/// <summary>
/// Sends the sounds of silence. If the destination is on the other side of a NAT this is useful to open
/// a pinhole and hopefully get the remote RTP stream through.
/// </summary>
/// <param name="rtpSocket">The socket we're using to send from.</param>
/// <param name="rtpSendSession">Our RTP sending session.</param>
/// <param name="cts">Cancellation token to stop the call.</param>
private static async void SendRtp(Socket rtpSocket, RTPSession rtpSendSession, CancellationTokenSource cts)
{
uint bufferSize = (uint)SILENCE_SAMPLE_PERIOD * 8; // PCM transmission rate is 64kbit/s.
uint rtpSamplePeriod = (uint)(1000 / SILENCE_SAMPLE_PERIOD);
uint rtpSendTimestamp = 0;
uint packetSentCount = 0;
uint bytesSentCount = 0;
while (cts.IsCancellationRequested == false)
{
byte[] sample = new byte[bufferSize / 2];
int sampleIndex = 0;
for (int index = 0; index < bufferSize; index += 2)
{
sample[sampleIndex] = PCMU_SILENCE_BYTE_ZERO;
sample[sampleIndex + 1] = PCMU_SILENCE_BYTE_ONE;
}
if (_remoteRtpEndPoint != null)
{
rtpSendSession.SendAudioFrame(rtpSocket, _remoteRtpEndPoint, rtpSendTimestamp, sample);
rtpSendTimestamp += rtpSamplePeriod;
packetSentCount++;
bytesSentCount += (uint)sample.Length;
}
await Task.Delay((int)rtpSamplePeriod);
}
}
/// <summary>
/// Sends the sounds of silence. If the destination is on the other side of a NAT this is useful to open
/// a pinhole and hopefully get the remote RTP stream through.
/// </summary>
/// <param name="rtpChannel">The RTP channel we're sending from.</param>
/// <param name="rtpSendSession">Our RTP sending session.</param>
/// <param name="cts">Cancellation token to stop the call.</param>
private static async void SendSilence(RTPSession rtpSession, CancellationTokenSource cts)
{
int samplingFrequency = rtpSession.MediaFormat.GetClockRate();
uint rtpTimestampStep = (uint)(samplingFrequency * SILENCE_SAMPLE_PERIOD / 1000);
uint bufferSize = (uint)SILENCE_SAMPLE_PERIOD;
uint rtpSampleTimestamp = 0;
while (cts.IsCancellationRequested == false)
{
if (rtpSession.DestinationEndPoint != null)
{
byte[] sample = new byte[bufferSize / 2];
int sampleIndex = 0;
for (int index = 0; index < bufferSize; index += 2)
{
sample[sampleIndex] = PCMU_SILENCE_BYTE_ZERO;
sample[sampleIndex + 1] = PCMU_SILENCE_BYTE_ONE;
}
rtpSession.SendAudioFrame(rtpSampleTimestamp, sample);
rtpSampleTimestamp += rtpTimestampStep;
}
await Task.Delay(SILENCE_SAMPLE_PERIOD);
}
}
private static void SendRtp(Socket rtpSocket, RTPSession rtpSendSession, CancellationTokenSource cts)
{
WaveFormat waveFormat = new WaveFormat(8000, 16, 1); // The format that both the input and output audio streams will use, i.e. PCMU.
// 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.
if (WaveInEvent.DeviceCount == 0)
{
Log.LogWarning("No audio input devices available. No audio will be sent.");
}
else
{
DateTime lastSendReportAt = DateTime.Now;
uint rtpSendTimestamp = 0;
uint packetSentCount = 0;
uint bytesSentCount = 0;
// Device used to get audio sample from, e.g. microphone.
using (WaveInEvent waveInEvent = new WaveInEvent())
{
waveInEvent.BufferMilliseconds = 20; // This sets the frequency of the RTP packets.
