/// <summary>
/// Used to hangup the call or indicate that the client hungup.
/// </summary>
/// <param name="clientHungup">True if the BYE request was received from the client. False if the hangup
/// needs to originate from this agent.</param>
public void Hangup(bool clientHungup)
{
m_isHungup = true;
if (m_sipDialogue == null)
{
return;
}
// Only need to send a BYE request if the client didn't already do so.
if (clientHungup == false)
{
try
{
SIPEndPoint localEndPoint = (m_outboundProxy != null) ?
m_sipTransport.GetDefaultSIPEndPoint(m_outboundProxy) :
m_sipTransport.GetDefaultSIPEndPoint(GetRemoteTargetEndpoint());
SIPRequest byeRequest = GetByeRequest(localEndPoint);
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(byeRequest, null, localEndPoint, m_outboundProxy);
byeTransaction.NonInviteTransactionFinalResponseReceived += ByeServerFinalResponseReceived;
byeTransaction.SendReliableRequest();
}
catch (Exception excp)
{
logger.LogError("Exception SIPServerUserAgent Hangup. " + excp.Message);
throw;
}
}
}
/// <summary>
/// Used to hangup the call or indicate that the client hungup.
/// </summary>
/// <param name="clientHungup">True if the BYE request was received from the client. False if the hangup
/// needs to originate from this agent.</param>
public void Hangup(bool clientHungup)
{
m_isHungup = true;
if (SIPDialogue == null)
{
return;
}
// Only need to send a BYE request if the client didn't already do so.
if (clientHungup == false)
{
try
{
// Cases found where the Contact in the INVITE was to a different protocol than the oringinal request.
var inviteContact = m_uasTransaction.TransactionRequest.Header.Contact.FirstOrDefault();
if (inviteContact == null)
{
logger.LogWarning("The Contact header on the INVITE request was missing, BYE request cannot be generated.");
}
else
{
SIPRequest byeRequest = GetByeRequest();
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(byeRequest, m_outboundProxy);
byeTransaction.NonInviteTransactionFinalResponseReceived += ByeServerFinalResponseReceived;
byeTransaction.SendReliableRequest();
}
}
catch (Exception excp)
{
logger.LogError("Exception SIPServerUserAgent Hangup. " + excp.Message);
throw;
}
}
}
public void Hangup()
{
if (m_sipDialogue == null)
{
return;
}
try
{
SIPEndPoint localEndPoint = (m_outboundProxy != null) ?
m_sipTransport.GetDefaultSIPEndPoint(m_outboundProxy) :
m_sipTransport.GetDefaultSIPEndPoint(GetRemoteTargetEndpoint());
SIPRequest byeRequest = GetByeRequest(localEndPoint);
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(byeRequest, null, localEndPoint, m_outboundProxy);
byeTransaction.NonInviteTransactionFinalResponseReceived += ByeServerFinalResponseReceived;
byeTransaction.SendReliableRequest();
}
catch (Exception excp)
{
logger.Error("Exception SIPServerUserAgent Hangup. " + excp.Message);
throw;
}
}
/// <summary>
/// Initiates a SUBSCRIBE request to a notification server.
/// </summary>
/// <param name="subscribeToURI">The SIP user that dialog notifications are being subscribed to.</param>
public void Subscribe(SIPURI subscribeURI, int expiry, SIPEventPackage sipEventPackage, string subscribeCallID, SIPURI contactURI)
{
try
{
if (m_attempts >= MAX_SUBSCRIBE_ATTEMPTS)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.NotifierClient, SIPMonitorEventTypesEnum.SubscribeFailed, "Susbcription to " + subscribeURI.ToString() + " reached the maximum number of allowed attempts without a failure condition.", null));
m_subscribed = false;
SubscriptionFailed(subscribeURI, SIPResponseStatusCodesEnum.InternalServerError, "Subscription reached the maximum number of allowed attempts.");
m_waitForSubscribeResponse.Set();
}
else
{
m_attempts++;
m_localCSeq++;
SIPRequest subscribeRequest = m_sipTransport.GetRequest(
SIPMethodsEnum.SUBSCRIBE,
m_resourceURI,
new SIPToHeader(null, subscribeURI, m_subscriptionToTag),
null);
if (contactURI != null)
{
subscribeRequest.Header.Contact = new List <SIPContactHeader>()
{
new SIPContactHeader(null, contactURI)
};
}
subscribeRequest.Header.From = new SIPFromHeader(null, new SIPURI(m_authUsername, m_authDomain, null, SIPSchemesEnum.sip, SIPProtocolsEnum.udp), m_subscriptionFromTag);
subscribeRequest.Header.CSeq = m_localCSeq;
subscribeRequest.Header.Expires = expiry;
subscribeRequest.Header.Event = sipEventPackage.ToString();
subscribeRequest.Header.CallId = subscribeCallID;
if (!m_filter.IsNullOrBlank())
{
subscribeRequest.Body = m_filter;
subscribeRequest.Header.ContentLength = m_filter.Length;
subscribeRequest.Header.ContentType = m_filterTextType;
