/// <summary>
/// Incoming call constructor.
/// </summary>
/// <param name="stack">Reference to SIP stack.</param>
/// <param name="dialog">Reference SIP dialog.</param>
/// <param name="session">Call RTP multimedia session.</param>
/// <param name="localSDP">Local SDP.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stack</b>,<b>dialog</b>,<b>session</b> or <b>localSDP</b> is null reference.</exception>
internal SIP_Call(SIP_Stack stack, SIP_Dialog dialog, RTP_MultimediaSession session, SDP_Message localSDP)
{
if (stack == null)
{
throw new ArgumentNullException("stack");
}
if (dialog == null)
{
throw new ArgumentNullException("dialog");
}
if (session == null)
{
throw new ArgumentNullException("session");
}
if (localSDP == null)
{
throw new ArgumentNullException("localSDP");
}
m_pStack = stack;
m_pDialog = (SIP_Dialog_Invite)dialog;
m_pRtpMultimediaSession = session;
m_pLocalSDP = localSDP;
m_StartTime = DateTime.Now;
m_pFlow = dialog.Flow;
dialog.StateChanged += new EventHandler(m_pDialog_StateChanged);
SetState(SIP_CallState.Active);
// Start ping timer.
m_pKeepAliveTimer = new TimerEx(40000);
m_pKeepAliveTimer.Elapsed += new System.Timers.ElapsedEventHandler(m_pKeepAliveTimer_Elapsed);
m_pKeepAliveTimer.Enabled = true;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="session">RTP multimedia session.</param>
public wfrm_RTP_Debug(RTP_MultimediaSession session)
{
if (session == null)
{
throw new ArgumentNullException("session");
}
m_pSession = session;
InitUI();
m_pSession.Error += m_pSession_Error;
m_pSession.SessionCreated += m_pSession_SessionCreated;
m_pSession.NewParticipant += m_pSession_NewParticipant;
m_pSession.LocalParticipant.SourceAdded += Participant_SourceAdded;
m_pSession.LocalParticipant.SourceRemoved += Participant_SourceRemoved;
m_pTimer = new Timer();
m_pTimer.Interval = 1000;
m_pTimer.Tick += m_pTimer_Tick;
m_pTimer.Enabled = true;
TreeNode nodeParticipant = new TreeNode(session.LocalParticipant.CNAME);
nodeParticipant.Tag = new RTP_ParticipantInfo(session.LocalParticipant);
nodeParticipant.Nodes.Add("Sources");
m_pParticipants.Nodes.Add(nodeParticipant);
}
/// <summary>
///
/// </summary>
/// <param name="ip"></param>
/// <param name="port"></param>
/// <param name="volume">The supplied value is a percentage of the maximum volume level of the device and must be in the range 0-100.</param>
/// <returns></returns>
static public Status Play(Execute execute, bool multicast, IPAddress source_ip, int port, uint?volume100 = null)
{
if (session != null)
{
if (!Message.YesNo("Would you like to drop the previous stream?"))
{
return(Status.BUSY);
}
session.Stop();
session.Dispose();
}
Rtp.execute = execute;
Rtp.source_ip = source_ip;
Rtp.volume100 = volume100;
Rtp.multicast = multicast;
session = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
if (multicast)
{
session.CreateMulticastSession(null, new RTP_Clock(0, 8000), new RTP_Address(source_ip, port, port + 1));
}
else
{
session.CreateSession(new RTP_Address(IPAddress.Any, port, port + 1), new RTP_Clock(0, 8000));
}
session.Sessions[0].NewReceiveStream += NewReceiveStream;
session.Sessions[0].Payload = payload;
session.Sessions[0].Start();
return(Status.ACCEPTED);
}
/// <summary>
/// Calling constructor.
