/// <summary>
/// Allows an arbitrary block of bytes to be sent on the RTP channel. This is mainly used for the Gingle
/// client which needs to send a STUN binding request to the Google Voice gateway.
/// </summary>
public void SendRTPRaw(byte[] buffer, int length)
{
if (_rtpAudioChannel != null)
{
_rtpAudioChannel.SendRTPRaw(buffer);
}
}
/// <summary>
/// Event handler for receiving an audio sample that is ready for encoding, packaging into RTP and sending to the remote end
/// of the VoIP call.
/// </summary>
private void RTPChannelSampleAvailable(object sender, WaveInEventArgs e)
{
TimeSpan samplePeriod = DateTime.Now.Subtract(_lastInputSampleReceivedAt);
_lastInputSampleReceivedAt = DateTime.Now;
_inputSampleCount++;
_audioLogger.Debug(_inputSampleCount + " sample period " + samplePeriod.TotalMilliseconds + "ms, sample bytes " + e.BytesRecorded + ".");
byte[] sample = new byte[e.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < e.Buffer.Length; index += 2)
{
var ulawByte = MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(e.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
m_rtpChannel.SendRTPRaw(sample);
}
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)
{
}
}