRtpSender("VOIP Listener", PayloadType.dynamicAudio, null);
}
/// <summary>
/// Leave the Venue and do all the cleanup work.
/// </summary>
public static void LeaveVenue()
{
if( Conference.ActiveVenue == null )
throw new InvalidOperationException(Strings.CannotExitNotInVenue);
lock(Conference.ActiveVenue)
{
StopDiagnosticMonitor();
// So nothing new is created from the network
UnhookRtpEvents();
// Clean up participants + owned capabilities
foreach(Participant p in new ArrayList(participants.Values))
{
// A dependency on the local participant dictates that we should always have the local participant
// in the venue.
if( p != Conference.localParticipant )
RemoveParticipant(p);
}
Debug.Assert(participants.Count == 1);
// Dispose remaining capabilities (channels)
foreach(ICapability ic in new ArrayList(capabilities.Values))
{
DisposeCapability(ic);
}
Debug.Assert(capabilities.Count == 0);
Debug.Assert(capabilityViewers.Count == 0);
Debug.Assert(capabilitySenders.Count == 0);
// Clear collection of form positions
capabilityForms.Clear();
// Leave the network
if (rtpSession != null)
{
rtpSession.Dispose();
rtpSession = null;
}
// andrew: termination of reflector sessions is now done in UdpListner. This should happen
// as a side-effect of disposing the RTP session.
Conference.activeVenue = null;
}
originalWindowPositions.Clear();
Utilities.SaveWindowPersistenceData(windowPersistenceData);
}
private void LeaveRtpSession()
{
if(rtpSession != null)
{
// Clean up all outstanding objects owned by the RtpSession
rtpSession.Dispose();
rtpSession = null;
rtpSender = null;
}
}
public RTPSendConnection(IPEndPoint ipe)
{
this.m_Participant = new RtpParticipant(Guid.NewGuid().ToString(), "Classroom Playback");
this.m_Session = new RtpSession(ipe, this.m_Participant, true, false);
this.m_Sender = this.m_Session.CreateRtpSenderFec("Classroom Presenter", PayloadType.dynamicPresentation, null, 0, 100);
this.m_Queue = new SendingQueue(this);
}
// CF2 Create participant, join session
// CF3 Retrieve RtpSender
private void JoinRtpSession(string name)
{
rtpSession = new RtpSession(ep, new RtpParticipant(name, name), true, true);
rtpSender = rtpSession.CreateRtpSenderFec(name, PayloadType.Chat, null, 0, 200);
}
internal static RtpSender CreateInstance(RtpSession rtpSession, string name, PayloadType pt, Hashtable priExns)
{
return CreateInstance(rtpSession, name, pt, priExns, 0);
}
public RtpListener(IPEndPoint nextHopEP, IPEndPoint groupEP, RtpSession rtpSession)
{
this.rtpSession = rtpSession;
this.nextHopEP = nextHopEP;
this.groupEP = groupEP;
returnBufferHandler = new ReturnBufferHandler(ReturnBuffer);
getBufferHandler = new GetBufferHandler(GetBuffer);
InitializeBufferPool();
InitializePerformanceCounters();
InitializeNetwork();
InitializeThreads();
}
public RtpListener(IPEndPoint nextHopEP, RtpSession rtpSession)
: this(nextHopEP, nextHopEP, rtpSession)
{
}
private void DisposeRtpSession()
{
if (rtpSession != null)
{
rtpSession.Dispose();
rtpSession = null;
rtpSender = null;
}
}
private void JoinRtpSession()
{
RtpParticipant participant = new RtpParticipant("Video" + pb1.Handle.ToInt32(), "Video");
rtpSession = new RtpSession(ep, participant, true, true);
}
public RtpListener(RtpSession rtpSession)
{
this.rtpSession = rtpSession;
returnBufferHandler = new ReturnBufferHandler(ReturnBuffer);
getBufferHandler = new GetBufferHandler(GetBuffer);
InitializeBufferPool();
InitializePerformanceCounters();
InitializeNetwork();
InitializeThreads();
}
public void Dispose()
{
lock(classLock)
{
try
{
disposed = true;
// Unhook events
RtpEvents.ReceiverReport -= new RtpEvents.ReceiverReportEventHandler(RtpReceiverReport);
RtpEvents.DuplicateCNameDetected -= new RtpEvents.DuplicateCNameDetectedEventHandler(DuplicateCNameDetected);
if(updateThread != null)
{
updateThread.Dispose();
updateThread = null;
}
if(rtpSession != null)
{
rtpSession.Dispose();
rtpSession = null;
rtpSender = null;
}
}
catch(Exception e)
{
eventLog.WriteEntry("Error disposing ConnectivityDetector. \n" + e.ToString(),
EventLogEntryType.Error, 0);
}
}
}
// CF1
private void LeaveRtpSession()
{
UnhookRtpEvents();
if(rtpSession != null)
{
rtpSession.Dispose();
rtpSession = null;
rtpSender = null;
rtpStream = null;
}
}
public HiddenSocketExceptionEventArgs( RtpSession session, System.Net.Sockets.SocketException ex )
{
Session = session;
Exception = ex;
}
/// <summary>
/// Join the Venue and start participating. AutoSend default devices if AutoSend == true
/// The optional password is used generate a symmetric encryption key...