}
SIPNonInviteTransaction subscribeTransaction = m_sipTransport.CreateNonInviteTransaction(subscribeRequest, m_outboundProxy);
subscribeTransaction.NonInviteTransactionFinalResponseReceived += SubscribeTransactionFinalResponseReceived;
subscribeTransaction.NonInviteTransactionTimedOut += SubsribeTransactionTimedOut;
m_sipTransport.SendSIPReliable(subscribeTransaction);
LastSubscribeAttempt = DateTime.Now;
}
}
catch (Exception excp)
{
logger.LogError("Exception SIPNotifierClient Subscribe. " + excp.Message);
SubscriptionFailed(m_resourceURI, SIPResponseStatusCodesEnum.InternalServerError, "Exception Subscribing. " + excp.Message);
m_waitForSubscribeResponse.Set();
}
}
private void SendInitialRegister()
{
try
{
if (m_attempts >= MAX_REGISTER_ATTEMPTS)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.UserAgentClient, SIPMonitorEventTypesEnum.ContactRegisterFailed, "Registration to " + m_sipAccountAOR.ToString() + " reached the maximum number of allowed attempts without a failure condition.", m_owner));
m_isRegistered = false;
if (RegistrationTemporaryFailure != null)
{
RegistrationTemporaryFailure(m_sipAccountAOR, "Registration reached the maximum number of allowed attempts.");
}
m_waitForRegistrationMRE.Set();
}
else
{
m_attempts++;
SIPEndPoint registrarSIPEndPoint = m_outboundProxy;
if (registrarSIPEndPoint == null)
{
SIPDNSLookupResult lookupResult = m_sipTransport.GetHostEndPoint(m_registrarHost, false);
if (lookupResult.LookupError != null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.UserAgentClient, SIPMonitorEventTypesEnum.ContactRegisterFailed, "Could not resolve " + m_registrarHost + ", " + lookupResult.LookupError, m_owner));
}
else
{
registrarSIPEndPoint = lookupResult.GetSIPEndPoint();
}
}
if (registrarSIPEndPoint == null && RegistrationFailed != null)
{
RegistrationFailed(m_sipAccountAOR, "Could not resolve " + m_registrarHost + ".");
}
else
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.UserAgentClient, SIPMonitorEventTypesEnum.ContactRegisterInProgress, "Initiating registration to " + m_registrarHost + " at " + registrarSIPEndPoint.ToString() + " for " + m_sipAccountAOR.ToString() + ".", m_owner));
SIPRequest regRequest = GetRegistrationRequest(m_localEndPoint);
SIPNonInviteTransaction regTransaction = m_sipTransport.CreateNonInviteTransaction(regRequest, registrarSIPEndPoint, m_localEndPoint, m_outboundProxy);
// These handlers need to be on their own threads to take the processing off the SIP transport layer.
regTransaction.NonInviteTransactionFinalResponseReceived += (lep, rep, tn, rsp) => { ThreadPool.QueueUserWorkItem(delegate { ServerResponseReceived(lep, rep, tn, rsp); }); };
regTransaction.NonInviteTransactionTimedOut += (tn) => { ThreadPool.QueueUserWorkItem(delegate { RegistrationTimedOut(tn); }); };
m_sipTransport.SendSIPReliable(regTransaction);
}
}
}
catch (Exception excp)
{
logger.Error("Exception SendInitialRegister to " + m_registrarHost + ". " + excp.Message);
if (RegistrationFailed != null)
{
RegistrationFailed(m_sipAccountAOR, "Exception SendInitialRegister to " + m_registrarHost + ". " + excp.Message);
}
}
}
/*public static string GetDialogueId(string callId, string localTag, string remoteTag)
* {
* return Crypto.GetSHAHashAsString(callId + localTag + remoteTag);
* }
*
* public static string GetDialogueId(SIPHeader sipHeader)
* {
* return Crypto.GetSHAHashAsString(sipHeader.CallId + sipHeader.To.ToTag + sipHeader.From.FromTag);
* }*/
public void Hangup(SIPTransport sipTransport, SIPEndPoint outboundProxy)
{
try
{
SIPEndPoint byeOutboundProxy = null;
if (outboundProxy != null && IPAddress.IsLoopback(outboundProxy.Address))
{
byeOutboundProxy = outboundProxy;
}
else if (!ProxySendFrom.IsNullOrBlank())
{
byeOutboundProxy =
new SIPEndPoint(new IPEndPoint(SIPEndPoint.ParseSIPEndPoint(ProxySendFrom).Address,
m_defaultSIPPort));
}
else if (outboundProxy != null)
{
byeOutboundProxy = outboundProxy;
}
SIPEndPoint localEndPoint = (byeOutboundProxy != null)
? sipTransport.GetDefaultSIPEndPoint(byeOutboundProxy.Protocol)
: sipTransport.GetDefaultSIPEndPoint(GetRemoteTargetProtocol());
SIPRequest byeRequest = GetByeRequest(localEndPoint);
SIPNonInviteTransaction byeTransaction =
sipTransport.CreateNonInviteTransaction(byeRequest, null, localEndPoint, byeOutboundProxy);
byeTransaction.SendReliableRequest();
}
catch (Exception excp)
{
logger.Error("Exception SIPDialogue Hangup. " + excp.Message);
throw;
}
}
/// <summary>
/// Generates a BYE request for this dialog and forwards it to the remote cal party.