waveInEvent.NumberOfBuffers = 1;
waveInEvent.DeviceNumber = 0;
waveInEvent.WaveFormat = waveFormat;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (_remoteRtpEndPoint != null)
{
rtpSendSession.SendAudioFrame(rtpSocket, _remoteRtpEndPoint, rtpSendTimestamp, sample);
rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
packetSentCount++;
bytesSentCount += (uint)sample.Length;
}
if (DateTime.Now.Subtract(lastSendReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
// This is typically where RTCP sender (SR) reports would be sent. Omitted here for brevity.
lastSendReportAt = DateTime.Now;
var remoteRtpEndPoint = _remoteRtpEndPoint as IPEndPoint;
Log.LogDebug($"RTP send report {rtpSocket.LocalEndPoint}->{remoteRtpEndPoint} pkts {packetSentCount} bytes {bytesSentCount}");
}
};
waveInEvent.StartRecording();
cts.Token.WaitHandle.WaitOne();
}
}
}
/// <summary>
/// Sends two separate RTP streams to an application like ffplay.
///
/// ffplay -i ffplay_av.sdp -protocol_whitelist "file,rtp,udp" -loglevel debug
///
/// The SDP that describes the streams is:
///
/// v=0
/// o=- 1129870806 2 IN IP4 127.0.0.1
/// s=-
/// c=IN IP4 192.168.11.50
/// t=0 0
/// m=audio 4040 RTP/AVP 0
/// a=rtpmap:0 PCMU/8000
/// m=video 4042 RTP/AVP 100
/// a=rtpmap:100 VP8/90000
/// </summary>
private void SendSamplesAsRtp(IPEndPoint dstBaseEndPoint)
{
try
{
Socket videoSrcRtpSocket = null;
Socket videoSrcControlSocket = null;
Socket audioSrcRtpSocket = null;
Socket audioSrcControlSocket = null;
// WebRtc multiplexes all the RTP and RTCP sessions onto a single UDP connection.
// The approach needed for ffplay is the original way where each media type has it's own UDP connection and the RTCP
// also require a separate UDP connection on RTP port + 1.
IPAddress localIPAddress = IPAddress.Any;
IPEndPoint audioRtpEP = dstBaseEndPoint;
IPEndPoint audioRtcpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 1);
IPEndPoint videoRtpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 2);
IPEndPoint videoRtcpEP = new IPEndPoint(dstBaseEndPoint.Address, dstBaseEndPoint.Port + 3);
RTPSession audioRtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
RTPSession videoRtpSession = new RTPSession(VP8_PAYLOAD_TYPE_ID, null, null);
DateTime lastRtcpSenderReportSentAt = DateTime.Now;
NetServices.CreateRtpSocket(localIPAddress, RAW_RTP_START_PORT_RANGE, RAW_RTP_END_PORT_RANGE, true, out audioSrcRtpSocket, out audioSrcControlSocket);
NetServices.CreateRtpSocket(localIPAddress, ((IPEndPoint)audioSrcRtpSocket.LocalEndPoint).Port, RAW_RTP_END_PORT_RANGE, true, out videoSrcRtpSocket, out videoSrcControlSocket);
OnMediaSampleReady += (mediaType, timestamp, sample) =>
{
if (mediaType == MediaSampleTypeEnum.VP8)
{
videoRtpSession.SendVp8Frame(videoSrcRtpSocket, videoRtpEP, timestamp, sample);
}
else
{
audioRtpSession.SendAudioFrame(audioSrcRtpSocket, audioRtpEP, timestamp, sample);
}
// Deliver periodic RTCP sender reports. This helps the receiver to sync the audio and video stream timestamps.
// If there are gaps in the media, silence supression etc. then the sender repors shouldn't be triggered from the media samples.
