/// <summary>
/// Sets the remote end point for the RTP channel. This will be set from the SDP packet received from the remote
/// end of the VoIP call.
/// </summary>
/// <param name="remoteEndPoint">The remote end point to send RTP to.</param>
public void SetRemoteRTPEndPoints(IPEndPoint audioRemoteEndPoint, IPEndPoint videoRemoteEndPoint)
{
if (audioRemoteEndPoint != null)
{
logger.Debug("Remote RTP audio end point set as " + audioRemoteEndPoint + ".");
_rtpAudioChannel.RemoteEndPoint = audioRemoteEndPoint;
_rtpAudioChannel.Start();
}
else if (videoRemoteEndPoint != null)
{
logger.Debug("Remote RTP video end point set as " + videoRemoteEndPoint + ".");
_remoteVideoEP = videoRemoteEndPoint;
_rtpVideoChannel.RemoteEndPoint = videoRemoteEndPoint;
_rtpVideoChannel.Start();
}
}
/// <summary>
/// 实时视频请求
/// </summary>
public void RealVideoRequest()
{
SipInitialize();
_mediaPort = _messageCore.SetMediaPort();
SIPRequest request = InviteRequest();
RealVideo real = new RealVideo()
{
Address = _cameraId,
Variable = VariableType.RealMedia,
Privilege = 90,
Format = "4CIF CIF QCIF 720p 1080p",
Video = "H.264",
Audio = "G.711",
MaxBitrate = 800,
Socket = this.ToString()
};
string xmlBody = RealVideo.Instance.Save <RealVideo>(real);
request.Body = xmlBody;
_m_sipTransport.SendRequest(_remoteEndPoint, request);
//启动RTP通道
_rtpChannel.IsClosed = false;
_rtpChannel.ReservePorts(_mediaPort[0], _mediaPort[1]);
_rtpChannel.Start();
}
/// <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 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.");
}
private void Uac_CallAnswered(ISIPClientUserAgent uac, SIPResponse sipResponse)
{
logger.Debug($"{ prefix } Call answered; { sipResponse.StatusCode } { sipResponse.Status }");
switch (sipResponse.StatusCode)
{
case 404:
logger.Error($"{ prefix } Received 404 Not Found from remote endpoint");
break;
case 486:
// Busy
logger.Error($"{ prefix } Received 486 Remote endpoint is busy; try again later");
break;
case 488:
// Possible audio format issue
logger.Error($"{ prefix } Received 488 Not Acceptable from remote endpoint; check audio format");
break;
case 503:
// Check Twilio service and geo-permissions
logger.Error($"{ prefix } Received 503 Service Unavailable from remote endpoint; check service permissions");
break;
case 200:
if (sipResponse.Header.ContentType != SDP.SDP_MIME_CONTENTTYPE)
{
logger.Error($"{ prefix } Received incorrect content type");
Stop();
return;
}
if (sipResponse.Body.IsNullOrBlank())
{
logger.Error($"{ prefix } Received an empty SDP payload");
Stop();
return;
}
var sdp = SDP.ParseSDPDescription(sipResponse.Body);
var ip = IPAddress.Parse(sdp.Connection.ConnectionAddress);
var announcement = sdp.Media.Where(x => x.Media == SDPMediaTypesEnum.audio).FirstOrDefault();
if (announcement != null)
{
if (announcement.Port != 0)
{
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };
rtpChannel.RemoteEndPoint = new IPEndPoint(ip, announcement.Port);
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
}
else
{
logger.Error($"{ prefix } Remote endpoint did not specify a port number");
return;
}
}
else
{
logger.Error($"{ prefix } Remote endpoint has not valid audio announcement");
return;
}
break;
}
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (transaction != null)
{
return;
}
logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);
transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
agent = new SIPServerUserAgent(
transport,
null,
sipRequest.Header.From.FromURI.User,
null,
SIPCallDirection.In,
null,
null,
null,
transaction);
agent.CallCancelled += Agent_CallCancelled;
agent.TransactionComplete += Agent_TransactionComplete;
agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answer = SDP.ParseSDPDescription(agent.CallRequest.Body);
var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
var port = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(address, port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
// TODO Fix hardcoded ports
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += Channel_OnFrameReady;
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{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)
{
}
}