/// </summary>
/// <param name="stack">Reference to SIP stack.</param>
/// <param name="sender">Initial INVITE sender.</param>
/// <param name="session">Call RTP multimedia session.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stack</b>,<b>sender</b> or <b>session</b> is null reference.</exception>
internal SIP_Call(SIP_Stack stack, SIP_RequestSender sender, RTP_MultimediaSession session)
{
if (stack == null)
{
throw new ArgumentNullException("stack");
}
if (sender == null)
{
throw new ArgumentNullException("sender");
}
if (session == null)
{
throw new ArgumentNullException("session");
}
m_pStack = stack;
m_pInitialInviteSender = sender;
m_pRtpMultimediaSession = session;
m_pTags = new Dictionary <string, object>();
m_pInitialInviteSender.Completed += new EventHandler(delegate(object s, EventArgs e){
m_pInitialInviteSender = null;
if (this.State == SIP_CallState.Terminating)
{
SetState(SIP_CallState.Terminated);
}
});
m_CallState = SIP_CallState.Calling;
}
public void Start()
{
if (IsDisposed)
{
throw new ObjectDisposedException(this.GetType().Name);
}
if (!Started)
{
_hTrigger.Init();
_multiSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
RTP_Session session = _multiSession.CreateSession(new RTP_Address(IPAddress.Parse(_localIP), _localPort, _localPort + 1), new RTP_Clock(0, VideoRate));
session.Payload = RTP_PayloadTypes.H264;
session.Start();
_sendStream = session.CreateSendStream();
_vSoure = VideoSourceCreator.Instance.GetVideoSource(_videoId);
if (_targets.Count > 0)
{
foreach (string key in _targets.Keys)
{
TargetItem ti = _targets[key];
session.AddTarget(new RTP_Address(ti.IP, ti.Port, ti.Port + 1));
}
startPlay();
}
Started = true;
}
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="session">RTP multimedia session.</param>
public wfrm_RTP_Debug(RTP_MultimediaSession session)
{
if (session == null)
{
throw new ArgumentNullException("session");
}
m_pSession = session;
InitUI();
// Windows must be visible, otherwise we may get "window handle not created" if RTP session rises events before window gets visible.
this.Visible = true;
m_pSession.Error += new EventHandler <LumiSoft.Net.ExceptionEventArgs>(m_pSession_Error);
m_pSession.SessionCreated += new EventHandler <LumiSoft.Net.EventArgs <RTP_Session> >(m_pSession_SessionCreated);
m_pSession.NewParticipant += new EventHandler <RTP_ParticipantEventArgs>(m_pSession_NewParticipant);
m_pSession.LocalParticipant.SourceAdded += new EventHandler <RTP_SourceEventArgs>(Participant_SourceAdded);
m_pSession.LocalParticipant.SourceRemoved += new EventHandler <RTP_SourceEventArgs>(Participant_SourceRemoved);
//m_pSession.Disposed
m_pTimer = new TimerEx(m_pTimer_Tick, 1000);
m_pTimer.Start();
foreach (RTP_Session s in m_pSession.Sessions)
{
ComboBoxItem item = new ComboBoxItem("Session: " + s.GetHashCode(), new RTP_SessionStatistics(s));
m_pSessions.Items.Add(item);
}
if (m_pSessions.Items.Count > 0)
{
m_pSessions.SelectedIndex = 0;
}
}
public void Stop()
{
if (Started)
{
stopPlay();
_sendStream.Close();
_multiSession.Close("User Close");
_multiSession = null;
Started = false;
}
}
static public void Stop()
{
if (session != null)
{
ao.Stop();
ao.Dispose();
ao = null;
session.Close("Closed.");
session.Dispose();
session = null;
execute = null;
}
}
/// <summary>
/// This method is called when this form is closed.