// In this case the samples are from an mp4 file which provides a constant uninterrupted stream.
if (DateTime.Now.Subtract(lastRtcpSenderReportSentAt).TotalSeconds >= RTCP_SR_PERIOD_SECONDS)
{
videoRtpSession.SendRtcpSenderReport(videoSrcControlSocket, videoRtcpEP, _vp8Timestamp);
audioRtpSession.SendRtcpSenderReport(audioSrcControlSocket, audioRtcpEP, _mulawTimestamp);
lastRtcpSenderReportSentAt = DateTime.Now;
}
};
}
catch (Exception excp)
{
logger.Error("Exception SendSamplesAsRtp. " + excp);
}
}
/// <summary>
/// Sends the sounds of silence. If the destination is on the other side of a NAT this is useful to open
/// a pinhole and hopefully get the remote RTP stream through.
/// </summary>
/// <param name="rtpSocket">The socket we're using to send from.</param>
/// <param name="rtpSendSession">Our RTP sending session.</param>
/// <param name="cts">Cancellation token to stop the call.</param>
private static async void SendRtp(Socket rtpSocket, RTPSession rtpSendSession, CancellationTokenSource cts)
{
int samplingFrequency = RTPPayloadTypes.GetSamplingFrequency((RTPPayloadTypesEnum)rtpSendSession.PayloadType);
uint rtpTimestampStep = (uint)(samplingFrequency * SILENCE_SAMPLE_PERIOD / 1000);
uint bufferSize = (uint)SILENCE_SAMPLE_PERIOD;
uint rtpSendTimestamp = 0;
uint packetSentCount = 0;
uint bytesSentCount = 0;
while (cts.IsCancellationRequested == false)
{
if (_remoteRtpEndPoint != null)
{
if (!_dtmfEvents.IsEmpty)
{
// Check if there are any DTMF events to send.
_dtmfEvents.TryDequeue(out var rtpEvent);
if (rtpEvent != null)
{
await rtpSendSession.SendDtmfEvent(rtpSocket, _remoteRtpEndPoint, rtpEvent, rtpSendTimestamp, (ushort)SILENCE_SAMPLE_PERIOD, (ushort)rtpTimestampStep, cts);
}
rtpSendTimestamp += rtpEvent.TotalDuration + rtpTimestampStep;
}
else
{
// If there are no DTMF events to send we'll send silence.
byte[] sample = new byte[bufferSize / 2];
int sampleIndex = 0;
for (int index = 0; index < bufferSize; index += 2)
{
sample[sampleIndex] = PCMU_SILENCE_BYTE_ZERO;
sample[sampleIndex + 1] = PCMU_SILENCE_BYTE_ONE;
}
rtpSendSession.SendAudioFrame(rtpSocket, _remoteRtpEndPoint, rtpSendTimestamp, sample);
rtpSendTimestamp += rtpTimestampStep;
packetSentCount++;
bytesSentCount += (uint)sample.Length;
}
}
await Task.Delay(SILENCE_SAMPLE_PERIOD);
}
}
public void SendMedia(MediaSampleTypeEnum mediaType, uint sampleTimestamp, byte[] sample)
{
var connectedIceCandidate = Peer.LocalIceCandidates.Where(y => y.RemoteRtpEndPoint != null).FirstOrDefault();
if (connectedIceCandidate != null)
{
var srcRtpEndPoint = connectedIceCandidate.LocalRtpSocket;
var dstRtpEndPoint = connectedIceCandidate.RemoteRtpEndPoint;
if (mediaType == MediaSampleTypeEnum.VP8)
{
_videoRtpSession.SendVp8Frame(srcRtpEndPoint, dstRtpEndPoint, sampleTimestamp, sample);
}
else
{
_audioRtpSession.SendAudioFrame(srcRtpEndPoint, dstRtpEndPoint, sampleTimestamp, sample);
}
}
}
/// <summary>
/// Sends the sounds of silence. If the destination is on the other side of a NAT this is useful to open
/// a pinhole and hopefully get the remote RTP stream through.