/// </summary>
public static void JoinVenue(IVenue venue,String password)
{
foreach (Participant p in participants.Values) {
//If there was an active warning on the local particpant at the time of the
//previous venue leave, it needs to be cleaned up.
p.ThroughputWarnings.Clear();
}
lock(venue)
{
if (Conference.ActiveVenue != null)
{
throw new InvalidOperationException(Strings.AlreadyJoinedToVenue);
}
if (venue.VenueData.VenueType == VenueType.Invalid)
{
throw new ArgumentException(Strings.InvalidVenueType);
}
// (pfb) Why are we checking this here? When would it happen?
if (rtpSession == null)
{
// Do this first, or else the locks in the eventHandlers will throw, due to null argument
Conference.activeVenue = venue;
HookRtpEvents();
#region Start the RtpSession
try
{
// Pri3: Add casting of a Participant to an RtpParticipant
Participant p = Conference.LocalParticipant;
RtpParticipant rp = new RtpParticipant(p.Identifier, p.Name);
rp.Email = p.Email;
rp.Location = Conference.Location;
rp.Phone = p.Phone;
if (Conference.ReflectorEnabled && MSR.LST.Net.Utility.IsMulticast(venue.EndPoint))
{
IPEndPoint refEP;
IPAddress reflectorIP;
// Resolve the reflector address
try
{
reflectorIP = System.Net.Dns.GetHostEntry(
Conference.ReflectorAddress).AddressList[0];
}
catch(Exception e)
{
//Starting with .Net 4, passing an IP address to GetHostEntry excepts. We'll check
//here to see if the user entered an IP address before rethowing.
if (!IPAddress.TryParse(Conference.ReflectorAddress, out reflectorIP)) {
throw new Exception(string.Format(CultureInfo.CurrentCulture,
Strings.ReflectorAddressNotFound, Conference.ReflectorAddress), e);
}
}
// andrew: reflector registation is now done via UDP, and has been
// pushed down into UDPListener...
refEP = new IPEndPoint(reflectorIP, ReflectorRtpPort);
StartDiagnosticMonitor(venue.Name, venue.EndPoint);
rtpSession = new RtpSession(venue.EndPoint, rp, true, true, refEP);
if (password != null)
{
PacketTransform xform = MakePacketTransform(password);
rtpSession.PacketTransform = xform;
}
}
else
{
StartDiagnosticMonitor(venue.Name, venue.EndPoint);
rtpSession = new RtpSession(venue.EndPoint, rp, true, true);
if (password != null)
{
PacketTransform xform = MakePacketTransform(password);
rtpSession.PacketTransform = xform;
}
}
// andrew: notify RTP of our venue name; this is for diagnostic purposes
rtpSession.VenueName = venue.Name;
}
catch (System.Net.Sockets.SocketException se)
{
eventLog.WriteEntry(se.ToString(), EventLogEntryType.Error, 99);
LeaveVenue();
if (se.ErrorCode == 10013)
{
throw new Exception(Strings.SecurityErrorOccurredConnecting);
}
else
{
throw;
}
}
catch (Exception e)
{
eventLog.WriteEntry(e.ToString(), EventLogEntryType.Error, 99);
LeaveVenue();
throw;
}
#endregion
}
}
}
