public async void ChecklistProcessingToFailStateUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPChannel rtpChannel = new RTPChannel(false, null);
var iceSession = new IceSession(rtpChannel, RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug($"host candidate: {hostCandidate}");
}
var remoteCandidate = RTCIceCandidate.Parse("candidate:408132416 1 udp 2113937151 192.168.11.50 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate);
iceSession.SetRemoteCredentials("CI7o", "xxxxxxxxxxxx");
iceSession.StartGathering();
logger.LogDebug($"ICE session retry interval {iceSession.RTO}ms.");
// The defaults are 5 STUN requests and for a checklist with one entry they will be 500ms apart.
await Task.Delay(4000);
Assert.Equal(IceSession.ChecklistEntryState.Failed, iceSession._checklist.Single().State);
Assert.Equal(IceSession.ChecklistState.Failed, iceSession._checklistState);
Assert.Equal(RTCIceConnectionState.failed, iceSession.ConnectionState);
}
public void GetHostCandidatesForRTPBindUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
var localAddress = NetServices.InternetDefaultAddress;
RTPSession rtpSession = new RTPSession(true, true, true, localAddress);
// Add a track to the session in order to initialise the RTPChannel.
MediaStreamTrack dummyTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(dummyTrack);
RTPChannel rtpChannel = rtpSession.GetRtpChannel(SDPMediaTypesEnum.audio);
logger.LogDebug($"RTP channel RTP socket local end point {rtpChannel.RTPLocalEndPoint}.");
var iceSession = new IceSession(rtpChannel, RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
Assert.True(localAddress.Equals(IPAddress.Parse(iceSession.Candidates.Single().address)));
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug(hostCandidate.ToString());
}
}
public async void ChecklistConstructionUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPChannel rtpChannel = new RTPChannel(false, null);
var iceSession = new IceSession(rtpChannel, RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug($"host candidate: {hostCandidate}");
}
var remoteCandidate = RTCIceCandidate.Parse("candidate:408132416 1 udp 2113937151 192.168.11.50 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate);
var remoteCandidate2 = RTCIceCandidate.Parse("candidate:408132417 1 udp 2113937150 192.168.11.50 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate2);
foreach (var entry in iceSession._checklist)
{
logger.LogDebug($"checklist entry: {entry.LocalCandidate} -> {entry.RemoteCandidate}");
}
Assert.Single(iceSession._checklist);
}
public async void ChecklistProcessingUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTPChannel rtpChannel = new RTPChannel(false, null);
var iceSession = new IceSession(rtpChannel, RTCIceComponent.rtp, null);
iceSession.StartGathering();
Assert.NotNull(iceSession);
Assert.NotEmpty(iceSession.Candidates);
foreach (var hostCandidate in iceSession.Candidates)
{
logger.LogDebug($"host candidate: {hostCandidate}");
}
var remoteCandidate = RTCIceCandidate.Parse("candidate:408132416 1 udp 2113937151 192.168.11.50 51268 typ host generation 0 ufrag CI7o network-cost 999");
await iceSession.AddRemoteCandidate(remoteCandidate);
iceSession.SetRemoteCredentials("CI7o", "xxxxxxxxxxxx");
iceSession.StartGathering();
await Task.Delay(2000);
var checklistEntry = iceSession._checklist.Single();
logger.LogDebug($"Checklist entry state {checklistEntry.State}, last check sent at {checklistEntry.LastCheckSentAt}.");
Assert.Equal(IceSession.ChecklistEntryState.InProgress, checklistEntry.State);
}
public AudioChannel()
{
// Set up the device that will play the audio from the RTP received from the remote end of the call.
m_waveOut = new WaveOut();
m_waveProvider = new BufferedWaveProvider(_waveFormat);
m_waveOut.Init(m_waveProvider);
m_waveOut.Play();
// Set up the input device that will provide audio samples that can be encoded, packaged into RTP and sent to
// the remote end of the call.
m_waveInEvent = new WaveInEvent();
m_waveInEvent.BufferMilliseconds = 20;
m_waveInEvent.NumberOfBuffers = 1;
m_waveInEvent.DeviceNumber = 0;
m_waveInEvent.DataAvailable += RTPChannelSampleAvailable;
m_waveInEvent.WaveFormat = _waveFormat;
// Create a UDP socket to use for sending and receiving RTP packets.