/// </summary>
/// <param name="rtpSocket">The socket we're using to send from.</param>
/// <param name="rtpSendSession">Our RTP sending session.</param>
/// <param name="cts">Cancellation token to stop the call.</param>
private static async void SendRtp(Socket rtpSocket, RTPSession rtpSendSession, CancellationTokenSource cts)
{
try
{
while (cts.IsCancellationRequested == false)
{
uint timestamp = 0;
using (StreamReader sr = new StreamReader(AUDIO_FILE_PCMU))
{
DateTime lastSendReportAt = DateTime.Now;
uint packetsSentCount = 0;
uint bytesSentCount = 0;
byte[] buffer = new byte[320];
int bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
while (bytesRead > 0 && !cts.IsCancellationRequested)
{
if (rtpSendSession.DestinationEndPoint != null)
{
packetsSentCount++;
bytesSentCount += (uint)bytesRead;
rtpSendSession.SendAudioFrame(rtpSocket, rtpSendSession.DestinationEndPoint, timestamp, buffer);
}
timestamp += (uint)buffer.Length;
if (DateTime.Now.Subtract(lastSendReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
lastSendReportAt = DateTime.Now;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP send report {rtpSocket.LocalEndPoint}->{rtpSendSession.DestinationEndPoint} pkts {packetsSentCount} bytes {bytesSentCount}");
}
await Task.Delay(40, cts.Token);
bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
}
}
}
}
catch (ObjectDisposedException) // Gets thrown when the RTP socket is closed. Can safely ignore.
{ }
}
private static async Task SendRecvRtp(Socket rtpSocket, RTPSession rtpSession, IPEndPoint dstRtpEndPoint, string audioFileName, CancellationTokenSource cts)
{
try
{
SIPSorcery.Sys.Log.Logger.LogDebug($"Sending from RTP socket {rtpSocket.LocalEndPoint} to {dstRtpEndPoint}.");
// Nothing is being done with the data being received from the client. But the remote rtp socket will
// be switched if it differs from the one in the SDP. This helps cope with NAT.
var rtpRecvTask = Task.Run(async() =>
{
DateTime lastRecvReportAt = DateTime.Now;
uint packetReceivedCount = 0;
uint bytesReceivedCount = 0;
byte[] buffer = new byte[512];
EndPoint remoteEP = new IPEndPoint(IPAddress.Any, 0);
SIPSorcery.Sys.Log.Logger.LogDebug($"Listening on RTP socket {rtpSocket.LocalEndPoint}.");
var recvResult = await rtpSocket.ReceiveFromAsync(buffer, SocketFlags.None, remoteEP);
while (recvResult.ReceivedBytes > 0 && !cts.IsCancellationRequested)
{
RTPPacket rtpPacket = new RTPPacket(buffer.Take(recvResult.ReceivedBytes).ToArray());
packetReceivedCount++;
bytesReceivedCount += (uint)rtpPacket.Payload.Length;
recvResult = await rtpSocket.ReceiveFromAsync(buffer, SocketFlags.None, remoteEP);
if (DateTime.Now.Subtract(lastRecvReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
lastRecvReportAt = DateTime.Now;
dstRtpEndPoint = recvResult.RemoteEndPoint as IPEndPoint;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP recv {rtpSocket.LocalEndPoint}<-{dstRtpEndPoint} pkts {packetReceivedCount} bytes {bytesReceivedCount}");
}
}
});
string audioFileExt = Path.GetExtension(audioFileName).ToLower();
switch (audioFileExt)
{
case ".g722":
case ".ulaw":
{
uint timestamp = 0;
using (StreamReader sr = new StreamReader(audioFileName))
{
DateTime lastSendReportAt = DateTime.Now;
uint packetReceivedCount = 0;
uint bytesReceivedCount = 0;
byte[] buffer = new byte[320];
int bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
while (bytesRead > 0 && !cts.IsCancellationRequested)
{
packetReceivedCount++;
bytesReceivedCount += (uint)bytesRead;
if (!dstRtpEndPoint.Address.Equals(IPAddress.Any))
{
rtpSession.SendAudioFrame(rtpSocket, dstRtpEndPoint, timestamp, buffer);
}
timestamp += (uint)buffer.Length;
if (DateTime.Now.Subtract(lastSendReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
