/// <summary>
/// Called when the media channels are no longer required, such as when the VoIP call using it has terminated, and all resources can be shutdown
/// and closed.
/// </summary>
public void Close()
{
try
{
logger.Debug("RTP Manager closing.");
_stop = true;
if (_rtpAudioChannel != null)
{
_rtpAudioChannel.OnFrameReady -= AudioFrameReady;
_rtpAudioChannel.Close();
}
if (_rtpVideoChannel != null)
{
_rtpVideoChannel.OnFrameReady -= VideoFrameReady;
_rtpVideoChannel.Close();
}
}
catch (Exception excp)
{
logger.Error("Exception RTP Manager Close. " + excp);
}
}
public void MultipleRtpChannelLoopbackUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
const int PACKET_LENGTH = 100;
List <Task> tasks = new List <Task>();
for (int i = 0; i < 3; i++)
{
var t = Task.Run(async() =>
{
RTPChannel channel1 = new RTPChannel(false, null);
bool testResult = false;
ManualResetEventSlim testCompleteEvent = new ManualResetEventSlim(false);
RTPChannel channel2 = new RTPChannel(false, null);
channel2.OnRTPDataReceived += (lep, rep, pkt) =>
{
logger.LogDebug($"RTP data receive packet length {pkt.Length}.");
testResult = pkt.Length == PACKET_LENGTH;
testCompleteEvent.Set();
};
channel1.Start();
channel2.Start();
// Give the socket receive tasks time to fire up.
await Task.Delay(2000);
IPAddress channel2Address = (channel2.RTPLocalEndPoint.AddressFamily == AddressFamily.InterNetworkV6) ? IPAddress.IPv6Loopback : IPAddress.Loopback;
IPEndPoint channel2Dst = new IPEndPoint(channel2Address, channel2.RTPPort);
logger.LogDebug($"Attempting to send packet from {channel1.RTPLocalEndPoint} to {channel2Dst}.");
var sendResult = channel1.SendAsync(RTPChannelSocketsEnum.RTP, channel2Dst, new byte[PACKET_LENGTH]);
logger.LogDebug($"Send result {sendResult}.");
testCompleteEvent.Wait(TimeSpan.FromSeconds(TEST_TIMEOUT_SECONDS));
Assert.True(testResult);
channel1.Close("normal");
channel2.Close("normal");
});
tasks.Add(t);
}
CancellationTokenSource cts = new CancellationTokenSource();
Assert.True(Task.WaitAll(tasks.ToArray(), 10000, cts.Token));
logger.LogDebug($"Test complete.");
}
public void Stop()
{
logger.Debug($"{ prefix } Stop consumer for { endpoint }");
if (uac.IsUACAnswered && uac.SIPDialogue != null)
{
uac.SIPDialogue.Hangup(transport, null);
}
rtpChannel.OnFrameReady -= RtpChannel_OnFrameReady;
if (!rtpChannel.IsClosed)
{
rtpChannel.Close();
}
StateChanged?.Invoke(this, new ConsumerStateEventArgs(State.Ready, endpoint, SessionId));
}
/// <summary>
/// 实时视频取消
/// </summary>
public void RealVideoBye()
{
SipInitialize();
_rtpChannel.Close();
if (_realReqSession == null)
{
return;
}
SIPRequest req = ByeRequest();
_m_sipTransport.SendRequest(_remoteEndPoint, req);
}
public async void RtpChannelLoopbackUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPChannel channel1 = new RTPChannel(false, null);
bool testResult = false;
ManualResetEventSlim testCompleteEvent = new ManualResetEventSlim(false);
RTPChannel channel2 = new RTPChannel(false, null);
channel2.OnRTPDataReceived += (lep, rep, pkt) =>
{
logger.LogDebug($"RTP data receive packet length {pkt.Length}.");
testResult = true;
testCompleteEvent.Set();
};
channel1.Start();
channel2.Start();
// Give the socket receive tasks time to fire up.