/// This has the effect of hanging up the call.
/// </summary>
/// <param name="sipTransport">The transport layer to use for sending the request.</param>
/// <param name="outboundProxy">Optional. If set an end point that the BYE request will be directly forwarded to.</param>
public void Hangup(SIPTransport sipTransport, SIPEndPoint outboundProxy)
{
try
{
SIPEndPoint byeOutboundProxy = null;
if (outboundProxy != null && IPAddress.IsLoopback(outboundProxy.Address))
{
byeOutboundProxy = outboundProxy;
}
else if (!ProxySendFrom.IsNullOrBlank())
{
byeOutboundProxy = new SIPEndPoint(new IPEndPoint(SIPEndPoint.ParseSIPEndPoint(ProxySendFrom).Address, m_defaultSIPPort));
}
else if (outboundProxy != null)
{
byeOutboundProxy = outboundProxy;
}
SIPRequest byeRequest = GetInDialogRequest(SIPMethodsEnum.BYE);
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(byeRequest, byeOutboundProxy);
byeTransaction.SendReliableRequest();
}
catch (Exception excp)
{
logger.LogError("Exception SIPDialogue Hangup. " + excp.Message);
}
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var sipEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
switch (sipRequest.Method)
{
case SIPMethodsEnum.BYE:
logger.Debug($"{ prefix } Hangup from { sipRequest.Header.From.FromURI.User }");
var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, sipEndPoint, null);
var response = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
noninvite.SendFinalResponse(response);
StateChanged?.Invoke(this, new ConsumerStateEventArgs(State.Finished, endpoint, SessionId));
rtpChannel.OnFrameReady -= RtpChannel_OnFrameReady;
rtpChannel.Close();
StateChanged?.Invoke(this, new ConsumerStateEventArgs(State.Ready, endpoint, SessionId));
return;
case SIPMethodsEnum.CANCEL:
break;
}
}
public void SendRequest(SIPMethodsEnum method)
{
try
{
SIPRequest req = GetRequest(method);
SIPNonInviteTransaction tran = m_sipTransport.CreateNonInviteTransaction(req, null, m_sipTransport.GetDefaultSIPEndPoint(), m_outboundProxy);
ManualResetEvent waitForResponse = new ManualResetEvent(false);
tran.NonInviteTransactionTimedOut += RequestTimedOut;
tran.NonInviteTransactionFinalResponseReceived += ServerResponseReceived;
tran.SendReliableRequest();
}
catch (Exception excp)
{
logger.LogError("Exception SIPNonInviteClientUserAgent SendRequest to " + m_callDescriptor.Uri + ". " + excp.Message);
throw;
}
}
/// <summary>
/// 处理注册请求消息
/// </summary>
/// <param name="localSIPEndPoint">本地终结点</param>
/// <param name="remoteEndPoint">远程终结点</param>
/// <param name="registerRequest">注册请求</param>
private void ProcessRegisterReqMessage(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest registerRequest)
{
SIPSorceryPerformanceMonitor.IncrementCounter(SIPSorceryPerformanceMonitor.REGISTRAR_REGISTRATION_REQUESTS_PER_SECOND);
int requestedExpiry = GetRequestedExpiry(registerRequest);
if (registerRequest.Header.To == null)
{
logger.Debug("Bad register request, no To header from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing To header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (registerRequest.Header.To.ToURI.User.IsNullOrBlank())
{
logger.Debug("Bad register request, no To user from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing username on To header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (registerRequest.Header.Contact == null || registerRequest.Header.Contact.Count == 0)
{
logger.Debug("Bad register request, no Contact header from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing Contact header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (requestedExpiry > 0 && requestedExpiry < MINIMUM_EXPIRY_SECONDS)
{
logger.Debug("Bad register request, no expiry of " + requestedExpiry + " to small from " + remoteEndPoint + ".");
SIPResponse tooFrequentResponse = GetErrorResponse(registerRequest, SIPResponseStatusCodesEnum.IntervalTooBrief, null);
tooFrequentResponse.Header.MinExpires = MINIMUM_EXPIRY_SECONDS;
m_sipTransport.SendResponse(tooFrequentResponse);
}
else
{
if (m_registerQueue.Count < MAX_REGISTER_QUEUE_SIZE)
{
SIPNonInviteTransaction registrarTransaction = m_sipTransport.CreateNonInviteTransaction(registerRequest, remoteEndPoint, localSIPEndPoint, null);
lock (m_registerQueue)
{
m_registerQueue.Enqueue(registrarTransaction);
}
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingInProgress, "Register queued for " + registerRequest.Header.To.ToURI.ToString() + ".", null));
}
else
{
logger.Error("Register queue exceeded max queue size " + MAX_REGISTER_QUEUE_SIZE + ", overloaded response sent.");
SIPResponse overloadedResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, "Registrar overloaded, please try again shortly");