int port = FreePort.FindNextAvailableUDPPort(DEFAULT_START_RTP_PORT);
_rtpEndPoint = new IPEndPoint(_defaultLocalAddress, port);
m_rtpChannel = new RTPChannel(_rtpEndPoint);
m_rtpChannel.OnFrameReady += RTPChannelSampleReceived;
_audioLogger.Debug("RTP channel endpoint " + _rtpEndPoint.ToString());
}
/// <summary>
/// 确认接收视频请求
/// </summary>
/// <param name="response">响应消息</param>
/// <returns></returns>
public void AckRequest(SIPResponse response)
{
_rtpChannel = new RTPChannel(_remoteEndPoint.GetIPEndPoint(), _mediaPort[0], _mediaPort[1], FrameTypesEnum.H264);
_rtpChannel.OnFrameReady += _rtpChannel_OnFrameReady;
_rtpChannel.Start();
SIPURI localUri = new SIPURI(_msgCore.LocalSIPId, _msgCore.LocalEndPoint.ToHost(), "");
SIPURI remoteUri = new SIPURI(_deviceId, _remoteEndPoint.ToHost(), "");
SIPRequest ackReq = _msgCore.Transport.GetRequest(SIPMethodsEnum.ACK, remoteUri);
SIPFromHeader from = new SIPFromHeader(null, response.Header.From.FromURI, response.Header.From.FromTag);
SIPToHeader to = new SIPToHeader(null, remoteUri, response.Header.To.ToTag);
SIPContactHeader contactHeader = new SIPContactHeader(null, localUri);
SIPHeader header = new SIPHeader(from, to, response.Header.CSeq, response.Header.CallId);
header.CSeqMethod = SIPMethodsEnum.ACK;
header.Contact = response.Header.Contact;
header.Contact.Clear();
header.Contact.Add(contactHeader);
header.Vias = response.Header.Vias;
header.MaxForwards = response.Header.MaxForwards;
header.ContentLength = response.Header.ContentLength;
header.UserAgent = _msgCore.UserAgent;
header.Allow = null;
ackReq.Header = header;
_okTag = response.Header.To.ToTag;
_contact = header.Contact.FirstOrDefault();
_via = header.Vias;
_msgCore.Transport.SendRequest(_remoteEndPoint, ackReq);
}
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($"Test complete.");
}
public void Start(string endpoint)
{
this.endpoint = endpoint;
var caller = "1003";
var password = passwords[0];
var port = FreePort.FindNextAvailableUDPPort(15090);
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(IPAddress.Parse(asterisk), port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += RtpChannel_OnFrameReady;
uac = new SIPClientUserAgent(transport, null, null, null, null);
var uri = SIPURI.ParseSIPURIRelaxed($"{ endpoint }@{ asterisk }");
var from = (new SIPFromHeader(caller, new SIPURI(caller, asterisk, null), null)).ToString();
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())
};
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);
var descriptor = new SIPCallDescriptor(caller, password, uri.ToString(), from, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null);
uac.CallTrying += Uac_CallTrying;
uac.CallRinging += Uac_CallRinging;
uac.CallAnswered += Uac_CallAnswered;
uac.CallFailed += Uac_CallFailed;
uac.Call(descriptor);
}
/// <summary>
/// rtcp包回调事件处理
/// </summary>
/// <param name="buffer"></param>
/// <param name="rtcpSocket"></param>
private void _rtpChannel_OnControlDataReceived(byte[] buffer, Socket rtcpSocket)
{
_rtcpSocket = rtcpSocket;
DateTime packetTimestamp = DateTime.Now;
_rtcpTimestamp = RTPChannel.DateTimeToNptTimestamp90K(DateTime.Now);
if (_rtcpRemoteEndPoint != null)
{
SendRtcpSenderReport(RTPChannel.DateTimeToNptTimestamp(packetTimestamp), _rtcpTimestamp);
}
}
/// <param name="rtpListenAddress">The local IP address to establish the RTP listener socket on.</param>
/// <param name="sdpAdvertiseAddress">The public IP address to put into the SDP sent back to the caller.</param>