lastSendReportAt = DateTime.Now;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP send {rtpSocket.LocalEndPoint}->{dstRtpEndPoint} pkts {packetReceivedCount} bytes {bytesReceivedCount}");
}
await Task.Delay(40, cts.Token);
bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
}
}
}
break;
case ".mp3":
{
DateTime lastSendReportAt = DateTime.Now;
uint packetReceivedCount = 0;
uint bytesReceivedCount = 0;
var pcmFormat = new WaveFormat(8000, 16, 1);
var ulawFormat = WaveFormat.CreateMuLawFormat(8000, 1);
uint timestamp = 0;
using (WaveFormatConversionStream pcmStm = new WaveFormatConversionStream(pcmFormat, new Mp3FileReader(audioFileName)))
{
using (WaveFormatConversionStream ulawStm = new WaveFormatConversionStream(ulawFormat, pcmStm))
{
byte[] buffer = new byte[320];
int bytesRead = ulawStm.Read(buffer, 0, buffer.Length);
while (bytesRead > 0 && !cts.IsCancellationRequested)
{
packetReceivedCount++;
bytesReceivedCount += (uint)bytesRead;
byte[] sample = new byte[bytesRead];
Array.Copy(buffer, sample, bytesRead);
if (dstRtpEndPoint.Address != IPAddress.Any)
{
rtpSession.SendAudioFrame(rtpSocket, dstRtpEndPoint, timestamp, buffer);
}
timestamp += (uint)buffer.Length;
if (DateTime.Now.Subtract(lastSendReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
lastSendReportAt = DateTime.Now;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP send {rtpSocket.LocalEndPoint}->{dstRtpEndPoint} pkts {packetReceivedCount} bytes {bytesReceivedCount}");
}
await Task.Delay(40, cts.Token);
bytesRead = ulawStm.Read(buffer, 0, buffer.Length);
}
}
}
}
break;
default:
throw new NotImplementedException($"The {audioFileExt} file type is not understood by this example.");
}
}
catch (OperationCanceledException) { }
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception sending RTP. {excp.Message}");
}
}
private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5; // Period at which to write RTP stats.
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource cts = new CancellationTokenSource();
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
IPAddress defaultAddr = LocalIPConfig.GetDefaultIPv4Address();
var sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
int port = FreePort.FindNextAvailableUDPPort(SIPConstants.DEFAULT_SIP_PORT + 2);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(defaultAddr, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += (uac, resp) => SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
SIPSorcery.Sys.Log.Logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
IPEndPoint remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
SIPSorcery.Sys.Log.Logger.LogDebug($"Sending initial RTP packet to remote RTP socket {remoteRtpEndPoint}.");
// Send a dummy packet to open the NAT session on the RTP path.
rtpSendSession.SendAudioFrame(rtpSocket, remoteRtpEndPoint, 0, new byte[] { 0x00 });
}
else
{
SIPSorcery.Sys.Log.Logger.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
cts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => SendRecvRtp(rtpSocket, rtpSendSession, cts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
DESTINATION_SIP_URI,
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += async delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
cts.Cancel();
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to clean up...");
await Task.Delay(1000);
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
/// <summary>
/// Handling packets received on the RTP socket. One of the simplest, if not the simplest, cases, is
/// PCMU audio packets. THe handling can get substantially more complicated if the RTP socket is being
/// used to multiplex different protocols. This is what WebRTC does with STUN, RTP and RTCP.