m_sipTransport.SendResponse(overloadedResponse);
}
m_registerARE.Set();
}
}
private void SendNotifyRequestForSubscription(SIPEventSubscription subscription)
{
try
{
subscription.SubscriptionDialogue.CSeq++;
//logger.Debug(DateTime.Now.ToString("HH:mm:ss:fff") + " Sending NOTIFY request for " + subscription.SubscriptionDialogue.Owner + " event " + subscription.SubscriptionEventPackage.ToString()
// + " and " + subscription.ResourceURI.ToString() + " to " + subscription.SubscriptionDialogue.RemoteTarget.ToString() + ", cseq=" + (subscription.SubscriptionDialogue.CSeq) + ".");
int secondsRemaining = Convert.ToInt32(subscription.LastSubscribe.AddSeconds(subscription.Expiry).Subtract(DateTime.Now).TotalSeconds % Int32.MaxValue);
SIPRequest notifyRequest = m_sipTransport.GetRequest(SIPMethodsEnum.NOTIFY, subscription.SubscriptionDialogue.RemoteTarget);
notifyRequest.Header.From = SIPFromHeader.ParseFromHeader(subscription.SubscriptionDialogue.LocalUserField.ToString());
notifyRequest.Header.To = SIPToHeader.ParseToHeader(subscription.SubscriptionDialogue.RemoteUserField.ToString());
notifyRequest.Header.Event = subscription.SubscriptionEventPackage.ToString();
notifyRequest.Header.CSeq = subscription.SubscriptionDialogue.CSeq;
notifyRequest.Header.CallId = subscription.SubscriptionDialogue.CallId;
notifyRequest.Body = subscription.GetNotifyBody();
notifyRequest.Header.ContentLength = notifyRequest.Body.Length;
notifyRequest.Header.SubscriptionState = "active;expires=" + secondsRemaining.ToString();
notifyRequest.Header.ContentType = subscription.NotifyContentType;
notifyRequest.Header.ProxySendFrom = subscription.SubscriptionDialogue.ProxySendFrom;
// If the outbound proxy is a loopback address, as it will normally be for local deployments, then it cannot be overriden.
SIPEndPoint dstEndPoint = m_outboundProxy;
if (m_outboundProxy != null && IPAddress.IsLoopback(m_outboundProxy.Address))
{
dstEndPoint = m_outboundProxy;
}
else if (subscription.SubscriptionDialogue.ProxySendFrom != null)
{
// The proxy will always be listening on UDP port 5060 for requests from internal servers.
dstEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(SIPEndPoint.ParseSIPEndPoint(subscription.SubscriptionDialogue.ProxySendFrom).Address, m_defaultSIPPort));
}
SIPNonInviteTransaction notifyTransaction = m_sipTransport.CreateNonInviteTransaction(notifyRequest, dstEndPoint, m_sipTransport.GetDefaultSIPEndPoint(dstEndPoint), m_outboundProxy);
notifyTransaction.NonInviteTransactionFinalResponseReceived += (local, remote, transaction, response) => { NotifyTransactionFinalResponseReceived(local, remote, transaction, response, subscription); };
m_sipTransport.SendSIPReliable(notifyTransaction);
//logger.Debug(notifyRequest.ToString());
MonitorLogEvent_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.NotifySent, "Notification sent for " + subscription.SubscriptionEventPackage.ToString() + " and " + subscription.ResourceURI.ToString() + " to " + subscription.SubscriptionDialogue.RemoteTarget.ToString() + " (cseq=" + notifyRequest.Header.CSeq + ").", subscription.SubscriptionDialogue.Owner));
subscription.NotificationSent();
}
catch (Exception excp)
{
logger.Error("Exception SendNotifyRequestForSubscription. " + excp.Message);
throw;
}
}
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
switch (sipRequest.Method)
{
case SIPMethodsEnum.OPTIONS:
SIPNonInviteTransaction optionsTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse optionsResponse = SipHelper.WG67ResponseNormalize(
SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null));
optionsTransaction.SendFinalResponse(optionsResponse);
OptionsReceived?.Invoke(remoteEndPoint.ToString());
break;
case SIPMethodsEnum.SUBSCRIBE:
m_sip_notifier.AddSubscribeRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
break;
case SIPMethodsEnum.PUBLISH:
default:
throw new NotImplementedException();
}
}
catch (NotImplementedException)
{
_logger.From().Debug(sipRequest.Method + " request processing not implemented for " + sipRequest.URI.ToParameterlessString() + " from " + remoteEndPoint + ".");
SIPNonInviteTransaction notImplTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse notImplResponse = SipHelper.WG67ResponseNormalize(SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotImplemented, null));
notImplTransaction.SendFinalResponse(notImplResponse);
}
catch (Exception x)
{
_logger.From().Debug($"SIPTransportRequestReceived Exception {x.Message}", x);
}
}
public void SendSIPRequest(SIPRequest sipRequest, string dstSocket)
{
//ResetSIPResponse();