/// <param name="request">The INVITE request that instigated the RTP diagnostics job.</param>
public RTPDiagnosticsJob(IPAddress rtpListenAddress, IPAddress sdpAdvertiseAddress, SIPServerUserAgent uas, SIPRequest request)
{
m_request = request;
m_remoteSDP = SDP.ParseSDPDescription(request.Body);
RemoteRTPEndPoint = new IPEndPoint(IPAddress.Parse(m_remoteSDP.Connection.ConnectionAddress), m_remoteSDP.Media[0].Port);
UAS = uas;
//m_rawSourceStream = new RawSourceWaveStream(m_outStream, WaveFormat.CreateMuLawFormat(8000, 1));
//m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
//m_outPCMStream = WaveFormatConversionStream.CreatePcmStream(m_rawSourceStream);
//m_rawRTPPayloadWriter = new StreamWriter("out.rtp");
//m_rawRTPPayloadReader = new StreamReader("in.rtp");
//IPEndPoint rtpListenEndPoint = null;
IPEndPoint rtpListenEndPoint = null;
NetServices.CreateRandomUDPListener(rtpListenAddress, RTP_PORTRANGE_START, RTP_PORTRANGE_END, m_inUsePorts, out rtpListenEndPoint);
RTPListenEndPoint = rtpListenEndPoint;
m_inUsePorts.Add(rtpListenEndPoint.Port);
//RTPListenEndPoint = new IPEndPoint(rtpListenAddress, RTP_PORTRANGE_START);
m_rtpChannel = new RTPChannel(RTPListenEndPoint);
m_rtpChannel.SampleReceived += SampleReceived;
ThreadPool.QueueUserWorkItem(delegate { GetAudioSamples(); });
LocalSDP = new SDP()
{
SessionId = Crypto.GetRandomString(6),
Address = sdpAdvertiseAddress.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(sdpAdvertiseAddress.ToString()),
Media = new List <SDPMediaAnnouncement>()
{
new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
Port = RTPListenEndPoint.Port,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU)
}
}
}
};
}
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.");
}
/// <summary>
/// sip初始化完成事件
/// </summary>
/// <param name="sipRequest">sip请求</param>
/// <param name="localEndPoint">本地终结点</param>
/// <param name="remoteEndPoint">远程终结点</param>
/// <param name="sipAccount">sip账户</param>
private void messageCore_SipRequestInited(SIPRequest sipRequest, SIPEndPoint localEndPoint, SIPEndPoint remoteEndPoint, SIPAccount sipAccount)
{
_sipInited = true;
_sipRequest = sipRequest;
_localEndPoint = localEndPoint;
_remoteEndPoint = remoteEndPoint;
_sipAccount = sipAccount;
_rtpRemoteEndPoint = new IPEndPoint(remoteEndPoint.Address, remoteEndPoint.Port);
_rtpChannel = new RTPChannel(_rtpRemoteEndPoint);
_rtpChannel.OnFrameReady += _rtpChannel_OnFrameReady;
_rtpChannel.OnControlDataReceived += _rtpChannel_OnControlDataReceived;
if (SipStatusHandler != null)
{
SipStatusHandler(SipServiceStatus.Inited);
}
_messageCore.SipRequestInited -= messageCore_SipRequestInited;
}
public event Action <byte[], int> OnRemoteVideoSampleReady; // Fires when a remote video sample is ready for display.
public RTPManager(bool includeAudio, bool includeVideo)
{
if (includeAudio)
{
// Create a UDP socket to use for sending and receiving RTP audio packets.
_rtpAudioChannel = new RTPChannel();
_rtpAudioChannel.SetFrameType(FrameTypesEnum.Audio);
_rtpAudioChannel.ReservePorts(DEFAULT_START_RTP_PORT, DEFAULT_END_RTP_PORT);
_rtpAudioChannel.OnFrameReady += AudioFrameReady;
}
if (includeVideo)
{
_rtpVideoChannel = new RTPChannel();
_rtpVideoChannel.SetFrameType(FrameTypesEnum.VP8);
_rtpVideoChannel.ReservePorts(DEFAULT_START_RTP_PORT, DEFAULT_END_RTP_PORT);
_rtpVideoChannel.OnFrameReady += VideoFrameReady;
_rtpVideoChannel.OnRTPSocketDisconnected += () => { };
}
}
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");
}
}
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)
{
}
}
