/// <summary>
/// Parses the destination command line option into:
/// - A SIPEndPoint, which is an IP end point and transport (udp, tcp or tls),
/// - A SIP URI.
/// The SIPEndPoint determines the remote network destination to send the request to.
/// The SIP URI is whe URI that will be set on the request.
/// </summary>
/// <param name="dstn">The destination string to parse.</param>
/// <returns>The SIPEndPoint and SIPURI parsed from the destination string.</returns>
private static (SIPEndPoint, SIPURI) ParseDestination(string dst)
{
var dstEp = SIPEndPoint.ParseSIPEndPoint(dst);
SIPURI dstUri = null;
// Don't attempt a SIP URI parse for serialised SIPEndPoints.
if (Regex.IsMatch(dst, "^(udp|tcp|tls)") == false && SIPURI.TryParse(dst))
{
dstUri = SIPURI.ParseSIPURIRelaxed(dst);
}
else
{
dstUri = new SIPURI(dstEp.Scheme, dstEp);
}
if (dstEp == null)
{
logger.LogDebug($"Could not extract IP end point from destination host of {dstUri.Host}.");
var result = SIPDNSManager.ResolveSIPService(dstUri, false);
if (result != null)
{
logger.LogDebug($"Resolved SIP URI {dstUri} to {result.GetSIPEndPoint()}.");
dstEp = result.GetSIPEndPoint();
}
}
return(dstEp, dstUri);
}
/// <summary>
/// Places an outgoing SIP call.
/// </summary>
/// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will
/// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will
/// be sent to the configured SIP server.</param>
public async Task Call(string destination)
{
// Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account.
SIPURI callURI = null;
string sipUsername = null;
string sipPassword = null;
string fromHeader = null;
if (destination.Contains("@") || m_sipServer == null)
{
// Anonymous call direct to SIP server specified in the URI.
callURI = SIPURI.ParseSIPURIRelaxed(destination);
fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString();
}
else
{
// This call will use the pre-configured SIP account.
callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer);
sipUsername = m_sipUsername;
sipPassword = m_sipPassword;
fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipUsername, m_sipServer, null), null)).ToString();
}
StatusMessage(this, $"Starting call to {callURI}.");
var lookupResult = await Task.Run(() =>
{
return(SIPDNSManager.ResolveSIPService(callURI, false));
});
if (lookupResult == null || lookupResult.LookupError != null)
{
StatusMessage(this, $"Call failed, could not resolve {callURI}.");
}
else
{
var dstEndpoint = lookupResult.GetSIPEndPoint();
StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}.");
System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}.");
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null);
var audioSrcOpts = new AudioOptions
{
AudioSource = AudioSourcesEnum.Microphone,
OutputDeviceIndex = m_audioOutDeviceIndex
};
var videoSrcOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
MediaSession = new RtpAVSession(audioSrcOpts, videoSrcOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.InitiateCallAsync(callDescriptor, MediaSession);
}
}
public void ResolveSIPServiceTest()
{
try
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPDNSManager.UseNAPTRLookups = true;
var result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed("sip:reg.sip-trunk.telekom.de;transport=tcp"), false);
SIPEndPoint resultEP = result.GetSIPEndPoint();
Assert.NotNull(resultEP);
logger.LogDebug($"resolved to SIP end point {resultEP}");
Assert.NotEmpty(result.SIPNAPTRResults);
Assert.NotEmpty(result.SIPSRVResults);
Assert.NotEmpty(result.EndPointResults);
result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed("sip:tel.t-online.de"), false);
Assert.NotNull(resultEP);
result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed("sips:hpbxsec.deutschland-lan.de:5061;transport=tls"), false);
Assert.NotNull(resultEP);
}
finally
{
SIPDNSManager.UseNAPTRLookups = false;
}
}
public SIPDNSLookupResult Resolve(SIPRequest sipRequest)
{
if (sipRequest.Header.Routes != null && sipRequest.Header.Routes.Length > 0 && !sipRequest.Header.Routes.TopRoute.IsStrictRouter)
{
return(SIPDNSManager.ResolveSIPService(sipRequest.Header.Routes.TopRoute.URI, true));
}
else
{
return(SIPDNSManager.ResolveSIPService(sipRequest.URI, true));
}
}
public void ResolveHostFromServiceTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
var result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed("sipsorcery.com"), false);
SIPEndPoint resultEP = result.GetSIPEndPoint();
Assert.NotNull(resultEP);
logger.LogDebug($"resolved to SIP end point {resultEP}");
}
public void LookupLocalHostnameTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
string hostname = System.Net.Dns.GetHostName();
var result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed(hostname), false);
SIPEndPoint resultEP = result.GetSIPEndPoint();
Assert.NotNull(resultEP);
logger.LogDebug($"resolved to SIP end point {resultEP}");
}
/// <summary>
/// Places an outgoing SIP call.
/// </summary>
/// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will
/// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will
/// be sent to the configured SIP server.</param>
public async Task Call(string destination)
{
// Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account.