/// </summary>
/// <param name="rtpSocket">The raw RTP socket.</param>
/// <param name="rtpSendSession">The session infor for the RTP pakcets being sent.</param>
private static async void SendRecvRtp(Socket rtpSocket, RTPSession rtpSendSession, CancellationTokenSource cts)
{
try
{
DateTime lastRecvReportAt = DateTime.Now;
uint packetReceivedCount = 0;
uint bytesReceivedCount = 0;
uint packetSentCount = 0;
uint bytesSentCount = 0;
byte[] buffer = new byte[512];
uint rtpSendTimestamp = 0;
IPEndPoint anyEndPoint = new IPEndPoint(IPAddress.Any, 0);
SIPSorcery.Sys.Log.Logger.LogDebug($"Listening on RTP socket {rtpSocket.LocalEndPoint}.");
using (var waveOutEvent = new WaveOutEvent())
{
var waveProvider = new BufferedWaveProvider(new WaveFormat(8000, 16, 1));
waveOutEvent.Init(waveProvider);
waveOutEvent.Play();
var recvResult = await rtpSocket.ReceiveFromAsync(buffer, SocketFlags.None, anyEndPoint);
SIPSorcery.Sys.Log.Logger.LogDebug($"Initial RTP packet recieved from {recvResult.RemoteEndPoint}.");
while (recvResult.ReceivedBytes > 0 && !cts.IsCancellationRequested)
{
var rtpPacket = new RTPPacket(buffer.Take(recvResult.ReceivedBytes).ToArray());
packetReceivedCount++;
bytesReceivedCount += (uint)rtpPacket.Payload.Length;
for (int index = 0; index < rtpPacket.Payload.Length; index++)
{
short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(rtpPacket.Payload[index]);
byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
waveProvider.AddSamples(pcmSample, 0, 2);
}
// Periodically Send a dummy packet to keep any NAT session that may be on the media path open.
if (DateTime.Now.Subtract(lastRecvReportAt).TotalSeconds > RTP_REPORTING_PERIOD_SECONDS)
{
// This is typically where RTCP reports would be sent. Omitted here for brevity.
lastRecvReportAt = DateTime.Now;
var remoteRtpEndPoint = recvResult.RemoteEndPoint as IPEndPoint;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP recv {rtpSocket.LocalEndPoint}<-{remoteRtpEndPoint} pkts {packetReceivedCount} bytes {bytesReceivedCount}");
rtpSendSession.SendAudioFrame(rtpSocket, recvResult.RemoteEndPoint as IPEndPoint, rtpSendTimestamp, new byte[] { 0x00 });
rtpSendTimestamp += 32000; // Arbitrary and not critical. Corresponds to 40ms payload at 25pps which means 4s for 100 packets.
packetSentCount++;
bytesSentCount++;
SIPSorcery.Sys.Log.Logger.LogDebug($"RTP sent {rtpSocket.LocalEndPoint}->{remoteRtpEndPoint} pkts {packetSentCount} bytes {bytesSentCount}");
}
recvResult = await rtpSocket.ReceiveFromAsync(buffer, SocketFlags.None, anyEndPoint);
}
}
}
catch (ObjectDisposedException) { } // This is how .Net deals with an in use socket being closed. Safe to ignore.
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception processing RTP. {excp}");
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
IPAddress localAddress = sipTransport.GetLocalAddress(IPAddress.Parse("8.8.8.8"));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(localAddress, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, port)));
//EnableTraceLogs(sipTransport);
// Select the IP address to use for RTP based on the destination SIP URI.
var endPointForCall = callUri.ToSIPEndPoint() == null?sipTransport.GetDefaultSIPEndPoint(callUri.Protocol) : sipTransport.GetDefaultSIPEndPoint(callUri.ToSIPEndPoint());
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
// TODO (find something better): If the SIP endpoint is using 0.0.0.0 for SIP use loopback for RTP.