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
//m_inviteRequest = sipRequest;
UACInviteTransaction inviteTransaction = m_sipTransport.CreateUACTransaction(sipRequest, IPSocket.GetIPEndPoint(dstSocket), m_sipTransport.GetTransportContact(null), SIPProtocolsEnum.UDP);
inviteTransaction.UACInviteTransactionInformationResponseReceived += new SIPTransactionResponseReceivedDelegate(TransactionInformationResponseReceived);
inviteTransaction.UACInviteTransactionFinalResponseReceived += new SIPTransactionResponseReceivedDelegate(TransactionFinalResponseReceived);
m_sipTransport.SendSIPReliable(inviteTransaction);
}
else
{
SIPNonInviteTransaction sipTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, IPSocket.GetIPEndPoint(dstSocket), m_sipTransport.GetTransportContact(null), SIPProtocolsEnum.UDP);
sipTransaction.NonInviteTransactionFinalResponseReceived += new SIPTransactionResponseReceivedDelegate(TransactionFinalResponseReceived);
m_sipTransport.SendSIPReliable(sipTransaction);
}
}
public void AddSubscribeRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest subscribeRequest)
{
try
{
if (subscribeRequest.Method != SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notSupportedResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, "Subscribe requests only");
m_sipTransport.SendResponse(notSupportedResponse);
}
else
{
#region Do as many validation checks as possible on the request before adding it to the queue.
if (subscribeRequest.Header.Event.IsNullOrBlank() ||
!(subscribeRequest.Header.Event.ToLower() == SIPEventPackage.Dialog.ToString().ToLower() || subscribeRequest.Header.Event.ToLower() == SIPEventPackage.Presence.ToString().ToLower()))
{
SIPResponse badEventResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.BadEvent, null);
m_sipTransport.SendResponse(badEventResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Event type " + subscribeRequest.Header.Event + " not supported for " + subscribeRequest.URI.ToString() + ".", null));
}
else if (subscribeRequest.Header.Expires > 0 && subscribeRequest.Header.Expires < MIN_SUBSCRIPTION_EXPIRY)
{
SIPResponse tooBriefResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.IntervalTooBrief, null);
tooBriefResponse.Header.MinExpires = MIN_SUBSCRIPTION_EXPIRY;
m_sipTransport.SendResponse(tooBriefResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request was rejected as interval too brief " + subscribeRequest.Header.Expires + ".", null));
}
else if (subscribeRequest.Header.Contact == null || subscribeRequest.Header.Contact.Count == 0)
{
SIPResponse noContactResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing Contact header");
m_sipTransport.SendResponse(noContactResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request was rejected due to no Contact header.", null));
}
#endregion
else
{
if (m_notifierQueue.Count < MAX_NOTIFIER_QUEUE_SIZE)
{
SIPNonInviteTransaction subscribeTransaction = m_sipTransport.CreateNonInviteTransaction(subscribeRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
lock (m_notifierQueue)
{
m_notifierQueue.Enqueue(subscribeTransaction);
}
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeQueued, "Subscribe queued for " + subscribeRequest.Header.To.ToURI.ToString() + ".", null));
}
else
{
logger.Error("Subscribe queue exceeded max queue size " + MAX_NOTIFIER_QUEUE_SIZE + ", overloaded response sent.");
SIPResponse overloadedResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, "Notifier overloaded, please try again shortly");
m_sipTransport.SendResponse(overloadedResponse);
}
m_notifierARE.Set();
}
}
}
catch (Exception excp)
{
logger.Error("Exception AddNotifierRequest (" + remoteEndPoint.ToString() + "). " + excp.Message);
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.G722)))
{
Log.LogDebug($"Using G722 RTP media type and audio file {AUDIO_FILE_G722}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.G722, null, null);
audioFile = AUDIO_FILE_G722;
}
else if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(dstRtpEndPoint.AddressFamily == AddressFamily.InterNetworkV6 ? IPAddress.IPv6Any : IPAddress.Any, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
IPEndPoint rtpEndPoint = new IPEndPoint(rtpAddress, (rtpSocket.LocalEndPoint as IPEndPoint).Port);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, audioFile, rtpCts))
.ContinueWith(_ =>
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
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();
}
}
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 rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
// Uncomment this line to see each SIP message sent and received.