await Task.Delay(1000);
IPAddress channel2Address = (channel2.RTPLocalEndPoint.AddressFamily == AddressFamily.InterNetworkV6) ? IPAddress.IPv6Loopback : IPAddress.Loopback;
IPEndPoint channel2Dst = new IPEndPoint(channel2Address, channel2.RTPPort);
logger.LogDebug($"Attempting to send packet from {channel1.RTPLocalEndPoint} to {channel2Dst}.");
var sendResult = channel1.SendAsync(RTPChannelSocketsEnum.RTP, channel2Dst, new byte[] { 0x00 });
logger.LogDebug($"Send result {sendResult}.");
testCompleteEvent.Wait(TimeSpan.FromSeconds(TEST_TIMEOUT_SECONDS));
Assert.True(testResult);
channel1.Close("normal");
channel2.Close("normal");
logger.LogDebug($"Test complete.");
}
public void RtpChannelCreateManyUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
List <RTPChannel> channels = new List <RTPChannel>();
for (int i = 0; i < 10; i++)
{
RTPChannel channel = new RTPChannel(true, null);
channels.Add(channel);
}
Assert.Equal(10, channels.Count);
foreach (var channel in channels)
{
channel.Close("normal");
}
}
/// <summary>
/// Called when the audo channel is no longer required, such as when the VoIP call using it has terminated, and all resources can be shutdown
/// and closed.
/// </summary>
public void Close()
{
try
{
_audioLogger.Debug("Closing audio channel.");
m_rtpChannel.Close();
if (m_recordingStarted)
{
m_waveInEvent.StopRecording();
}
if (m_waveOut.PlaybackState == PlaybackState.Playing)
{
m_waveOut.Stop(); m_waveOut.Dispose();
}
}
catch (Exception excp)
{
_audioLogger.Error("Exception AudioChannel Close. " + excp.Message);
}
}
/// <summary>
/// 停止计时器/关闭RTP通道
/// </summary>
public void Stop()
{
if (_realTask != null)
{
_realTask.OnCloseRTPChannel -= Task_OnCloseRTPChannel;
_realTask.Stop();
}
if (_byeTask != null)
{
_byeTask.OnCloseRTPChannel -= Task_OnCloseRTPChannel;
_byeTask.Stop();
}
if (_rtpChannel != null)
{
_rtpChannel.OnFrameReady -= _rtpChannel_OnFrameReady;
_rtpChannel.Close();
}
if (m_fs != null)
{
m_fs.Close();
m_fs.Dispose();
m_fs = null;
}
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (transaction != null)
{
return;
}
logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);
transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
agent = new SIPServerUserAgent(
transport,
null,
sipRequest.Header.From.FromURI.User,
null,
SIPCallDirection.In,
null,
null,
null,
transaction);
agent.CallCancelled += Agent_CallCancelled;
agent.TransactionComplete += Agent_TransactionComplete;
agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answer = SDP.ParseSDPDescription(agent.CallRequest.Body);
var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
var port = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(address, port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
// TODO Fix hardcoded ports
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += Channel_OnFrameReady;
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
SetState(State.Listening, sipRequest.Header.From.FromURI.User);
agent.Progress(SIPResponseStatusCodesEnum.Accepted, null, null, null, null);
agent.Answer(SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
SetState(State.Busy, "");
return;
}
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (State != State.Busy)
{
return;
}
logger.DebugFormat("{0} Hangup from {1}", prefix, sipRequest.Header.From.FromURI.User);
var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, endpoint, null);
var response = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
noninvite.SendFinalResponse(response);
SetState(State.Finished, Endpoint);
rtpChannel.OnFrameReady -= Channel_OnFrameReady;
rtpChannel.Close();
agent.TransactionComplete -= Agent_TransactionComplete;
agent.CallCancelled -= Agent_CallCancelled;
agent = null;
transaction = null;
SetState(State.Ready, Endpoint);
return;
}
if (sipRequest.Method == SIPMethodsEnum.ACK)
{
}
if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
}
}