IPAddress rtpAddress = localAddress;
NetServices.CreateRtpSocket(rtpAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
IPEndPoint remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Sending initial RTP packet to remote RTP socket {remoteRtpEndPoint}.");
// Send a dummy packet to open the NAT session on the RTP path.
rtpSendSession.SendAudioFrame(rtpSocket, remoteRtpEndPoint, 0, new byte[] { 0x00 });
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => SendRecvRtp(rtpSocket, rtpSendSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
rtpCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
private static async Task SendRtp(RTPSession rtpSession, IPEndPoint dstRtpEndPoint, string audioFileName, CancellationTokenSource cts)
{
try
{
string audioFileExt = Path.GetExtension(audioFileName).ToLower();
switch (audioFileExt)
{
case ".g722":
case ".ulaw":
{
uint timestamp = 0;
using (StreamReader sr = new StreamReader(audioFileName))
{
byte[] buffer = new byte[320];
int bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
while (bytesRead > 0 && !cts.IsCancellationRequested)
{
if (!dstRtpEndPoint.Address.Equals(IPAddress.Any))
{
rtpSession.SendAudioFrame(timestamp, buffer);
}
timestamp += (uint)buffer.Length;
await Task.Delay(40, cts.Token);
bytesRead = sr.BaseStream.Read(buffer, 0, buffer.Length);
}
}
}
break;
case ".mp3":
{
var pcmFormat = new WaveFormat(8000, 16, 1);
var ulawFormat = WaveFormat.CreateMuLawFormat(8000, 1);
uint timestamp = 0;
using (WaveFormatConversionStream pcmStm = new WaveFormatConversionStream(pcmFormat, new Mp3FileReader(audioFileName)))
{
using (WaveFormatConversionStream ulawStm = new WaveFormatConversionStream(ulawFormat, pcmStm))
{
byte[] buffer = new byte[320];
int bytesRead = ulawStm.Read(buffer, 0, buffer.Length);
while (bytesRead > 0 && !cts.IsCancellationRequested)
{
byte[] sample = new byte[bytesRead];
Array.Copy(buffer, sample, bytesRead);
if (dstRtpEndPoint.Address != IPAddress.Any)
{
rtpSession.SendAudioFrame(timestamp, buffer);
}
timestamp += (uint)buffer.Length;
await Task.Delay(40, cts.Token);
bytesRead = ulawStm.Read(buffer, 0, buffer.Length);
}
}
}
}
break;
default:
throw new NotImplementedException($"The {audioFileExt} file type is not understood by this example.");
}
}
catch (OperationCanceledException) { }
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception sending RTP. {excp.Message}");
}
}
private static int INPUT_SAMPLE_PERIOD_MILLISECONDS = 20; // This sets the frequency of the RTP packets.
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
ManualResetEvent exitMre = new ManualResetEvent(false);
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
// Get the IP address the RTP will be sent from. While we can listen on IPAddress.Any | IPv6Any
// we can't put 0.0.0.0 or [::0] in the SDP or the callee will ignore us.
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RTPSession((int)SDPMediaFormatsEnum.PCMU, null, null, true, localIPAddress.AddressFamily);
var offerSDP = rtpSession.GetSDP(localIPAddress);
// Get the audio input device.
WaveInEvent waveInEvent = GetAudioInputDevice();
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// Only set the remote RTP end point if there hasn't already been a packet received on it.
if (rtpSession.DestinationEndPoint == null)
{
rtpSession.DestinationEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {rtpSession.DestinationEndPoint}.");
}
rtpSession.SetRemoteSDP(SDP.ParseSDPDescription(resp.Body));
waveInEvent.StartRecording();
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitMre.Set();
}
}
};
// Wire up the RTP receive session to the audio output device.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
rtpSession.OnReceivedSampleReady += (sample) =>
{
for (int index = 0; index < sample.Length; index++)
{
short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]);
byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
audioOutProvider.AddSamples(pcmSample, 0, 2);
}
};
// Wire up the RTP send session to the audio input device.
uint rtpSendTimestamp = 0;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (rtpSession.DestinationEndPoint != null)
{
rtpSession.SendAudioFrame(rtpSendTimestamp, sample);
rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
}
};
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor);
uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
rtpSession.CloseSession(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}