SIPURI callURI = null;
string sipUsername = null;
string sipPassword = null;
string fromHeader = null;
if (destination.Contains("@") || m_sipServer == null)
{
// Anonymous call direct to SIP server specified in the URI.
callURI = SIPURI.ParseSIPURIRelaxed(destination);
fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString();
}
else
{
// This call will use the pre-configured SIP account.
callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer);
sipUsername = m_sipUsername;
sipPassword = m_sipPassword;
fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipUsername, m_sipServer, null), null)).ToString();
}
StatusMessage(this, $"Starting call to {callURI}.");
var lookupResult = await Task.Run(() =>
{
return(SIPDNSManager.ResolveSIPService(callURI, false));
});
if (lookupResult == null || lookupResult.LookupError != null)
{
StatusMessage(this, $"Call failed, could not resolve {callURI}.");
}
else
{
var dstEndpoint = lookupResult.GetSIPEndPoint();
StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}.");
System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}.");
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null);
m_rtpMediaSessionManager.Create(dstEndpoint.Address.AddressFamily);
m_rtpMediaSessionManager.RTPMediaSession.RemotePutOnHold += OnRemotePutOnHold;
m_rtpMediaSessionManager.RTPMediaSession.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.InitiateCall(callDescriptor, m_rtpMediaSessionManager.RTPMediaSession);
}
}
public void ResolveSIPServiceFromCacheTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPURI lookupURI = SIPURI.ParseSIPURIRelaxed("sip:tel.t-online.de");
var result = SIPDNSManager.ResolveSIPService(lookupURI, false);
Assert.NotNull(result);
SIPEndPoint resultEP = result.GetSIPEndPoint();
Assert.NotNull(resultEP);
logger.LogDebug($"resolved to SIP end point {resultEP}");
Assert.NotEmpty(result.SIPSRVResults);
Assert.NotEmpty(result.EndPointResults);
// Do the same look up again immediately to check the result when it comes from the in-memory cache.
var resultCache = SIPDNSManager.ResolveSIPService(lookupURI, false);
Assert.NotNull(resultCache);
Assert.NotNull(resultCache.GetSIPEndPoint());
logger.LogDebug($"cache resolved to SIP end point {resultCache.GetSIPEndPoint()}");
}
public void LookupLocalHostnameTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
string hostname = System.Net.Dns.GetHostName();
if (hostname.EndsWith(SIPDNSManager.MDNS_TLD))
{
// TODO: Look into why DNS calls on macos cannot resolve domains ending in ".local"
// RFC6762 domains.
logger.LogWarning("Skipping unit test LookupLocalHostnameTest due to RFC6762 domain.");
}
else
{
var result = SIPDNSManager.ResolveSIPService(SIPURI.ParseSIPURIRelaxed(hostname), false);
SIPEndPoint resultEP = result.GetSIPEndPoint();
Assert.NotNull(resultEP);
logger.LogDebug($"resolved to SIP end point {resultEP}");
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
// Check whether an override desination has been entered on the command line.
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(_sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"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.
_ourRtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out _ourRtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
exitCts.Cancel();
};
userAgent.ClientCallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{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 (_remoteRtpEndPoint == null)
{
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
userAgent.CallHungup += () =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.OnReinviteRequest += ReinviteRequestReceived;
// The only incoming requests that need to be explicitly in this example program are in-dialog
// re-INVITE requests that are being used to place the call on/off hold.
_sipTransport.SIPTransportRequestReceived += (localSIPEndPoint, remoteEndPoint, sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
_sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception excp)
{
Log.LogError($"Exception processing request. {excp.Message}");
}
};
// 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(() => RecvRtp(_ourRtpSocket, rtpRecvSession, exitCts));
Task.Run(() => SendRtp(_ourRtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
_ourSDP.ToString(),
null);
userAgent.Call(callDescriptor);
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsAnswered)
{
if (_holdStatus == HoldStatus.None)
{
Log.LogInformation("Placing the remote call party on hold.");
_holdStatus = HoldStatus.WePutOnHold;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDONLY);
userAgent.SendReInviteRequest(_ourSDP);
}
else if (_holdStatus == HoldStatus.WePutOnHold)
{
Log.LogInformation("Removing the remote call party from hold.");
_holdStatus = HoldStatus.None;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
userAgent.SendReInviteRequest(_ourSDP);
}
else
{
Log.LogInformation("Sorry we're already on hold by the remote call party.");
}
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
_ourRtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.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();
}
#endregion
}
static async Task 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 treat our RTP stream as inactive.
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RtpAVSession(dstAddress.AddressFamily, new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
var offerSDP = await rtpSession.createOffer(new RTCOfferOptions { RemoteSignallingAddress = dstAddress });
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{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($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.answer, sdp = SDP.ParseSDPDescription(resp.Body)
});
rtpSession.Start();
}
else
{
Log.LogWarning($"{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();
}
}
};
// 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($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.CloseSession(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
public SIPDNSLookupResult Resolve(SIPURI sipURI)
{
return(SIPDNSManager.ResolveSIPService(sipURI, true));
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP trnasport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{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($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// Only set the remote RTP end point if there hasn't already been a packet received on it.
if (_remoteRtpEndPoint == null)
{
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
}
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, 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;
exitCts.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(() => RecvRtp(rtpSocket, rtpRecvSession, exitCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
}
else if (keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
uac.Hangup();
exitCts.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.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 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($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RTPMediaSession((int)SDPMediaFormatsEnum.PCMU, localIPAddress.AddressFamily);
var offerSDP = rtpSession.GetSDP(localIPAddress);
// Get the audio input device.
WaveInEvent waveInEvent = GetAudioInputDevice();
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{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();
}
}