EnableTraceLogs(sipTransport);
// Send an OPTIONS request to determine the local IP address to use for the RTP socket.
var optionsTask = SendOptionsTaskAsync(sipTransport, callUri);
var result = Task.WhenAny(optionsTask, Task.Delay(SIP_REQUEST_TIMEOUT_MILLISECONDS));
result.Wait();
if (optionsTask.IsCompletedSuccessfully == false || optionsTask.Result == null)
{
Log.LogError($"OPTIONS request to {callUri} failed.");
}
else
{
IPAddress localIPAddress = optionsTask.Result;
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 49000, 49100, 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($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
Log.LogDebug(resp.ToString());
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{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}.");
_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, 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(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(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 is established. We'll wait for a few seconds and then transfer.
Task.Delay(DELAY_UNTIL_TRANSFER_MILLISECONDS).Wait();
SIPRequest referRequest = GetReferRequest(uac, SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI));
SIPNonInviteTransaction referTx = sipTransport.CreateNonInviteTransaction(referRequest, referRequest.RemoteSIPEndPoint, referRequest.LocalSIPEndPoint, null);
referTx.NonInviteTransactionFinalResponseReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPTransaction sipTransaction, SIPResponse sipResponse) =>
{
if (sipResponse.Header.CSeqMethod == SIPMethodsEnum.REFER && sipResponse.Status == SIPResponseStatusCodesEnum.Accepted)
{
Log.LogInformation("Call transfer was accepted by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
};
referTx.SendReliableRequest();
// At this point the call transfer 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 ConcurrentQueue <RTPEvent> _dtmfEvents = new ConcurrentQueue <RTPEvent>(); // Add a DTMF event to this queue to have the it sent
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 rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// Note this relies on the callURI host being an IP address. If it's a hostname a DNS lookup is required.
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(callUri.ToSIPEndPoint().Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{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($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"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;
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(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(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, RTPPayloadTypesEnum.PCMU).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.
Task.Delay(3000).Wait();
// Add some DTMF events to the queue. These will be transmitted by the SendRtp thread.
_dtmfEvents.Enqueue(new RTPEvent(0x05, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
_dtmfEvents.Enqueue(new RTPEvent(0x09, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
_dtmfEvents.Enqueue(new RTPEvent(0x02, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
Log.LogInformation("Exiting...");
rtpCts.Cancel();
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();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// 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.
var sipTransport = new SIPTransport();
sipTransport.ContactHost = Dns.GetHostName();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//if (File.Exists("localhost.pfx"))
//{
// var certificate = new X509Certificate2(@"localhost.pfx", "");
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.IPv6Any, SIPS_LISTEN_PORT)));
//}
//EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
// In practice there could be a number of reasons to reject the call. Unsupported extensions, mo matching codecs etc. etc.
if (sipRequest.Header.HasUnknownRequireExtension)
{
// The caller requires an extension that we don't support.
SIPSorcery.Sys.Log.Logger.LogWarning($"Rejecting incoming call due to one or more required exensions not being supported, required extensions: {sipRequest.Header.Require}.");
uas.Reject(SIPResponseStatusCodesEnum.NotImplemented, "Unsupported Require Extension", null);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Simulating answer delay to test provisional response retransmits.
await Task.Delay(2000);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localSIPEndPoint.Address, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, rtpCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
try
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{localSIPEndPoint.ToString()}<-{remoteEndPoint.ToString()}: {sipRequest.StatusLine}");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
catch (Exception optionsExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Failed to send SIP OPTIONS response. {optionsExcp.Message}");
}
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpCts?.Cancel();
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to hangup...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
/// <summary>
/// Handler for processing incoming SIP requests.
/// </summary>
/// <param name="localSIPEndPoint">The end point the request was received on.</param>
/// <param name="remoteEndPoint">The end point the request came from.</param>
/// <param name="sipRequest">The SIP request received.</param>
private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
SIPNonInviteTransaction optionsTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
optionsTransaction.SendFinalResponse(optionsResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
_sipTransport.SendResponse(tryingResponse);
SIPResponseStatusCodesEnum registerResponse = SIPResponseStatusCodesEnum.Ok;
if (sipRequest.Header.Contact != null && sipRequest.Header.Contact.Count > 0)
{
int expiry = sipRequest.Header.Contact[0].Expires > 0 ? sipRequest.Header.Contact[0].Expires : sipRequest.Header.Expires;
var sipAccount = new SIPAccount(null, sipRequest.Header.From.FromURI.Host, sipRequest.Header.From.FromURI.User, null, null);
SIPAccountBinding binding = new SIPAccountBinding(sipAccount, sipRequest.Header.Contact[0].ContactURI, remoteEndPoint, localSIPEndPoint, expiry);
if (_sipRegistrations.ContainsKey(sipAccount.SIPUsername))
{
_sipRegistrations.Remove(sipAccount.SIPUsername);
}
_sipRegistrations.Add(sipAccount.SIPUsername, binding);
logger.LogDebug("Registered contact for " + sipAccount.SIPUsername + " as " + binding.RegisteredContact.ToString() + ".");
}
else
{
registerResponse = SIPResponseStatusCodesEnum.BadRequest;
}
SIPNonInviteTransaction registerTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, registerResponse, null);
registerTransaction.SendFinalResponse(okResponse);
}
else
{
logger.LogDebug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
}
}
catch (NotImplementedException)
{
logger.LogDebug(sipRequest.Method + " request processing not implemented for " + sipRequest.URI.ToParameterlessString() + " from " + remoteEndPoint + ".");
SIPNonInviteTransaction notImplTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse notImplResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotImplemented, null);
notImplTransaction.SendFinalResponse(notImplResponse);
}
}
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 server example.");
Console.WriteLine("Press ctrl-c to exit.");
// 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);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource uasCts = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept a little it simpler.