/// </summary>
/// <param name="sender">Sender.</param>
/// <param name="e">Event data.</param>
private void wfrm_Main_FormClosed(object sender, FormClosedEventArgs e)
{
m_IsRunning = false;
if (m_pRtpSession != null)
{
m_pRtpSession.Close(null);
m_pRtpSession = null;
}
if (m_pWaveOut != null)
{
m_pWaveOut.Dispose();
m_pWaveOut = null;
}
if (m_pRecordStream != null)
{
m_pRecordStream.Dispose();
m_pRecordStream = null;
}
}
static void Main(string[] args)
{
string ip = "127.0.0.1";
//string ip = "224.0.0.1";
int port = 20700;
RTP_MultimediaSession session = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
session.CreateMulticastSession(null, new RTP_Clock(0, 8000), new RTP_Address(IPAddress.Parse(ip), port, port + 1));
//session.CreateSession(new RTP_Address(IPAddress.Any, port, port + 1), new RTP_Clock(0, 8000));
session.Sessions[0].NewReceiveStream += new EventHandler <RTP_ReceiveStreamEventArgs>(m_pRtpSession_NewReceiveStream);
session.Sessions[0].Payload = payload;
session.Sessions[0].Start();
Console.ReadKey();
}
private void StartRtpSession(IPAddress remoteIp, IPAddress localIp, int rtpPort, int rtcpPort)
{
Dictionary <int, AudioCodec> m_pAudioCodecs = new Dictionary <int, AudioCodec>
{
{ 0, new PCMU() }
};
RTP_MultimediaSession rtpMultimediaSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
sessionRtp = rtpMultimediaSession.CreateSession(new RTP_Address(localIp, rtpPort, rtcpPort), new RTP_Clock(1, 8000));
sessionRtp.AddTarget(new RTP_Address(remoteIp, rtpPort, rtcpPort));
sessionRtp.Payload = 0;
sessionRtp.StreamMode = RTP_StreamMode.SendReceive;
sessionRtp.NewReceiveStream += delegate(object s, RTP_ReceiveStreamEventArgs e2)
{
if (audioOut != null)
{
audioOutRtp = new AudioOut_RTP(audioOut, e2.Stream, m_pAudioCodecs);
if (speakerMute == false)
{
audioOutRtp.Start();
}
}
};
if (audioIn != null)
{
audioInRtp = new AudioIn_RTP(audioIn, 10, m_pAudioCodecs, sessionRtp.CreateSendStream());
if (micMute == false)
{
audioInRtp.Start();
}
}
sessionRtp.Start();
}
private void btnRestartConnection_Click_1(object sender, RoutedEventArgs e)
{
if (cbLocalIp.SelectedIndex < 0)
{
MessageBox.Show("Choose local IP");
return;
}
IsConnected = true;
InitializeClient();
InitializeServer();
/*InitializeSoundSender();
* InitializeSoundReceiver();*/
if (m_IsRunning)
{
m_IsRunning = false;
m_IsSendingTest = false;
m_pRtpSession.Dispose();
m_pRtpSession = null;
m_pWaveOut.Dispose();
m_pWaveOut = null;
}
else
{
m_IsRunning = true;
switch (_selectedCodec)
{
case "PCMU":
m_pActiveCodec = new PCMU();
break;
case "PCMA":
default:
m_pActiveCodec = new PCMA();
break;
}
var selectedOutDevice = cbAudioOutDevices.SelectedItem as AudioOutDevice;
m_pWaveOut = new AudioOut(selectedOutDevice, _samplesPerSecond, _bitsPerSample, 1); // 1 - one channel (mono)
m_pRtpSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
string localIp = cbLocalIp.SelectedItem.ToString();
string partnerIp = tbxPartnerIp.Text;
int k = string.Compare(localIp, partnerIp);
m_pRtpSession.CreateSession(new RTP_Address(IPAddress.Parse(cbLocalIp.SelectedItem.ToString()), (int)10000 + k * 500 /*m_pLocalPort.Value*/, (int)/*m_pLocalPort.Value*/ 10000 + k * 500 + 1), new RTP_Clock(0, _samplesPerSecond));
m_pRtpSession.Sessions[0].AddTarget(new RTP_Address(IPAddress.Parse(tbxPartnerIp.Text), (int)/*m_pRemotePort.Value*/ 10000 - k * 500, (int)/*m_pRemotePort.Value*/ 10000 - k * 500 + 1));
m_pRtpSession.Sessions[0].NewSendStream += new EventHandler <RTP_SendStreamEventArgs>(m_pRtpSession_NewSendStream);
m_pRtpSession.Sessions[0].NewReceiveStream += new EventHandler <RTP_ReceiveStreamEventArgs>(m_pRtpSession_NewReceiveStream);
m_pRtpSession.Sessions[0].Payload = 0;
m_pRtpSession.Sessions[0].Start();
m_pAudioCodecs = new Dictionary <int, AudioCodec>();