uas?.Hangup();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uasCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// 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);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, uasCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup();
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup();
uasCts?.Cancel();
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
if (uas?.IsHungup == false)
{
uas?.Hangup();
}
uasCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
public void ProcessNotifyRequest(SIPRequest sipRequest)
{
try
{
// Hack to work around MWI request from callcentric not having a trailing CRLF and breaking some softphones like the Bria.
if (sipRequest.Header.Event == MWI_EVENT_TYPE && sipRequest.Body.NotNullOrBlank() && sipRequest.Body.Substring(sipRequest.Body.Length - 2) != m_crlf)
{
sipRequest.Body += m_crlf;
}
string fromURI = (sipRequest.Header.From != null && sipRequest.Header.From.FromURI != null) ? sipRequest.Header.From.FromURI.ToString() : "unknown";
string domain = GetCanonicalDomain_External(sipRequest.URI.Host, true);
if (domain != null)
{
SIPAccountAsset sipAccount = GetSIPAccount_External(s => s.SIPUsername == sipRequest.URI.User && s.SIPDomain == domain);
if (sipAccount != null)
{
List <SIPRegistrarBinding> bindings = GetSIPAccountBindings_External(b => b.SIPAccountId == sipAccount.Id, null, 0, MAX_FORWARD_BINDINGS);
if (bindings != null)
{
foreach (SIPRegistrarBinding binding in bindings)
{
SIPURI dstURI = binding.MangledContactSIPURI;
SIPEndPoint localSIPEndPoint = (m_outboundProxy != null) ? m_sipTransport.GetDefaultSIPEndPoint(m_outboundProxy.Protocol) : m_sipTransport.GetDefaultSIPEndPoint(dstURI.Protocol);
SIPEndPoint dstSIPEndPoint = null;
// If the outbound proxy is a loopback address, as it will normally be for local deployments, then it cannot be overriden.
if (m_outboundProxy != null && IPAddress.IsLoopback(m_outboundProxy.Address))
{
dstSIPEndPoint = m_outboundProxy;
}
else if (binding.ProxySIPEndPoint != null)
{
// If the binding has a specific proxy end point sent then the request needs to be forwarded to the proxy's default end point for it to take care of.
dstSIPEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(binding.ProxySIPEndPoint.Address, m_defaultSIPPort));
}
else if (m_outboundProxy != null)
{
dstSIPEndPoint = m_outboundProxy;
}
else
{
SIPDNSLookupResult lookupResult = m_sipTransport.GetURIEndPoint(dstURI, false);
if (lookupResult.LookupError != null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "A NOTIFY request from " + fromURI + " was not forwarded due to DNS failure for " + dstURI.Host + ", " + lookupResult.LookupError + ".", sipAccount.Owner));
}
else
{
dstSIPEndPoint = lookupResult.GetSIPEndPoint();
}
}
if (dstSIPEndPoint != null)
{
// Rather than create a brand new request copy the received one and modify the headers that need to be unique.
SIPRequest notifyRequest = sipRequest.Copy();
notifyRequest.URI = dstURI;
notifyRequest.Header.Contact = SIPContactHeader.CreateSIPContactList(new SIPURI(dstURI.Scheme, localSIPEndPoint));
notifyRequest.Header.To = new SIPToHeader(null, dstURI, null);
notifyRequest.Header.CallId = CallProperties.CreateNewCallId();
SIPViaHeader viaHeader = new SIPViaHeader(localSIPEndPoint, CallProperties.CreateBranchId());
notifyRequest.Header.Vias = new SIPViaSet();
notifyRequest.Header.Vias.PushViaHeader(viaHeader);
// If the binding has a proxy socket defined set the header to ask the upstream proxy to use it.
if (binding.ProxySIPEndPoint != null)
{
notifyRequest.Header.ProxySendFrom = binding.ProxySIPEndPoint.ToString();
// If the binding has a specific proxy end point sent then the request needs to be forwarded to the proxy's default end point for it to take care of.
dstSIPEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(binding.ProxySIPEndPoint.Address, m_defaultSIPPort));
}
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "Forwarding NOTIFY request from " + fromURI + " to registered binding at " + dstURI.ToString() + ", proxy " + dstSIPEndPoint.ToString() + ".", sipAccount.Owner));
SIPNonInviteTransaction notifyTransaction = m_sipTransport.CreateNonInviteTransaction(notifyRequest, dstSIPEndPoint, localSIPEndPoint, dstSIPEndPoint);
notifyTransaction.SendReliableRequest();
}
else
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "A NOTIFY request from " + fromURI + " was not forwarded as no destination end point was resolved.", sipAccount.Owner));
}
}
// Send OK response to server.
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
}
else
{
// Send unavailable response to server.
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for " + sipAccount.SIPUsername + "@" + sipAccount.SIPDomain + " but no bindings available, responding with temporarily unavailable.", sipAccount.Owner));
SIPResponse notAvailableResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, null);
m_sipTransport.SendResponse(notAvailableResponse);
}
}
else
{
// Send Not found response to server.
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for " + sipRequest.URI.ToString() + " but no matching SIP account, responding with not found.", null));
SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, null);
m_sipTransport.SendResponse(notFoundResponse);
}
}
else
{
// Send Not Serviced response to server.
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for a non-serviced domain responding with not found.", null));
SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced");
m_sipTransport.SendResponse(notServicedResponse);
}
}
catch (Exception excp)
{
logger.Error("Exception SIPNotifyManager ProcessNotifyRequest. " + 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();
}
};
}
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();
}
}
/// <summary>
/// Handler for when an in dialog request is received on an established call.
/// Typical types of request will be re-INVITES for things like putting a call on or
/// off hold and REFER requests for transfers. Some in dialog request types, such
/// as re-INVITES have specific events so they can be bubbled up to the
/// application to deal with.
/// </summary>
/// <param name="request">The in dialog request received.</param>
public async Task InDialogRequestReceivedAsync(SIPRequest sipRequest)
{
// Make sure the request matches our dialog and is not a stray.
// A dialog request should match on to tag, from tag and call ID. We'll be more
// accepting just in case the sender got the tags wrong.
if (Dialogue == null || sipRequest.Header.CallId != Dialogue.CallId)
{
var noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
var sendResult = await SendResponse(noCallLegResponse);
if (sendResult != SocketError.Success)
{
logger.LogWarning($"SIPUserAgent send response failed in InCallRequestReceivedAsync with {sendResult}.");
}
}
else
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
logger.LogDebug($"Matching dialogue found for {sipRequest.StatusLine}.");
SIPNonInviteTransaction byeTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallHungup?.Invoke();
m_uac = null;
m_uas = null;
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
logger.LogDebug($"Re-INVITE request received {sipRequest.StatusLine}.");
UASInviteTransaction reInviteTransaction = m_transport.CreateUASTransaction(sipRequest, m_outboundProxy);
if (OnReinviteRequest == null)
{
// The application isn't prepared to accept re-INVITE requests. We'll reject as gently as we can to try and not lose the call.
SIPResponse notAcceptableResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptable, null);
reInviteTransaction.SendFinalResponse(notAcceptableResponse);
}
else
{
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
reInviteTransaction.SendProvisionalResponse(tryingResponse);
OnReinviteRequest(reInviteTransaction);
}
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "OPTIONS request for established dialogue " + dialogue.DialogueName + ".", dialogue.Owner));
SIPNonInviteTransaction optionsTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Body = Dialogue.RemoteSDP;
okResponse.Header.ContentLength = okResponse.Body.Length;
okResponse.Header.ContentType = m_sdpContentType;
optionsTransaction.SendFinalResponse(okResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "MESSAGE for call " + sipRequest.URI.ToString() + ": " + sipRequest.Body + ".", dialogue.Owner));
SIPNonInviteTransaction messageTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
messageTransaction.SendFinalResponse(okResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "REFER received on dialogue " + dialogue.DialogueName + ", transfer mode is " + dialogue.TransferMode + ".", dialogue.Owner));
SIPNonInviteTransaction referTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
if (sipRequest.Header.ReferTo.IsNullOrBlank())
{
// A REFER request must have a Refer-To header.
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Bad REFER request, no Refer-To header.", dialogue.Owner));
SIPResponse invalidResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing mandatory Refer-To header");
referTransaction.SendFinalResponse(invalidResponse);
}
else
{
//TODO: Add handling logic for indialog REFER requests.
}
}
}
}
/// <summary>
/// Handler for processing incomign SIP requests.
/// </summary>
/// <param name="localSIPEndPoint">The end point the request was received on.</param>
/// <param name="remoteEndPoint">The end point the request came from.</param>
/// <param name="sipRequest">The SIP request received.</param>
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (m_uac != null && m_uac.SIPDialogue != null && sipRequest.Header.CallId == m_uac.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup by remote end.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else if (m_uas != null && m_uas.SIPDialogue != null && sipRequest.Header.CallId == m_uas.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup.");
m_uas.SIPDialogue.Hangup(m_sipTransport, null);
CallFinished();
}
else
{
logger.Debug("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
StatusMessage("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
m_uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
m_uas.CallCancelled += UASCallCancelled;
IncomingCall();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
StatusMessage("Call was cancelled by remote end.");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
logger.Debug("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
CallFinished();
}
else
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
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)
{
}
}