m_pAudioCodecs.Add(0, new PCMU());
m_pAudioCodecs.Add(8, new PCMA());
}
}
private void m_pToggleRun_Click(object sender, EventArgs e)
{
if (m_IsRunning)
{
m_IsRunning = false;
m_IsSendingTest = false;
m_pRtpSession.Dispose();
m_pRtpSession = null;
m_pWaveOut.Dispose();
m_pWaveOut = null;
if (m_pRecordStream != null)
{
m_pRecordStream.Dispose();
m_pRecordStream = null;
}
m_pOutDevices.Enabled = true;
m_pToggleRun.Text = "Start";
m_pRecord.Enabled = true;
m_pRecordFile.Enabled = true;
m_pRecordFileBrowse.Enabled = true;
m_pRemoteIP.Enabled = false;
m_pRemotePort.Enabled = false;
m_pCodec.Enabled = false;
m_pToggleMic.Text = "Send";
m_pToggleMic.Enabled = false;
m_pSendTestSound.Enabled = false;
m_pSendTestSound.Text = "Send";
m_pPlayTestSound.Enabled = false;
m_pPlayTestSound.Text = "Play";
}
else
{
if (m_pOutDevices.SelectedIndex == -1)
{
MessageBox.Show(this, "Please select output device !", "Error:", MessageBoxButtons.OK, MessageBoxIcon.Error);
return;
}
if (m_pRecord.Checked && m_pRecordFile.Text == "")
{
MessageBox.Show(this, "Please specify record file !", "Error:", MessageBoxButtons.OK, MessageBoxIcon.Error);
return;
}
if (m_pRecord.Checked)
{
m_pRecordStream = File.Create(m_pRecordFile.Text);
}
m_IsRunning = true;
m_pWaveOut = new AudioOut(AudioOut.Devices[m_pOutDevices.SelectedIndex], 8000, 16, 1);
m_pRtpSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
// --- Debug -----
wfrm_RTP_Debug frmRtpDebug = new wfrm_RTP_Debug(m_pRtpSession);
frmRtpDebug.Show();
//-----------------
// m_pRtpSession.CreateSession(new RTP_Address(IPAddress.Parse(m_pLocalIP.Text),(int)m_pLocalPort.Value,(int)m_pLocalPort.Value + 1),new RTP_Clock(0,8000));
string ip = "127.0.0.1";
int port = 1100;
int remote_port = 20700;
//RTP_Session session = m_pRtpSession.CreateSession(new RTP_Address(IPAddress.Parse(ip), port, port + 1), new RTP_Clock(0, 8000));
//session.AddTarget(new RTP_Address(IPAddress.Parse("127.0.0.1"), remote_port, remote_port + 1));
RTP_Session session = m_pRtpSession.CreateMulticastSession(new RTP_Address(IPAddress.Parse(ip), port, port + 1), new RTP_Clock(0, 8000), null);
session.AddTarget(new RTP_Address(IPAddress.Parse("224.0.0.1"), remote_port, remote_port + 1));
session.Payload = 0;
session.Start();
m_pOutDevices.Enabled = false;
m_pToggleRun.Text = "Stop";
m_pRecord.Enabled = false;
m_pRecordFile.Enabled = false;
m_pRecordFileBrowse.Enabled = false;
m_pRemoteIP.Enabled = true;
m_pRemotePort.Enabled = true;
m_pCodec.Enabled = true;
m_pToggleMic.Enabled = true;
m_pSendTestSound.Enabled = true;
m_pSendTestSound.Text = "Send";
m_pPlayTestSound.Enabled = true;
m_pPlayTestSound.Text = "Play";
}
wfrm_SendAudio frm = new wfrm_SendAudio(this, m_pRtpSession.Sessions[0], @"D:\_d\_PROJECTS\CisteraDesktopNotificationService\Lumisoft.Net.Rtp\Rtp Audio Demo\Rtp Audio Demo\bin\Debug\audio\futurama.raw");
frm.Show();
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="session">RTP multimedia session.</param>
public wfrm_RTP_Debug(RTP_MultimediaSession session)
{
if (session == null)
{
throw new ArgumentNullException("session");
}
m_pSession = session;
InitUI();
m_pSession.Error += m_pSession_Error;
m_pSession.SessionCreated += m_pSession_SessionCreated;
m_pSession.NewParticipant += m_pSession_NewParticipant;
m_pSession.LocalParticipant.SourceAdded += Participant_SourceAdded;
m_pSession.LocalParticipant.SourceRemoved += Participant_SourceRemoved;
m_pTimer = new Timer();
m_pTimer.Interval = 1000;
m_pTimer.Tick += m_pTimer_Tick;
m_pTimer.Enabled = true;
TreeNode nodeParticipant = new TreeNode(session.LocalParticipant.CNAME);
nodeParticipant.Tag = new RTP_ParticipantInfo(session.LocalParticipant);
nodeParticipant.Nodes.Add("Sources");
m_pParticipants.Nodes.Add(nodeParticipant);
}
